From khalid_khursheed at hotmail.com  Mon Oct  1 02:22:14 2007
From: khalid_khursheed at hotmail.com (Syed Khalid Ali)
Date: Mon, 1 Oct 2007 11:22:14 +0500
Subject: [cisco-voip] CME-AA using TCL Script
In-Reply-To: <46FD0383.901@saskin.net>
References: <BAY113-W2118950FA1DB449CD7F3D5E8B20@phx.gbl>
	<46FD0383.901@saskin.net>
Message-ID: <BAY113-W37D360CA6CD743B89051ACE8AD0@phx.gbl>


hi,
 
actually i'm using it in my lab, first i tried it to recieved all calls to receptionist console using the connection plar opx 100, my ccme 3.3 is connected to a panasonic hybrid key system has 8-lines with are connected to 8-FXO's on my ccme router. hope u have  better understanding now.
 
i'm also confused about the incoming called-number and answered-address command or should i say i don't understand the purpose. glad if u could help....
 
Regarding the CCME-AA, could u point out the mistake in the config.
 
 
 
 
 
Syed Khalid Ali Future Technology 311, Park Avenue, Shahrae Faisal, Karachi - 75400, Pakistan Ph: +92 21 4311908-9, Fax: +92 21 4536571, Cell: +92 333 3354164> Date: Fri, 28 Sep 2007 09:37:07 -0400> From: matt at saskin.net> To: khalid_khursheed at hotmail.com> CC: cisco-voip at puck.nether.net> Subject: Re: [cisco-voip] CME-AA using TCL Script> > To my understanding, the aa-pilot parameter is the DN which the > auto-attendant application will answer on. Basically, you want to > direct inbound calls (either from your voice-ports directly via a PLAR > command or from other dial-peers) to the aa-pilot number as their > destination.> > see this for reference:> http://cciev.wordpress.com/2006/06/09/cme-aa-using-tcl/> > -matt> > Syed Khalid Ali wrote:> > > > hi,> > > > i'm using cme 3.3. we have 8 fxo trunks (number 201 to 208). previously > > the calls on trunks were recieved by the receptionist. But now we want > > to move Auto Attandant. I tried to configure CME-AA but in vain. I am > > confused about the AA Pilot Number. What it should be ?(either a number > > from fxo trunks or some pot/voip dial-peer)> > > > I guess any one in this forum could help.> > Included is the CME router config:> > application> > service cme-aa-queue flash:app-b-acd-2.1.0.0.tcl > > param number-of-hunt-grps 1> > param queue-manager-debugs 1> > param queue-len 30> > param queue-cme-debugs 1> > param aa-hunt1 700> > !> > service cme-aa flash:app-b-acd-aa-2.1.0.0.tcl> > param voice-mail 202> > param max-time-call-retry 20> > paramspace english index 0> > param service-name cme-aa-queue> > param number-of-hunt-grps 1> > param handoff-string cme-aa> > paramspace english language en> > param max-time-vm-retry 0> > paramspace english location flash:> > param second-greeting-time 20> > param welcome-prompt en_bacd_welcome.au> > paramspace english prefix en> > param call-retry-timer 20> > param aa-pilot 100> > !> > dial-peer voice 1 pots> > description ** FXO pots dial-peer **> > destination-pattern 9[2]..> > port 0/0/0> > !> > !> > dial-peer voice 100 voip> > service cme-aa> > incoming called-number 2..> > dtmf-relay h245-alphanumeric> > codec g711ulaw> > no vad> > !> > voice-port 0/0/0> > connection plar opx 100> > caller-id enable> > no shut> > !> > telephony-service> > load 7912 CP7912060000SCCP050124A> > max-ephones 40> > max-dn 60> > ip source-address 192.168.0.1 port 2000> > system message CME Testing> > time-zone 38> > date-format dd-mm-yy> > create cnf-files version-stamp 7960 Sep 27 2007 10:28:53> > max-conferences 8 gain -6> > call-forward pattern .T> > moh music-on-hold.au> > web admin system name ftech password ftech311> > dn-webedit> > time-webedit> > transfer-system full-consult dss> > transfer-pattern .T> > transfer-pattern 600> > secondary-dialtone 9> > directory entry 1 101 name Manager> > directory entry 2 102 name Cashier> > directory entry 3 103 name Reception> > !> > Syed Khalid Ali > > ------------------------------------------------------------------------> > Connect to the next generation of MSN Messenger Get it now! > > <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline>> > > > > > ------------------------------------------------------------------------> > > > _______________________________________________> > cisco-voip mailing list> > cisco-voip at puck.nether.net> > https://puck.nether.net/mailman/listinfo/cisco-voip> 
_________________________________________________________________
Discover the new Windows Vista
http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE
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From ErasmuE4 at telkom.co.za  Mon Oct  1 06:16:15 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 1 Oct 2007 12:16:15 +0200
Subject: [cisco-voip] call manager express AA - prompt question
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01A920FC@CNTRRA20-XCS01.telkom.co.za>


I have a Call manager Express solution with Auto Attendant / Qing
using a drop through option. All works fine but have a problem getting
the drop through to play my custom prompt instead of the prompt with
the cisco lady saying " all agents are currently busy assisting other
customers ........... etc.

 

I have a prompt I recorded in flash ....... en_dt_prompt.au  ... see
show flash below.    I also have the config where I specify param
drop-through-prompt en_dt_prompt.au and also tried param
drop-through-prompt _dt_prompt.au without the leading en. But cannot
get it to work. Not even sure where the system is getting the prompt
it plays because it is not in flash but must be embedded in the system


 

 

I did issue the following commands after making changes........

 

audio-promp load flash:en_dt_prompt.au

call application voice load aa

call application voice load queue

 

 

 

Any advise will be appreciated.

ERIK ERASMUS

 

 

!-----configs and show flash below
---------------------------------------

!

!

application

 service queue flash:app-b-acd-2.1.2.2.tcl

  param queue-len 10

  param aa-hunt1 9999

  param number-of-hunt-grps 1

  param queue-manager-debugs 1

 !

 service aa flash:app-b-acd-aa-2.1.2.2.tcl

  paramspace english index 0

  param number-of-hunt-grps 1

  param drop-through-option 1

  param handoff-string aa

  paramspace english language en

  param max-time-vm-retry 2

  param aa-pilot 8700

  paramspace english location flash:

  param drop-through-prompt en_dt_prompt.au

  param second-greeting-time 20

  param call-retry-timer 15

  param voice-mail 8798

  param max-time-call-retry 100

  param service-name queue

 !

!

!         

!

!

 

 

Sh flash

 

===========

 

cmepelab#sh flash: 

 

System CompactFlash directory:

File  Length   Name/status

  1   40804132  c3845-ipvoicek9-mz.124-11.XW1.bin  

  2   4627     admin_user.html  

  3   653758   admin_user.js  

  4   1029     CiscoLogo.gif  

  5   618      CME_GUI_README.TXT  

  6   953      Delete.gif  

  7   16344    dom.js  

  8   864      downarrow.gif  

  9   6146     ephone_admin.html  

 10   4658     logohome.gif  

 11   3724     normal_user.html  

 12   80033    normal_user.js  

 13   1347     Plus.gif  

 14   843      sxiconad.gif  

 15   174      Tab.gif  

 16   2410     telephony_service.html  

 17   870      uparrow.gif  

 18   9968     xml-test.html  

 19   3412     xml.template  

 20   18836    app-b-acd-2.1.2.2-ReadMe.txt  

 21   24985    app-b-acd-2.1.2.2.tcl  

 22   35485    app-b-acd-aa-2.1.2.2.tcl  

 23   75650    en_bacd_allagentsbusy.au  

 24   83291    en_bacd_disconnect.au  

 25   63055    en_bacd_enter_dest.au  

 26   37952    en_bacd_invalidoption.au  

 27   496521   en_bacd_music_on_hold.au  

 28   123446   en_bacd_options_menu.au  

 29   42978    en_bacd_welcome.au  

 30   34794    en_bacd_xferto_operator.au  

 31   2453315  apps41.8-2-2TR2.sbn  

 32   435144   cnu41.8-2-2TR2.sbn  

 33   2141080  cvm41sccp.8-2-2TR2.sbn  

 34   509329   dsp41.8-2-2TR2.sbn  

 35   229333   jar41sccp.8-2-2TR2.sbn  

 36   642      SCCP41.8-2-2SR2S.loads  

 37   642      term41.default.loads  

 38   642      term61.default.loads  

 39   496521   music-on-hold.au  

 40   341088   CP7912080002SCCP060817A.sbin  

 41   281506   ATA030100SCCP040211A.zup  

 42   916      speeddial.xml  

 43   61978    en_dt_prompt.au  

[49587856 bytes used, 14637420 available, 64225276 total]

62720K bytes of ATA System CompactFlash (Read/Write)

 

 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail is subject to the Telkom SA electronic communication legal notice, available at :
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From Sechaba.Mokoena at btgroup.co.za  Mon Oct  1 10:33:13 2007
From: Sechaba.Mokoena at btgroup.co.za (Sechaba Mokoena)
Date: Mon, 1 Oct 2007 16:33:13 +0200
Subject: [cisco-voip] CCME and External SIP provider
References: <472E195E692EE74BAE4E7B2880A78C1F01A91F87@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <A086A6B787B09E4B9110CA837AF892D00150D039@midms01.btg>

Hi Erik,

 

I managed to get this working,  it took an IOS upgrade and opening some
ports on the firewall, thanks for the help.

 

Best regards,

 

Sechaba Mokoena

Systems Engineer

Bytes Technology Group - Systems Integration


Tel : (+27) (11) 205-7000 |  Fax: (+27) (11) 205-7584  |  Cell : (+27)
84 798-6803 | msn:sechaba.mokoena at hotmail.com
Web  : www.btgroup.co.za  |  Press Office : www.itweb.co.za/office/bytes
<blocked::http://www.itweb.co.za/office/bytes>    |   Licensing :
www.purelicensing.co.za <blocked::http://www.purelicensing.co.za/>  

 

Bytes Systems Integration (Pty) Ltd : Registration No: 1995/012031/07
A subsidiary of Bytes Technology Group SA (Pty) Ltd,   In association
with KAGISO
P O Box 5905, Halfway House, 1685, Midrand;  Bytes Business Park, Block
B, 241 Third Road, Halfway Gardens, Midrand, South Africa 

E-Mail Disclaimer: http://www.altron.co.za/email.asp
<blocked::blocked::http://www.altron.co.za/email.asp>  or phone: (+27)
(11) 205-7000

 

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za] 
Sent: Wednesday, September 26, 2007 8:58 AM
To: Sechaba Mokoena
Subject: RE: [cisco-voip] CCME and External SIP provider

 

On viatalk they also mention this

 

 

How do I make audio work?

If you experience audio not working with your line on either outgoing or
incoming calls, it is because your ISP (or self purchased) supplied
modem/firewall/router is blocking various ports needed for our specific
service. 

First you'll need to find your router/modem/firewall's manual, or look
it up at the following link:
http://www.portforward.com/english/applications/port_forwarding/Viatalk/
Viatalkindex.htm

Next, login to your router. Forward the following port sets directly to
the ViaTalk supplied device:
Public/WAN PORT Private/LAN PORT
*8080 80
*69 69
*5060 - 5081 5060 - 5081
*10000 - 20000 10000 - 20000

The ports are for (in same order as above list):
*Remote Admin (make sure the VT supplied device has a admin password
set, otherwise leave this unforwarded until a VT tech sets you up on the
phone)
*TFTP (Our configuration servers use this)
*SIP (VOIP Data ports)
*SIP Audio (VOIP Audio ports for WAN communications)

If this does not solve your problem, please issue a ticket.

 

 

________________________________

From: Sechaba Mokoena [mailto:Sechaba.Mokoena at btgroup.co.za] 
Sent: Tuesday, September 25, 2007 4:24 PM
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CCME and External SIP provider

 

Hi Erik,

 

I've attached a copy of my config. I can place calls from the PSTN to
the IP Phone, but I have no audio, I attribute this to routing issues.
I'm still ironing out the small issues. 

 

Best regards,

 

Sechaba Mokoena

Systems Engineer

Bytes Technology Group - Systems Integration


Tel : (+27) (11) 205-7000 |  Fax: (+27) (11) 205-7584  |  Cell : (+27)
84 798-6803 | msn:sechaba.mokoena at hotmail.com
Web  : www.btgroup.co.za  |  Press Office : www.itweb.co.za/office/bytes
<blocked::http://www.itweb.co.za/office/bytes>    |   Licensing :
www.purelicensing.co.za <blocked::http://www.purelicensing.co.za/>  

 

Bytes Systems Integration (Pty) Ltd : Registration No: 1995/012031/07
A subsidiary of Bytes Technology Group SA (Pty) Ltd,   In association
with KAGISO
P O Box 5905, Halfway House, 1685, Midrand;  Bytes Business Park, Block
B, 241 Third Road, Halfway Gardens, Midrand, South Africa 

E-Mail Disclaimer: http://www.altron.co.za/email.asp
<blocked::blocked::http://www.altron.co.za/email.asp>  or phone: (+27)
(11) 205-7000

 

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za] 
Sent: Tuesday, September 25, 2007 4:00 PM
To: Sechaba Mokoena; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CCME and External SIP provider

 

What does your config look like on CCME

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sechaba Mokoena
Sent: Tuesday, September 25, 2007 3:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCME and External SIP provider

 

Hi,

 

Has anyone successfully setup CCME to use a SIP provider as a link to
the PSTN. If I look at registration status my number is registered to
the SIP proxy.

 

Line          peer           expires(sec)  registered 

============  =============  ============  ===========

0*            1                135                      yes         

2001        20004      134                      yes         

27877000089   20005            135     yes  

 

I cannot place calls to the PSTN, reason for failure is Last Disconnect
Text is "no route to destination (3)". I have IP connectivity to the SIP
server, I can ping the server's hostname. I'm not sure where to go from
here, any help will be greatly appreciated.

 

Kind regards,

 

Sechaba Mokoena

Systems Engineer

Bytes Technology Group - Systems Integration


Tel : (+27) (11) 205-7000 |  Fax: (+27) (11) 205-7584  |  Cell : (+27)
84 798-6803 | msn:sechaba.mokoena at hotmail.com
Web  : www.btgroup.co.za  |  Press Office : www.itweb.co.za/office/bytes
<blocked::http://www.itweb.co.za/office/bytes>    |   Licensing :
www.purelicensing.co.za <blocked::http://www.purelicensing.co.za/>  

 

Bytes Systems Integration (Pty) Ltd : Registration No: 1995/012031/07
A subsidiary of Bytes Technology Group SA (Pty) Ltd,   In association
with KAGISO
P O Box 5905, Halfway House, 1685, Midrand;  Bytes Business Park, Block
B, 241 Third Road, Halfway Gardens, Midrand, South Africa 

E-Mail Disclaimer: http://www.altron.co.za/email.asp
<blocked::blocked::http://www.altron.co.za/email.asp>  or phone: (+27)
(11) 205-7000

 


The information contained in this message and or attachments is intended
only for the person or entity to which it is addressed and may contain
confidential and/or privileged material.  Any review, retransmission,
dissemination or other use of, or taking of any action in reliance upon,
this information by persons or entities other than the intended
recipient is prohibited. If you received this in error, please contact
the sender and delete the material from any system and destroy any
copies.

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail is subject to the Telkom SA electronic communication legal
notice, available at :
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

 


The information contained in this message and or attachments is intended
only for the person or entity to which it is addressed and may contain
confidential and/or privileged material.  Any review, retransmission,
dissemination or other use of, or taking of any action in reliance upon,
this information by persons or entities other than the intended
recipient is prohibited. If you received this in error, please contact
the sender and delete the material from any system and destroy any
copies.


The information contained in this message and or attachments is intended
only for the person or entity to which it is addressed and may contain
confidential and/or privileged material.  Any review, retransmission,
dissemination or other use of, or taking of any action in reliance upon,
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From SIngram at clayton.com  Mon Oct  1 10:59:36 2007
From: SIngram at clayton.com (Scott Ingram)
Date: Mon, 1 Oct 2007 10:59:36 -0400
Subject: [cisco-voip] CCM_4.13 media kit CM4.1-K9-MIG0 failure to install.
Message-ID: <A9604D7A0239984EAD184D9A42206FDE01DA9134@EXCHVS01.cgi.clayton.com>

I have a HP proliant G4 380 duel Xeon 3.4 procs with 4 gig of memory
with 4 72.0 gig hard drives <HP smart array controller p600>.  When we
install the media 2000 2.7a <hardware detection cd> then disc II  media
2000 2.7 and the server reboots and crashs <inaccessible boot disc>.  We
have had Cisco and the VAR send us 2 media kits and both are doing the
same issue.  Has anyone ever seen this issue?


IMPORTANT NOTICE:

This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law.  If you have received this message in error, you are hereby notified that we do not consent to any reading, dissemination, distribution or copying of this message.  If you have received this communication in error, please notify the sender immediately and destroy the transmitted information.

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From dustin.s.fowler at gmail.com  Mon Oct  1 11:13:12 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Mon, 1 Oct 2007 11:13:12 -0400
Subject: [cisco-voip] Cisco DN Configuration
In-Reply-To: <A086A6B787B09E4B9110CA837AF892D00150D039@midms01.btg>
References: <472E195E692EE74BAE4E7B2880A78C1F01A91F87@CNTRRA20-XCS01.telkom.co.za>
	<A086A6B787B09E4B9110CA837AF892D00150D039@midms01.btg>
Message-ID: <47010e96.1187460a.6fbe.349e@mx.google.com>

Good Afternoon,

 

Has anyone seen this error while on a CCM 4.1(3)?

 

Automation server can't create object (-2146827859)

 

 

Thanks 

 

Dustin

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From wwmaples at cisco.com  Mon Oct  1 11:24:19 2007
From: wwmaples at cisco.com (Wayne Maples (wwmaples))
Date: Mon, 1 Oct 2007 11:24:19 -0400
Subject: [cisco-voip] Cisco DN Configuration
In-Reply-To: <47010e96.1187460a.6fbe.349e@mx.google.com>
References: <472E195E692EE74BAE4E7B2880A78C1F01A91F87@CNTRRA20-XCS01.telkom.co.za><A086A6B787B09E4B9110CA837AF892D00150D039@midms01.btg>
	<47010e96.1187460a.6fbe.349e@mx.google.com>
Message-ID: <2C7C3316358DC24C8A99DAAE1182E1D20406DA5E@xmb-rtp-212.amer.cisco.com>

Try this and see if it applies.
 
http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K13282137

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dustin S Fowler
Sent: Monday, October 01, 2007 11:13 AM
Cc: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco DN Configuration



Good Afternoon,

 

Has anyone seen this error while on a CCM 4.1(3)?

 

Automation server can't create object (-2146827859)

 

 

Thanks 

 

Dustin

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From jonvoip at gmail.com  Mon Oct  1 11:37:57 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 1 Oct 2007 10:37:57 -0500
Subject: [cisco-voip] TAPS question...
Message-ID: <5d093f9a0710010837g5832ff3cg117a408f857a1b32@mail.gmail.com>

I am doing a phone deployment with IPCC, and I want to use TAPS...

We have a pair of CCM4.3 7845s and a pair of 7845 IPCC 4.1 servers
(High Availability)...

Where should I install TAPS?

The Cisco documentation implies I should install another instance of
IPCC on the CCM Publisher (this seems dumb to me), but I have seen
posts implying problems installing it on the IPCC cluster...


Any ideas?



Jonathan

From CarlosOrtiz at bayviewfinancial.com  Mon Oct  1 11:44:04 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Mon, 1 Oct 2007 11:44:04 -0400
Subject: [cisco-voip] TAPS question...
In-Reply-To: <5d093f9a0710010837g5832ff3cg117a408f857a1b32@mail.gmail.com>
Message-ID: <OF2E0A9157.575C9671-ON85257367.00561EEF-85257367.00566EE6@bftg.com>

Jonathan,

It's been a while and it was on 4.1.3 but I think you need to install TAPS 
on both PUB and IPCC.  You DO NOT need to install an instance of IPCC on 
the Pub.  Read the docs again and make sure you are not mixing 
documentation for a coresident IPCC install.  It can be tricky getting 
TAPS to work properly but when you do I think it helps a lot.  Being able 
to drop phones anywhere instead of assigning to user via MAC works well in 
large deployments.....

Carlos



"Jonathan Charles" <jonvoip at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
10/01/2007 11:37 AM

To
"cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] TAPS question...






I am doing a phone deployment with IPCC, and I want to use TAPS...

We have a pair of CCM4.3 7845s and a pair of 7845 IPCC 4.1 servers
(High Availability)...

Where should I install TAPS?

The Cisco documentation implies I should install another instance of
IPCC on the CCM Publisher (this seems dumb to me), but I have seen
posts implying problems installing it on the IPCC cluster...


Any ideas?



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From dustin.s.fowler at gmail.com  Mon Oct  1 11:49:57 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Mon, 1 Oct 2007 11:49:57 -0400
Subject: [cisco-voip] Cisco DN Configuration
In-Reply-To: <2C7C3316358DC24C8A99DAAE1182E1D20406DA5E@xmb-rtp-212.amer.cisco.com>
References: <472E195E692EE74BAE4E7B2880A78C1F01A91F87@CNTRRA20-XCS01.telkom.co.za><A086A6B787B09E4B9110CA837AF892D00150D039@midms01.btg>
	<47010e96.1187460a.6fbe.349e@mx.google.com>
	<2C7C3316358DC24C8A99DAAE1182E1D20406DA5E@xmb-rtp-212.amer.cisco.com>
Message-ID: <47011732.033a360a.5529.4893@mx.google.com>

That fixed it. Thank You very much.

 

From: Wayne Maples (wwmaples) [mailto:wwmaples at cisco.com] 
Sent: Monday, October 01, 2007 11:24 AM
To: Dustin S Fowler
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Cisco DN Configuration

 

Try this and see if it applies.

 

http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K13282137

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dustin S Fowler
Sent: Monday, October 01, 2007 11:13 AM
Cc: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco DN Configuration

Good Afternoon,

 

Has anyone seen this error while on a CCM 4.1(3)?

 

Automation server can't create object (-2146827859)

 

 

Thanks 

 

Dustin

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From sgrana at urudata.com  Mon Oct  1 12:06:44 2007
From: sgrana at urudata.com (=?iso-8859-1?Q?Sebasti=E1n_Gra=F1a?=)
Date: Mon, 1 Oct 2007 13:06:44 -0300
Subject: [cisco-voip] RTMT on CCM5.1
Message-ID: <77B19B7818C8DF43BA6BE948BFE66550333B2D@urusrvexc.urudata.com.uy>

Hi,
 
    I'm trying to configure alert email sending with this tool but it is not working.
    This is what I've done so far:
 
    In RTMT, Tools-Alert-Config Mail Server. There I put the IP address of the SMTP server I want to use, and the port.
    In Tools-Alert-Config Alert Action, in Default added email recipients.
 
    I have no activity in the SMTP server form any of the call managers.
 
    Has anyone seen this problem?
 
Thank you
 
Best Regards 
 
Sebasti?n Gra?a
MCSE - CCNA 
Departamento T?cnico
Urudata Infraestructura
Canelones 1611 
Montevideo / Uruguay 
Tel.: (+598 2) 419 6457
 
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From alessandro.bertacco at lanservice.it  Mon Oct  1 11:38:31 2007
From: alessandro.bertacco at lanservice.it (Alessandro Bertacco)
Date: Mon, 1 Oct 2007 17:38:31 +0200
Subject: [cisco-voip] CUCM 6.0.(1) Installation Problem
Message-ID: <BABE7297CCC23245A867092258C59507DFC558@LS-MAIL.lanservice.local>

Hi every one.

Have you ever had a problem during the CUCM 6.0.(1) DVD installation? The problem is that.

After the server DL 320 G2 boot up from the DVD drive of the CUCM 6.0.(1) System Disk, the system ask me to configure all requested parameter, Ip address, NTP, ecc...the installation should begin, but, Xserver windows appear and I see a Red Hat Enterprise installation ask me the partition configuration of the HardDrive, after that the installation show me a unrecoverable error and stop.

Any idea???

Thank you very much



Alessandro Bertacco
Network Engineer
alessandro.bertacco at lanservice.it  




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From Edward.Countryman at provena.org  Mon Oct  1 12:21:00 2007
From: Edward.Countryman at provena.org (Countryman, Edward)
Date: Mon, 1 Oct 2007 11:21:00 -0500
Subject: [cisco-voip] FXO Port for overhead paging
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006EFA9DF@USNAEXCH.na.didata.local>
Message-ID: <898237D7FE163A40A3257E385562F27E0160B23E@AEXMV02.phnet.phroot.local>

I actually was contemplating ordering the E&M card because of the CCO
article.

 

Thank you for the feedback.

 

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com] 
Sent: Sunday, September 30, 2007 10:00 PM
To: Ted Nugent; Lelio Fulgenzi
Cc: Countryman, Edward; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] FXO Port for overhead paging

 

The V9970 with 2FXS is the unit I commonly see with problems. About once
a month in customer X 40 warehouses one acts up and you have to shut/no
shut the fxs interface or reset the 9970. Valcom and Cisco tech support
are pretty clear FXS has limited disconnect supervision, and FXS
treatment varies by product (CME vs CCM).

 

Thus I stick with FXO/Valcom 2003A.

 

Whoever wrote the CCO article on E&M I don't think had enough real-world
experience with paging. When I walk into a customer using E&M for
paging,  I ask why E&M, they state they found the article on CCO and
ordered the E&M card.  

 

From: Ted Nugent [mailto:tednugent73 at gmail.com] 
Sent: Sunday, September 30, 2007 10:36 PM
To: Lelio Fulgenzi
Cc: Jason Aarons (US); Countryman, Edward; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] FXO Port for overhead paging

 

 

 

I've personally had zero issues with fxs (I typically push for the
valcom V9970, cheap easy) and only had one pain in the ass with fxo and
that was probably a disconnect between the customer and I and not an
issue with the adapter (I typically push for the valcom V2000 or better
for FXO). I've done a bunch of both fxs and fxo and not had any notable
issues with either once you determine what they have or even better if I
had the luxury of a recommendation. Once you determine what you're
interfacing with (fxo,fxs, loop ground etc...) the configuration is
pretty much self explanatory or has been for me. 

 

 



 

On 9/26/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote: 

is this with the use of an adapter? the adapter has all these dials and
knobs that protect agains that sort of stuff. 

although a 130db screech would be interesting to see in our animal
barns.

----- Original Message -----
From: "Jason Aarons (US)" <jason.aarons at us.didata.com >
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>; "Countryman, Edward"
<Edward.Countryman at provena.org >; <cisco-voip at puck.nether.net>
Sent: Wednesday, September 26, 2007 2:58 PM
Subject: RE: [cisco-voip] FXO Port for overhead paging


>I would disagree, FXS has limited supervision disconnect, stick with a 
> FXO as best practice. I've seen FXS interfaces get stuck offhook and
the
> paging units screech 130db noise....also ATA behaves differently
between
> CCM and CME, FXO doesn't.
>
> -jason 
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Lelio
Fulgenzi
> Sent: Wednesday, September 26, 2007 2:54 PM
> To: Countryman, Edward; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] FXO Port for overhead paging 
>
> save yourself the trouble....get an adapter and use an FXS port. ;)
>
> we use the Bogen TAM-B.
>
> ----- Original Message -----
> From: "Countryman, Edward" < Edward.Countryman at provena.org
<mailto:Edward.Countryman at provena.org> >
> To: <cisco-voip at puck.nether.net>
> Sent: Wednesday, September 26, 2007 2:41 PM
> Subject: [cisco-voip] FXO Port for overhead paging 
>
>
>> Has anyone successfully configured and used an FXO port to access an
>> overhead paging system?  All I can seem to find are E&M examples.
>> _______________________________________________ 
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck..nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck..nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the 
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you. 
>
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck..nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
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you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
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From jason.aarons at us.didata.com  Mon Oct  1 12:43:07 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 1 Oct 2007 12:43:07 -0400
Subject: [cisco-voip] FXO Port for overhead paging
In-Reply-To: <898237D7FE163A40A3257E385562F27E0160B23E@AEXMV02.phnet.phroot.local>
References: <C1FE15183DA37645BC0633BC604E44F006EFA9DF@USNAEXCH.na.didata.local>
	<898237D7FE163A40A3257E385562F27E0160B23E@AEXMV02.phnet.phroot.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006EFB086@USNAEXCH.na.didata.local>

Once you decide on the Paging vendor, call their technical support and
ask them how they would integrate with your phone system -jason

 

From: Countryman, Edward [mailto:Edward.Countryman at provena.org] 
Sent: Monday, October 01, 2007 12:21 PM
To: Jason Aarons (US); Ted Nugent; Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] FXO Port for overhead paging

 

I actually was contemplating ordering the E&M card because of the CCO
article.

 

Thank you for the feedback.

 

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com] 
Sent: Sunday, September 30, 2007 10:00 PM
To: Ted Nugent; Lelio Fulgenzi
Cc: Countryman, Edward; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] FXO Port for overhead paging

 

The V9970 with 2FXS is the unit I commonly see with problems. About once
a month in customer X 40 warehouses one acts up and you have to shut/no
shut the fxs interface or reset the 9970. Valcom and Cisco tech support
are pretty clear FXS has limited disconnect supervision, and FXS
treatment varies by product (CME vs CCM).

 

Thus I stick with FXO/Valcom 2003A.

 

Whoever wrote the CCO article on E&M I don't think had enough real-world
experience with paging. When I walk into a customer using E&M for
paging,  I ask why E&M, they state they found the article on CCO and
ordered the E&M card.  

 

From: Ted Nugent [mailto:tednugent73 at gmail.com] 
Sent: Sunday, September 30, 2007 10:36 PM
To: Lelio Fulgenzi
Cc: Jason Aarons (US); Countryman, Edward; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] FXO Port for overhead paging

 

 

 

I've personally had zero issues with fxs (I typically push for the
valcom V9970, cheap easy) and only had one pain in the ass with fxo and
that was probably a disconnect between the customer and I and not an
issue with the adapter (I typically push for the valcom V2000 or better
for FXO). I've done a bunch of both fxs and fxo and not had any notable
issues with either once you determine what they have or even better if I
had the luxury of a recommendation. Once you determine what you're
interfacing with (fxo,fxs, loop ground etc...) the configuration is
pretty much self explanatory or has been for me. 

 

 



 

On 9/26/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote: 

is this with the use of an adapter? the adapter has all these dials and
knobs that protect agains that sort of stuff. 

although a 130db screech would be interesting to see in our animal
barns.

----- Original Message -----
From: "Jason Aarons (US)" <jason.aarons at us.didata.com >
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>; "Countryman, Edward"
<Edward.Countryman at provena.org >; <cisco-voip at puck.nether.net>
Sent: Wednesday, September 26, 2007 2:58 PM
Subject: RE: [cisco-voip] FXO Port for overhead paging


>I would disagree, FXS has limited supervision disconnect, stick with a 
> FXO as best practice. I've seen FXS interfaces get stuck offhook and
the
> paging units screech 130db noise....also ATA behaves differently
between
> CCM and CME, FXO doesn't.
>
> -jason 
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Lelio
Fulgenzi
> Sent: Wednesday, September 26, 2007 2:54 PM
> To: Countryman, Edward; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] FXO Port for overhead paging 
>
> save yourself the trouble....get an adapter and use an FXS port. ;)
>
> we use the Bogen TAM-B.
>
> ----- Original Message -----
> From: "Countryman, Edward" < Edward.Countryman at provena.org
<mailto:Edward.Countryman at provena.org> >
> To: <cisco-voip at puck.nether.net>
> Sent: Wednesday, September 26, 2007 2:41 PM
> Subject: [cisco-voip] FXO Port for overhead paging 
>
>
>> Has anyone successfully configured and used an FXO port to access an
>> overhead paging system?  All I can seem to find are E&M examples.
>> _______________________________________________ 
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck..nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck..nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the 
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you. 
>
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck..nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From Jean-Victor.Brun at ins.com  Mon Oct  1 13:55:06 2007
From: Jean-Victor.Brun at ins.com (Jean-Victor.Brun at ins.com)
Date: Mon, 1 Oct 2007 12:55:06 -0500
Subject: [cisco-voip] IPCC Question
In-Reply-To: <A56BD242744BF540B15D1672E0E3E87501E57686@LON-INT-MSGVS01.internal.vancocorp.com>
References: <94a1afde0702070516y6fcfab63web3d6f5f0d1a3c1c@mail.gmail.com>
	<A56BD242744BF540B15D1672E0E3E87501E57686@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <A3C7243F8807F440A996B5556A8AD87201026228@usemp05.ins.com>

Here you go.  Hope that is what you are looking for.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johnny Crothers
(c)
Sent: Friday, September 28, 2007 3:16 AM
To: Ed Leatherman; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC Question

 

I'm trying to find this Holiday script, has anyone got a copy of the
.aef? I can use the below to create the XML.

 

Thanks all!! t

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: 07 February 2007 13:16
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Question

 

Hi,

We're using the HolidayDoc script out of the IPCC repository for an ICD
script. Pretty much following it verbatim, but script isnt working. When
i do a reactive debug on the script, when I get to a "Get XML Document
Data" step, its coming back with the following error: 
"invalid byte 1 of 1-byte UTF-8 sequence (0x94)"
Anyone see that before? my xml file _looks correct based on the sample
give in the repository. i'm sure i'm missing something obvious here but
can't figure out what. 

<?xml version="1.0" encoding="ISO-8859-1"?>
<!-- NO LEADING ZEROES IN DATE NUMBERS !!!!-->
<Holidays>
<holidayDate1>4/6/2007</holidayDate1>
<Holidays>
-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 



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From wsisk at cisco.com  Mon Oct  1 17:19:06 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 01 Oct 2007 17:19:06 -0400
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <77B19B7818C8DF43BA6BE948BFE66550333B2D@urusrvexc.urudata.com.uy>
References: <77B19B7818C8DF43BA6BE948BFE66550333B2D@urusrvexc.urudata.com.uy>
Message-ID: <4701644A.8050700@cisco.com>

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From wsisk at cisco.com  Mon Oct  1 17:21:18 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 01 Oct 2007 17:21:18 -0400
Subject: [cisco-voip] CUCM 6.0.(1) Installation Problem
In-Reply-To: <BABE7297CCC23245A867092258C59507DFC558@LS-MAIL.lanservice.local>
References: <BABE7297CCC23245A867092258C59507DFC558@LS-MAIL.lanservice.local>
Message-ID: <470164CE.7090202@cisco.com>

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From jonvoip at gmail.com  Mon Oct  1 20:54:22 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 1 Oct 2007 19:54:22 -0500
Subject: [cisco-voip] CCM_4.13 media kit CM4.1-K9-MIG0 failure to
	install.
In-Reply-To: <A9604D7A0239984EAD184D9A42206FDE01DA9134@EXCHVS01.cgi.clayton.com>
References: <A9604D7A0239984EAD184D9A42206FDE01DA9134@EXCHVS01.cgi.clayton.com>
Message-ID: <5d093f9a0710011754k19881be3j4b4fcf625bd94fc5@mail.gmail.com>

I know this is going to sound like a pointless waste of time (good web
site BTW), try installing just Windows 2000 or 2003 (using a 2000
install install disc)...

Make sure there is nothing wrong with your drives, controller, CPU, RAM, etc...



Jonathan

On 10/1/07, Scott Ingram <SIngram at clayton.com> wrote:
>
>
>
> I have a HP proliant G4 380 duel Xeon 3.4 procs with 4 gig of memory with 4
> 72.0 gig hard drives <HP smart array controller p600>.  When we install the
> media 2000 2.7a <hardware detection cd> then disc II  media 2000 2.7 and the
> server reboots and crashs <inaccessible boot disc>.  We have had Cisco and
> the VAR send us 2 media kits and both are doing the same issue.  Has anyone
> ever seen this issue?
>
>  IMPORTANT NOTICE:
>
> This message is intended only for the use of the individual or entity to
> which it is addressed and may contain information that is privileged,
> confidential and exempt from disclosure under applicable law. If you have
> received this message in error, you are hereby notified that we do not
> consent to any reading, dissemination, distribution or copying of this
> message. If you have received this communication in error, please notify the
> sender immediately and destroy the transmitted information.
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From alessandro.bertacco at lanservice.it  Tue Oct  2 02:40:42 2007
From: alessandro.bertacco at lanservice.it (Alessandro Bertacco)
Date: Tue, 2 Oct 2007 08:40:42 +0200
Subject: [cisco-voip] R:  CUCM 6.0.(1) Installation Problem
In-Reply-To: <470164CE.7090202@cisco.com>
References: <BABE7297CCC23245A867092258C59507DFC558@LS-MAIL.lanservice.local>
	<470164CE.7090202@cisco.com>
Message-ID: <BABE7297CCC23245A867092258C59507DFC563@LS-MAIL.lanservice.local>

Hi Wes,
 
So you think it will be an hardware misconfiguration problem?
 
Ok, i'll try with a server that match exatly the supported config.
 
Thank you very much
 
Alessandro

________________________________

Da: Wes Sisk [mailto:wsisk at cisco.com] 
Inviato: luned? 1 ottobre 2007 23.21
A: Alessandro Bertacco
Cc: cisco-voip at puck.nether.net
Oggetto: Re: [cisco-voip] CUCM 6.0.(1) Installation Problem


verify your hardware matches the specified hardware buildout exactly.
http://www.cisco.com/en/US/products/hw/voiceapp/ps378/prod_brochure_list.html

/Wes

Alessandro Bertacco wrote: 

	Hi every one. 

	Have you ever had a problem during the CUCM 6.0.(1) DVD installation? The problem is that. 

	After the server DL 320 G2 boot up from the DVD drive of the CUCM 6.0.(1) System Disk, the system ask me to configure all requested parameter, Ip address, NTP, ecc...the installation should begin, but, Xserver windows appear and I see a Red Hat Enterprise installation ask me the partition configuration of the HardDrive, after that the installation show me a unrecoverable error and stop.

	Any idea??? 

	Thank you very much 



	Alessandro Bertacco
	Network Engineer 
	alessandro.bertacco at lanservice.it <mailto:alessandro.bertacco at lanservice.it>  
	



	
________________________________


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From aelnagar at ACT-EG.COM  Tue Oct  2 04:15:55 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 2 Oct 2007 10:15:55 +0200
Subject: [cisco-voip] Webdialer+Outlook+Forced Authorization Code
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED01380E68@mail.ACT-EG.COM>

Yes I had the same problem with attendant console and she had to dial the FAC from its telephone pad.

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Wednesday, September 26, 2007 8:36 PM
To: Cucumber Green
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Webdialer+Outlook+Forced Authorization Code

 

generally will not be possible as FAC/CMC was not implemented for CTI in initial implementation.

even after it was implemented it is a separate CTI message/call.  Outlook would have to make specific system calls to invoke fac/cmc feature

You can see info on the FAC specific messages here:
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_programming_usage_guide_chapter09186a00802e2aa0.html

CTIERR_FAC_CMC_REASON_CMC_NEEDED 

Client Matter Code (CMC) is required to offer the call 


/Wes


Cucumber Green wrote: 

Currently have Webdialer + Outlook working, but when users try to dial telephone numbers that require FAC they need to enter the FAC code on the IP telephone pad, 

 

How can I send the FAC from Outlook contact?s phone number?

 

I have tried adding some commas after the telephone number, then the FAC, but outlook (2003) says the phone number has invalid digits... :(

 

 

________________________________

Yahoo! oneSearch: Finally, mobile search that gives answers <http://us.rd.yahoo.com/evt=48252/*http:/mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC> , not web links. 

 



________________________________



 
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From Nick.Kassel at Charles-Stanley.co.uk  Tue Oct  2 04:20:05 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 2 Oct 2007 09:20:05 +0100
Subject: [cisco-voip] RTMT on CCM5.1
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC03B482D8@CSEXCH01.charles-stanley.co.uk>

I never got this working either, Wes could you provide some more details on those bug id's as they only available to Cisco employees on CCO.

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: 01 October 2007 22:19
To: Sebasti?n Gra?a
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RTMT on CCM5.1

 

Watch out for CSCsf08244 and CSCsh33877.

/Wes

Sebasti?n Gra?a wrote: 

Hi,

 

    I'm trying to configure alert email sending with this tool but it is not working.

    This is what I've done so far:

 

    In RTMT, Tools-Alert-Config Mail Server. There I put the IP address of the SMTP server I want to use, and the port.

    In Tools-Alert-Config Alert Action, in Default added email recipients.

 

    I have no activity in the SMTP server form any of the call managers.

 

    Has anyone seen this problem?

 

Thank you

 

Best Regards 

 

Sebasti?n Gra?a

MCSE - CCNA 

Departamento T?cnico

Urudata Infraestructura
Canelones 1611 
Montevideo / Uruguay 
Tel.: (+598 2) 419 6457

 

 



________________________________



 
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https://puck.nether.net/mailman/listinfo/cisco-voip
********************SAVE PAPER - THINK BEFORE YOU PRINT**************************************************

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From aelnagar at ACT-EG.COM  Tue Oct  2 04:46:35 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 2 Oct 2007 10:46:35 +0200
Subject: [cisco-voip] how to allow users to enable/disable Voicemail
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED01380EA9@mail.ACT-EG.COM>

Hello;

 

I think you can leave there phone without any fwd to voice mail, when they want to fwd there calls to voicemail they can fwdall (from the IP Phone) to the voicemail pilot number (but this will transfer all calls). Or you can use the IP Forwarder and forward to the voice mail pilot number (when busy when no answer as he/she wish).

It just will not be nice that your end users know your voicemail pilot number.

Try it and give us a feedback

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Michael Hamann
Sent: Thursday, September 20, 2007 5:47 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to allow users to enable/disable Voicemail

 

Hello,

 

is it possible to allow the end user to enable or disable his voicemail

somehow? We are using CCM 4.2 with Asterisk as Voicemail System. At the

moment we are configuring the voicemail on the users extension and I don?t

see a chance to allow the user itself to switch on/off his voicemailbox.

 

I found the IP Forwarder IP Service from my myvoipsource.org which allows

the users to change his forwarding settings. But as I can see I can?t

change the forward settings to the voicemail...

 

Are there any other ways? Any hints?

 

thank you

Michael

 

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From aelnagar at ACT-EG.COM  Tue Oct  2 05:02:53 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 2 Oct 2007 11:02:53 +0200
Subject: [cisco-voip] 3570 POE ports
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED01380ECE@mail.ACT-EG.COM>

You can calculate it using the power calculator in Cisco Tools; I
remember that you cannot fill up this switch with all ports connected to
7970 IP Phones (it will exceed the maximum power supply output); correct
me if I am wrong guys.

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Wednesday, September 19, 2007 4:54 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 3570 POE ports

 

Does anyone know the maximum amount of 7970 phones that can be used for
POE on 48 port 3750 switches?

 


********************SAVE PAPER - THINK BEFORE YOU
PRINT**************************************************

The information contained in this e-mail is strictly confidential, some
or all of which may be legally privileged. 
Access to this e-mail by any person other than the recipient is
prohibited. If you have received this message in
error, any use, disclosure, copying, printing, distribution of, replying
to or any action taken or omitted to be
taken in reliance on this e-mail, is prohibited. Please advise the
sender immediately should this e-mail have been 
incorrectly addressed or transmitted, and then delete the email and any
attachment sent with it from your computer.

You are advised that urgent, time sensitive and confidential
communications should not be sent by e-mail. You 
accept that any instructions are deemed to have been given at the time
the recipient(s) accesses them and that 
delivery receipt does not constitute acknowledgement or receipt by the
intended recipient(s). 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised 
interference, mis-delivery or delay. Charles Stanley reserves the right
to monitor and/or record emails sent and
received via its network for any lawful business purpose in accordance
with applicable law and regulations.
************************************************************************
*******************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
of Charles Stanley & Co. Ltd  

Authorised and Regulated by the Financial Services Authority, Member of
the London Stock Exchange, The 
International Capital Market Association and The London International
Financial Futures & Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee VirusScan and SurfControl Email 
Filter software. 

 

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From aelnagar at ACT-EG.COM  Tue Oct  2 05:13:24 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 2 Oct 2007 11:13:24 +0200
Subject: [cisco-voip] Disaster Recovery: Using ghost to restore MCS
	Servers maybe?
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED01380EDF@mail.ACT-EG.COM>

AFAIK it is not supported with TAC, but have you tried it several times?

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
Sent: Tuesday, September 18, 2007 12:37 PM
To: Cucumber Green
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Disaster Recovery: Using ghost to restore MCS
Servers maybe?

 

Ghost definitely works, however it's not worthwhile to make frequent 

ghosts. So (just as others have put it) make an image after major 

changes and use BARS in between.

 

Major changes being software updates, major component changes in the 

system (new gateways, VM, etc.)

 

regards,

  Zoltan

 

Cucumber Green wrote:

> Hello

>  

> Has anyone used Ghost to keep an image of your MCS servers in order to


> use it in case of a server crash? If so, did it work, are there any 

> caveats, tips?

>  

> If not using Ghost, whats your disaster recovery procedure to restore 

> an MCS crashed server ( other than a full reinstall ).

>  

>  

>  

> 

>
------------------------------------------------------------------------

> Boardwalk for $500? In 2007? Ha!

> Play Monopoly Here and Now 

>
<http://us.rd.yahoo.com/evt=48223/*http://get.games.yahoo.com/proddesc?g
amekey=monopolyherenow> 

> (it's updated for today's economy) at Yahoo! Games.

>
------------------------------------------------------------------------

> 

> _______________________________________________

> cisco-voip mailing list

> cisco-voip at puck.nether.net

> https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From keli at carocomp.ro  Tue Oct  2 05:30:10 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Tue, 02 Oct 2007 12:30:10 +0300
Subject: [cisco-voip] AAR vs. Route Lists
Message-ID: <47020FA2.1090105@carocomp.ro>

Hi,

this may be a noob question to some of you, but there's something I 
don't quite understand:
 don't Route Lists provide the same (or rather - similar) functionality 
as AAR? If not (and I suppose not) what's the difference?

Case in point:
  We have a client with three CCM clusters, all connected through ICTs. 
We haven't used AAR so far, but the calls are routed through Route 
Lists: the trunk is the first route group, but if that fails, calls are 
redirected to the PSTN route group with appropriate transform masks for 
called and calling numbers.

If the IP trunk is congested, shouldn't the calls be rerouted to the 
PSTN anyway? (even though AAR is not set)

thanks,
  Zoltan

From aelnagar at ACT-EG.COM  Tue Oct  2 05:47:23 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 2 Oct 2007 11:47:23 +0200
Subject: [cisco-voip] ASA Configuration
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED01380F4B@mail.ACT-EG.COM>

You can use a configuration called (as I remember) NAT zero, it can help
you when you don't need to NAT on the firewall while you still need to
pass traffic from it. But really no need to but two firewalls behind
each others (unless you are using different vendors)

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Lyon
Sent: Friday, September 14, 2007 12:07 AM
To: Kevin Dunn
Cc: cisco voip
Subject: Re: [cisco-voip] ASA Configuration

 

Why are you double firewalling it?

 

-Mike

 

 

On 9/13/07, Kevin Dunn <kevinddunn at gmail.com> wrote:

> We currently have a Firewall in place with a public address.

> I need to place an ASA between the Internet router and the original

> firewall.

> 

> But those devices are on the same subnet and I can't have the outside
and

> inside interfaces on the same subnet in the ASA.

> 

> Can I get around that without double NATing the incoming packets?

> 

> ** I know this isn't a VOIP question, but we are placing the ASA to
allow

> VOIP traffic...sorry guys just looking for clues**

> 

> Kevin D. Dunn

> Network Engineer

> Marathon Cheese

> _______________________________________________

> cisco-voip mailing list

> cisco-voip at puck.nether.net

> https://puck.nether.net/mailman/listinfo/cisco-voip

> 

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From keli at carocomp.ro  Tue Oct  2 06:26:19 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Tue, 02 Oct 2007 13:26:19 +0300
Subject: [cisco-voip] Disaster Recovery: Using ghost to restore MCS
 Servers maybe?
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED01380EDF@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED01380EDF@mail.ACT-EG.COM>
Message-ID: <47021CCB.8060301@carocomp.ro>

We have used ghost with 3.x and 4.x servers,several times, successfully. 
It worked. (then again, why shouldn't it?)

However, try to keep the same (or similar) hardware underneath, as you 
*might* have problems with different hardware setups between source and 
destination. We managed to rewrite a ghost to a different make (and thus 
slightly different sized) HDD on a 7815 (since no appropriate spare was 
on hand when the original crashed and it was urgent), but it might not 
work out always as expected.

regards,
  Zoltan

Ahmed Elnagar wrote:
>
> AFAIK it is not supported with TAC, but have you tried it several times?
>
>  
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
> Sent: Tuesday, September 18, 2007 12:37 PM
> To: Cucumber Green
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Disaster Recovery: Using ghost to restore 
> MCS Servers maybe?
>
>  
>
> Ghost definitely works, however it's not worthwhile to make frequent
>
> ghosts. So (just as others have put it) make an image after major
>
> changes and use BARS in between.
>
>  
>
> Major changes being software updates, major component changes in the
>
> system (new gateways, VM, etc.)
>
>  
>
> regards,
>
>   Zoltan
>
>  
>
> Cucumber Green wrote:
>
> > Hello
>
> > 
>
> > Has anyone used Ghost to keep an image of your MCS servers in order to
>
> > use it in case of a server crash? If so, did it work, are there any
>
> > caveats, tips?
>
> > 
>
> > If not using Ghost, whats your disaster recovery procedure to restore
>
> > an MCS crashed server ( other than a full reinstall ).
>
> > 
>
> > 
>
> > 
>
> > 
>
> > ------------------------------------------------------------------------
>
> > Boardwalk for $500? In 2007? Ha!
>
> > Play Monopoly Here and Now
>
> > <http://us.rd.yahoo.com/evt=48223/*http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow> 
>
>
> > (it's updated for today's economy) at Yahoo! Games.
>
> > ------------------------------------------------------------------------
>
> > 
>
> > _______________________________________________
>
> > cisco-voip mailing list
>
> > cisco-voip at puck.nether.net
>
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>  
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From aelnagar at ACT-EG.COM  Tue Oct  2 07:16:01 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 2 Oct 2007 13:16:01 +0200
Subject: [cisco-voip] AAR vs. Route Lists
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED013810D0@mail.ACT-EG.COM>

Hello;

 

I never use the AAR feature before, but something came to my mind. Maybe
the AAR can achieve the following (that RL cannot):

 

When the call is rejected due to not enough bandwidth on the WAN link
don't send the call through the PSTN.

When the call is rejected due to WAN failure send the call through the
PSTN.

 

I am not sure of it, I am just assuming.

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
Sent: Tuesday, October 02, 2007 11:30 AM
To: Cisco VoIP
Subject: [cisco-voip] AAR vs. Route Lists

 

Hi,

 

this may be a noob question to some of you, but there's something I 

don't quite understand:

 don't Route Lists provide the same (or rather - similar) functionality 

as AAR? If not (and I suppose not) what's the difference?

 

Case in point:

  We have a client with three CCM clusters, all connected through ICTs. 

We haven't used AAR so far, but the calls are routed through Route 

Lists: the trunk is the first route group, but if that fails, calls are 

redirected to the PSTN route group with appropriate transform masks for 

called and calling numbers.

 

If the IP trunk is congested, shouldn't the calls be rerouted to the 

PSTN anyway? (even though AAR is not set)

 

thanks,

  Zoltan

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

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From jsteinberg at gmail.com  Tue Oct  2 08:39:31 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 2 Oct 2007 08:39:31 -0400
Subject: [cisco-voip] AAR vs. Route Lists
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED013810D0@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED013810D0@mail.ACT-EG.COM>
Message-ID: <f1b65f880710020539x633243adyc6292db809c64b23@mail.gmail.com>

AAR works in conjunction with CCM Locations bandwidth.  Think two IP
phones located at two remote sites that attempt to dial each other and
exceed the Locations bandwidth.  AAR is invoked and the call is
rerouted out a local PSTN gateway.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803edabe.html#wp1056007

On 10/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>
>
>
> Hello;
>
>
>
> I never use the AAR feature before, but something came to my mind. Maybe the
> AAR can achieve the following (that RL cannot):
>
>
>
> When the call is rejected due to not enough bandwidth on the WAN link don't
> send the call through the PSTN.
>
> When the call is rejected due to WAN failure send the call through the PSTN.
>
>
>
> I am not sure of it, I am just assuming.
>
>
>
>
> -----Original Message-----
>  From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Kelemen Zoltan
>  Sent: Tuesday, October 02, 2007 11:30 AM
>  To: Cisco VoIP
>  Subject: [cisco-voip] AAR vs. Route Lists
>
>
>
> Hi,
>
>
>
> this may be a noob question to some of you, but there's something I
>
> don't quite understand:
>
>  don't Route Lists provide the same (or rather - similar) functionality
>
> as AAR? If not (and I suppose not) what's the difference?
>
>
>
> Case in point:
>
>   We have a client with three CCM clusters, all connected through ICTs.
>
> We haven't used AAR so far, but the calls are routed through Route
>
> Lists: the trunk is the first route group, but if that fails, calls are
>
> redirected to the PSTN route group with appropriate transform masks for
>
> called and calling numbers.
>
>
>
> If the IP trunk is congested, shouldn't the calls be rerouted to the
>
> PSTN anyway? (even though AAR is not set)
>
>
>
> thanks,
>
>   Zoltan
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From aelnagar at ACT-EG.COM  Tue Oct  2 08:41:53 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 2 Oct 2007 14:41:53 +0200
Subject: [cisco-voip] AAR vs. Route Lists
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED01381200@mail.ACT-EG.COM>

Yes, then I am wrong

-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Tuesday, October 02, 2007 2:40 PM
To: Ahmed Elnagar
Cc: Kelemen Zoltan; Cisco VoIP
Subject: Re: [cisco-voip] AAR vs. Route Lists

AAR works in conjunction with CCM Locations bandwidth.  Think two IP
phones located at two remote sites that attempt to dial each other and
exceed the Locations bandwidth.  AAR is invoked and the call is
rerouted out a local PSTN gateway.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administra
tion_guide_chapter09186a00803edabe.html#wp1056007

On 10/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>
>
>
> Hello;
>
>
>
> I never use the AAR feature before, but something came to my mind.
Maybe the
> AAR can achieve the following (that RL cannot):
>
>
>
> When the call is rejected due to not enough bandwidth on the WAN link
don't
> send the call through the PSTN.
>
> When the call is rejected due to WAN failure send the call through the
PSTN.
>
>
>
> I am not sure of it, I am just assuming.
>
>
>
>
> -----Original Message-----
>  From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Kelemen Zoltan
>  Sent: Tuesday, October 02, 2007 11:30 AM
>  To: Cisco VoIP
>  Subject: [cisco-voip] AAR vs. Route Lists
>
>
>
> Hi,
>
>
>
> this may be a noob question to some of you, but there's something I
>
> don't quite understand:
>
>  don't Route Lists provide the same (or rather - similar)
functionality
>
> as AAR? If not (and I suppose not) what's the difference?
>
>
>
> Case in point:
>
>   We have a client with three CCM clusters, all connected through
ICTs.
>
> We haven't used AAR so far, but the calls are routed through Route
>
> Lists: the trunk is the first route group, but if that fails, calls
are
>
> redirected to the PSTN route group with appropriate transform masks
for
>
> called and calling numbers.
>
>
>
> If the IP trunk is congested, shouldn't the calls be rerouted to the
>
> PSTN anyway? (even though AAR is not set)
>
>
>
> thanks,
>
>   Zoltan
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From keli at carocomp.ro  Tue Oct  2 08:38:32 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Tue, 02 Oct 2007 15:38:32 +0300
Subject: [cisco-voip] AAR vs. Route Lists
In-Reply-To: <f1b65f880710020539x633243adyc6292db809c64b23@mail.gmail.com>
References: <88C8D019DA547C4BABECB9A806ED22ED013810D0@mail.ACT-EG.COM>
	<f1b65f880710020539x633243adyc6292db809c64b23@mail.gmail.com>
Message-ID: <47023BC8.6020609@carocomp.ro>

So a RL won't fall back to the second route group if the network is 
congested (that basically should send a busy as well, not?)?

Example:
RouteList IP_PSTN
 RG1: IPTrunk
 RG2: PSTTrunk

if RG1 is congested (Locations bandwidth exceeded) it won't go over to RG2?

Zoltan

Justin Steinberg wrote:
> AAR works in conjunction with CCM Locations bandwidth.  Think two IP
> phones located at two remote sites that attempt to dial each other and
> exceed the Locations bandwidth.  AAR is invoked and the call is
> rerouted out a local PSTN gateway.
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803edabe.html#wp1056007
>
> On 10/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>   
>>
>>
>> Hello;
>>
>>
>>
>> I never use the AAR feature before, but something came to my mind. Maybe the
>> AAR can achieve the following (that RL cannot):
>>
>>
>>
>> When the call is rejected due to not enough bandwidth on the WAN link don't
>> send the call through the PSTN.
>>
>> When the call is rejected due to WAN failure send the call through the PSTN.
>>
>>
>>
>> I am not sure of it, I am just assuming.
>>
>>
>>
>>
>> -----Original Message-----
>>  From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
>> Kelemen Zoltan
>>  Sent: Tuesday, October 02, 2007 11:30 AM
>>  To: Cisco VoIP
>>  Subject: [cisco-voip] AAR vs. Route Lists
>>
>>
>>
>> Hi,
>>
>>
>>
>> this may be a noob question to some of you, but there's something I
>>
>> don't quite understand:
>>
>>  don't Route Lists provide the same (or rather - similar) functionality
>>
>> as AAR? If not (and I suppose not) what's the difference?
>>
>>
>>
>> Case in point:
>>
>>   We have a client with three CCM clusters, all connected through ICTs.
>>
>> We haven't used AAR so far, but the calls are routed through Route
>>
>> Lists: the trunk is the first route group, but if that fails, calls are
>>
>> redirected to the PSTN route group with appropriate transform masks for
>>
>> called and calling numbers.
>>
>>
>>
>> If the IP trunk is congested, shouldn't the calls be rerouted to the
>>
>> PSTN anyway? (even though AAR is not set)
>>
>>
>>
>> thanks,
>>
>>   Zoltan
>>
>> _______________________________________________
>>
>> cisco-voip mailing list
>>
>> cisco-voip at puck.nether.net
>>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>     


From jsteinberg at gmail.com  Tue Oct  2 08:58:21 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 2 Oct 2007 08:58:21 -0400
Subject: [cisco-voip] AAR vs. Route Lists
In-Reply-To: <47023BC8.6020609@carocomp.ro>
References: <88C8D019DA547C4BABECB9A806ED22ED013810D0@mail.ACT-EG.COM>
	<f1b65f880710020539x633243adyc6292db809c64b23@mail.gmail.com>
	<47023BC8.6020609@carocomp.ro>
Message-ID: <f1b65f880710020558o77608810hb781011427393f1c@mail.gmail.com>

There is a callmanager service parameter that controls this action:

 Stop Routing on Out of Bandwidth Flag

By default it is set to false, and it will try RG2.

On 10/2/07, Kelemen Zoltan <keli at carocomp.ro> wrote:
> So a RL won't fall back to the second route group if the network is
> congested (that basically should send a busy as well, not?)?
>
> Example:
> RouteList IP_PSTN
>  RG1: IPTrunk
>  RG2: PSTTrunk
>
> if RG1 is congested (Locations bandwidth exceeded) it won't go over to RG2?
>
> Zoltan
>
> Justin Steinberg wrote:
> > AAR works in conjunction with CCM Locations bandwidth.  Think two IP
> > phones located at two remote sites that attempt to dial each other and
> > exceed the Locations bandwidth.  AAR is invoked and the call is
> > rerouted out a local PSTN gateway.
> >
> > http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803edabe.html#wp1056007
> >
> > On 10/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
> >
> >>
> >>
> >> Hello;
> >>
> >>
> >>
> >> I never use the AAR feature before, but something came to my mind. Maybe the
> >> AAR can achieve the following (that RL cannot):
> >>
> >>
> >>
> >> When the call is rejected due to not enough bandwidth on the WAN link don't
> >> send the call through the PSTN.
> >>
> >> When the call is rejected due to WAN failure send the call through the PSTN.
> >>
> >>
> >>
> >> I am not sure of it, I am just assuming.
> >>
> >>
> >>
> >>
> >> -----Original Message-----
> >>  From: cisco-voip-bounces at puck.nether.net
> >> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> >> Kelemen Zoltan
> >>  Sent: Tuesday, October 02, 2007 11:30 AM
> >>  To: Cisco VoIP
> >>  Subject: [cisco-voip] AAR vs. Route Lists
> >>
> >>
> >>
> >> Hi,
> >>
> >>
> >>
> >> this may be a noob question to some of you, but there's something I
> >>
> >> don't quite understand:
> >>
> >>  don't Route Lists provide the same (or rather - similar) functionality
> >>
> >> as AAR? If not (and I suppose not) what's the difference?
> >>
> >>
> >>
> >> Case in point:
> >>
> >>   We have a client with three CCM clusters, all connected through ICTs.
> >>
> >> We haven't used AAR so far, but the calls are routed through Route
> >>
> >> Lists: the trunk is the first route group, but if that fails, calls are
> >>
> >> redirected to the PSTN route group with appropriate transform masks for
> >>
> >> called and calling numbers.
> >>
> >>
> >>
> >> If the IP trunk is congested, shouldn't the calls be rerouted to the
> >>
> >> PSTN anyway? (even though AAR is not set)
> >>
> >>
> >>
> >> thanks,
> >>
> >>   Zoltan
> >>
> >> _______________________________________________
> >>
> >> cisco-voip mailing list
> >>
> >> cisco-voip at puck.nether.net
> >>
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >>
>
>

From Andrius-conf at elsis.lt  Tue Oct  2 08:55:38 2007
From: Andrius-conf at elsis.lt (Andrius Kislas)
Date: Tue, 02 Oct 2007 15:55:38 +0300
Subject: [cisco-voip] AAR vs. Route Lists
In-Reply-To: <47020FA2.1090105@carocomp.ro>
References: <47020FA2.1090105@carocomp.ro>
Message-ID: <47023FCA.7010507@elsis.lt>

a) AAR will work inside one CallManager cluster without any IP or PSTN
trunks.
b) AAR will save your time because it will automatically take external
called phone number from phones configuration and reroute calls.

Andrius

Kelemen Zoltan wrote:
> Hi,
> 
> this may be a noob question to some of you, but there's something I 
> don't quite understand:
>  don't Route Lists provide the same (or rather - similar) functionality 
> as AAR? If not (and I suppose not) what's the difference?
> 
> Case in point:
>   We have a client with three CCM clusters, all connected through ICTs. 
> We haven't used AAR so far, but the calls are routed through Route 
> Lists: the trunk is the first route group, but if that fails, calls are 
> redirected to the PSTN route group with appropriate transform masks for 
> called and calling numbers.
> 
> If the IP trunk is congested, shouldn't the calls be rerouted to the 
> PSTN anyway? (even though AAR is not set)
> 
> thanks,
>   Zoltan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From Sechaba.Mokoena at btgroup.co.za  Tue Oct  2 09:48:38 2007
From: Sechaba.Mokoena at btgroup.co.za (Sechaba Mokoena)
Date: Tue, 2 Oct 2007 15:48:38 +0200
Subject: [cisco-voip]  Error Message in Attendant Console.
Message-ID: <A086A6B787B09E4B9110CA837AF892D00150DCCB@midms01.btg>

Hi,

 

Has anyone come across the following error message in Attendant Console:

 

"Failed To Make A New Call"

 

I receive this message if a call isn't answered by the receptionist.

 

Best regards,

 

Sechaba Mokoena

Systems Engineer

Bytes Technology Group - Systems Integration


Tel : (+27) (11) 205-7000 |  Fax: (+27) (11) 205-7584  |  Cell : (+27)
84 798-6803 | msn:sechaba.mokoena at hotmail.com
Web  : www.btgroup.co.za  |  Press Office : www.itweb.co.za/office/bytes
<blocked::http://www.itweb.co.za/office/bytes>    |   Licensing :
www.purelicensing.co.za <blocked::http://www.purelicensing.co.za/>  

 

Bytes Systems Integration (Pty) Ltd : Registration No: 1995/012031/07
A subsidiary of Bytes Technology Group SA (Pty) Ltd,   In association
with KAGISO
P O Box 5905, Halfway House, 1685, Midrand;  Bytes Business Park, Block
B, 241 Third Road, Halfway Gardens, Midrand, South Africa 

E-Mail Disclaimer: http://www.altron.co.za/email.asp
<blocked::blocked::http://www.altron.co.za/email.asp>  or phone: (+27)
(11) 205-7000

 


The information contained in this message and or attachments is intended
only for the person or entity to which it is addressed and may contain
confidential and/or privileged material.  Any review, retransmission,
dissemination or other use of, or taking of any action in reliance upon,
this information by persons or entities other than the intended recipient
is prohibited. If you received this in error, please contact the sender and
delete the material from any system and destroy any copies.
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From akoul at dataways.gr  Tue Oct  2 10:39:28 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Tue, 02 Oct 2007 17:39:28 +0300
Subject: [cisco-voip] CallManager express: display caller name from a lookup
	in the directory
Message-ID: <47025820.5040303@dataways.gr>

Hi,
as far as i know, for CallManager, if you want to see a caller's name in 
an incoming call you need a third party tapi application that reads from 
a local database.
But, i have seen that many things are much easier in CME (i remember CME 
supported night service for example but in CCM you needed Unity!!)
So, is it possible for CME to present to the called IPPhone the name of 
the caller by looking up a database or txt directory stored at the 
router's flash?

Thanks
Anthony

From dustin.s.fowler at gmail.com  Tue Oct  2 11:05:04 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Tue, 2 Oct 2007 11:05:04 -0400
Subject: [cisco-voip] Error Message in Attendant Console.
In-Reply-To: <A086A6B787B09E4B9110CA837AF892D00150DCCB@midms01.btg>
References: <A086A6B787B09E4B9110CA837AF892D00150DCCB@midms01.btg>
Message-ID: <47025e33.1219360a.16fd.1ba1@mx.google.com>

Typically this is happens when the phone is not hung up properly and it
errors out. 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sechaba Mokoena
Sent: Tuesday, October 02, 2007 9:49 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Error Message in Attendant Console.

 

Hi,

 

Has anyone come across the following error message in Attendant Console:

 

"Failed To Make A New Call"

 

I receive this message if a call isn't answered by the receptionist.

 

Best regards,

 

Sechaba Mokoena

Systems Engineer

Bytes Technology Group - Systems Integration


Tel : (+27) (11) 205-7000 |  Fax: (+27) (11) 205-7584  |  Cell : (+27) 84
798-6803 | msn:sechaba.mokoena at hotmail.com
Web  : www.btgroup.co.za  |  Press Office :
<blocked::http://www.itweb.co.za/office/bytes> www.itweb.co.za/office/bytes
|   Licensing :  <blocked::http://www.purelicensing.co.za/>
www.purelicensing.co.za 

 

Bytes Systems Integration (Pty) Ltd : Registration No: 1995/012031/07
A subsidiary of Bytes Technology Group SA (Pty) Ltd,   In association with
KAGISO
P O Box 5905, Halfway House, 1685, Midrand;  Bytes Business Park, Block B,
241 Third Road, Halfway Gardens, Midrand, South Africa 

E-Mail Disclaimer:  <blocked::blocked::http://www.altron.co.za/email.asp>
http://www.altron.co.za/email.asp or phone: (+27) (11) 205-7000

 


The information contained in this message and or attachments is intended
only for the person or entity to which it is addressed and may contain
confidential and/or privileged material.  Any review, retransmission,
dissemination or other use of, or taking of any action in reliance upon,
this information by persons or entities other than the intended recipient is
prohibited. If you received this in error, please contact the sender and
delete the material from any system and destroy any copies.

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From svoll.voip at gmail.com  Tue Oct  2 11:05:51 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 2 Oct 2007 08:05:51 -0700
Subject: [cisco-voip] anyone using Exchange 2007 UM?
Message-ID: <f84a38d30710020805x788c5f04tf90ed41d5d6bad7a@mail.gmail.com>

I've had a TAC case open for weeks and I'm not getting anywhere.

does anyone have exchange 2007 working as there Unified messaging?

All works great except I setup the star+extension to go directly to VM.  The
problem is when you try to transfer someone to star+extension the call
drops.  Anyone seen this?

CM 5.1.2

Thanks

Scott
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From jason.aarons at us.didata.com  Tue Oct  2 11:05:02 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 2 Oct 2007 11:05:02 -0400
Subject: [cisco-voip] CUAdmin not responding
Message-ID: <C1FE15183DA37645BC0633BC604E44F006F53A49@USNAEXCH.na.didata.local>

Unity Connection 2.0(1)

 

I type https://server/cuadmin

 

I get a warning about certificate, I accept, but then never get the
splash screen with username login.  Stuck on
https://X.X.X.X/cuadmin/home.do;jsessionid=B2D360D8685EB0AC414D4AAB83C8C
F77

 

In Unity 4.x I would do a iisreset or stop/start IIS Admin.

 

CUAdmin has worked previously and users Voicemail button works fine.
What do I do in Platform OS to restart the web front end?

 




-----------------------------------------
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This e-mail communication and any attachments may contain
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From jsteinberg at gmail.com  Tue Oct  2 11:38:47 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 2 Oct 2007 11:38:47 -0400
Subject: [cisco-voip] anyone using Exchange 2007 UM?
In-Reply-To: <f84a38d30710020805x788c5f04tf90ed41d5d6bad7a@mail.gmail.com>
References: <f84a38d30710020805x788c5f04tf90ed41d5d6bad7a@mail.gmail.com>
Message-ID: <f1b65f880710020838p40172e62qe0f4f6065b49537d@mail.gmail.com>

Where did you setup the *+extension?   Typically, with Unity I've set
that up as a CTI route point and set cfwdall to VM.  The VM pilot mask
is set to XXXX.

Maybe with Exchange/sip trunk you should move the *+ext to a route
pattern and setup *.XXXX with predot discard and point to your
exchange trunk?

Just a guess here...

On 10/2/07, Scott Voll <svoll.voip at gmail.com> wrote:
> I've had a TAC case open for weeks and I'm not getting anywhere.
>
> does anyone have exchange 2007 working as there Unified messaging?
>
> All works great except I setup the star+extension to go directly to VM.  The
> problem is when you try to transfer someone to star+extension the call
> drops.  Anyone seen this?
>
> CM 5.1.2
>
> Thanks
>
> Scott
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From rduric at uoguelph.ca  Tue Oct  2 12:01:53 2007
From: rduric at uoguelph.ca (Rasim Duric)
Date: Tue, 2 Oct 2007 12:01:53 -0400
Subject: [cisco-voip] CSCsg83588
Message-ID: <BD89624EE83BD349BD31A08B453DE24401C67CB6@uogexb201.cfs.uoguelph.ca>

I'm planning to upgrade my CCM 4.1(3)4d cluster to 4.1(3)5d in order to
resolve this bug: CSCsg83588:

 

If Unity completes a blind transfer too quickly to a consultation leg
that is busy, the Unity port may stay busy in Callmanager and will no
longer be able to process incoming calls.  

 

If this happens over and over the result may be that all Unity ports
stop being able to process calls and calls to Unity will get busy tone. 

 

 

It appears that the Callmanager feature layer does not check for this
condition. 

 

 

Workaround: 

 

Stop/start Unity services.  This will re-register the Unity ports and
clear the faulty condition. 

 

 

This software defect was resolved in software release Cisco Callmanager
4.1(3) SR 5B

 

 

 

Has anyone seen this defect and/or already upgraded their CCM to
4.1(3)5d? Any issues? Thanks.

 

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

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From sgrana at urudata.com  Tue Oct  2 13:40:49 2007
From: sgrana at urudata.com (=?iso-8859-1?Q?Sebasti=E1n_Gra=F1a?=)
Date: Tue, 2 Oct 2007 14:40:49 -0300
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <470175D6.5090004@cisco.com>
Message-ID: <77B19B7818C8DF43BA6BE948BFE66550333B3F@urusrvexc.urudata.com.uy>

Wes

DNS is not configured, I'm using SMTP's IP address.

I've done the capture and there is nothing on the wire coming from any of the call managers with dst port TCP/25.
About the logs, wich service log should I look for? 

Thanks

Sebas

-----Mensaje original-----
De: Wes Sisk [mailto:wsisk at cisco.com] 
Enviado el: Lunes, 01 de Octubre de 2007 07:34 p.m.
Para: Sebasti?n Gra?a
Asunto: Re: [cisco-voip] RTMT on CCM5.1

it still applies.  I just update a release-note on the 2nd cdets. the 1st tells you most of what you need.  do you have dns configured?  have you checked the platform logs or performed a packet capture to see the smtp exchange?

Sebasti?n Gra?a wrote:
> Hi Wes
> Thanks for yuor reply.
>  
> I'm using CCM 5.1.1.3000-5 so I believe first Bug does not apply, or 
> it does?
> Not able to view information for CSCsh33877, can you tell me what is 
> it about?
>  
> Thanks
>
> ----------------------------------------------------------------------
> --
> *De:* Wes Sisk [mailto:wsisk at cisco.com] *Enviado el:* Lunes, 01 de 
> Octubre de 2007 06:19 p.m.
> *Para:* Sebasti?n Gra?a
> *CC:* cisco-voip at puck.nether.net
> *Asunto:* Re: [cisco-voip] RTMT on CCM5.1
>
> Watch out for CSCsf08244 and CSCsh33877.
>
> /Wes
>
> Sebasti?n Gra?a wrote:
>> Hi,
>>  
>>     I'm trying to configure alert email sending with this tool but it 
>> is not working.
>>     This is what I've done so far:
>>  
>>     In RTMT, Tools-Alert-Config Mail Server. There I put the IP 
>> address of the SMTP server I want to use, and the port.
>>     In Tools-Alert-Config Alert Action, in Default added email 
>> recipients.
>>  
>>     I have no activity in the SMTP server form any of the call managers.
>>  
>>     Has anyone seen this problem?
>>  
>> Thank you
>>  
>> Best Regards
>>  
>> Sebasti?n Gra?a
>> MCSE - CCNA
>> Departamento T?cnico
>> *Urudata Infraestructura*
>> Canelones 1611
>> Montevideo / Uruguay
>> Tel.: (+598 2) 419 6457
>>  
>> ---------------------------------------------------------------------
>> ---
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip

From wsisk at cisco.com  Tue Oct  2 14:17:09 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 02 Oct 2007 14:17:09 -0400
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <77B19B7818C8DF43BA6BE948BFE66550333B3F@urusrvexc.urudata.com.uy>
References: <77B19B7818C8DF43BA6BE948BFE66550333B3F@urusrvexc.urudata.com.uy>
Message-ID: <47028B25.4060404@cisco.com>

Hello Sebastian,

You are encountering the defect CSCsf08244. Please follow the 
instructions in the workaround.

If you want a permanent fix without enabling DNS open a TAC case and 
request the ES where that defect is resolved.

/Wes

Sebasti?n Gra?a wrote:
> Wes
>
> DNS is not configured, I'm using SMTP's IP address.
>
> I've done the capture and there is nothing on the wire coming from any of the call managers with dst port TCP/25.
> About the logs, wich service log should I look for? 
>
> Thanks
>
> Sebas
>
> -----Mensaje original-----
> De: Wes Sisk [mailto:wsisk at cisco.com] 
> Enviado el: Lunes, 01 de Octubre de 2007 07:34 p.m.
> Para: Sebasti?n Gra?a
> Asunto: Re: [cisco-voip] RTMT on CCM5.1
>
> it still applies.  I just update a release-note on the 2nd cdets. the 1st tells you most of what you need.  do you have dns configured?  have you checked the platform logs or performed a packet capture to see the smtp exchange?
>
> Sebasti?n Gra?a wrote:
>   
>> Hi Wes
>> Thanks for yuor reply.
>>  
>> I'm using CCM 5.1.1.3000-5 so I believe first Bug does not apply, or 
>> it does?
>> Not able to view information for CSCsh33877, can you tell me what is 
>> it about?
>>  
>> Thanks
>>
>> ----------------------------------------------------------------------
>> --
>> *De:* Wes Sisk [mailto:wsisk at cisco.com] *Enviado el:* Lunes, 01 de 
>> Octubre de 2007 06:19 p.m.
>> *Para:* Sebasti?n Gra?a
>> *CC:* cisco-voip at puck.nether.net
>> *Asunto:* Re: [cisco-voip] RTMT on CCM5.1
>>
>> Watch out for CSCsf08244 and CSCsh33877.
>>
>> /Wes
>>
>> Sebasti?n Gra?a wrote:
>>     
>>> Hi,
>>>  
>>>     I'm trying to configure alert email sending with this tool but it 
>>> is not working.
>>>     This is what I've done so far:
>>>  
>>>     In RTMT, Tools-Alert-Config Mail Server. There I put the IP 
>>> address of the SMTP server I want to use, and the port.
>>>     In Tools-Alert-Config Alert Action, in Default added email 
>>> recipients.
>>>  
>>>     I have no activity in the SMTP server form any of the call managers.
>>>  
>>>     Has anyone seen this problem?
>>>  
>>> Thank you
>>>  
>>> Best Regards
>>>  
>>> Sebasti?n Gra?a
>>> MCSE - CCNA
>>> Departamento T?cnico
>>> *Urudata Infraestructura*
>>> Canelones 1611
>>> Montevideo / Uruguay
>>> Tel.: (+598 2) 419 6457
>>>  
>>> ---------------------------------------------------------------------
>>> ---
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>       

From wsisk at cisco.com  Tue Oct  2 14:20:34 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 02 Oct 2007 14:20:34 -0400
Subject: [cisco-voip] CSCsg83588
In-Reply-To: <BD89624EE83BD349BD31A08B453DE24401C67CB6@uogexb201.cfs.uoguelph.ca>
References: <BD89624EE83BD349BD31A08B453DE24401C67CB6@uogexb201.cfs.uoguelph.ca>
Message-ID: <47028BF2.7050608@cisco.com>

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From wsisk at cisco.com  Tue Oct  2 14:21:44 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 02 Oct 2007 14:21:44 -0400
Subject: [cisco-voip] Disaster Recovery: Using ghost to restore
 MCS	Servers maybe?
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED01380EDF@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED01380EDF@mail.ACT-EG.COM>
Message-ID: <47028C38.4080806@cisco.com>

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From wsisk at cisco.com  Tue Oct  2 14:24:52 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 02 Oct 2007 14:24:52 -0400
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC03B482D8@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482D8@CSEXCH01.charles-stanley.co.uk>
Message-ID: <47028CF4.4030005@cisco.com>

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From svoll.voip at gmail.com  Tue Oct  2 14:58:10 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 2 Oct 2007 11:58:10 -0700
Subject: [cisco-voip] VGW over VPN
Message-ID: <f84a38d30710021158i5a279d90id48f2878b6dc2c7f@mail.gmail.com>

I have the following topology

LAN -- VGW -- MetroE -- VPN Concentrator -- layer 3 switch -- CM

I have two ethernet interfaces on the VGW.  I have bound the voice stuff to
the LAN side of the VGW.  all works fine except I can't update the VGW
config via MGCP.  I'm guessing that the ccm-manager stuff is binding to the
wrong interface as all voice stuff works correctly.  it's just new changes
like what phone to ring.  The are all FXO ports.

Give me an idea as to why the configs don't download.

I have all the normal ccm, mgcp, commands.  its just changes i'm having
issues with.

Thanks

Scott
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From svoll.voip at gmail.com  Tue Oct  2 14:58:10 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 2 Oct 2007 11:58:10 -0700
Subject: [cisco-voip] VGW over VPN
Message-ID: <f84a38d30710021158i5a279d90id48f2878b6dc2c7f@mail.gmail.com>

I have the following topology

LAN -- VGW -- MetroE -- VPN Concentrator -- layer 3 switch -- CM

I have two ethernet interfaces on the VGW.  I have bound the voice stuff to
the LAN side of the VGW.  all works fine except I can't update the VGW
config via MGCP.  I'm guessing that the ccm-manager stuff is binding to the
wrong interface as all voice stuff works correctly.  it's just new changes
like what phone to ring.  The are all FXO ports.

Give me an idea as to why the configs don't download.

I have all the normal ccm, mgcp, commands.  its just changes i'm having
issues with.

Thanks

Scott
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From Becky.Baddorf at calence.com  Tue Oct  2 11:12:33 2007
From: Becky.Baddorf at calence.com (Baddorf, Becky)
Date: Tue, 2 Oct 2007 08:12:33 -0700
Subject: [cisco-voip] Archive Directory
In-Reply-To: <mailman.2677.1191329904.1331.cisco-voip@puck.nether.net>
Message-ID: <1322EB8CED7F624EBD28EA4C04E36EAB04541857@phxml05.calence.com>

 I am attempting to download the archives to search for a Phone Bug that
I believe was listed here.  I continue to get errors every time I
attempt to view the archives.

Any Suggestions?

Thanks!!


Becky Baddorf
IPT Engineer - Management Services
Calence, LLC
Primary business hours: 5:00am to 2:00pm MST
IP Phone: 480.889.9738
Fax: 480.889.9599
NOC: 877.682.9600
www.calence.com
2005 Cisco Global Channel Partner of the Year - US/Canada
_________________________
The information contained in this message and any attachment may contain
privileged or confidential information protected from disclosure. If you
are not the intended recipient, or an employee or agent responsible for
delivering this message to the intended recipient, you are hereby
notified that any review, dissemination, distribution or copying of this
information is strictly prohibited.  If you have received this
transmission in error, please notify the sender immediately by replying
to this message and destroying the original and all copies.  Thank you.


From jsteinberg at gmail.com  Tue Oct  2 15:35:27 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 2 Oct 2007 15:35:27 -0400
Subject: [cisco-voip] VGW over VPN
In-Reply-To: <f84a38d30710021158i5a279d90id48f2878b6dc2c7f@mail.gmail.com>
References: <f84a38d30710021158i5a279d90id48f2878b6dc2c7f@mail.gmail.com>
Message-ID: <f1b65f880710021235r7d516c3dx9f96ceeb19edc7d9@mail.gmail.com>

tftp is used for mgcp config download and by default tftp is sourcing
from the egress interface and if that isnt routable to ccm then the
mgcp download fails.

bind your tftp to the same LAN interface via

ip tftp source-interface fax/x

Justin


On 10/2/07, Scott Voll <svoll.voip at gmail.com> wrote:
> I have the following topology
>
> LAN -- VGW -- MetroE -- VPN Concentrator -- layer 3 switch -- CM
>
> I have two ethernet interfaces on the VGW.  I have bound the voice stuff to
> the LAN side of the VGW.  all works fine except I can't update the VGW
> config via MGCP.  I'm guessing that the ccm-manager stuff is binding to the
> wrong interface as all voice stuff works correctly.  it's just new changes
> like what phone to ring.  The are all FXO ports.
>
> Give me an idea as to why the configs don't download.
>
> I have all the normal ccm, mgcp, commands.  its just changes i'm having
> issues with.
>
> Thanks
>
> Scott
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jason.aarons at us.didata.com  Tue Oct  2 16:09:51 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 2 Oct 2007 16:09:51 -0400
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <47028B25.4060404@cisco.com>
References: <77B19B7818C8DF43BA6BE948BFE66550333B3F@urusrvexc.urudata.com.uy>
	<47028B25.4060404@cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006F54167@USNAEXCH.na.didata.local>

If I choose no to DNS during installation, how do I go back and re-enable DNS after the face?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Tuesday, October 02, 2007 2:17 PM
To: Sebasti?n Gra?a
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RTMT on CCM5.1

Hello Sebastian,

You are encountering the defect CSCsf08244. Please follow the 
instructions in the workaround.

If you want a permanent fix without enabling DNS open a TAC case and 
request the ES where that defect is resolved.

/Wes

Sebasti?n Gra?a wrote:
> Wes
>
> DNS is not configured, I'm using SMTP's IP address.
>
> I've done the capture and there is nothing on the wire coming from any of the call managers with dst port TCP/25.
> About the logs, wich service log should I look for? 
>
> Thanks
>
> Sebas
>
> -----Mensaje original-----
> De: Wes Sisk [mailto:wsisk at cisco.com] 
> Enviado el: Lunes, 01 de Octubre de 2007 07:34 p.m.
> Para: Sebasti?n Gra?a
> Asunto: Re: [cisco-voip] RTMT on CCM5.1
>
> it still applies.  I just update a release-note on the 2nd cdets. the 1st tells you most of what you need.  do you have dns configured?  have you checked the platform logs or performed a packet capture to see the smtp exchange?
>
> Sebasti?n Gra?a wrote:
>   
>> Hi Wes
>> Thanks for yuor reply.
>>  
>> I'm using CCM 5.1.1.3000-5 so I believe first Bug does not apply, or 
>> it does?
>> Not able to view information for CSCsh33877, can you tell me what is 
>> it about?
>>  
>> Thanks
>>
>> ----------------------------------------------------------------------
>> --
>> *De:* Wes Sisk [mailto:wsisk at cisco.com] *Enviado el:* Lunes, 01 de 
>> Octubre de 2007 06:19 p.m.
>> *Para:* Sebasti?n Gra?a
>> *CC:* cisco-voip at puck.nether.net
>> *Asunto:* Re: [cisco-voip] RTMT on CCM5.1
>>
>> Watch out for CSCsf08244 and CSCsh33877.
>>
>> /Wes
>>
>> Sebasti?n Gra?a wrote:
>>     
>>> Hi,
>>>  
>>>     I'm trying to configure alert email sending with this tool but it 
>>> is not working.
>>>     This is what I've done so far:
>>>  
>>>     In RTMT, Tools-Alert-Config Mail Server. There I put the IP 
>>> address of the SMTP server I want to use, and the port.
>>>     In Tools-Alert-Config Alert Action, in Default added email 
>>> recipients.
>>>  
>>>     I have no activity in the SMTP server form any of the call managers.
>>>  
>>>     Has anyone seen this problem?
>>>  
>>> Thank you
>>>  
>>> Best Regards
>>>  
>>> Sebasti?n Gra?a
>>> MCSE - CCNA
>>> Departamento T?cnico
>>> *Urudata Infraestructura*
>>> Canelones 1611
>>> Montevideo / Uruguay
>>> Tel.: (+598 2) 419 6457
>>>  
>>> ---------------------------------------------------------------------
>>> ---
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>       
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
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this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From cbomba at s4nets.com  Tue Oct  2 15:14:27 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Tue, 2 Oct 2007 15:14:27 -0400 (EDT)
Subject: [cisco-voip] MIBs for monitoring PRIs
Message-ID: <48945.192.168.1.70.1191352467.webmail@192.168.1.70>

Does anyone know of any MIBs a Cisco router has to monitor the activity of a PRI?  I am trying to use CDR to get some information and I know one of my PRIs was being overrun with calls yesterday from 1pm to 3pm but when I do a utilization report in CDR it says it was only 19 percent utilized during those times?

Anyone have any other way to collect this information?

Thanks,
Chris


From wsisk at cisco.com  Tue Oct  2 16:31:23 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 02 Oct 2007 16:31:23 -0400
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006F54167@USNAEXCH.na.didata.local>
References: <77B19B7818C8DF43BA6BE948BFE66550333B3F@urusrvexc.urudata.com.uy>
	<47028B25.4060404@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F006F54167@USNAEXCH.na.didata.local>
Message-ID: <4702AA9B.40702@cisco.com>

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From svoll.voip at gmail.com  Tue Oct  2 16:31:39 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 2 Oct 2007 13:31:39 -0700
Subject: [cisco-voip] VGW over VPN
In-Reply-To: <f1b65f880710021235r7d516c3dx9f96ceeb19edc7d9@mail.gmail.com>
References: <f84a38d30710021158i5a279d90id48f2878b6dc2c7f@mail.gmail.com>
	<f1b65f880710021235r7d516c3dx9f96ceeb19edc7d9@mail.gmail.com>
Message-ID: <f84a38d30710021331vf882d0lfe048ed5f6c0e531@mail.gmail.com>

your the man!

Thanks

Scott


On 10/2/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> tftp is used for mgcp config download and by default tftp is sourcing
> from the egress interface and if that isnt routable to ccm then the
> mgcp download fails.
>
> bind your tftp to the same LAN interface via
>
> ip tftp source-interface fax/x
>
> Justin
>
>
> On 10/2/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > I have the following topology
> >
> > LAN -- VGW -- MetroE -- VPN Concentrator -- layer 3 switch -- CM
> >
> > I have two ethernet interfaces on the VGW.  I have bound the voice stuff
> to
> > the LAN side of the VGW.  all works fine except I can't update the VGW
> > config via MGCP.  I'm guessing that the ccm-manager stuff is binding to
> the
> > wrong interface as all voice stuff works correctly.  it's just new
> changes
> > like what phone to ring.  The are all FXO ports.
> >
> > Give me an idea as to why the configs don't download.
> >
> > I have all the normal ccm, mgcp, commands.  its just changes i'm having
> > issues with.
> >
> > Thanks
> >
> > Scott
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
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From CarlosOrtiz at bayviewfinancial.com  Tue Oct  2 16:25:46 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 2 Oct 2007 16:25:46 -0400
Subject: [cisco-voip] Archive Directory
In-Reply-To: <1322EB8CED7F624EBD28EA4C04E36EAB04541857@phxml05.calence.com>
Message-ID: <OF8CA8343E.0C2A6E8A-ON85257368.00702007-85257368.007038E5@bftg.com>

Don't download them......Try this link and just search for it.......

http://www.gossamer-threads.com/lists/cisco/voip/






"Baddorf, Becky" <Becky.Baddorf at calence.com> 
Sent by: cisco-voip-bounces at puck.nether.net
10/02/2007 11:12 AM

To
<cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] Archive Directory






 I am attempting to download the archives to search for a Phone Bug that
I believe was listed here.  I continue to get errors every time I
attempt to view the archives.

Any Suggestions?

Thanks!!


Becky Baddorf
IPT Engineer - Management Services
Calence, LLC
Primary business hours: 5:00am to 2:00pm MST
IP Phone: 480.889.9738
Fax: 480.889.9599
NOC: 877.682.9600
www.calence.com
2005 Cisco Global Channel Partner of the Year - US/Canada
_________________________
The information contained in this message and any attachment may contain
privileged or confidential information protected from disclosure. If you
are not the intended recipient, or an employee or agent responsible for
delivering this message to the intended recipient, you are hereby
notified that any review, dissemination, distribution or copying of this
information is strictly prohibited.  If you have received this
transmission in error, please notify the sender immediately by replying
to this message and destroying the original and all copies.  Thank you.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From erickbee at gmail.com  Tue Oct  2 16:42:05 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Tue, 2 Oct 2007 15:42:05 -0500
Subject: [cisco-voip] 12.4(11)T3 IOS MGCP issues
Message-ID: <f4445faf0710021342h6383b3d9l32dc309dd11fc2d9@mail.gmail.com>

Does anyone know of an issue with MGCP in 12.4(11)T3 IOS?

Had a 2800 ISR series gateway stop accepting/placing calls the other
day, and had to reboot the gateway to fix the issue.  No MGCP / MGCP
did not work. When you reset the gateway from call manager it did not
reset (ccm-manager config is in router config). Shutting down the T1
controller on the gateway and bringing it back up did not resolve it
either. When we tried to switch it over to backup call manage
(ccm-manager switchover-) it kept trying to register on the other call
managers but never did. Other gateways were registered to these just
fine, so after all this it seemed MGCP went whacko and lost its mind
so we rebooted the gateway and all has been fine since. Doing a bug
search doesn't come up with any bugs matching this behavior either.

We were on 12.4(9)T before, but each time the gateway lost it's MGCP
connection MGCP would not come up cleanly and needed to manually
bounced (no mgcp/mgcp) to come back. So we upgraded to a newer IOS
version and MGCP has been recovering on it's own when it lost
connection over WAN until the issue yesterday.  I've had issues with
12.4(15)T and 12.4(15)T1 places and used 12.4(11)T3 here because that
has worked elsewhere without problems until now.

In the ISDN q931 debug, the call would come in and immediately get an
unallocated/unassigned as a cause code back to the carrier from the
Call Manager. But the CCM Detailed trace from time did not show any
inbound MGCP activity (yea, I was on the right call manager and looked
at all the traces from both the servers to make sure).

Erick

From sgrana at urudata.com  Tue Oct  2 16:47:25 2007
From: sgrana at urudata.com (=?iso-8859-1?Q?Sebasti=E1n_Gra=F1a?=)
Date: Tue, 2 Oct 2007 17:47:25 -0300
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <4702AA9B.40702@cisco.com>
Message-ID: <77B19B7818C8DF43BA6BE948BFE66550333B41@urusrvexc.urudata.com.uy>

Sorry Wes, this is what I get when searching for this bug in the bugtoolkit.
 
CSCsf08244 Bug Details 
Information contained within Bug ID CSCsf08244 is currently under review for publishing on Bug Toolkit and will be available soon. The publishing team has been automatically alerted to expedite the review of this bug. Try back in an hour or two as we prioritize bugs based on this notification. Further, you may add this bug to your watch group to be notified when the bug is made available to you. 

Thank you very much for your post.
 
Regards


________________________________

De: Wes Sisk [mailto:wsisk at cisco.com] 
Enviado el: Martes, 02 de Octubre de 2007 05:31 p.m.
Para: Jason Aarons (US)
CC: Sebasti?n Gra?a; cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] RTMT on CCM5.1


come 'on guys,  bugtoolkit on Cisco.com:
CSCsf08244

<B>Symptom:</B>
 Email alerts from RTMT may not be delivered to recipients.
 
 <B>Conditions:</B>
 RTMT alerts currently do not supply a domain in the from field of the SMTP
header.  
 Example packet capture of problem:
 Simple Mail Transfer Protocol
     Command: MAIL FROM:<RTMT_Admin>\r\n
         Command: MAIL
         Request parameter: FROM:<RTMT_Admin>
 

CM logs show:
alertmgr.SendMail - SendMail::send: got address exception=
javax.mail.internet.AddressException: Missing domain in string ``RTMT_Admin@''  

 <B>Workaround:</B>
 assign the fully qualified domain name to the servers involved using the
following CLI:
 
 set network dns primary <your primary dns IP address>
 set network dns secondary <your secondary dns IP address>
 set network domain <node.company.com>

/Wes


Jason Aarons (US) wrote: 

	If I choose no to DNS during installation, how do I go back and re-enable DNS after the face?
	
	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
	Sent: Tuesday, October 02, 2007 2:17 PM
	To: Sebasti?n Gra?a
	Cc: cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] RTMT on CCM5.1
	
	Hello Sebastian,
	
	You are encountering the defect CSCsf08244. Please follow the 
	instructions in the workaround.
	
	If you want a permanent fix without enabling DNS open a TAC case and 
	request the ES where that defect is resolved.
	
	/Wes
	
	Sebasti?n Gra?a wrote:
	  

		Wes
		
		DNS is not configured, I'm using SMTP's IP address.
		
		I've done the capture and there is nothing on the wire coming from any of the call managers with dst port TCP/25.
		About the logs, wich service log should I look for? 
		
		Thanks
		
		Sebas
		
		-----Mensaje original-----
		De: Wes Sisk [mailto:wsisk at cisco.com] 
		Enviado el: Lunes, 01 de Octubre de 2007 07:34 p.m.
		Para: Sebasti?n Gra?a
		Asunto: Re: [cisco-voip] RTMT on CCM5.1
		
		it still applies.  I just update a release-note on the 2nd cdets. the 1st tells you most of what you need.  do you have dns configured?  have you checked the platform logs or performed a packet capture to see the smtp exchange?
		
		Sebasti?n Gra?a wrote:
		  
		    

			Hi Wes
			Thanks for yuor reply.
			 
			I'm using CCM 5.1.1.3000-5 so I believe first Bug does not apply, or 
			it does?
			Not able to view information for CSCsh33877, can you tell me what is 
			it about?
			 
			Thanks
			
			----------------------------------------------------------------------
			--
			*De:* Wes Sisk [mailto:wsisk at cisco.com] *Enviado el:* Lunes, 01 de 
			Octubre de 2007 06:19 p.m.
			*Para:* Sebasti?n Gra?a
			*CC:* cisco-voip at puck.nether.net
			*Asunto:* Re: [cisco-voip] RTMT on CCM5.1
			
			Watch out for CSCsf08244 and CSCsh33877.
			
			/Wes
			
			Sebasti?n Gra?a wrote:
			    
			      

				Hi,
				 
				    I'm trying to configure alert email sending with this tool but it 
				is not working.
				    This is what I've done so far:
				 
				    In RTMT, Tools-Alert-Config Mail Server. There I put the IP 
				address of the SMTP server I want to use, and the port.
				    In Tools-Alert-Config Alert Action, in Default added email 
				recipients.
				 
				    I have no activity in the SMTP server form any of the call managers.
				 
				    Has anyone seen this problem?
				 
				Thank you
				 
				Best Regards
				 
				Sebasti?n Gra?a
				MCSE - CCNA
				Departamento T?cnico
				*Urudata Infraestructura*
				Canelones 1611
				Montevideo / Uruguay
				Tel.: (+598 2) 419 6457
				 
				---------------------------------------------------------------------
				---
				
				_______________________________________________
				cisco-voip mailing list
				cisco-voip at puck.nether.net
				https://puck.nether.net/mailman/listinfo/cisco-voip
				      
				        

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	
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From svoll.voip at gmail.com  Tue Oct  2 19:18:32 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 2 Oct 2007 16:18:32 -0700
Subject: [cisco-voip] Unity upgrade license
Message-ID: <f84a38d30710021618w55339c56u814ed7bff6809621@mail.gmail.com>

I'm upgrading from Unity 4.0.5 to 5.x.  the upgrade doc talks about new
license files.  it also talks about PAK.  since I already have 4.0 licenses
do I need to contact licensing and they will get me the new license files or
does the upgrade take care of this?  same hardware.

Thanks

Scott
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From erickbee at gmail.com  Tue Oct  2 19:22:13 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Tue, 2 Oct 2007 18:22:13 -0500
Subject: [cisco-voip] 12.4(11)T3 IOS MGCP issues
In-Reply-To: <BAY126-DAV14695E86D9AC1F62254F41BDAE0@phx.gbl>
References: <f4445faf0710021342h6383b3d9l32dc309dd11fc2d9@mail.gmail.com>
	<BAY126-DAV14695E86D9AC1F62254F41BDAE0@phx.gbl>
Message-ID: <f4445faf0710021622s4047eb44h6b0483203f54f94b@mail.gmail.com>

Call Manager 4.1(3)sr2
There are also multiple sites (gateways) using this server with no problem.

On 10/2/07, Dennis Hartmann <dennisjhartmann at hotmail.com> wrote:
>
>         I'm using IOS 12.4(11XJ) and Call Manager 5.0(4) in some of my Call
> Manager courses.  The ccm-manaager config download is giving me LOTS of
> intermittent problems.  When I hard code dial-peer entries with the service
> mgcpapp command, the problem goes away, but I can not rely on the TFTP
> download.
>
>         What version of Call Manager are you using?
>
>
> Sincerely,
>
> Dennis J. Hartmann
> White Pine Communications
> CCIE# 15651
> CCVP/CCSI/CCIP/CCNP/MCSE
> Phone: (914) 980-0316
> Fax: (914) 470-0121
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
> Sent: Tuesday, October 02, 2007 4:42 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] 12.4(11)T3 IOS MGCP issues
>
> Does anyone know of an issue with MGCP in 12.4(11)T3 IOS?
>
> Had a 2800 ISR series gateway stop accepting/placing calls the other
> day, and had to reboot the gateway to fix the issue.  No MGCP / MGCP
> did not work. When you reset the gateway from call manager it did not
> reset (ccm-manager config is in router config). Shutting down the T1
> controller on the gateway and bringing it back up did not resolve it
> either. When we tried to switch it over to backup call manage
> (ccm-manager switchover-) it kept trying to register on the other call
> managers but never did. Other gateways were registered to these just
> fine, so after all this it seemed MGCP went whacko and lost its mind
> so we rebooted the gateway and all has been fine since. Doing a bug
> search doesn't come up with any bugs matching this behavior either.
>
> We were on 12.4(9)T before, but each time the gateway lost it's MGCP
> connection MGCP would not come up cleanly and needed to manually
> bounced (no mgcp/mgcp) to come back. So we upgraded to a newer IOS
> version and MGCP has been recovering on it's own when it lost
> connection over WAN until the issue yesterday.  I've had issues with
> 12.4(15)T and 12.4(15)T1 places and used 12.4(11)T3 here because that
> has worked elsewhere without problems until now.
>
> In the ISDN q931 debug, the call would come in and immediately get an
> unallocated/unassigned as a cause code back to the carrier from the
> Call Manager. But the CCM Detailed trace from time did not show any
> inbound MGCP activity (yea, I was on the right call manager and looked
> at all the traces from both the servers to make sure).
>
> Erick
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From Roconnell at unislumin.com  Tue Oct  2 19:54:13 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Tue, 2 Oct 2007 19:54:13 -0400
Subject: [cisco-voip] CUCM 5.X/6.X and LDAP
In-Reply-To: <54DD8B39132EFA4293A5C0565004E54609BB9ED2@SBEMAIL1.au.baesystems.com>
References: <20070828172818.116950@gmx.net> <46D46B0C.2060308@cisco.com>
	<54DD8B39132EFA4293A5C0565004E54609BB9ED2@SBEMAIL1.au.baesystems.com>
Message-ID: <18082B35CF0EBD4C8D265874508A615C070EDB11@hermes.unislumin.com>

My problem is the formatting of the phone numbers. The Active Directory
has already been populated with phone number but when they were
populated the were so with brackets and dashes and spaces etc, which
don't work for CM. Has anyone else found a way to script a mass edit for
these fields in AD?

Ryan

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of MILLS, Mark
Sent: Monday, September 03, 2007 1:10 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CUCM 5.X/6.X and LDAP


Hi,

I am trying to filter based on users in a specific group within AD, but
have an issue using the & operator in the ldap filter, has anyone else
successfully managed to do it?

I am trying to use an ldap filter of:

(&(objectclass=user)(!(UserAccountControl:1.2.840.113556.1.4.
803:=2))(memberOf=CN=VOIP,OU=SecurityGroups,OU=General,DC=au,DC=baesyste
ms,DC=com))

I receive the following error:

[Fatal Error] :16:47: The entity name must immediately follow the '&' in
the entity reference.
org.xml.sax.SAXParseException: The entity name must immediately follow
the '&' in the entity reference.

We can do anything with the filter as long as we don't specify the &
operator ... Anyone able to point out what I have wrong?

Cheers,
  Mark



> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
> Sent: Wednesday, 29 August 2007 4:06 AM
> To: rschuknecht at gmx.de
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CUCM 5.X/6.X and LDAP
> 
> Robert,
> 
> Yes.  requires setting attributes in your directory and then 
> applying a 
> filter on CM. I will unicast a doc.
> 
> /Wes
> 
> rschuknecht at gmx.de wrote:
> > Hi List,
> >
> > is it possible to filter out "unwanted" LDAP Users from the 
> Callmanager End-Users, or in other words from the corporate 
> directory? 
> >
> > I have an Callmanager 5.X and 6.X, in LAB for testing, the 
> are both connected (synced) with an mircosoft AD. And i see 
> all my AD-Users. The only users which are filtered out 
> automatically are the default Mircrosoft Users.
> >
> > Any ideas? Thanks in Advance.
> >
> > Robert
> >   
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
Attention:

Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.

From Roconnell at unislumin.com  Tue Oct  2 19:22:47 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Tue, 2 Oct 2007 19:22:47 -0400
Subject: [cisco-voip] CallFwd NOAN and CallFwd Busy not using VM Profile
Message-ID: <18082B35CF0EBD4C8D265874508A615C070EDB0A@hermes.unislumin.com>

Hello all,

 

I have a setup as follows.

 

 

 

Analog Line ------->

PlAR'ed to ext 5555 ------->

Voip peer pointing 5555 to CM----->

Hunt Pilot in CM pointing 5555 to Unity---->

Routing rule sends 5555 to Site Auto Attendant CallHandler ----> 

When Caller selects option "1" the call is transferred "release to
switch" to the Subscriber at that extenstion ---> 

if the on NoAnswer or Busy it is suppose to be sent back to Unity to
that specific users mailbox. This is where it breaks, it is send me back
to Unity but I'm getting "Open Tree" instead of the subscriber mailbox.

 

 

The subscriber mailbox's are actually 10 digits but their CallManager
extensions are 4 digits. I am using a VoiceMail Profile Mask to prefix
the 4 digit local to 10 digits before it goes back into Unity. This
seems to be what is not working, because when I use Unity Tools
CallViewer, I see the forwarding station as the 4 digit local instead of
the 10 digit local. This behavior only occurs with inbound calls through
the h.323 gw. When you call the same pilot from an IP phone everything
works fine. Also if you CFWDALL from the Subscriber extension it also
works fine, it just fails on CFWD BUSY and CFWD Noan

 

This is a CallManager 5.1(2)b to Unity 4.2 setup.

 

Ryan

 

 

 

 

 


Attention:

Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.
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From tednugent73 at gmail.com  Tue Oct  2 20:34:07 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Tue, 2 Oct 2007 20:34:07 -0400
Subject: [cisco-voip] Unity upgrade license
In-Reply-To: <f84a38d30710021618w55339c56u814ed7bff6809621@mail.gmail.com>
References: <f84a38d30710021618w55339c56u814ed7bff6809621@mail.gmail.com>
Message-ID: <22e6380b0710021734p3bcdce5cj13b00608dffea62c@mail.gmail.com>

If you have UCSS then you can order the upgrade online and they will send
you the PAK and Software. The PAK basically just allows you to use your
Unity 4.x .lic files with the 5.x server. If you don;t go this route I'm not
sure if licensing will provide the upgrade .lic file?




On 10/2/07, Scott Voll <svoll.voip at gmail.com> wrote:
>
> I'm upgrading from Unity 4.0.5 to 5.x.  the upgrade doc talks about new
> license files.  it also talks about PAK.  since I already have 4.0licenses do I need to contact licensing and they will get me the new license
> files or does the upgrade take care of this?  same hardware.
>
> Thanks
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jason.aarons at us.didata.com  Tue Oct  2 23:06:32 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 2 Oct 2007 23:06:32 -0400
Subject: [cisco-voip] Unity Connection 2.x Direct Routing Rule failed to
	find routing rule
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E8F1@USNAEXCH.na.didata.local>

Unity Connection 2.0 I have a direct routing rule for 5067 to go to a custom callhandler, it was failing so I changed from that call handler to the directory handler; in short anywhere i send the call I get 'failsafe'.
 
I have 2 other Direct Routing Rules to custom callhandlers that work fine.
 
 
Here is what I get in macro trace;
 
10/02/2007 22:42:23.129 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,Arbiter,13,SetPortActive: port='5:Default Switch-1-001'; switch='0:'; caps=0xff; xtrainfo=0x0, available=0 |

10/02/2007 22:42:23.129 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,Arbiter,-1,Incoming Call: Caller=9543840175, Called=5067, Redirect=5067, Reason=8, Port=Default Switch-1-001, PortsInUse=1, AnsPortsFree=15, CallGuid=0C7400D2E8244B57B4268842764175B0|

10/02/2007 22:42:23.160 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-016' returning 0xFF|

10/02/2007 22:42:23.161 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-012' returning 0xFF|

10/02/2007 22:42:23.161 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-007' returning 0xFF|

10/02/2007 22:42:23.161 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-002' returning 0xFF|

10/02/2007 22:42:23.161 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-001' returning 0xFF|

10/02/2007 22:42:23.161 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-005' returning 0xFF|

10/02/2007 22:42:23.161 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-011' returning 0xFF|

10/02/2007 22:42:23.161 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-006' returning 0xFF|

10/02/2007 22:42:23.162 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-014' returning 0xFF|

10/02/2007 22:42:23.162 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-008' returning 0xFF|

10/02/2007 22:42:23.162 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-010' returning 0xFF|

10/02/2007 22:42:23.162 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-009' returning 0xFF|

10/02/2007 22:42:23.162 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-003' returning 0xFF|

10/02/2007 22:42:23.162 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-013' returning 0xFF|

10/02/2007 22:42:23.163 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-004' returning 0xFF|

10/02/2007 22:42:23.163 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-015' returning 0xFF|

10/02/2007 22:42:23.197 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,CDE,10,CCsStateMachine::Execute : Starting state machine with root state [/opt/cisco/connection/share/tui/AttemptForward.cde!Dummy] [Src/CsStateMachine.cpp:126]|

10/02/2007 22:42:23.197 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,CDE,10,CCsStateMachine::Execute : Calling OnEntry for state [/opt/cisco/connection/share/tui/AttemptForward.cde!Dummy] [Src/CsStateMachine.cpp:138]|

10/02/2007 22:42:23.198 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,CDE,10,CCsStateMachine::Execute : OnEntry for state [/opt/cisco/connection/share/tui/AttemptForward.cde!Dummy] returned 0x00000000 S_OK [Src/CsStateMachine.cpp:142]|

10/02/2007 22:42:23.198 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,CDE,10,CCsStateMachine::Execute : Event is [NULL] [Src/CsStateMachine.cpp:165]|

10/02/2007 22:42:23.198 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,CDE,10,CCsStateMachine::Execute : Next state is [NULL] [Src/CsStateMachine.cpp:187]|

10/02/2007 22:42:23.198 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,CDE,10,CCsStateMachine::Execute : Exiting state machine. [Src/CsStateMachine.cpp:223]|

10/02/2007 22:42:23.199 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,RoutingRules,-1,FindFirstTrueRule: No Rule evaluates to TRUE with start index = 4|

10/02/2007 22:42:23.199 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,Arbiter,-1,Failed to find Routing Rule; port=Default Switch-1-001, from index=-1.|

10/02/2007 22:42:23.199 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,Arbiter,-1,Routing table error; using failsafe conversation 'Failsafe'|

10/02/2007 22:42:23.199 |10181,Default Switch-1-001,0C7400D2E8244B57B4268842764175B0,-1,-1,An unexpected error occured while a caller was on the phone with Unity Connection, and the caller heard the failsafe conversation.|

10/02/2007 22:42:28.164 |10177,,,Arbiter,13,GetPortCaps() for port 'Default Switch-1-016' returning 0xFF|




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From Nick.Kassel at Charles-Stanley.co.uk  Wed Oct  3 06:37:50 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed, 3 Oct 2007 11:37:50 +0100
Subject: [cisco-voip] RTMT on CCM5.1
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC03B482DF@CSEXCH01.charles-stanley.co.uk>

That's what I get as well.

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sebasti?n Gra?a
Sent: 02 October 2007 21:47
To: Wes Sisk; Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RTMT on CCM5.1

 

Sorry Wes, this is what I get when searching for this bug in the bugtoolkit.

 

CSCsf08244 Bug Details 

Information contained within Bug ID CSCsf08244 is currently under review for publishing on Bug Toolkit and will be available soon. The publishing team has been automatically alerted to expedite the review of this bug. Try back in an hour or two as we prioritize bugs based on this notification. Further, you may add this bug to your watch group to be notified when the bug is made available to you. 

Thank you very much for your post.

 

Regards

 

 

________________________________

De: Wes Sisk [mailto:wsisk at cisco.com] 
Enviado el: Martes, 02 de Octubre de 2007 05:31 p.m.
Para: Jason Aarons (US)
CC: Sebasti?n Gra?a; cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] RTMT on CCM5.1

come 'on guys,  bugtoolkit on Cisco.com:
CSCsf08244

<B>Symptom:</B>
 Email alerts from RTMT may not be delivered to recipients.
 
 <B>Conditions:</B>
 RTMT alerts currently do not supply a domain in the from field of the SMTP
header.  
 Example packet capture of problem:
 Simple Mail Transfer Protocol
     Command: MAIL FROM:<RTMT_Admin>\r\n
         Command: MAIL
         Request parameter: FROM:<RTMT_Admin>
 

CM logs show:
alertmgr.SendMail - SendMail::send: got address exception=
javax.mail.internet.AddressException: Missing domain in string ``RTMT_Admin@''  

 <B>Workaround:</B>
 assign the fully qualified domain name to the servers involved using the
following CLI:
 
 set network dns primary <your primary dns IP address>
 set network dns secondary <your secondary dns IP address>
 set network domain <node.company.com>

/Wes


Jason Aarons (US) wrote: 

If I choose no to DNS during installation, how do I go back and re-enable DNS after the face?
 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Tuesday, October 02, 2007 2:17 PM
To: Sebasti?n Gra?a
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RTMT on CCM5.1
 
Hello Sebastian,
 
You are encountering the defect CSCsf08244. Please follow the 
instructions in the workaround.
 
If you want a permanent fix without enabling DNS open a TAC case and 
request the ES where that defect is resolved.
 
/Wes
 
Sebasti?n Gra?a wrote:
  

	Wes
	 
	DNS is not configured, I'm using SMTP's IP address.
	 
	I've done the capture and there is nothing on the wire coming from any of the call managers with dst port TCP/25.
	About the logs, wich service log should I look for? 
	 
	Thanks
	 
	Sebas
	 
	-----Mensaje original-----
	De: Wes Sisk [mailto:wsisk at cisco.com] 
	Enviado el: Lunes, 01 de Octubre de 2007 07:34 p.m.
	Para: Sebasti?n Gra?a
	Asunto: Re: [cisco-voip] RTMT on CCM5.1
	 
	it still applies.  I just update a release-note on the 2nd cdets. the 1st tells you most of what you need.  do you have dns configured?  have you checked the platform logs or performed a packet capture to see the smtp exchange?
	 
	Sebasti?n Gra?a wrote:
	  
	    

		Hi Wes
		Thanks for yuor reply.
		 
		I'm using CCM 5.1.1.3000-5 so I believe first Bug does not apply, or 
		it does?
		Not able to view information for CSCsh33877, can you tell me what is 
		it about?
		 
		Thanks
		 
		----------------------------------------------------------------------
		--
		*De:* Wes Sisk [mailto:wsisk at cisco.com] *Enviado el:* Lunes, 01 de 
		Octubre de 2007 06:19 p.m.
		*Para:* Sebasti?n Gra?a
		*CC:* cisco-voip at puck.nether.net
		*Asunto:* Re: [cisco-voip] RTMT on CCM5.1
		 
		Watch out for CSCsf08244 and CSCsh33877.
		 
		/Wes
		 
		Sebasti?n Gra?a wrote:
		    
		      

			Hi,
			 
			    I'm trying to configure alert email sending with this tool but it 
			is not working.
			    This is what I've done so far:
			 
			    In RTMT, Tools-Alert-Config Mail Server. There I put the IP 
			address of the SMTP server I want to use, and the port.
			    In Tools-Alert-Config Alert Action, in Default added email 
			recipients.
			 
			    I have no activity in the SMTP server form any of the call managers.
			 
			    Has anyone seen this problem?
			 
			Thank you
			 
			Best Regards
			 
			Sebasti?n Gra?a
			MCSE - CCNA
			Departamento T?cnico
			*Urudata Infraestructura*
			Canelones 1611
			Montevideo / Uruguay
			Tel.: (+598 2) 419 6457
			 
			---------------------------------------------------------------------
			---
			 
			_______________________________________________
			cisco-voip mailing list
			cisco-voip at puck.nether.net
			https://puck.nether.net/mailman/listinfo/cisco-voip
			      
			        

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
 
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From martind at cae.com  Wed Oct  3 06:49:03 2007
From: martind at cae.com (Daniel Martin)
Date: Wed, 3 Oct 2007 06:49:03 -0400
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <4702AA9B.40702@cisco.com>
References: <77B19B7818C8DF43BA6BE948BFE66550333B3F@urusrvexc.urudata.com.uy><47028B25.4060404@cisco.com><C1FE15183DA37645BC0633BC604E44F006F54167@USNAEXCH.na.didata.local>
	<4702AA9B.40702@cisco.com>
Message-ID: <09881513E25DD549B26BF7F38ECAF8DA3DC4DF@caemex01.caecorp.cae.com>

I had the same problem and opened a case with TAC. They supplied me version 5.1.1-3119 which fixed it.

 

Daniel.

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Tuesday, October 02, 2007 4:31 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net; Sebasti?n Gra?a
Subject: Re: [cisco-voip] RTMT on CCM5.1

 

come 'on guys,  bugtoolkit on Cisco.com:
CSCsf08244

<B>Symptom:</B>
 Email alerts from RTMT may not be delivered to recipients.
 
 <B>Conditions:</B>
 RTMT alerts currently do not supply a domain in the from field of the SMTP
header.  
 Example packet capture of problem:
 Simple Mail Transfer Protocol
     Command: MAIL FROM:<RTMT_Admin>\r\n
         Command: MAIL
         Request parameter: FROM:<RTMT_Admin>
 

CM logs show:
alertmgr.SendMail - SendMail::send: got address exception=
javax.mail.internet.AddressException: Missing domain in string ``RTMT_Admin@''  

 <B>Workaround:</B>
 assign the fully qualified domain name to the servers involved using the
following CLI:
 
 set network dns primary <your primary dns IP address>
 set network dns secondary <your secondary dns IP address>
 set network domain <node.company.com>

/Wes


Jason Aarons (US) wrote: 

If I choose no to DNS during installation, how do I go back and re-enable DNS after the face?
 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Tuesday, October 02, 2007 2:17 PM
To: Sebasti?n Gra?a
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RTMT on CCM5.1
 
Hello Sebastian,
 
You are encountering the defect CSCsf08244. Please follow the 
instructions in the workaround.
 
If you want a permanent fix without enabling DNS open a TAC case and 
request the ES where that defect is resolved.
 
/Wes
 
Sebasti?n Gra?a wrote:
  

	Wes
	 
	DNS is not configured, I'm using SMTP's IP address.
	 
	I've done the capture and there is nothing on the wire coming from any of the call managers with dst port TCP/25.
	About the logs, wich service log should I look for? 
	 
	Thanks
	 
	Sebas
	 
	-----Mensaje original-----
	De: Wes Sisk [mailto:wsisk at cisco.com] 
	Enviado el: Lunes, 01 de Octubre de 2007 07:34 p.m.
	Para: Sebasti?n Gra?a
	Asunto: Re: [cisco-voip] RTMT on CCM5.1
	 
	it still applies.  I just update a release-note on the 2nd cdets. the 1st tells you most of what you need.  do you have dns configured?  have you checked the platform logs or performed a packet capture to see the smtp exchange?
	 
	Sebasti?n Gra?a wrote:
	  
	    

		Hi Wes
		Thanks for yuor reply.
		 
		I'm using CCM 5.1.1.3000-5 so I believe first Bug does not apply, or 
		it does?
		Not able to view information for CSCsh33877, can you tell me what is 
		it about?
		 
		Thanks
		 
		----------------------------------------------------------------------
		--
		*De:* Wes Sisk [mailto:wsisk at cisco.com] *Enviado el:* Lunes, 01 de 
		Octubre de 2007 06:19 p.m.
		*Para:* Sebasti?n Gra?a
		*CC:* cisco-voip at puck.nether.net
		*Asunto:* Re: [cisco-voip] RTMT on CCM5.1
		 
		Watch out for CSCsf08244 and CSCsh33877.
		 
		/Wes
		 
		Sebasti?n Gra?a wrote:
		    
		      

			Hi,
			 
			    I'm trying to configure alert email sending with this tool but it 
			is not working.
			    This is what I've done so far:
			 
			    In RTMT, Tools-Alert-Config Mail Server. There I put the IP 
			address of the SMTP server I want to use, and the port.
			    In Tools-Alert-Config Alert Action, in Default added email 
			recipients.
			 
			    I have no activity in the SMTP server form any of the call managers.
			 
			    Has anyone seen this problem?
			 
			Thank you
			 
			Best Regards
			 
			Sebasti?n Gra?a
			MCSE - CCNA
			Departamento T?cnico
			*Urudata Infraestructura*
			Canelones 1611
			Montevideo / Uruguay
			Tel.: (+598 2) 419 6457
			 
			---------------------------------------------------------------------
			---
			 
			_______________________________________________
			cisco-voip mailing list
			cisco-voip at puck.nether.net
			https://puck.nether.net/mailman/listinfo/cisco-voip
			      
			        

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https://puck.nether.net/mailman/listinfo/cisco-voip
 
-----------------------------------------
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This e-mail communication and any attachments may contain
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From voip at mrga.ch  Wed Oct  3 09:55:45 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Wed, 03 Oct 2007 15:55:45 +0200
Subject: [cisco-voip] Update Phone Button Template with BAT
Message-ID: <1191419745.47039f61ebcdf@www.mail2web.ch>

Hello

I try to update the Phone Button Template with BAT on my Callmanager 4.2(3).
However I can not find an entry in BAT to update the Phone Button Template.

Has anyone an idea how I can update the Phone Button Template with BAT?

Thanks Reto


From alessandro.bertacco at lanservice.it  Wed Oct  3 10:18:09 2007
From: alessandro.bertacco at lanservice.it (Alessandro Bertacco)
Date: Wed, 3 Oct 2007 16:18:09 +0200
Subject: [cisco-voip] CUCM 6.0.(1) critical service Don't Start after
	migration from CCM 4.1.3
Message-ID: <BABE7297CCC23245A867092258C59507DFC5D8@LS-MAIL.lanservice.local>

Hi Every one,
    Cisco make me Crazy...:O)

The situation:

HP DL320 G2 certified HW.

Migrated copy of CUCM 6.0.(1).

Installed Valid migration license file obtained from http://www.cisco.com/go/license (exatly that link: https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet?FormId=806 )

The page ask me the Publisger MAC Address and the cut & paste of the migrated license file......I've done it successfully!!

Cisco has send me the .lic file!!!

I've successfully installed  the license file on my server under System -> License -> License File Upload and I restarted the publisher.

CCM service doesn't start and RTMT said me:

Date: Oct 3 15:59:47
Machine Name: ls-ccm
Severity: Error
App ID: Cisco CallManager
Message:  : 0: Oct 03 13:59:47.716 UTC :  %CCM_CALLMANAGER-CALLMANAGER-3-NoFeatureLicense: No feature license found. Cluster ID:StandAloneCluster Node ID:ls-ccm

The license is installed, but for test reason I've changed the ip address of the publisher server. Is it a possible problem?

Thank you

Bye


Alessandro Bertacco
Network Engineer
alessandro.bertacco at lanservice.it  
 <<Immagine (Metafile)>> 
 via Brodolini, 80
 15033 CASALE MONFERRATO (AL)
 Tel +39 0142 330500
 Fax +39 0142 330655
 web http://www.lanservice.it



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From patrick.diener at gmail.com  Wed Oct  3 11:27:01 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Wed, 3 Oct 2007 17:27:01 +0200
Subject: [cisco-voip] CUCM 6.0.(1) critical service Don't Start after
	migration from CCM 4.1.3
In-Reply-To: <BABE7297CCC23245A867092258C59507DFC5D8@LS-MAIL.lanservice.local>
References: <BABE7297CCC23245A867092258C59507DFC5D8@LS-MAIL.lanservice.local>
Message-ID: <c0deb5b80710030827r52cfccd6ie8e61b54d1b8be4c@mail.gmail.com>

what can you see in the License Unit Report? (System -> License Unit Report)

if the installation of the lics. was really successful then the node
license(s) should appear there...

Regards
Patrick

On 10/3/07, Alessandro Bertacco <alessandro.bertacco at lanservice.it> wrote:
>
>  Hi Every one,
>     Cisco make me Crazy?:O)
>
> The situation:
>
> HP DL320 G2 certified HW.
>
> Migrated copy of CUCM 6.0.(1).
>
> Installed Valid migration license file obtained from *
> http://www.cisco.com/go/license* <http://www.cisco.com/go/license> (exatly
> that link: *
> https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet?FormId=806
> *<https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet?FormId=806>)
>
> The page ask me the Publisger MAC Address and the cut & paste of the
> migrated license file??I've done it successfully!!
>
> Cisco has send me the .lic file!!!
>
> I've successfully installed  the license file on my server under System ->
> License -> License File Upload and I restarted the publisher.
>
> CCM service doesn't start and RTMT said me:
>
> Date: Oct 3 15:59:47
> Machine Name: ls-ccm
> Severity: Error
> App ID: Cisco CallManager
> Message:  : 0: Oct 03 13:59:47.716 UTC :
> %CCM_CALLMANAGER-CALLMANAGER-3-NoFeatureLicense: No feature license found.
> Cluster ID:StandAloneCluster Node ID:ls-ccm
>
> The license is installed, but for test reason I've changed the ip address
> of the publisher server. Is it a possible problem?
>
> Thank you
>
> Bye
>
> *Alessandro Bertacco*
> Network Engineer
> ***alessandro.bertacco at lanservice.it***<alessandro.bertacco at lanservice.it>
> **
>  [image: Immagine (Metafile)]
>  via Brodolini, 80
>  15033 CASALE MONFERRATO (AL)
>  Tel +39 0142 330500
>  Fax +39 0142 330655
>  web *http://www.lanservice.it* <http://www.lanservice.it>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From wsisk at cisco.com  Wed Oct  3 10:59:05 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 03 Oct 2007 10:59:05 -0400
Subject: [cisco-voip] CUCM 5.X/6.X and LDAP
In-Reply-To: <54DD8B39132EFA4293A5C0565004E54609BB9ED2@SBEMAIL1.au.baesystems.com>
References: <20070828172818.116950@gmx.net> <46D46B0C.2060308@cisco.com>
	<54DD8B39132EFA4293A5C0565004E54609BB9ED2@SBEMAIL1.au.baesystems.com>
Message-ID: <4703AE39.8020002@cisco.com>

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From wsisk at cisco.com  Wed Oct  3 10:56:56 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 03 Oct 2007 10:56:56 -0400
Subject: [cisco-voip] RTMT on CCM5.1
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC03B482DF@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482DF@CSEXCH01.charles-stanley.co.uk>
Message-ID: <4703ADB8.4010709@cisco.com>

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From jason.aarons at us.didata.com  Wed Oct  3 12:10:11 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 3 Oct 2007 12:10:11 -0400
Subject: [cisco-voip] Adding/Removing Ring tones in 5.x ?
Message-ID: <C1FE15183DA37645BC0633BC604E44F006F95AD5@USNAEXCH.na.didata.local>

In 4.x I could edit the RingList.xml and put other .raw rings in the
ringlist, can I do this in 5.x and how?

 




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From jonvoip at gmail.com  Wed Oct  3 12:18:26 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 3 Oct 2007 11:18:26 -0500
Subject: [cisco-voip] Unity upgrade license
In-Reply-To: <22e6380b0710021734p3bcdce5cj13b00608dffea62c@mail.gmail.com>
References: <f84a38d30710021618w55339c56u814ed7bff6809621@mail.gmail.com>
	<22e6380b0710021734p3bcdce5cj13b00608dffea62c@mail.gmail.com>
Message-ID: <5d093f9a0710030918l4c3148e3y599fe99fef46a3d4@mail.gmail.com>

You don't need to wait for them to send you a PAK.

As soon as you order Unity 5 thru the PUT, you will get a UG number
and a Sale Order number.

Call TAC, give them those, and you will get the Unity 5 license...

Just note, the Unity 5 upgrade license does NOT contain any licensed
features... it just says that your Unity 4 licenses can be used for
Unity 5.


Jonathan

On 10/2/07, Ted Nugent <tednugent73 at gmail.com> wrote:
> If you have UCSS then you can order the upgrade online and they will send
> you the PAK and Software. The PAK basically just allows you to use your
> Unity 4.x .lic files with the 5.x server. If you don;t go this route I'm not
> sure if licensing will provide the upgrade .lic file?
>
>
>
>
> On 10/2/07, Scott Voll <svoll.voip at gmail.com> wrote:
> >
> >
> > I'm upgrading from Unity 4.0.5 to 5.x.  the upgrade doc talks about new
> license files.  it also talks about PAK.  since I already have 4.0 licenses
> do I need to contact licensing and they will get me the new license files or
> does the upgrade take care of this?  same hardware.
> >
> > Thanks
> >
> > Scott
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From Nunnally at Harding.edu  Wed Oct  3 13:22:16 2007
From: Nunnally at Harding.edu (John Nunnally)
Date: Wed, 03 Oct 2007 12:22:16 -0500
Subject: [cisco-voip] Call handler question
Message-ID: <040201c805e1$f1d0e9c0$c906010a@harding.edu>

Is there a way in Unity to allow a caller to enter an extension number and
leave a message in a mailbox without ringing the phone?  Nortel calls this
feature "express messaging".
 
In other words, I would like to use "Caller System Transfer" and go directly
to the subscribers greeting instead of transferring the call.
 
Is there a way to do that?
 
JWN
 
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From wsisk at cisco.com  Wed Oct  3 13:41:55 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 03 Oct 2007 13:41:55 -0400
Subject: [cisco-voip] CUCM 5.X/6.X and LDAP
In-Reply-To: <18082B35CF0EBD4C8D265874508A615C070EDB11@hermes.unislumin.com>
References: <20070828172818.116950@gmx.net> <46D46B0C.2060308@cisco.com>
	<54DD8B39132EFA4293A5C0565004E54609BB9ED2@SBEMAIL1.au.baesystems.com>
	<18082B35CF0EBD4C8D265874508A615C070EDB11@hermes.unislumin.com>
Message-ID: <4703D463.8090905@cisco.com>

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From erickbee at gmail.com  Wed Oct  3 13:47:54 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Wed, 3 Oct 2007 12:47:54 -0500
Subject: [cisco-voip] 7920 Phone USB cable for Phone to USB type A
Message-ID: <f4445faf0710031047k2a0122abn9ffab396ef4e7db8@mail.gmail.com>

Does anyone know where I can buy the cable to connect from USB port on
computer directly to the 7920 phone without the cradle?

Cisco's website saids this and has a picture, but I've googled for
this and similar searches and haven't come up with a place that has
said cable up for sale.

Phone to A-Type 48-Inch USB Cable

From CDAmore at jackhenry.com  Wed Oct  3 13:58:49 2007
From: CDAmore at jackhenry.com (Chris D'amore)
Date: Wed, 3 Oct 2007 12:58:49 -0500
Subject: [cisco-voip] Call handler question
In-Reply-To: <040201c805e1$f1d0e9c0$c906010a@harding.edu>
References: <040201c805e1$f1d0e9c0$c906010a@harding.edu>
Message-ID: <2B3745BC9D31514381A6FCC5E72D9373026AA0A8@SMOMAIL.jhacorp.com>

Follow this link.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a00800dea82.shtml

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Nunnally
Sent: Wednesday, October 03, 2007 12:22 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call handler question

 

Is there a way in Unity to allow a caller to enter an extension number
and leave a message in a mailbox without ringing the phone?  Nortel
calls this feature "express messaging".

 

In other words, I would like to use "Caller System Transfer" and go
directly to the subscribers greeting instead of transferring the call.

 

Is there a way to do that?

 

JWN

 

NOTICE: This electronic mail message and any files transmitted with it are intended
exclusively for the individual or entity to which it is addressed. The message, 
together with any attachment, may contain confidential and/or privileged information.
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From jonvoip at gmail.com  Wed Oct  3 14:01:44 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 3 Oct 2007 13:01:44 -0500
Subject: [cisco-voip] AAR vs. Route Lists
In-Reply-To: <47020FA2.1090105@carocomp.ro>
References: <47020FA2.1090105@carocomp.ro>
Message-ID: <5d093f9a0710031101v7d5082bdp93dfba708c05daf0@mail.gmail.com>

If the IP Trunks go to a Gatekeeper and you set the Service Parameter
to make CCM BRQ aware...

However, that still won't kick off AAR (cuz AAR only kicks in when
Locations-based CAC is exhausted), however, it will kick off your
route-list 'next Route Group' thing...



Jonathan

On 10/2/07, Kelemen Zoltan <keli at carocomp.ro> wrote:
> Hi,
>
> this may be a noob question to some of you, but there's something I
> don't quite understand:
>  don't Route Lists provide the same (or rather - similar) functionality
> as AAR? If not (and I suppose not) what's the difference?
>
> Case in point:
>   We have a client with three CCM clusters, all connected through ICTs.
> We haven't used AAR so far, but the calls are routed through Route
> Lists: the trunk is the first route group, but if that fails, calls are
> redirected to the PSTN route group with appropriate transform masks for
> called and calling numbers.
>
> If the IP trunk is congested, shouldn't the calls be rerouted to the
> PSTN anyway? (even though AAR is not set)
>
> thanks,
>   Zoltan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Wed Oct  3 14:04:12 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 3 Oct 2007 13:04:12 -0500
Subject: [cisco-voip] anyone using Exchange 2007 UM?
In-Reply-To: <f84a38d30710020805x788c5f04tf90ed41d5d6bad7a@mail.gmail.com>
References: <f84a38d30710020805x788c5f04tf90ed41d5d6bad7a@mail.gmail.com>
Message-ID: <5d093f9a0710031104t205c8a9ehc000aae1641749d1@mail.gmail.com>

I have it working at home (in a test environment) works fine...

My star+extension xfer to vmail works fine... Did you set the mask
correctly so it doesn't send the star?



Jonathan

On 10/2/07, Scott Voll <svoll.voip at gmail.com> wrote:
> I've had a TAC case open for weeks and I'm not getting anywhere.
>
> does anyone have exchange 2007 working as there Unified messaging?
>
> All works great except I setup the star+extension to go directly to VM.  The
> problem is when you try to transfer someone to star+extension the call
> drops.  Anyone seen this?
>
> CM 5.1.2
>
> Thanks
>
> Scott
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From svoll.voip at gmail.com  Wed Oct  3 15:18:34 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 3 Oct 2007 12:18:34 -0700
Subject: [cisco-voip] anyone using Exchange 2007 UM?
In-Reply-To: <5d093f9a0710031104t205c8a9ehc000aae1641749d1@mail.gmail.com>
References: <f84a38d30710020805x788c5f04tf90ed41d5d6bad7a@mail.gmail.com>
	<5d093f9a0710031104t205c8a9ehc000aae1641749d1@mail.gmail.com>
Message-ID: <f84a38d30710031218j5edd7cb2xb85e7f0d277ed0db@mail.gmail.com>

the *extension works fine ..... it's the "transfer" *extension "transfer"
that does not work fine.  Try that and let me know.

Scott


On 10/3/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> I have it working at home (in a test environment) works fine...
>
> My star+extension xfer to vmail works fine... Did you set the mask
> correctly so it doesn't send the star?
>
>
>
> Jonathan
>
> On 10/2/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > I've had a TAC case open for weeks and I'm not getting anywhere.
> >
> > does anyone have exchange 2007 working as there Unified messaging?
> >
> > All works great except I setup the star+extension to go directly to
> VM.  The
> > problem is when you try to transfer someone to star+extension the call
> > drops.  Anyone seen this?
> >
> > CM 5.1.2
> >
> > Thanks
> >
> > Scott
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
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From jvw at medicineforthedefense.com  Wed Oct  3 16:43:47 2007
From: jvw at medicineforthedefense.com (Weigand, John V.)
Date: Wed, 3 Oct 2007 16:43:47 -0400
Subject: [cisco-voip] Amount of Time for CUC Updates
Message-ID: <1E196438D2BA76448E5A077C8C99329D0570BAC0@lmiexmb01.corp.medicineforthedefense.com>

Serious Medicine for the Defense R 

Any estimate on total amount of downtime it would take to run the Cisco
Unity Server Updates Wizard, and Service Releases 1 and 2, for a Unity
Connection 1.2.1 server? We haven't updated it since it was installed in
February, want to get it current, but need to know how much of a
maintenance window we'll need. 

 

Thanks!


John V. Weigand
Help Desk Support

Litigation Management Inc.
300 Allen-Bradley Drive
Suite 200
Mayfield Heights, OH 44124





Tel: 
Fax: 440-484-2020
Cell: 
email: jvw at medicineforthedefense.com 




PRIVILEGE AND CONFIDENTIALITY NOTICE 

The information in this electronic mail is intended for the named recipients only. It may contain privileged and confidential material and may be protected under law by the Health Insurance Portability and Accountability Act. Any use of this information by anyone other than the intended receiver is prohibited. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, copying, or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately by replying to this electronic e-mail or by calling (800) 778-5424. Please delete it from your computer. Thank you.
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From c3voip at nc.rr.com  Wed Oct  3 16:55:24 2007
From: c3voip at nc.rr.com (c3voip)
Date: Wed, 3 Oct 2007 16:55:24 -0400
Subject: [cisco-voip] Excluding DC Directory users from Corporate Directory
Message-ID: <000001c805ff$b8409a20$28c1ce60$@rr.com>

I know in an AD Integrated CCM 4.1 cluster you can exclude users from the
Corporate Directory/Global Directory by giving the user the Description of
"CiscoPrivateUser".  Is there a way to do the same thing with DC Directory?

 

Thanks,

-C

 

 

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From jonvoip at gmail.com  Wed Oct  3 16:55:55 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 3 Oct 2007 15:55:55 -0500
Subject: [cisco-voip] Adding/Removing Ring tones in 5.x ?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006F95AD5@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F006F95AD5@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0710031355r606c974dm47c1be9245255223@mail.gmail.com>

Yes, but.

Here's how I did it (kinda sad, but it worked)...

Download the RingList.xml file to a router (yes, a router) using a
copy tftp flash

then copy that to a tftp server on your PC...

Edit away.

Upload file and overwrite...


Done.


Jonathan

On 10/3/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> In 4.x I could edit the RingList.xml and put other ..raw rings in the
> ringlist, can I do this in 5.x and how?
>
>
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jason.aarons at us.didata.com  Wed Oct  3 17:01:37 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 3 Oct 2007 17:01:37 -0400
Subject: [cisco-voip] BAT in CCM 5-1 to remove service
Message-ID: <C1FE15183DA37645BC0633BC604E44F006F96040@USNAEXCH.na.didata.local>

Customer has a service they want removed from all IP Phones, can I do
this with BAT in CallManager 5.1 ?

 




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Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
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From jsteinberg at gmail.com  Wed Oct  3 17:18:11 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 3 Oct 2007 17:18:11 -0400
Subject: [cisco-voip] BAT in CCM 5-1 to remove service
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006F96040@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F006F96040@USNAEXCH.na.didata.local>
Message-ID: <f1b65f880710031418y7340d63bi18c13019e267f14d@mail.gmail.com>

easiest way to do this is to delete the service from callmanager
altogether and it will be deleted from all the phones.  (you can
always add it back if need be.)

On 10/3/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> Customer has a service they want removed from all IP Phones, can I do this
> with BAT in CallManager 5.1 ?
>
>
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jason.aarons at us.didata.com  Wed Oct  3 17:19:21 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 3 Oct 2007 17:19:21 -0400
Subject: [cisco-voip] 7961 ring volume
Message-ID: <C1FE15183DA37645BC0633BC604E44F006F96077@USNAEXCH.na.didata.local>

Customer wants me to force the ring volume of all phones lower, I don't
think it's possible..

 

"Is it possible for the default ringer and speaker volume to be lower? "

 

 




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From jason.aarons at us.didata.com  Wed Oct  3 17:27:00 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 3 Oct 2007 17:27:00 -0400
Subject: [cisco-voip] BAT in CCM 5-1 to remove service
In-Reply-To: <f1b65f880710031418y7340d63bi18c13019e267f14d@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F006F96040@USNAEXCH.na.didata.local>
	<f1b65f880710031418y7340d63bi18c13019e267f14d@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006F96093@USNAEXCH.na.didata.local>

I was scared of this method, I could see CCM hitting a caveat and
breaking. I'll try deleting the service after hours tonight -thanks
-jason

-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Wednesday, October 03, 2007 5:18 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] BAT in CCM 5-1 to remove service

easiest way to do this is to delete the service from callmanager
altogether and it will be deleted from all the phones.  (you can
always add it back if need be.)

On 10/3/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> Customer has a service they want removed from all IP Phones, can I do
this
> with BAT in CallManager 5.1 ?
>
>
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
designated
> addressee(s) named above only. If you are not the intended addressee,
you
> are hereby notified that you have received this communication in error
and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this
communication in
> error, please notify us immediately by replying to this message and
deleting
> it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
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this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From patrick.diener at gmail.com  Wed Oct  3 18:10:06 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Thu, 4 Oct 2007 00:10:06 +0200
Subject: [cisco-voip] 7961 ring volume
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006F96077@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F006F96077@USNAEXCH.na.didata.local>
Message-ID: <c0deb5b80710031510p50a76e2fy50c856b89926c321@mail.gmail.com>

you can use ip phone xml and the built in web server of the ip phones
to post (HTTP POST) a (or several) CiscoIPPhoneExecute XML Object with
ExecuteItem Key:VolDwn to the phones...
it will look like a ghost is pressing the volume key on the phone ;-)

It's a bit cumbersome but it works!

here some more info:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programming_usage_guide_book09186a00807a34b9.html

Regards
Patrick

On 10/3/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> Customer wants me to force the ring volume of all phones lower, I don't
> think it's possible..
>
>
>
> "Is it possible for the default ringer and speaker volume to be lower? "
>
>
>
>
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From tman701 at gmail.com  Wed Oct  3 18:33:07 2007
From: tman701 at gmail.com (Joel Perez)
Date: Wed, 3 Oct 2007 18:33:07 -0400
Subject: [cisco-voip] MCU question.
Message-ID: <470418ac.1735640a.7cac.4cee@mx.google.com>

Hey Guys,


Two hopefully quick questions.

1) Im setting up an MCU 3515 and wanted to know what is the preference to
use as the protocol, H323 or SCCP? This will be my first time setting one
up.

2) I had a customer ask me if there is a way to make 2 IPT's with two
different ext's ring at the same time?
Im mapping a DID to one of the IPT's ext's and the other ext does not have a
DID. That same Ext with the DID is also a last resort for the AA if no one
presses any menu options in AA. So it would have to work that way also. The
AA is running on CUE 2.3.4 if that makes a difference.

This is on CUCM 5.1

Thanks,

Joel P


From jason.aarons at us.didata.com  Wed Oct  3 18:40:21 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 3 Oct 2007 18:40:21 -0400
Subject: [cisco-voip] 7961 ring volume
In-Reply-To: <c0deb5b80710031510p50a76e2fy50c856b89926c321@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F006F96077@USNAEXCH.na.didata.local>
	<c0deb5b80710031510p50a76e2fy50c856b89926c321@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006F96136@USNAEXCH.na.didata.local>

Would this scale to force 500 7940 handsets to a lower default out of
box ring volume? If the end users raises the ring volume I would have to
re-run the tool/script?  They want to force a volume limit on ring tones
centrally.

-----Original Message-----
From: Patrick Diener [mailto:patrick.diener at gmail.com] 
Sent: Wednesday, October 03, 2007 6:10 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7961 ring volume

you can use ip phone xml and the built in web server of the ip phones
to post (HTTP POST) a (or several) CiscoIPPhoneExecute XML Object with
ExecuteItem Key:VolDwn to the phones...
it will look like a ghost is pressing the volume key on the phone ;-)

It's a bit cumbersome but it works!

here some more info:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programmin
g_usage_guide_book09186a00807a34b9.html

Regards
Patrick

On 10/3/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> Customer wants me to force the ring volume of all phones lower, I
don't
> think it's possible..
>
>
>
> "Is it possible for the default ringer and speaker volume to be lower?
"
>
>
>
>
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
designated
> addressee(s) named above only. If you are not the intended addressee,
you
> are hereby notified that you have received this communication in error
and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this
communication in
> error, please notify us immediately by replying to this message and
deleting
> it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From jd80301 at yahoo.com  Wed Oct  3 19:56:35 2007
From: jd80301 at yahoo.com (Jake Doe)
Date: Wed, 3 Oct 2007 16:56:35 -0700 (PDT)
Subject: [cisco-voip] anyone using Exchange 2007 UM?
In-Reply-To: <f84a38d30710031218j5edd7cb2xb85e7f0d277ed0db@mail.gmail.com>
Message-ID: <616979.97072.qm@web50803.mail.re2.yahoo.com>

First, you need to configure Media Termination Point
(MTP) resources.  Second, make sure you check the
Media Termination Point Required checkbox on your SIP
trunk configuration.  After that, then the transfers
should work (including iDivert).

--- Scott Voll <svoll.voip at gmail.com> wrote:

> the *extension works fine ..... it's the "transfer"
> *extension "transfer"
> that does not work fine.  Try that and let me know.
> 
> Scott
> 
> 
> On 10/3/07, Jonathan Charles <jonvoip at gmail.com>
> wrote:
> >
> > I have it working at home (in a test environment)
> works fine...
> >
> > My star+extension xfer to vmail works fine... Did
> you set the mask
> > correctly so it doesn't send the star?
> >
> >
> >
> > Jonathan
> >
> > On 10/2/07, Scott Voll <svoll.voip at gmail.com>
> wrote:
> > > I've had a TAC case open for weeks and I'm not
> getting anywhere.
> > >
> > > does anyone have exchange 2007 working as there
> Unified messaging?
> > >
> > > All works great except I setup the
> star+extension to go directly to
> > VM.  The
> > > problem is when you try to transfer someone to
> star+extension the call
> > > drops.  Anyone seen this?
> > >
> > > CM 5.1.2
> > >
> > > Thanks
> > >
> > > Scott
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
____________________________________________________________________________________
Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545469

From wsisk at cisco.com  Wed Oct  3 20:26:30 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 03 Oct 2007 20:26:30 -0400
Subject: [cisco-voip] 7961 ring volume
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006F96136@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F006F96077@USNAEXCH.na.didata.local>	<c0deb5b80710031510p50a76e2fy50c856b89926c321@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F006F96136@USNAEXCH.na.didata.local>
Message-ID: <47043336.1070202@cisco.com>

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From wsisk at cisco.com  Wed Oct  3 20:29:06 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 03 Oct 2007 20:29:06 -0400
Subject: [cisco-voip] Excluding DC Directory users from Corporate
	Directory
In-Reply-To: <000001c805ff$b8409a20$28c1ce60$@rr.com>
References: <000001c805ff$b8409a20$28c1ce60$@rr.com>
Message-ID: <470433D2.7010701@cisco.com>

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From kenny.kant at running-config.com  Thu Oct  4 00:30:27 2007
From: kenny.kant at running-config.com (Kenny Kant)
Date: Wed, 03 Oct 2007 23:30:27 -0500
Subject: [cisco-voip] ASA Configuration
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED01380F4B@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED01380F4B@mail.ACT-EG.COM>
Message-ID: <1191472227.809.2.camel@akant-desktop>

Not sure what you are trying to do with the ASA here.  But you are
concerned with IP issues have you considered running the ASA in
transparent mode?

Kenny



On Tue, 2007-10-02 at 11:47 +0200, Ahmed Elnagar wrote:
> You can use a configuration called (as I remember) NAT zero, it can
> help you when you don?t need to NAT on the firewall while you still
> need to pass traffic from it. But really no need to but two firewalls
> behind each others (unless you are using different vendors)
> 
>  
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Lyon
> Sent: Friday, September 14, 2007 12:07 AM
> To: Kevin Dunn
> Cc: cisco voip
> Subject: Re: [cisco-voip] ASA Configuration
> 
>  
> 
> Why are you double firewalling it?
> 
>  
> 
> -Mike
> 
>  
> 
>  
> 
> On 9/13/07, Kevin Dunn <kevinddunn at gmail.com> wrote:
> 
> > We currently have a Firewall in place with a public address.
> 
> > I need to place an ASA between the Internet router and the original
> 
> > firewall.
> 
> > 
> 
> > But those devices are on the same subnet and I can't have the
> outside and
> 
> > inside interfaces on the same subnet in the ASA.
> 
> > 
> 
> > Can I get around that without double NATing the incoming packets?
> 
> > 
> 
> > ** I know this isn't a VOIP question, but we are placing the ASA to
> allow
> 
> > VOIP traffic...sorry guys just looking for clues**
> 
> > 
> 
> > Kevin D. Dunn
> 
> > Network Engineer
> 
> > Marathon Cheese
> 
> > _______________________________________________
> 
> > cisco-voip mailing list
> 
> > cisco-voip at puck.nether.net
> 
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> > 
> 
> _______________________________________________
> 
> cisco-voip mailing list
> 
> cisco-voip at puck.nether.net
> 
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From Jordi.Riba at eu.didata.com  Thu Oct  4 06:15:08 2007
From: Jordi.Riba at eu.didata.com (Jordi Riba (Europe))
Date: Thu, 4 Oct 2007 12:15:08 +0200
Subject: [cisco-voip] Very Strange Behaviour
Message-ID: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>

Hi all,
 
I'm gonna try to explain a very strange situation/behaviour I found yesterday in a customer site... I think this is not the way it should work but ... I would like some of you to re-check it.
 
Thanks in advance...
 
 
Initial Situation:
- CCM 5.1.2
- One headquarters office. (CCM Cluster on it 1 publisher - 3 subscribers)
- Three branchoffices (h323-srst gateway on them) (12.4.9-T3)
- All phones and gateways in the branch are registerd to the publisher.
- Route-patterns and route-list registered to the publisher too.
- Gateways in the branch offices have two voip dialpeer (one pointing to the publisher (precedence 1)  and another pointing to one subscriber (precedence 2))

Action Done:
Changed callmanager group in the branch offices device pool to get branch gateways and phones registered to a subscriber CCM.

Problem:
After this change, Call coming from the PSTN to the gateway can not contact IP Phones. In q931 debug we get a "Network out of order" message.
 
Workaround:
If we deactivate the voice dialpeer pointing to the Publisher --> We work only with the dialpeer pointing to the subscriber. All works fine.

If we deactivate the voice dialpeer pointing to the Subscriber --> We work only with the dialpeer pointing to the Publisher. We get network out of order.

I think this is not the way it should work, is it??
 
Thanks again,
 
Jordi!
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From zohaibshabir at gmail.com  Thu Oct  4 08:02:08 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Thu, 4 Oct 2007 17:02:08 +0500
Subject: [cisco-voip] Very Strange Behaviour
In-Reply-To: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <1ff12970710040502k1a7c93d9s33c2db734e5f3e01@mail.gmail.com>

What you should do is to change the preference of the dial-peer pointing to
the publisher to preference 1 or above. It will work hopefully.

On 10/4/07, Jordi Riba (Europe) <Jordi.Riba at eu.didata.com> wrote:
>
>  Hi all,
>
> I'm gonna try to explain a very strange situation/behaviour I found
> yesterday in a customer site... I think this is not the way it should work
> but ... I would like some of you to re-check it.
>
> Thanks in advance...
>
>
> Initial Situation:
> - CCM 5.1.2
> - One headquarters office. (CCM Cluster on it 1 publisher - 3 subscribers)
> - Three branchoffices (h323-srst gateway on them) (12.4.9-T3)
> - All phones and gateways in the branch are registerd to the publisher.
> - Route-patterns and route-list registered to the publisher too.
> - Gateways in the branch offices have two voip dialpeer (one pointing to
> the publisher (precedence 1)  and another pointing to one subscriber
> (precedence 2))
>
> Action Done:
> Changed callmanager group in the branch offices device pool to get branch
> gateways and phones registered to a subscriber CCM.
>
> Problem:
> After this change, Call coming from the PSTN to the gateway can not
> contact IP Phones. In q931 debug we get a "Network out of order" message.
>
> Workaround:
> If we deactivate the voice dialpeer pointing to the Publisher --> We work
> only with the dialpeer pointing to the subscriber. All works fine.
>
> If we deactivate the voice dialpeer pointing to the Subscriber --> We work
> only with the dialpeer pointing to the Publisher. We get network out of
> order.
>
> I think this is not the way it should work, is it??
>
> Thanks again,
>
> Jordi!
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Regards,
Zohaib Shabir
Network Engineer(Voice)
DWP Group, TECH Division
Ph:+92-302-8232689
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From Jordi.Riba at eu.didata.com  Thu Oct  4 08:13:33 2007
From: Jordi.Riba at eu.didata.com (Jordi Riba (Europe))
Date: Thu, 4 Oct 2007 14:13:33 +0200
Subject: [cisco-voip] Very Strange Behaviour
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
	<1ff12970710040502k1a7c93d9s33c2db734e5f3e01@mail.gmail.com>
Message-ID: <FA233791F398B848B978464507739E8F094701@EUBEBRUSVEX1.eu.didata.local>

Hi Zohaib
 
But, what about if I only want to have a voip dial-peer pointing to my publisher? That situation was not working either.
 
Does the way you configure VOIP dial-peers depends to which CCM (Publisher or Subscriber) is a gateway registered to?
 
Thanks,
 
Jordi!
 
 

________________________________

De: zohaib shabir [mailto:zohaibshabir at gmail.com]
Enviado el: jue 04/10/2007 14:02
Para: Jordi Riba (Europe)
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Very Strange Behaviour


What you should do is to change the preference of the dial-peer pointing to the publisher to preference 1 or above. It will work hopefully.


On 10/4/07, Jordi Riba (Europe) <Jordi.Riba at eu.didata.com> wrote: 

	Hi all,
	 
	I'm gonna try to explain a very strange situation/behaviour I found yesterday in a customer site... I think this is not the way it should work but ... I would like some of you to re-check it. 
	 
	Thanks in advance...
	 
	 
	Initial Situation:
	- CCM 5.1.2
	- One headquarters office. (CCM Cluster on it 1 publisher - 3 subscribers)
	- Three branchoffices (h323-srst gateway on them) (12.4.9-T3)
	- All phones and gateways in the branch are registerd to the publisher.
	- Route-patterns and route-list registered to the publisher too.
	- Gateways in the branch offices have two voip dialpeer (one pointing to the publisher (precedence 1)  and another pointing to one subscriber (precedence 2))

	Action Done:
	Changed callmanager group in the branch offices device pool to get branch gateways and phones registered to a subscriber CCM.

	Problem:
	After this change, Call coming from the PSTN to the gateway can not contact IP Phones. In q931 debug we get a "Network out of order" message.
	 
	Workaround:
	If we deactivate the voice dialpeer pointing to the Publisher --> We work only with the dialpeer pointing to the subscriber. All works fine.

	If we deactivate the voice dialpeer pointing to the Subscriber --> We work only with the dialpeer pointing to the Publisher. We get network out of order.

	I think this is not the way it should work, is it??
	 
	Thanks again,
	
	 
	Jordi!

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net 
	https://puck.nether.net/mailman/listinfo/cisco-voip
	




-- 
Regards,
Zohaib Shabir
Network Engineer(Voice)
DWP Group, TECH Division
Ph:+92-302-8232689 
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From alessandro.bertacco at lanservice.it  Thu Oct  4 08:23:49 2007
From: alessandro.bertacco at lanservice.it (Alessandro Bertacco)
Date: Thu, 4 Oct 2007 14:23:49 +0200
Subject: [cisco-voip] R:  Very Strange Behaviour
In-Reply-To: <FA233791F398B848B978464507739E8F094701@EUBEBRUSVEX1.eu.didata.local>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local><1ff12970710040502k1a7c93d9s33c2db734e5f3e01@mail.gmail.com>
	<FA233791F398B848B978464507739E8F094701@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <BABE7297CCC23245A867092258C59507DFC624@LS-MAIL.lanservice.local>

Hi,
 
i can give my contribution...
 
I've a similar situation with central site with one publiser and one subscriber.
 
All phone registered on the subscriber
 
The remote SRTS gateway of the brranch office configured with two dialpeer pointing subscriber and publisher, but WITH OUT preference.
 
That's works fine in my network
 
Let me know
 
Regards
Alessandro

________________________________

Da: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] Per conto di Jordi Riba (Europe)
Inviato: gioved? 4 ottobre 2007 14.14
A: zohaib shabir
Cc: cisco-voip at puck.nether.net
Oggetto: Re: [cisco-voip] Very Strange Behaviour


Hi Zohaib
 
But, what about if I only want to have a voip dial-peer pointing to my publisher? That situation was not working either.
 
Does the way you configure VOIP dial-peers depends to which CCM (Publisher or Subscriber) is a gateway registered to?
 
Thanks,
 
Jordi!
 
 

________________________________

De: zohaib shabir [mailto:zohaibshabir at gmail.com]
Enviado el: jue 04/10/2007 14:02
Para: Jordi Riba (Europe)
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Very Strange Behaviour


What you should do is to change the preference of the dial-peer pointing to the publisher to preference 1 or above. It will work hopefully.


On 10/4/07, Jordi Riba (Europe) <Jordi.Riba at eu.didata.com> wrote: 

	Hi all,
	 
	I'm gonna try to explain a very strange situation/behaviour I found yesterday in a customer site... I think this is not the way it should work but ... I would like some of you to re-check it. 
	 
	Thanks in advance...
	 
	 
	Initial Situation:
	- CCM 5.1.2
	- One headquarters office. (CCM Cluster on it 1 publisher - 3 subscribers)
	- Three branchoffices (h323-srst gateway on them) (12.4.9-T3)
	- All phones and gateways in the branch are registerd to the publisher.
	- Route-patterns and route-list registered to the publisher too.
	- Gateways in the branch offices have two voip dialpeer (one pointing to the publisher (precedence 1)  and another pointing to one subscriber (precedence 2))

	Action Done:
	Changed callmanager group in the branch offices device pool to get branch gateways and phones registered to a subscriber CCM.

	Problem:
	After this change, Call coming from the PSTN to the gateway can not contact IP Phones. In q931 debug we get a "Network out of order" message.
	 
	Workaround:
	If we deactivate the voice dialpeer pointing to the Publisher --> We work only with the dialpeer pointing to the subscriber. All works fine.

	If we deactivate the voice dialpeer pointing to the Subscriber --> We work only with the dialpeer pointing to the Publisher. We get network out of order.

	I think this is not the way it should work, is it??
	 
	Thanks again,
	
	 
	Jordi!

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net 
	https://puck.nether.net/mailman/listinfo/cisco-voip
	




-- 
Regards,
Zohaib Shabir
Network Engineer(Voice)
DWP Group, TECH Division
Ph:+92-302-8232689 
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From patrick.diener at gmail.com  Thu Oct  4 08:38:13 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Thu, 4 Oct 2007 14:38:13 +0200
Subject: [cisco-voip] Very Strange Behaviour
In-Reply-To: <FA233791F398B848B978464507739E8F094701@EUBEBRUSVEX1.eu.didata.local>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
	<1ff12970710040502k1a7c93d9s33c2db734e5f3e01@mail.gmail.com>
	<FA233791F398B848B978464507739E8F094701@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <c0deb5b80710040538p2dc6b74cg811eafa8f3662c28@mail.gmail.com>

CCMs configured as session targets in the dial-peers on the GW have to
be in the CallManager Group which is assigned to that GW.

If the Gateway sends a call setup to another CCM (which is not in the
GWs CCM Group) the call cannot be completed! I think the GW is not
even alowed to complete the TCP session setup needed to send
subsequent H225 messages...

Regards
Patrick

On 10/4/07, Jordi Riba (Europe) <Jordi.Riba at eu.didata.com> wrote:
>
>
>
> Hi Zohaib
>
> But, what about if I only want to have a voip dial-peer pointing to my
> publisher? That situation was not working either.
>
> Does the way you configure VOIP dial-peers depends to which CCM (Publisher
> or Subscriber) is a gateway registered to?
>
> Thanks,
>
> Jordi!
>
>
>
>  ________________________________
>  De: zohaib shabir [mailto:zohaibshabir at gmail.com]
> Enviado el: jue 04/10/2007 14:02
> Para: Jordi Riba (Europe)
> CC: cisco-voip at puck.nether.net
> Asunto: Re: [cisco-voip] Very Strange Behaviour
>
>
> What you should do is to change the preference of the dial-peer pointing to
> the publisher to preference 1 or above. It will work hopefully.
>
>
> On 10/4/07, Jordi Riba (Europe) <Jordi.Riba at eu.didata.com> wrote:
> >
> >
> > Hi all,
> >
> > I'm gonna try to explain a very strange situation/behaviour I found
> yesterday in a customer site... I think this is not the way it should work
> but ... I would like some of you to re-check it.
> >
> > Thanks in advance...
> >
> >
> > Initial Situation:
> > - CCM 5.1.2
> > - One headquarters office. (CCM Cluster on it 1 publisher - 3 subscribers)
> > - Three branchoffices (h323-srst gateway on them) (12.4.9-T3)
> > - All phones and gateways in the branch are registerd to the publisher.
> > - Route-patterns and route-list registered to the publisher too.
> > - Gateways in the branch offices have two voip dialpeer (one pointing to
> the publisher (precedence 1)  and another pointing to one subscriber
> (precedence 2))
> >
> > Action Done:
> > Changed callmanager group in the branch offices device pool to get branch
> gateways and phones registered to a subscriber CCM.
> >
> > Problem:
> > After this change, Call coming from the PSTN to the gateway can not
> contact IP Phones. In q931 debug we get a "Network out of order" message.
> >
> > Workaround:
> > If we deactivate the voice dialpeer pointing to the Publisher --> We work
> only with the dialpeer pointing to the subscriber. All works fine.
> >
> > If we deactivate the voice dialpeer pointing to the Subscriber --> We work
> only with the dialpeer pointing to the Publisher. We get network out of
> order.
> >
> > I think this is not the way it should work, is it??
> >
> > Thanks again,
> >
> > Jordi!
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> --
> Regards,
> Zohaib Shabir
> Network Engineer(Voice)
> DWP Group, TECH Division
> Ph:+92-302-8232689
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From patrick.diener at gmail.com  Thu Oct  4 08:40:54 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Thu, 4 Oct 2007 14:40:54 +0200
Subject: [cisco-voip] R: CUCM 6.0.(1) critical service Don't Start after
	migration from CCM 4.1.3
In-Reply-To: <BABE7297CCC23245A867092258C59507DFC5DF@LS-MAIL.lanservice.local>
References: <BABE7297CCC23245A867092258C59507DFC5D8@LS-MAIL.lanservice.local>
	<c0deb5b80710030827r52cfccd6ie8e61b54d1b8be4c@mail.gmail.com>
	<BABE7297CCC23245A867092258C59507DFC5DF@LS-MAIL.lanservice.local>
Message-ID: <c0deb5b80710040540m13b0d29bg84e9d7f876389b00@mail.gmail.com>

that looks quit alright to me...
you just have one server right?!?

Regards
Patrick


On 10/3/07, Alessandro Bertacco <alessandro.bertacco at lanservice.it> wrote:
>
>  Hi patrick, I've attached to these e_mail the snapshot of the License
> Unite Report. It seems ok?
>
> I think yes!!
>
> Regards
> Alessandro
>
>  ------------------------------
> *Da:* Patrick Diener [mailto:patrick.diener at gmail.com]
> *Inviato:* mercoled? 3 ottobre 2007 17.27
> *A:* Alessandro Bertacco; cisco-voip at puck.nether.net
> *Oggetto:* Re: [cisco-voip] CUCM 6.0.(1) critical service Don't Start
> after migration from CCM 4.1.3
>
> what can you see in the License Unit Report? (System -> License Unit
> Report)
>
> if the installation of the lics. was really successful then the node
> license(s) should appear there...
>
> Regards
> Patrick
>
> On 10/3/07, Alessandro Bertacco <alessandro.bertacco at lanservice.it> wrote:
> >
> >  Hi Every one,
> >     Cisco make me Crazy?:O)
> >
> > The situation:
> >
> > HP DL320 G2 certified HW.
> >
> > Migrated copy of CUCM 6.0.(1).
> >
> > Installed Valid migration license file obtained from *
> > http://www.cisco.com/go/license* <http://www.cisco.com/go/license>(exatly that link:
> > *
> > https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet?FormId=806
> > *<https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet?FormId=806>)
> >
> > The page ask me the Publisger MAC Address and the cut & paste of the
> > migrated license file??I've done it successfully!!
> >
> > Cisco has send me the .lic file!!!
> >
> > I've successfully installed  the license file on my server under System
> > -> License -> License File Upload and I restarted the publisher.
> >
> > CCM service doesn't start and RTMT said me:
> >
> > Date: Oct 3 15:59:47
> > Machine Name: ls-ccm
> > Severity: Error
> > App ID: Cisco CallManager
> > Message:  : 0: Oct 03 13:59:47.716 UTC :
> > %CCM_CALLMANAGER-CALLMANAGER-3-NoFeatureLicense: No feature license found.
> > Cluster ID:StandAloneCluster Node ID:ls-ccm
> >
> > The license is installed, but for test reason I've changed the ip
> > address of the publisher server. Is it a possible problem?
> >
> > Thank you
> >
> > Bye
> >
> > *Alessandro Bertacco*
> > Network Engineer
> > ***alessandro.bertacco at lanservice.it ***<alessandro.bertacco at lanservice.it>
> > **
> > [image: Immagine (Metafile)]
> >  via Brodolini, 80
> >  15033 CASALE MONFERRATO (AL)
> >  Tel +39 0142 330500
> >  Fax +39 0142 330655
> >  web *http://www.lanservice.it* <http://www.lanservice.it>
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
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From zohaibshabir at gmail.com  Thu Oct  4 08:50:32 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Thu, 4 Oct 2007 17:50:32 +0500
Subject: [cisco-voip] R:  Very Strange Behaviour
In-Reply-To: <BABE7297CCC23245A867092258C59507DFC624@LS-MAIL.lanservice.local>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
	<1ff12970710040502k1a7c93d9s33c2db734e5f3e01@mail.gmail.com>
	<FA233791F398B848B978464507739E8F094701@EUBEBRUSVEX1.eu.didata.local>
	<BABE7297CCC23245A867092258C59507DFC624@LS-MAIL.lanservice.local>
Message-ID: <1ff12970710040550o17a39a76t874b5e9e5ba0b604@mail.gmail.com>

Actually gateways are registered to the cluster not to individual CM, At
least this is the case in CM 4.X but from your working it seems in CM
5.Xthis has changed. I have in my environment both CM
4.2 registered to gateway and it works with or without using the preference
command, so probably i think there is a change in behavior as far as i can
tell in CM 5x

On 10/4/07, Alessandro Bertacco <alessandro.bertacco at lanservice.it> wrote:
>
>  Hi,
>
> i can give my contribution...
>
> I've a similar situation with central site with one publiser and one
> subscriber.
>
> All phone registered on the subscriber
>
> The remote SRTS gateway of the brranch office configured with two dialpeer
> pointing subscriber and publisher, but WITH OUT preference.
>
> That's works fine in my network
>
> Let me know
>
> Regards
> Alessandro
>
>  ------------------------------
> *Da:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *Per conto di *Jordi Riba (Europe)
> *Inviato:* gioved? 4 ottobre 2007 14.14
> *A:* zohaib shabir
> *Cc:* cisco-voip at puck.nether.net
> *Oggetto:* Re: [cisco-voip] Very Strange Behaviour
>
>  Hi Zohaib
>
> But, what about if I only want to have a voip dial-peer pointing to my
> publisher? That situation was not working either.
>
> Does the way you configure VOIP dial-peers depends to which CCM (Publisher
> or Subscriber) is a gateway registered to?
>
> Thanks,
>
> Jordi!
>
>
>
> ------------------------------
> *De:* zohaib shabir [mailto:zohaibshabir at gmail.com]
> *Enviado el:* jue 04/10/2007 14:02
> *Para:* Jordi Riba (Europe)
> *CC:* cisco-voip at puck.nether.net
> *Asunto:* Re: [cisco-voip] Very Strange Behaviour
>
> What you should do is to change the preference of the dial-peer pointing
> to the publisher to preference 1 or above. It will work hopefully.
>
> On 10/4/07, Jordi Riba (Europe) <Jordi.Riba at eu.didata.com> wrote:
> >
> >  Hi all,
> >
> > I'm gonna try to explain a very strange situation/behaviour I found
> > yesterday in a customer site... I think this is not the way it should work
> > but ... I would like some of you to re-check it.
> >
> > Thanks in advance...
> >
> >
> > Initial Situation:
> > - CCM 5.1.2
> > - One headquarters office. (CCM Cluster on it 1 publisher - 3
> > subscribers)
> > - Three branchoffices (h323-srst gateway on them) (12.4.9-T3)
> > - All phones and gateways in the branch are registerd to the publisher.
> > - Route-patterns and route-list registered to the publisher too.
> > - Gateways in the branch offices have two voip dialpeer (one pointing to
> > the publisher (precedence 1)  and another pointing to one subscriber
> > (precedence 2))
> >
> > Action Done:
> > Changed callmanager group in the branch offices device pool to get
> > branch gateways and phones registered to a subscriber CCM.
> >
> > Problem:
> > After this change, Call coming from the PSTN to the gateway can not
> > contact IP Phones. In q931 debug we get a "Network out of order" message.
> >
> > Workaround:
> > If we deactivate the voice dialpeer pointing to the Publisher --> We
> > work only with the dialpeer pointing to the subscriber. All works fine.
> >
> > If we deactivate the voice dialpeer pointing to the Subscriber --> We
> > work only with the dialpeer pointing to the Publisher. We get network out of
> > order.
> >
> > I think this is not the way it should work, is it??
> >
> > Thanks again,
> >
> > Jordi!
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> --
> Regards,
> Zohaib Shabir
> Network Engineer(Voice)
> DWP Group, TECH Division
> Ph:+92-302-8232689
>



-- 
Regards,
Zohaib Shabir
Network Engineer(Voice)
DWP Group, TECH Division
Ph:+92-302-8232689
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From jsteinberg at gmail.com  Thu Oct  4 08:51:45 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 4 Oct 2007 08:51:45 -0400
Subject: [cisco-voip] Very Strange Behaviour
In-Reply-To: <1ff12970710040502k1a7c93d9s33c2db734e5f3e01@mail.gmail.com>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
	<1ff12970710040502k1a7c93d9s33c2db734e5f3e01@mail.gmail.com>
Message-ID: <f1b65f880710040551h25f7dfbbxf4172a736c0d9780@mail.gmail.com>

your h323 dial-peers should point to callmanagers defined in the
CMgroup assigned to the h323 gateway in ccmadmin.

since you removed the publisher from the cmgroup assigned to the h323
gw you should also remove the pub from the h323 dialpeer.  hopefully
you added another sub to the cmgroup and if so, add a new dialpeer for
the sub.

justin

On 10/4/07, zohaib shabir <zohaibshabir at gmail.com> wrote:
> What you should do is to change the preference of the dial-peer pointing to
> the publisher to preference 1 or above. It will work hopefully.
>
>
> On 10/4/07, Jordi Riba (Europe) <Jordi.Riba at eu.didata.com> wrote:
> >
> >
> >
> > Hi all,
> >
> > I'm gonna try to explain a very strange situation/behaviour I found
> yesterday in a customer site... I think this is not the way it should work
> but ... I would like some of you to re-check it.
> >
> > Thanks in advance...
> >
> >
> > Initial Situation:
> > - CCM 5.1.2
> > - One headquarters office. (CCM Cluster on it 1 publisher - 3 subscribers)
> > - Three branchoffices (h323-srst gateway on them) (12.4.9-T3)
> > - All phones and gateways in the branch are registerd to the publisher.
> > - Route-patterns and route-list registered to the publisher too.
> > - Gateways in the branch offices have two voip dialpeer (one pointing to
> the publisher (precedence 1)  and another pointing to one subscriber
> (precedence 2))
> >
> > Action Done:
> > Changed callmanager group in the branch offices device pool to get branch
> gateways and phones registered to a subscriber CCM.
> >
> > Problem:
> > After this change, Call coming from the PSTN to the gateway can not
> contact IP Phones. In q931 debug we get a "Network out of order" message.
> >
> > Workaround:
> > If we deactivate the voice dialpeer pointing to the Publisher --> We work
> only with the dialpeer pointing to the subscriber. All works fine.
> >
> > If we deactivate the voice dialpeer pointing to the Subscriber --> We work
> only with the dialpeer pointing to the Publisher. We get network out of
> order.
> >
> > I think this is not the way it should work, is it??
> >
> > Thanks again,
> >
> > Jordi!
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> --
> Regards,
> Zohaib Shabir
> Network Engineer(Voice)
> DWP Group, TECH Division
> Ph:+92-302-8232689
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From patrick.diener at gmail.com  Thu Oct  4 09:03:00 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Thu, 4 Oct 2007 15:03:00 +0200
Subject: [cisco-voip] R: Very Strange Behaviour
In-Reply-To: <1ff12970710040550o17a39a76t874b5e9e5ba0b604@mail.gmail.com>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
	<1ff12970710040502k1a7c93d9s33c2db734e5f3e01@mail.gmail.com>
	<FA233791F398B848B978464507739E8F094701@EUBEBRUSVEX1.eu.didata.local>
	<BABE7297CCC23245A867092258C59507DFC624@LS-MAIL.lanservice.local>
	<1ff12970710040550o17a39a76t874b5e9e5ba0b604@mail.gmail.com>
Message-ID: <c0deb5b80710040603w5bf99d0ei27019e45c6df54a1@mail.gmail.com>

H323 GW do not register AT ALL on all CCM and CUCM versions

preference configuration on the GW just tells the GW which dial-peer
to take first if destination-pattern on two or more dial-peers have
equal length matching.
If no preference is configured (which is pref. 0 by default) you leave
it up to the IOS software to choose one of these equal matching
dial-peers. This is not dependable on the CCM or CUCM version in use,
in fact the GW is not even aware that the dial-peers point to a CCM,
for the GW the CCMs are just H323 terminals...

Regards
Patrick

On 10/4/07, zohaib shabir <zohaibshabir at gmail.com> wrote:
> Actually gateways are registered to the cluster not to individual CM, At
> least this is the case in CM 4.X but from your working it seems in CM 5.X
> this has changed. I have in my environment both CM 4.2 registered to gateway
> and it works with or without using the preference command, so probably i
> think there is a change in behavior as far as i can tell in CM 5x
>
> On 10/4/07, Alessandro Bertacco
> <alessandro.bertacco at lanservice.it> wrote:
> >
> >
> > Hi,
> >
> > i can give my contribution...
> >
> > I've a similar situation with central site with one publiser and one
> subscriber.
> >
> > All phone registered on the subscriber
> >
> > The remote SRTS gateway of the brranch office configured with two dialpeer
> pointing subscriber and publisher, but WITH OUT preference.
> >
> > That's works fine in my network
> >
> > Let me know
> >
> > Regards
> > Alessandro
> >
> > ________________________________
>  Da: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] Per conto di
> Jordi Riba (Europe)
> > Inviato: gioved? 4 ottobre 2007 14.14
> > A: zohaib shabir
> > Cc: cisco-voip at puck.nether.net
> > Oggetto: Re: [cisco-voip] Very Strange Behaviour
> >
> >
> >
> >
> > Hi Zohaib
> >
> > But, what about if I only want to have a voip dial-peer pointing to my
> publisher? That situation was not working either.
> >
> > Does the way you configure VOIP dial-peers depends to which CCM (Publisher
> or Subscriber) is a gateway registered to?
> >
> > Thanks,
> >
> > Jordi!
> >
> >
> >
> > ________________________________
>  De: zohaib shabir [mailto:zohaibshabir at gmail.com]
> > Enviado el: jue 04/10/2007 14:02
> > Para: Jordi Riba (Europe)
> > CC: cisco-voip at puck.nether.net
> > Asunto: Re: [cisco-voip] Very Strange Behaviour
> >
> >
> > What you should do is to change the preference of the dial-peer pointing
> to the publisher to preference 1 or above. It will work hopefully.
> >
> >
> > On 10/4/07, Jordi Riba (Europe) <Jordi.Riba at eu.didata.com> wrote:
> > >
> > >
> > > Hi all,
> > >
> > > I'm gonna try to explain a very strange situation/behaviour I found
> yesterday in a customer site... I think this is not the way it should work
> but ... I would like some of you to re-check it.
> > >
> > > Thanks in advance...
> > >
> > >
> > > Initial Situation:
> > > - CCM 5.1.2
> > > - One headquarters office. (CCM Cluster on it 1 publisher - 3
> subscribers)
> > > - Three branchoffices (h323-srst gateway on them) (12.4.9-T3)
> > > - All phones and gateways in the branch are registerd to the publisher.
> > > - Route-patterns and route-list registered to the publisher too.
> > > - Gateways in the branch offices have two voip dialpeer (one pointing to
> the publisher (precedence 1)  and another pointing to one subscriber
> (precedence 2))
> > >
> > > Action Done:
> > > Changed callmanager group in the branch offices device pool to get
> branch gateways and phones registered to a subscriber CCM.
> > >
> > > Problem:
> > > After this change, Call coming from the PSTN to the gateway can not
> contact IP Phones. In q931 debug we get a "Network out of order" message.
> > >
> > > Workaround:
> > > If we deactivate the voice dialpeer pointing to the Publisher --> We
> work only with the dialpeer pointing to the subscriber. All works fine.
> > >
> > > If we deactivate the voice dialpeer pointing to the Subscriber --> We
> work only with the dialpeer pointing to the Publisher. We get network out of
> order.
> > >
> > > I think this is not the way it should work, is it??
> > >
> > > Thanks again,
> > >
> > > Jordi!
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> >
> > --
> > Regards,
> > Zohaib Shabir
> > Network Engineer(Voice)
> > DWP Group, TECH Division
> > Ph:+92-302-8232689
>
>
>
> --
> Regards,
> Zohaib Shabir
> Network Engineer(Voice)
> DWP Group, TECH Division
> Ph:+92-302-8232689
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From SCASPER at mtb.com  Thu Oct  4 10:07:44 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Thu, 04 Oct 2007 10:07:44 -0400
Subject: [cisco-voip]  SIP Addressing Change
In-Reply-To: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <4704BB73.9874.0039.0@mtb.com>

I am trying to implement a SIP trunking test with a carrier. I am sending them 
 
 From: "Cisco Test One" <sip:5422137 at 10.x.x.33> ( mailto:14104541137 at 10.99.0.33> ). 
 
They want to see
 
 From: "Cisco Test One" <sip:5422137@ ( mailto:14104541137 at 10.99.0.33> )test.ipttrunk.com> 
 
Is there a way to make this change on the router or Call Manager? The gateway is set up in Call Manager as a H323 gateway.
 
 
 
 
Thanks,

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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From deadpool.slayer.telecomm at gmail.com  Thu Oct  4 09:47:07 2007
From: deadpool.slayer.telecomm at gmail.com (Andrew F. Wolfe)
Date: Thu, 4 Oct 2007 08:47:07 -0500
Subject: [cisco-voip] CUE not picking up on outside calls
Message-ID: <913929690710040647k620f433dhb7f52cd1728028c7@mail.gmail.com>

Good Morning,

I have an issue that is odd.  My first CUE and CME implementation.  Internal
calls will CFB and CFNA to voicemail but outside callers never get
transferred to voicemail.  This is a native SIP trunk bringing in the calls
and they are delivering 4 digits.  I have the allow-connection sip to sip
command in the voice service voip section.  From what I have found on CCO
that should allow it but looking for any insight out there.  All help is
appreciated.  If you need to see any configs I can strip them down and send
you what you need.  Thanks.

Andrew
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From david.ponsdesserre at sgcib.com  Thu Oct  4 10:14:08 2007
From: david.ponsdesserre at sgcib.com (david.ponsdesserre at sgcib.com)
Date: Thu, 4 Oct 2007 15:14:08 +0100
Subject: [cisco-voip] Using ERSPAN to mirror voice traffic for voice
	recording [C1]
Message-ID: <OF802F0BFA.45D1D405-ONC125736A.004E2BA4-8025736A.004E31F2@fr.world.socgen>

Hello list . 

We are actually deploying IPT in here and i am working on the voice 
recording part of this project . 
I am actually thinking of using ERSPAN to mirror the voice traffic to our 
voice recorders on each of the our different building . 
Each individual access switches will have two ERSPAN session (s1 and s2) 
to send the voice traffic  the way down to the destinations switches where 
the Voice recorders will be plugged . 

Does anyone of you has already done the same thing ? Any input will be 
more than welcome . 

Thanks 

David Ponsdesserre
-------------------------------------------------------------------------
London Telecom and Network
Societe Generale Corporate and Investment Banking

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From malhatma at gmx.net  Thu Oct  4 10:12:10 2007
From: malhatma at gmx.net (malhatma)
Date: Thu, 04 Oct 2007 16:12:10 +0200
Subject: [cisco-voip] MCU question.
In-Reply-To: <470418ac.1735640a.7cac.4cee@mx.google.com>
References: <470418ac.1735640a.7cac.4cee@mx.google.com>
Message-ID: <4704F4BA.8080206@gmx.net>

Joel Perez schrieb:
> Hey Guys,
> 
> 
> Two hopefully quick questions.
> 
> 1) Im setting up an MCU 3515 and wanted to know what is the preference to
> use as the protocol, H323 or SCCP? This will be my first time setting one
> up.
> 
> 2) I had a customer ask me if there is a way to make 2 IPT's with two
> different ext's ring at the same time?
> Im mapping a DID to one of the IPT's ext's and the other ext does not have a
> DID. That same Ext with the DID is also a last resort for the AA if no one
> presses any menu options in AA. So it would have to work that way also. The
> AA is running on CUE 2.3.4 if that makes a difference.
> 
> This is on CUCM 5.1
> 
> Thanks,
> 
> Joel P
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

1) There is no real choice between H.323 or SCCP. It depends on the features you need.
SCCP supports Adhoc and Meetme conferencing.
H.323 or Sip supports "Dialin Conferencing"

2) Hunt-group with broadcast hunting?

regards
andre

From asobihoudai at yahoo.com  Thu Oct  4 10:32:53 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 4 Oct 2007 07:32:53 -0700 (PDT)
Subject: [cisco-voip] CUE not picking up on outside calls
In-Reply-To: <913929690710040647k620f433dhb7f52cd1728028c7@mail.gmail.com>
Message-ID: <201415.20422.qm@web57211.mail.re3.yahoo.com>

What codecs are you using for outside calls?

Paul

--- "Andrew F. Wolfe"
<deadpool.slayer.telecomm at gmail.com> wrote:

> Good Morning,
> 
> I have an issue that is odd.  My first CUE and CME
> implementation.  Internal
> calls will CFB and CFNA to voicemail but outside
> callers never get
> transferred to voicemail.  This is a native SIP
> trunk bringing in the calls
> and they are delivering 4 digits.  I have the
> allow-connection sip to sip
> command in the voice service voip section.  From
> what I have found on CCO
> that should allow it but looking for any insight out
> there.  All help is
> appreciated.  If you need to see any configs I can
> strip them down and send
> you what you need.  Thanks.
> 
> Andrew
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
____________________________________________________________________________________
Pinpoint customers who are looking for what you sell. 
http://searchmarketing.yahoo.com/

From greencucumber2007 at yahoo.com  Thu Oct  4 10:43:21 2007
From: greencucumber2007 at yahoo.com (Cucumber Green)
Date: Thu, 4 Oct 2007 07:43:21 -0700 (PDT)
Subject: [cisco-voip] Antw: Analog phones cant transfer calls
	on2811acting asH323 gw to UCM 4.2 (3)
In-Reply-To: <004601c7fc04$b40abda0$6501a8c0@digibox>
Message-ID: <867721.68229.qm@web44811.mail.sp1.yahoo.com>

Hey guys, my IOS doesnt support SCCP for FXS ports (12.4(3g)). So I did this to make it work:
   
  First, I downloaded the following TCL script 
             app-h450-transfer.2.0.0.9.zip
from
             http://www.cisco.com/pcgi-bin/tablebuild.pl/ip-iostsp

  Then added the following configuration
   
  !
  voice service voip 
 allow-connections h323 to h323 
 h323 
  !
  voice class h323 1 
 h225 timeout tcp establish 3 
  !
  application 
 service transfer_app flash:app-h450-transfer.2.0.0.9.tcl 
  paramspace english language en 
  paramspace english index 1 
  paramspace english location flash:prompts/en 
  paramspace english prefix en 
  tftp-server flash:prompts/en/en_music_xferee_ulaw.au 
tftp-server flash:prompts/en/en_author_fail_ulaw.au 
tftp-server flash:prompts/en/en_busy_tone_ulaw.au 
tftp-server flash:prompts/en/en_dest_blocked_ulaw.au 
tftp-server flash:prompts/en/en_dest_busy_ulaw.au 
tftp-server flash:prompts/en/en_disconnected_ulaw.au 
tftp-server flash:prompts/en/en_enter_dest_ulaw.au 
tftp-server flash:prompts/en/en_fast_busy_tone_ulaw.au 
tftp-server flash:prompts/en/en_music_xferto_ulaw.au 
tftp-server flash:prompts/en/en_no_dest_entered_ulaw.au 
tftp-server flash:prompts/en/en_no_service_available_ulaw.au 
tftp-server flash:prompts/en/en_operator_is_busy_ulaw.au 
tftp-server flash:prompts/en/en_reenter_dest_ulaw.au 
tftp-server flash:prompts/en/en_ringback_tone_ulaw.au 
  !
  dial-peer voice 10 pots 
 service transfer_app 
 destination-pattern 50001 
 port 0/1/0 
! 
dial-peer voice 20 pots 
 service transfer_app 
 destination-pattern 50002 
 port 0/1/1 
! 
dial-peer voice 1 voip 
 preference 1 
 service transfer_app 
 destination-pattern 1... 
 session target ipv4:x.x.x.x      
 dtmf-relay h245-alphanumeric 
 codec g729r8 bytes 40 
 ip qos dscp cs3 media 
 ip qos dscp cs3 signaling 
! 
dial-peer voice 4 voip 
 preference 2 
 service transfer_app 
 destination-pattern 1... 
 session target ipv4:y.y.y.y  
 dtmf-relay h245-alphanumeric 
 codec g729r8 bytes 40 
 ip qos dscp cs3 media 
 ip qos dscp cs3 signaling 
!
   
  Hookflash works now but I think I will  try downloading a better IOS later, and will use SCCP to get all the features working.
   
  Thanks for all the replies.
   


Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
          oh great....the various feature modes, one of which is called feature.
   
  i couldn't get over that the first time i read it. now it makes me laugh.
    ----- Original Message ----- 
  From: Ted Nugent 
  To: Jason Aarons (US) 
  Cc: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Friday, September 21, 2007 12:01 AM
  Subject: Re: [cisco-voip] Antw: Analog phones cant transfer calls on2811acting asH323 gw to UCM 4.2 (3)
  

  With Feature codes as you would with an ATA or VG248... check out table 3
   
  http://www.cisco.com/en/US/products/ps6706/products_feature_guide09186a00806cab75.html#wp1049432
  

 
  On 9/20/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:         How are you supposed to transfer a call using an analog phone on a VG224 that is SCCP? Can this be done? I didn't realize it was a feature!
   
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
Sent: Thursday, September 20, 2007 11:45 PM
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] Antw: Analog phones cant transfer calls on2811acting asH323 gw to UCM 4.2 (3)

     
    Yep same here and I wondered about that also until i tried to add an ISR FXS SCCP port to CM 4.1... its not an option like in 5.1 for example, I'm guessing thats what they're referring to...?


 

    On 9/20/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote: 
      I think the documentation is misleading...I'm using supplementary services w/ SCCP on the VG224 with CCM 4.1(3) with no problems.

        ----- Original Message ----- 

    From: Ted Nugent 

    To: Cucumber Green 

    Cc: cisco-voip at puck.nether.net 

    Sent: Thursday, September 20, 2007 10:21 PM

    Subject: Re: [cisco-voip] Antw: Analog phones cant transfer calls on 2811acting asH323 gw to UCM 4.2 (3)

    
 

    If you need features with fxs you might want to look into SCCP for FXS controller as you would with a VG224 and h323 for and PSTN/PRI access etc. (see the link you need CM4.2+ and a certain rev of IOS depending on your hardware). 

     

    http://www.cisco.com/en/US/partner/products/ps6441/products_feature_guide09186a0080483a76.html#wp1304455 

     

     

     

    

 

    On 9/20/07, Cucumber Green <greencucumber2007 at yahoo.com > wrote: 
    Hello!

     

    Yeah I have tried this without luck, I guess it only works if the H323 gateway has a FXO

    connection to a PBX

Robert Schuknecht <rschuknecht at gmx.de> wrote: 

    Hi Cucumber,

did you try to adjust the the hook-flash timer under the voice-port?
This timer has to equal on the voice-port and the connected telephone. 

/Robert

>>> Cucumber Green schrieb am Donnerstag, 20. 
    
September 2007 um 17:53 in Nachricht 3ab0adee131ef58c7afb0ea50017a410:
> Analog phones can send and receive calls from IP phones, but
> when pressing the hookflash key at the analog phone, the analog phone does 
> not get a dial tone in order to transfer the call :(
> 
> Phones are hooked into FXS ports on a 2811 and will use G729 codec.
> 
> I have already checked the "Media Termination Point required" check box in 
> the gateway configuration at call manager (using unified call manager 4.2(3) 
> ). I thought that would do the trick, but it didnt, any ideas?
> 
> This is the gateway config (IOS version 12.4(3g) spservicesK9) 
> 
> !
> version 12.4
> service timestamps debug datetime msec
> service timestamps log datetime msec
> service password-encryption
> !
> ip subnet-zero
> !
> ! 
> ip cef
> !
> !
> no ip domain lookup
> !
> voice-card 0
> no dspfarm
> !
> !
> !
> !
> interface Loopback0
> ip address X.X.X.X 255.255.255.255
> h323-gateway voip bind srcaddr X.X.X.X
> !
> !
> ip classless
> ip route 0.0.0.0 0.0.0.0 X.X.X.X 254
> !
> !
> control-plane
> !
> !
> !
> voice-port 0/1/0 
> signal loopStart
> !
> voice-port 0/1/1
> signal loopStart
> !
> dial-peer voice 10 pots
> destination-pattern 50001
> port 0/1/0
> !
> dial-peer voice 20 pots 
> destination-pattern 50002
> port 0/1/1
> !
> dial-peer voice 1 voip
> preference 1
> destination-pattern 1...
> session target ipv4:X.X.X.X
> dtmf-relay h245-alphanumeric 
> codec g729r8 bytes 40
> ip qos dscp cs3 media
> ip qos dscp cs3 signaling
> !
> dial-peer voice 4 voip
> preference 2
> destination-pattern 1...
> session target ipv4:X.X.X.X 
> dtmf-relay h245-alphanumeric
> codec g729r8 bytes 40
> ip qos dscp cs3 media
> ip qos dscp cs3 signaling
> !
> 
> 
> 
> ---------------------------------
> Be a better Globetrotter. Get better travel answers from someone who knows. 
> Yahoo! Answers - Check it out.

   
    
---------------------------------
  
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_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip

   

    
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https://puck.nether.net/mailman/listinfo/cisco-voip 


   



      
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From rduric at uoguelph.ca  Thu Oct  4 10:52:34 2007
From: rduric at uoguelph.ca (Rasim Duric)
Date: Thu, 4 Oct 2007 10:52:34 -0400
Subject: [cisco-voip] CUE not picking up on outside calls
In-Reply-To: <913929690710040647k620f433dhb7f52cd1728028c7@mail.gmail.com>
References: <913929690710040647k620f433dhb7f52cd1728028c7@mail.gmail.com>
Message-ID: <BD89624EE83BD349BD31A08B453DE24401C67E55@uogexb201.cfs.uoguelph.ca>

Check your system AA script parameter 'allowExternalTransfers'.  Should
be 'TRUE'.

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew F. Wolfe
Sent: Thursday, October 04, 2007 9:47 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CUE not picking up on outside calls

 

Good Morning,

I have an issue that is odd.  My first CUE and CME implementation.
Internal calls will CFB and CFNA to voicemail but outside callers never
get transferred to voicemail.  This is a native SIP trunk bringing in
the calls and they are delivering 4 digits.  I have the allow-connection
sip to sip command in the voice service voip section.  From what I have
found on CCO that should allow it but looking for any insight out there.
All help is appreciated.  If you need to see any configs I can strip
them down and send you what you need.  Thanks.  

Andrew

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From deadpool.slayer.telecomm at gmail.com  Thu Oct  4 11:12:18 2007
From: deadpool.slayer.telecomm at gmail.com (Andrew F. Wolfe)
Date: Thu, 4 Oct 2007 10:12:18 -0500
Subject: [cisco-voip] CUE not picking up on outside calls
In-Reply-To: <BD89624EE83BD349BD31A08B453DE24401C67E55@uogexb201.cfs.uoguelph.ca>
References: <913929690710040647k620f433dhb7f52cd1728028c7@mail.gmail.com>
	<BD89624EE83BD349BD31A08B453DE24401C67E55@uogexb201.cfs.uoguelph.ca>
Message-ID: <913929690710040812m182f7212o5695522b99d3a88@mail.gmail.com>

CODEC is G.711.  I will check that parameter.  Thanks guys.

On 10/4/07, Rasim Duric <rduric at uoguelph.ca> wrote:
>
>  Check your system AA script parameter 'allowExternalTransfers'.  Should
> be 'TRUE'.
>
>
>
> Rasim Duric
> Network Analyst (CCS)
> University of Guelph
> Guelph, N1G 2W1, ON
> 519-824-4120x53146
> rduric at uoguelph.ca
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Andrew F. Wolfe
> *Sent:* Thursday, October 04, 2007 9:47 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] CUE not picking up on outside calls
>
>
>
> Good Morning,
>
> I have an issue that is odd.  My first CUE and CME implementation.
> Internal calls will CFB and CFNA to voicemail but outside callers never get
> transferred to voicemail.  This is a native SIP trunk bringing in the calls
> and they are delivering 4 digits.  I have the allow-connection sip to sip
> command in the voice service voip section.  From what I have found on CCO
> that should allow it but looking for any insight out there.  All help is
> appreciated.  If you need to see any configs I can strip them down and send
> you what you need.  Thanks.
>
> Andrew
>
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From asobihoudai at yahoo.com  Thu Oct  4 11:18:17 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 4 Oct 2007 08:18:17 -0700 (PDT)
Subject: [cisco-voip] CUE not picking up on outside calls
In-Reply-To: <913929690710040812m182f7212o5695522b99d3a88@mail.gmail.com>
Message-ID: <662614.19704.qm@web57209.mail.re3.yahoo.com>

I don't think that AA parameter has anything to do
with this issue since that is more related to enabling
whether callers can transfer to DNs outside of the
range specified within CCME/CUE.

Does the phone keep ringing after your CNFA timeout or
does it just go to fast busy?

Paul

--- "Andrew F. Wolfe"
<deadpool.slayer.telecomm at gmail.com> wrote:

> CODEC is G.711.  I will check that parameter. 
> Thanks guys.
> 
> On 10/4/07, Rasim Duric <rduric at uoguelph.ca> wrote:
> >
> >  Check your system AA script parameter
> 'allowExternalTransfers'.  Should
> > be 'TRUE'.
> >
> >
> >
> > Rasim Duric
> > Network Analyst (CCS)
> > University of Guelph
> > Guelph, N1G 2W1, ON
> > 519-824-4120x53146
> > rduric at uoguelph.ca
> >
> > *From:* cisco-voip-bounces at puck.nether.net
> [mailto:
> > cisco-voip-bounces at puck.nether.net] *On Behalf Of
> *Andrew F. Wolfe
> > *Sent:* Thursday, October 04, 2007 9:47 AM
> > *To:* cisco-voip at puck.nether.net
> > *Subject:* [cisco-voip] CUE not picking up on
> outside calls
> >
> >
> >
> > Good Morning,
> >
> > I have an issue that is odd.  My first CUE and CME
> implementation.
> > Internal calls will CFB and CFNA to voicemail but
> outside callers never get
> > transferred to voicemail.  This is a native SIP
> trunk bringing in the calls
> > and they are delivering 4 digits.  I have the
> allow-connection sip to sip
> > command in the voice service voip section.  From
> what I have found on CCO
> > that should allow it but looking for any insight
> out there.  All help is
> > appreciated.  If you need to see any configs I can
> strip them down and send
> > you what you need.  Thanks.
> >
> > Andrew
> >
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



      ____________________________________________________________________________________
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From ealeatherman at gmail.com  Thu Oct  4 11:32:14 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 4 Oct 2007 11:32:14 -0400
Subject: [cisco-voip] IPCC Express Reporting access
Message-ID: <94a1afde0710040832u7daead82jd4244b1d0c3aa16e@mail.gmail.com>

Curious how people handle reporting with IPCC Express.. we run a number of
smaller call centers off of our IPCC Express system. Up till now we've just
been generating reports using the historical reporting tool, but there are
some units now that would like the reports to either be in a different
format or totally different reports altogether than what we can get from the
canned tool.

Do most people use Crystal to accomplish this? Anyone provided an ODBC
connection to more database savvy users? My database knowledge is pretty
limited and I know zilch about crystal, trying to figure out the best way
(security and performance wise) to handle it and still provide the info
these folks are asking for. They already understand they will be doing
anything special with reporting after-hours.

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From pfabian at telesphere.com  Thu Oct  4 11:37:37 2007
From: pfabian at telesphere.com (Pete Fabian)
Date: Thu, 4 Oct 2007 08:37:37 -0700
Subject: [cisco-voip] FW:   SIP Addressing Change
Message-ID: <006d01c8069c$7ff86830$b3d2a8c0@scf1.telesphere.com>

I am not sure I understand your topology

Are you doing:

CCM -> SIP Trunk -> SIP Provider or

CCM -> H323 -> H323Gateway -> SIP Trunk -> Sip Provider

Also, what version of call manager/IOS version are you running

In CCM 5.1 you can configure under Enterprise Parameters a Cluster Fully
Qualified Domain Name, that is probably what you are looking for.  Also, I
have found in many cases we have had to configure the SIP Trunk on CCM to
Require MTP

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of STEVEN CASPER
Sent: Thursday, October 04, 2007 7:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SIP Addressing Change

 

I am trying to implement a SIP trunking test with a carrier. I am sending
them 

 

 From:  <mailto:14104541137 at 10.99.0.33%3e> "Cisco Test One"
<sip:5422137 at 10.x.x.33>. 

 

They want to see

 

 From:  <mailto:14104541137 at 10.99.0.33%3e> "Cisco Test One"
<sip:5422137 at test.ipttrunk.com> 

 

Is there a way to make this change on the router or Call Manager? The
gateway is set up in Call Manager as a H323 gateway.

 

 

 

 

Thanks,

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026 

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This email may contain privileged and/or confidential information that is
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From deadpool.slayer.telecomm at gmail.com  Thu Oct  4 11:46:24 2007
From: deadpool.slayer.telecomm at gmail.com (Andrew F. Wolfe)
Date: Thu, 4 Oct 2007 10:46:24 -0500
Subject: [cisco-voip] CUE not picking up on outside calls
In-Reply-To: <662614.19704.qm@web57209.mail.re3.yahoo.com>
References: <913929690710040812m182f7212o5695522b99d3a88@mail.gmail.com>
	<662614.19704.qm@web57209.mail.re3.yahoo.com>
Message-ID: <913929690710040846x757ad7bbo75606ad3d7449e9c@mail.gmail.com>

it keeps ringing.

On 10/4/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> I don't think that AA parameter has anything to do
> with this issue since that is more related to enabling
> whether callers can transfer to DNs outside of the
> range specified within CCME/CUE.
>
> Does the phone keep ringing after your CNFA timeout or
> does it just go to fast busy?
>
> Paul
>
> --- "Andrew F. Wolfe"
> <deadpool.slayer.telecomm at gmail.com> wrote:
>
> > CODEC is G.711.  I will check that parameter.
> > Thanks guys.
> >
> > On 10/4/07, Rasim Duric <rduric at uoguelph.ca> wrote:
> > >
> > >  Check your system AA script parameter
> > 'allowExternalTransfers'.  Should
> > > be 'TRUE'.
> > >
> > >
> > >
> > > Rasim Duric
> > > Network Analyst (CCS)
> > > University of Guelph
> > > Guelph, N1G 2W1, ON
> > > 519-824-4120x53146
> > > rduric at uoguelph.ca
> > >
> > > *From:* cisco-voip-bounces at puck.nether.net
> > [mailto:
> > > cisco-voip-bounces at puck.nether.net] *On Behalf Of
> > *Andrew F. Wolfe
> > > *Sent:* Thursday, October 04, 2007 9:47 AM
> > > *To:* cisco-voip at puck.nether.net
> > > *Subject:* [cisco-voip] CUE not picking up on
> > outside calls
> > >
> > >
> > >
> > > Good Morning,
> > >
> > > I have an issue that is odd.  My first CUE and CME
> > implementation.
> > > Internal calls will CFB and CFNA to voicemail but
> > outside callers never get
> > > transferred to voicemail.  This is a native SIP
> > trunk bringing in the calls
> > > and they are delivering 4 digits.  I have the
> > allow-connection sip to sip
> > > command in the voice service voip section.  From
> > what I have found on CCO
> > > that should allow it but looking for any insight
> > out there.  All help is
> > > appreciated.  If you need to see any configs I can
> > strip them down and send
> > > you what you need.  Thanks.
> > >
> > > Andrew
> > >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ____________________________________________________________________________________
> Tonight's top picks. What will you watch tonight? Preview the hottest
> shows on Yahoo! TV.
> http://tv.yahoo.com/
>
>
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From erickbee at gmail.com  Thu Oct  4 12:03:27 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Thu, 4 Oct 2007 11:03:27 -0500
Subject: [cisco-voip] MGCP FXO port and sending no digits in CCM 5.x
Message-ID: <f4445faf0710040903l1b0027dcn4625203e478a85ad@mail.gmail.com>

Is there another trick to doing this in CCM 5.x then 4.x? In 4.x I
would set up the route pattern as say 1234. and do a PreDot digit
strip. I tried this in CCM 5.x for a MGCP FXO port going to a paging
system and it did not work, said unknown number and gave reorder tone
when route pattern was 1234.  but 1234 works and connects but sends
1234 out port which doesn't work with the paging system.  I made it
H323 but was wondering how to just have a FXO port go offhook and send
no digits with MGCP in 5.x.

From greencucumber2007 at yahoo.com  Thu Oct  4 12:35:48 2007
From: greencucumber2007 at yahoo.com (Cucumber Green)
Date: Thu, 4 Oct 2007 09:35:48 -0700 (PDT)
Subject: [cisco-voip] IPCC Express Reporting access
In-Reply-To: <94a1afde0710040832u7daead82jd4244b1d0c3aa16e@mail.gmail.com>
Message-ID: <592843.1986.qm@web44803.mail.sp1.yahoo.com>

I think you should take a look into the db_cra database. Specifically the table ContactCallDetail.
  If the data of your contact centers is stored in the same database, in the same table, there would be a security concern by giving one of your users access through ODBC, as he/she would be able to look at records of other contact centers.
   
  I have done custom reports using Crystal Reports (which rely on SQL Server stored procedures), never used ODBC to extract the data directly.

Ed Leatherman <ealeatherman at gmail.com> wrote:
  Curious how people handle reporting with IPCC Express.. we run a number of smaller call centers off of our IPCC Express system. Up till now we've just been generating reports using the historical reporting tool, but there are some units now that would like the reports to either be in a different format or totally different reports altogether than what we can get from the canned tool. 

Do most people use Crystal to accomplish this? Anyone provided an ODBC connection to more database savvy users? My database knowledge is pretty limited and I know zilch about crystal, trying to figure out the best way (security and performance wise) to handle it and still provide the info these folks are asking for. They already understand they will be doing anything special with reporting after-hours. 

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

       
---------------------------------
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From jgrace at adcapnet.com  Thu Oct  4 12:44:49 2007
From: jgrace at adcapnet.com (James Grace)
Date: Thu, 4 Oct 2007 12:44:49 -0400
Subject: [cisco-voip] IPCCX realtime monitor page
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E86A@mail3.adcapnetsvc.local>

?
Do any body know what service runs the realtime monitor page in IPCCX.  It keeps locking up and i want to reset it without having to reboot the server.
 
James Grace 
 
Systems Engineer
<http://www.adcapnet.com/>  
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
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From Thorsten.Mayr at barclayscapital.com  Thu Oct  4 12:08:24 2007
From: Thorsten.Mayr at barclayscapital.com (Thorsten.Mayr at barclayscapital.com)
Date: Thu, 4 Oct 2007 17:08:24 +0100
Subject: [cisco-voip] Meeting Place question
Message-ID: <0D198DEE3BD3174682375A4FD219F8290ADB95EE@LDNPCMEU304VEUA.INTRANET.BARCAPINT.COM>

Not sure if this is the right group to discuss this, but has anybody
come across the following behaviour on meeting place:

When participants disconnect from a conference a constant tone appears
and won't clear. This behaviour is random.
Test calls from various systems have been made and the only thing we
have been able to proof so far is that it definitely is absolute random
behaviour..

The meeting place has 3 directly connected trunks into it...

grateful for any hints... 

cheers
T
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From mjdsy9 at gmail.com  Thu Oct  4 13:13:14 2007
From: mjdsy9 at gmail.com (Michael Sy)
Date: Thu, 4 Oct 2007 10:13:14 -0700
Subject: [cisco-voip] ATA
Message-ID: <228dc9bd0710041013p73fcc61ch4cf2f524013a69f@mail.gmail.com>

ATA new install not coming up at all, red blinkies.  any assistance pls.
ccm 4.2.3 es44

-- 
Mike Sy .
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From patrick.diener at gmail.com  Thu Oct  4 15:44:25 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Thu, 4 Oct 2007 21:44:25 +0200
Subject: [cisco-voip] ATA
In-Reply-To: <228dc9bd0710041013p73fcc61ch4cf2f524013a69f@mail.gmail.com>
References: <228dc9bd0710041013p73fcc61ch4cf2f524013a69f@mail.gmail.com>
Message-ID: <c0deb5b80710041244r7b3444d1j38af009fa4009241@mail.gmail.com>

check the dhcp server to see if the ata gets a ip address and stuff
(default gw, tftp)

Regards
Patrick


On 10/4/07, Michael Sy <mjdsy9 at gmail.com> wrote:
> ATA new install not coming up at all, red blinkies.  any assistance pls.
> ccm 4.2.3 es44
>
> --
> Mike Sy .
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From CDAmore at jackhenry.com  Thu Oct  4 15:50:50 2007
From: CDAmore at jackhenry.com (Chris D'amore)
Date: Thu, 4 Oct 2007 14:50:50 -0500
Subject: [cisco-voip] ATA
In-Reply-To: <c0deb5b80710041244r7b3444d1j38af009fa4009241@mail.gmail.com>
References: <228dc9bd0710041013p73fcc61ch4cf2f524013a69f@mail.gmail.com>
	<c0deb5b80710041244r7b3444d1j38af009fa4009241@mail.gmail.com>
Message-ID: <2B3745BC9D31514381A6FCC5E72D9373026AABFA@SMOMAIL.jhacorp.com>

You might try forcing the switch port to 10/half.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Patrick Diener
Sent: Thursday, October 04, 2007 2:44 PM
To: Michael Sy; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ATA

check the dhcp server to see if the ata gets a ip address and stuff
(default gw, tftp)

Regards
Patrick


On 10/4/07, Michael Sy <mjdsy9 at gmail.com> wrote:
> ATA new install not coming up at all, red blinkies.  any assistance
pls.
> ccm 4.2.3 es44
>
> --
> Mike Sy .
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
NOTICE: This electronic mail message and any files transmitted with it are intended
exclusively for the individual or entity to which it is addressed. The message, 
together with any attachment, may contain confidential and/or privileged information.
Any unauthorized review, use, printing, saving, copying, disclosure or distribution 
is strictly prohibited. If you have received this message in error, please 
immediately advise the sender by reply email and delete all copies.

From jonvoip at gmail.com  Thu Oct  4 17:17:42 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 4 Oct 2007 16:17:42 -0500
Subject: [cisco-voip] ATA
In-Reply-To: <228dc9bd0710041013p73fcc61ch4cf2f524013a69f@mail.gmail.com>
References: <228dc9bd0710041013p73fcc61ch4cf2f524013a69f@mail.gmail.com>
Message-ID: <5d093f9a0710041417w12b6d3e8je4dd11ad5caff333@mail.gmail.com>

Attach a phone.

Press the button on top of the ATA.

When the phone rings, answer it.

Enter:

#80

And it will tell you what IP address it is...


Jonathan

On 10/4/07, Michael Sy <mjdsy9 at gmail.com> wrote:
> ATA new install not coming up at all, red blinkies.  any assistance pls.
> ccm 4.2.3 es44
>
> --
> Mike Sy .
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From dustin.s.fowler at gmail.com  Thu Oct  4 22:25:42 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Thu, 4 Oct 2007 22:25:42 -0400
Subject: [cisco-voip] IPCC/UCCX 3.5
Message-ID: <4705a0b2.0786460a.46bc.ffffe76a@mx.google.com>

 

 Good Evening,

 

Has anyone seen the following error when trying to have a supervisor monitor an agent:

 

Supervisor has failed to start voice monitor on the device: 5962

 

Any assistance is greatly appreciated.

 

 

Thank You,

 

Dustin S. Fowler 

Senior Cisco Engineer 

Data Corporation 

18503 Pines Blvd, Suite 213, 

Pembroke Pines, FL,33029 

954-671-5667

dfowler at data-corporation.com

 

 

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From keli at carocomp.ro  Fri Oct  5 02:52:13 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Fri, 05 Oct 2007 09:52:13 +0300
Subject: [cisco-voip] Disaster Recovery: Using ghost to restore
 MCS	Servers maybe?
In-Reply-To: <47028C38.4080806@cisco.com>
References: <88C8D019DA547C4BABECB9A806ED22ED01380EDF@mail.ACT-EG.COM>
	<47028C38.4080806@cisco.com>
Message-ID: <4705DF1D.10303@carocomp.ro>

I was referring to / using ghost strictly as a backup tool, so a dead 
CCM could be recovered from an image created from *that* CCM earlier. I 
never assumed to create multiple installations from the same ghost 
image, or to recover a server from a different server's ghost. It seems 
most useful for RAID-less 7815 installations, but it could come in handy 
with larger servers as well, if something goes terribly wrong :)

Thus there shouldn't be a case of duplicate GUID's.

Or am I mistaken?

Zoltan

Wes Sisk wrote:
> duplicate GUID's causes problems for SQL replication and DCOM 
> connectivity.
>
> ghost is not recommended.
>
> Ahmed Elnagar wrote:
>>
>> AFAIK it is not supported with TAC, but have you tried it several times?
>>
>>  
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net 
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
>> Sent: Tuesday, September 18, 2007 12:37 PM
>> To: Cucumber Green
>> Cc: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] Disaster Recovery: Using ghost to restore 
>> MCS Servers maybe?
>>
>>  
>>
>> Ghost definitely works, however it's not worthwhile to make frequent
>>
>> ghosts. So (just as others have put it) make an image after major
>>
>> changes and use BARS in between.
>>
>>  
>>
>> Major changes being software updates, major component changes in the
>>
>> system (new gateways, VM, etc.)
>>
>>  
>>
>> regards,
>>
>>   Zoltan
>>
>>  
>>
>> Cucumber Green wrote:
>>
>> > Hello
>>
>> > 
>>
>> > Has anyone used Ghost to keep an image of your MCS servers in order to
>>
>> > use it in case of a server crash? If so, did it work, are there any
>>
>> > caveats, tips?
>>
>> > 
>>
>> > If not using Ghost, whats your disaster recovery procedure to restore
>>
>> > an MCS crashed server ( other than a full reinstall ).
>>
>> > 
>>
>> > 
>>
>> > 
>>
>> > 
>>
>> > ------------------------------------------------------------------------
>>
>> > Boardwalk for $500? In 2007? Ha!
>>
>> > Play Monopoly Here and Now
>>
>> > <http://us.rd.yahoo.com/evt=48223/*http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow> 
>>
>>
>> > (it's updated for today's economy) at Yahoo! Games.
>>
>> > ------------------------------------------------------------------------
>>
>> > 
>>
>> > _______________________________________________
>>
>> > cisco-voip mailing list
>>
>> > cisco-voip at puck.nether.net
>>
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>  
>>
>> _______________________________________________
>>
>> cisco-voip mailing list
>>
>> cisco-voip at puck.nether.net
>>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip


From tele at plexialab.it  Fri Oct  5 07:56:27 2007
From: tele at plexialab.it (tele)
Date: Fri, 05 Oct 2007 13:56:27 +0200
Subject: [cisco-voip] SIP Addressing Change
In-Reply-To: <4704BB73.9874.0039.0@mtb.com>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
	<4704BB73.9874.0039.0@mtb.com>
Message-ID: <1191585387.6476.33.camel@mda>

10.x.x.33 is the IP of your cisco?
I'm not sure but you can try to manually set the host, you should see
the hostname and not the IP.

ip host test.ipttrun.com 10.x.x.33

if this not work you can do it with custom TCL script:

http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/tclappa.htm#wp1007665

:tele

On Thu, 2007-10-04 at 10:07 -0400, STEVEN CASPER wrote:
> I am trying to implement a SIP trunking test with a carrier. I am
> sending them 
>  
>  From: "Cisco Test One" <sip:5422137 at 10.x.x.33>. 
>  
> They want to see
>  
>  From: "Cisco Test One" <sip:5422137 at test.ipttrunk.com> 
>  
> Is there a way to make this change on the router or Call Manager? The
> gateway is set up in Call Manager as a H323 gateway.
>  
>  
>  
>  
> Thanks,
> 
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026 
> 
> ************************************
> This email may contain privileged and/or confidential information that
> is intended solely for the use of the addressee. If you are not the
> intended recipient or entity, you are strictly prohibited from
> disclosing, copying, distributing or using any of the information
> contained in the transmission. If you received this communication in
> error, please contact the sender immediately and destroy the material
> in its entirety, whether electronic or hard copy. This communication
> may contain nonpublic personal information about consumers subject to
> the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley
> Act. You may not directly or indirectly reuse or disclose such
> information for any purpose other than to provide the services for
> which you are receiving the information.
> There are risks associated with the use of electronic transmission.
> The sender of this information does not control the method of
> transmittal or service providers and assumes no duty or obligation for
> the security, receipt, or third party interception of this
> transmission.
> ************************************
> 
> 
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jonvoip at gmail.com  Fri Oct  5 08:27:47 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 5 Oct 2007 07:27:47 -0500
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
Message-ID: <5d093f9a0710050527u3608da18gdbebca41544fc0de@mail.gmail.com>

So, I am curious why all of us are putting up with this.

For years we were all told that Linux was superior, that Linux was how
a real web server should function...

Now, we have CCM 5 and CCM 6... and the web server performance, is,
er, well, how do we put this...

Here we go. It is far worse than anything Microsoft has ever released.
I think Microsoft BOB would be a better web server than whatever crap
is on the backend of this RedHat monstrosity.

In CallManger 4.3, using Microsoft WINDOWS(!!!!!!)  IIS and Cisco
Tomcat, I can scream through all of the web pages; do DB reads in
nanoseconds; do DB inserts almost as quick, and I have never seen a
message on my web browser saying, 'Loading, Please Wait'

I mean, seriously, this is godawful slow.

I just installed CCM5.1.3 on an HP 7845, 4GB of RAM, dual dual core
Xeons... and all I see at every mouse click is 'Loading, Please
Wait'..

Is Cisco punishing us for something? Did we offend them in some way? I
am at a point where I am wondering if I should tell customers about
these performance issues prior to picking a version ('if you want to
be able to administer this and not slash your wrists, waiting for the
home page to load, you may want to go with 4.3...') because I am tired
of hearing about it afterwards...

Seriously guys, I mean, this pretty much kills whatever argument you
Linux freaks had that this was better than Windows... add the whole
SFTP requirement, and you can see why the Microsoft side won this
argument...



Jonathan

From MillerS at DicksteinShapiro.COM  Fri Oct  5 09:27:33 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Fri, 5 Oct 2007 09:27:33 -0400
Subject: [cisco-voip] POWER SHUTDOWNS and verifying PRI Function Afterwards
Message-ID: <421CAA1A5300B84A87D200D7099857820F4399@DCEX2.DSMO.COM>

Would someone please let me know how to check the PRIs after a power
outage to see if they are enabled or disabled?  Is the gateway
registering to the Call Manager a good thing to look at?  Our power will
be off for about 12 hours this Saturday and I have notified the carrier,
but I need to be sure there is nothing wrong on our end if the phones
can't dial and I also need to know how to re-enable from our end.  Any
assistance would be greatly appreciated. Thank you!
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
responsible for delivering this confidential communication 
to the intended recipient, you have received this 
communication in error, and any review, use, dissemination, 
forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
network.  If you have received this confidential 
communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
the original message. 

To reply to our email administrator directly, send an email 
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Dickstein Shapiro LLP
http://www.DicksteinShapiro.com

==============================================================================
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From asobihoudai at yahoo.com  Fri Oct  5 09:29:34 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 06:29:34 -0700 (PDT)
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <5d093f9a0710050527u3608da18gdbebca41544fc0de@mail.gmail.com>
Message-ID: <538320.58722.qm@web57202.mail.re3.yahoo.com>

Linux is just an OS. What people put on top of it is
largely their choice. If Cisco installed wine and ran
IIS on top of that for their web server, then that
would likely run slowly. 

I am not informed on the Linux-based CCM boxen but I'd
certainly hope that Cisco is using apache to serve its
webpages. I've heard that the DB is Informix or
something like that.

My point being, it doesn't matter what OS you are
using - if you put a slow application on it, it'll
still run slow no matter what you do to it.

Paul

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> So, I am curious why all of us are putting up with
> this.
> 
> For years we were all told that Linux was superior,
> that Linux was how
> a real web server should function...
> 
> Now, we have CCM 5 and CCM 6... and the web server
> performance, is,
> er, well, how do we put this...
> 
> Here we go. It is far worse than anything Microsoft
> has ever released.
> I think Microsoft BOB would be a better web server
> than whatever crap
> is on the backend of this RedHat monstrosity.
> 
> In CallManger 4.3, using Microsoft WINDOWS(!!!!!!) 
> IIS and Cisco
> Tomcat, I can scream through all of the web pages;
> do DB reads in
> nanoseconds; do DB inserts almost as quick, and I
> have never seen a
> message on my web browser saying, 'Loading, Please
> Wait'
> 
> I mean, seriously, this is godawful slow.
> 
> I just installed CCM5.1.3 on an HP 7845, 4GB of RAM,
> dual dual core
> Xeons... and all I see at every mouse click is
> 'Loading, Please
> Wait'..
> 
> Is Cisco punishing us for something? Did we offend
> them in some way? I
> am at a point where I am wondering if I should tell
> customers about
> these performance issues prior to picking a version
> ('if you want to
> be able to administer this and not slash your
> wrists, waiting for the
> home page to load, you may want to go with 4.3...')
> because I am tired
> of hearing about it afterwards...
> 
> Seriously guys, I mean, this pretty much kills
> whatever argument you
> Linux freaks had that this was better than
> Windows... add the whole
> SFTP requirement, and you can see why the Microsoft
> side won this
> argument...
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  

From asobihoudai at yahoo.com  Fri Oct  5 09:31:15 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 06:31:15 -0700 (PDT)
Subject: [cisco-voip] POWER SHUTDOWNS and verifying PRI Function
	Afterwards
In-Reply-To: <421CAA1A5300B84A87D200D7099857820F4399@DCEX2.DSMO.COM>
Message-ID: <638678.59365.qm@web57202.mail.re3.yahoo.com>

Call into the office where the PRI is connected. At
the same time, inspect the router with 'debug isdn
q931' with 'terminal monitor' on. If the PRI is dead,
you will see nothing.

Paul

--- "Miller, Steve" <MillerS at DicksteinShapiro.COM>
wrote:

> Would someone please let me know how to check the
> PRIs after a power
> outage to see if they are enabled or disabled?  Is
> the gateway
> registering to the Call Manager a good thing to look
> at?  Our power will
> be off for about 12 hours this Saturday and I have
> notified the carrier,
> but I need to be sure there is nothing wrong on our
> end if the phones
> can't dial and I also need to know how to re-enable
> from our end.  Any
> assistance would be greatly appreciated. Thank you!
>  
> 
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> MillerS at dicksteinshapiro.com 
> 
>  
> 
>
--------------------------------------------------------
> This e-mail message and any attached files are
> confidential
> and are intended solely for the use of the
> addressee(s) 
> named above. This communication may contain material
> 
> protected by attorney-client, work product, or other
> 
> privileges. If you are not the intended recipient or
> person 
> responsible for delivering this confidential
> communication 
> to the intended recipient, you have received this 
> communication in error, and any review, use,
> dissemination, 
> forwarding, printing, copying, or other distribution
> of 
> this e-mail message and any attached files is
> strictly 
> prohibited. Dickstein Shapiro reserves the right to
> monitor 
> any communication that is created, received, or sent
> on its 
> network.  If you have received this confidential 
> communication in error, please notify the sender 
> immediately by reply e-mail message and permanently
> delete 
> the original message. 
> 
> To reply to our email administrator directly, send
> an email 
> to postmaster at dicksteinshapiro.com
> 
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
> 
>
==============================================================================
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
____________________________________________________________________________________
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  

From pfabian at telesphere.com  Fri Oct  5 09:34:27 2007
From: pfabian at telesphere.com (Pete Fabian)
Date: Fri, 5 Oct 2007 06:34:27 -0700
Subject: [cisco-voip] POWER SHUTDOWNS and verifying PRI Function
	Afterwards
In-Reply-To: <421CAA1A5300B84A87D200D7099857820F4399@DCEX2.DSMO.COM>
Message-ID: <001901c80754$73e2f480$bbd2a8c0@scf1.telesphere.com>

Show isdn status and look for the layer 1 state as active and layer 2 state
as Mulitple frame established

You can run a debug isdn q921 to see if sapi messages in both directions:

            Oct  5 06:33:56: ISDN Se1/0/0:23 Q921: User TX -> RRp sapi=0
tei=0 nr=60

Oct  5 06:33:56: ISDN Se1/0/0:23 Q921: User RX <- RRf sapi=0 tei=0 nr=15

Oct  5 06:34:02: ISDN Se1/0/1:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0

Oct  5 06:34:02: ISDN Se1/0/1:23 Q921: User RX <- RRf sapi=0 tei=0 nr=0

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Friday, October 05, 2007 6:28 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] POWER SHUTDOWNS and verifying PRI Function Afterwards

 

Would someone please let me know how to check the PRIs after a power outage
to see if they are enabled or disabled?  Is the gateway registering to the
Call Manager a good thing to look at?  Our power will be off for about 12
hours this Saturday and I have notified the carrier, but I need to be sure
there is nothing wrong on our end if the phones can't dial and I also need
to know how to re-enable from our end.  Any assistance would be greatly
appreciated. Thank you!

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
responsible for delivering this confidential communication 
to the intended recipient, you have received this 
communication in error, and any review, use, dissemination, 
forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
network.  If you have received this confidential 
communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
the original message. 
 
To reply to our email administrator directly, send an email 
to postmaster at dicksteinshapiro.com
 
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
 
============================================================================
==
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From cbomba at s4nets.com  Fri Oct  5 09:08:47 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Fri, 5 Oct 2007 09:08:47 -0400 (EDT)
Subject: [cisco-voip] QIP anyone?
Message-ID: <43017.192.168.1.35.1191589727.webmail@192.168.1.35>

Anyone using QIP for DHCP?  I have a client that will be implementing Cisco IPT and is using QIP for their DHPC application.  Wanted to know if there were any gotchas or anything to look out for before we get started.

Thanks,

Chris




From wsisk at cisco.com  Fri Oct  5 10:14:34 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 05 Oct 2007 10:14:34 -0400
Subject: [cisco-voip] Disaster Recovery: Using ghost to restore
 MCS	Servers maybe?
In-Reply-To: <4705DF1D.10303@carocomp.ro>
References: <88C8D019DA547C4BABECB9A806ED22ED01380EDF@mail.ACT-EG.COM>
	<47028C38.4080806@cisco.com> <4705DF1D.10303@carocomp.ro>
Message-ID: <470646CA.4070002@cisco.com>

in that case you are going to break SQL replication of the CCM database.

Kelemen Zoltan wrote:
> I was referring to / using ghost strictly as a backup tool, so a dead 
> CCM could be recovered from an image created from *that* CCM earlier. 
> I never assumed to create multiple installations from the same ghost 
> image, or to recover a server from a different server's ghost. It 
> seems most useful for RAID-less 7815 installations, but it could come 
> in handy with larger servers as well, if something goes terribly wrong :)
>
> Thus there shouldn't be a case of duplicate GUID's.
>
> Or am I mistaken?
>
> Zoltan
>
> Wes Sisk wrote:
>> duplicate GUID's causes problems for SQL replication and DCOM 
>> connectivity.
>>
>> ghost is not recommended.
>>
>> Ahmed Elnagar wrote:
>>>
>>> AFAIK it is not supported with TAC, but have you tried it several 
>>> times?
>>>
>>>  
>>>
>>> -----Original Message-----
>>> From: cisco-voip-bounces at puck.nether.net 
>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
>>> Sent: Tuesday, September 18, 2007 12:37 PM
>>> To: Cucumber Green
>>> Cc: cisco-voip at puck.nether.net
>>> Subject: Re: [cisco-voip] Disaster Recovery: Using ghost to restore 
>>> MCS Servers maybe?
>>>
>>>  
>>>
>>> Ghost definitely works, however it's not worthwhile to make frequent
>>>
>>> ghosts. So (just as others have put it) make an image after major
>>>
>>> changes and use BARS in between.
>>>
>>>  
>>>
>>> Major changes being software updates, major component changes in the
>>>
>>> system (new gateways, VM, etc.)
>>>
>>>  
>>>
>>> regards,
>>>
>>>   Zoltan
>>>
>>>  
>>>
>>> Cucumber Green wrote:
>>>
>>> > Hello
>>>
>>> >
>>> > Has anyone used Ghost to keep an image of your MCS servers in 
>>> order to
>>>
>>> > use it in case of a server crash? If so, did it work, are there any
>>>
>>> > caveats, tips?
>>>
>>> >
>>> > If not using Ghost, whats your disaster recovery procedure to restore
>>>
>>> > an MCS crashed server ( other than a full reinstall ).
>>>
>>> >
>>> >
>>> >
>>> >
>>> > 
>>> ------------------------------------------------------------------------ 
>>>
>>>
>>> > Boardwalk for $500? In 2007? Ha!
>>>
>>> > Play Monopoly Here and Now
>>>
>>> > 
>>> <http://us.rd.yahoo.com/evt=48223/*http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow> 
>>>
>>>
>>> > (it's updated for today's economy) at Yahoo! Games.
>>>
>>> > 
>>> ------------------------------------------------------------------------ 
>>>
>>>
>>> >
>>> > _______________________________________________
>>>
>>> > cisco-voip mailing list
>>>
>>> > cisco-voip at puck.nether.net
>>>
>>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>  
>>>
>>> _______________________________________________
>>>
>>> cisco-voip mailing list
>>>
>>> cisco-voip at puck.nether.net
>>>
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>> ------------------------------------------------------------------------ 
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip

From wsisk at cisco.com  Fri Oct  5 10:17:08 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 05 Oct 2007 10:17:08 -0400
Subject: [cisco-voip] QIP anyone?
In-Reply-To: <43017.192.168.1.35.1191589727.webmail@192.168.1.35>
References: <43017.192.168.1.35.1191589727.webmail@192.168.1.35>
Message-ID: <47064764.8070602@cisco.com>

seen a few and it seems to work.  the installations i've worked on set 
some interesting dhcp options like decreasing arp timeout which 
sometimes cause issues.  generally don't set any arp timeout or set it 
hight like 1 hour.  as with all it depends on how qip is configured.

/Wes

Christopher M. Bomba wrote:
> Anyone using QIP for DHCP?  I have a client that will be implementing Cisco IPT and is using QIP for their DHPC application.  Wanted to know if there were any gotchas or anything to look out for before we get started.
>
> Thanks,
>
> Chris
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Fri Oct  5 10:20:04 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 05 Oct 2007 10:20:04 -0400
Subject: [cisco-voip] POWER SHUTDOWNS and verifying PRI Function
	Afterwards
In-Reply-To: <421CAA1A5300B84A87D200D7099857820F4399@DCEX2.DSMO.COM>
References: <421CAA1A5300B84A87D200D7099857820F4399@DCEX2.DSMO.COM>
Message-ID: <47064814.1000501@cisco.com>

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From svoll.voip at gmail.com  Fri Oct  5 11:11:39 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 5 Oct 2007 08:11:39 -0700
Subject: [cisco-voip] QIP anyone?
In-Reply-To: <47064764.8070602@cisco.com>
References: <43017.192.168.1.35.1191589727.webmail@192.168.1.35>
	<47064764.8070602@cisco.com>
Message-ID: <f84a38d30710050811x7d9ca40er1b7887808f7c816a@mail.gmail.com>

excuse my stupid question......... But what is QIP?

Thanks

Scott


On 10/5/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> seen a few and it seems to work.  the installations i've worked on set
> some interesting dhcp options like decreasing arp timeout which
> sometimes cause issues.  generally don't set any arp timeout or set it
> hight like 1 hour.  as with all it depends on how qip is configured.
>
> /Wes
>
> Christopher M. Bomba wrote:
> > Anyone using QIP for DHCP?  I have a client that will be implementing
> Cisco IPT and is using QIP for their DHPC application.  Wanted to know if
> there were any gotchas or anything to look out for before we get started.
> >
> > Thanks,
> >
> > Chris
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From cbomba at s4nets.com  Fri Oct  5 11:46:58 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Fri, 5 Oct 2007 11:46:58 -0400 (EDT)
Subject: [cisco-voip] QIP anyone?
Message-ID: <45245.192.168.1.71.1191599218.webmail@192.168.1.71>

QIP is an enterprise platform for DHCP.  It allows you to have multiple servers with multiple scopes.


-----Original Message-----
From: Scott Voll <svoll.voip at gmail.com>
Sent: Fri, October 5, 2007 11:11 am
To: Wes Sisk <wsisk at cisco.com>
Cc: cbomba at s4nets.com, cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] QIP anyone?

excuse my stupid question......... But what is QIP?

Thanks

Scott


On 10/5/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> seen a few and it seems to work.  the installations i've worked on set
> some interesting dhcp options like decreasing arp timeout which
> sometimes cause issues.  generally don't set any arp timeout or set it
> hight like 1 hour.  as with all it depends on how qip is configured.
>
> /Wes
>
> Christopher M. Bomba wrote:
> > Anyone using QIP for DHCP?  I have a client that will be implementing
> Cisco IPT and is using QIP for their DHPC application.  Wanted to know if
> there were any gotchas or anything to look out for before we get started.
> >
> > Thanks,
> >
> > Chris
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



From bills at psu.edu  Fri Oct  5 11:13:37 2007
From: bills at psu.edu (Bill Simon)
Date: Fri, 05 Oct 2007 11:13:37 -0400
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
 Please Wait' (a rant)
In-Reply-To: <538320.58722.qm@web57202.mail.re3.yahoo.com>
References: <538320.58722.qm@web57202.mail.re3.yahoo.com>
Message-ID: <470654A1.5050906@psu.edu>

Yeah, just a WAG but I'd suspect the DB, not the web server.

CCM 4.x remains, in my opinion, the best out there.  Why are you all 
moving to the higher versions, anyway?  Just for SIP endpoints?


Paul Choi wrote:
> Linux is just an OS. What people put on top of it is
> largely their choice. If Cisco installed wine and ran
> IIS on top of that for their web server, then that
> would likely run slowly. 
> 
> I am not informed on the Linux-based CCM boxen but I'd
> certainly hope that Cisco is using apache to serve its
> webpages. I've heard that the DB is Informix or
> something like that.
> 
> My point being, it doesn't matter what OS you are
> using - if you put a slow application on it, it'll
> still run slow no matter what you do to it.
> 
> Paul
> 
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
> 
>> So, I am curious why all of us are putting up with
>> this.
>>
>> For years we were all told that Linux was superior,
>> that Linux was how
>> a real web server should function...
>>
>> Now, we have CCM 5 and CCM 6... and the web server
>> performance, is,
>> er, well, how do we put this...
>>
>> Here we go. It is far worse than anything Microsoft
>> has ever released.
>> I think Microsoft BOB would be a better web server
>> than whatever crap
>> is on the backend of this RedHat monstrosity.
>>
>> In CallManger 4.3, using Microsoft WINDOWS(!!!!!!) 
>> IIS and Cisco
>> Tomcat, I can scream through all of the web pages;
>> do DB reads in
>> nanoseconds; do DB inserts almost as quick, and I
>> have never seen a
>> message on my web browser saying, 'Loading, Please
>> Wait'
>>
>> I mean, seriously, this is godawful slow.
>>
>> I just installed CCM5.1.3 on an HP 7845, 4GB of RAM,
>> dual dual core
>> Xeons... and all I see at every mouse click is
>> 'Loading, Please
>> Wait'..
>>
>> Is Cisco punishing us for something? Did we offend
>> them in some way? I
>> am at a point where I am wondering if I should tell
>> customers about
>> these performance issues prior to picking a version
>> ('if you want to
>> be able to administer this and not slash your
>> wrists, waiting for the
>> home page to load, you may want to go with 4.3...')
>> because I am tired
>> of hearing about it afterwards...
>>
>> Seriously guys, I mean, this pretty much kills
>> whatever argument you
>> Linux freaks had that this was better than
>> Windows... add the whole
>> SFTP requirement, and you can see why the Microsoft
>> side won this
>> argument...

From asobihoudai at yahoo.com  Fri Oct  5 11:46:24 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 08:46:24 -0700 (PDT)
Subject: [cisco-voip] QIP anyone?
In-Reply-To: <f84a38d30710050811x7d9ca40er1b7887808f7c816a@mail.gmail.com>
Message-ID: <253144.45095.qm@web57201.mail.re3.yahoo.com>

Since I'm stoopider than Scott, I'd like to know too.

Paul

--- Scott Voll <svoll.voip at gmail.com> wrote:

> excuse my stupid question......... But what is QIP?
> 
> Thanks
> 
> Scott
> 
> 
> On 10/5/07, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > seen a few and it seems to work.  the
> installations i've worked on set
> > some interesting dhcp options like decreasing arp
> timeout which
> > sometimes cause issues.  generally don't set any
> arp timeout or set it
> > hight like 1 hour.  as with all it depends on how
> qip is configured.
> >
> > /Wes
> >
> > Christopher M. Bomba wrote:
> > > Anyone using QIP for DHCP?  I have a client that
> will be implementing
> > Cisco IPT and is using QIP for their DHPC
> application.  Wanted to know if
> > there were any gotchas or anything to look out for
> before we get started.
> > >
> > > Thanks,
> > >
> > > Chris
> > >
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
____________________________________________________________________________________
Need a vacation? Get great deals
to amazing places on Yahoo! Travel.
http://travel.yahoo.com/

From svoll.voip at gmail.com  Fri Oct  5 12:47:02 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 5 Oct 2007 09:47:02 -0700
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <470654A1.5050906@psu.edu>
References: <538320.58722.qm@web57202.mail.re3.yahoo.com>
	<470654A1.5050906@psu.edu>
Message-ID: <f84a38d30710050947n4ee5e6oc0692c76cdfbb471@mail.gmail.com>

upgrade ease. no java issues.  BLF/Mobility/presence.

Scott


On 10/5/07, Bill Simon <bills at psu.edu> wrote:
>
> Yeah, just a WAG but I'd suspect the DB, not the web server.
>
> CCM 4.x remains, in my opinion, the best out there.  Why are you all
> moving to the higher versions, anyway?  Just for SIP endpoints?
>
>
> Paul Choi wrote:
> > Linux is just an OS. What people put on top of it is
> > largely their choice. If Cisco installed wine and ran
> > IIS on top of that for their web server, then that
> > would likely run slowly.
> >
> > I am not informed on the Linux-based CCM boxen but I'd
> > certainly hope that Cisco is using apache to serve its
> > webpages. I've heard that the DB is Informix or
> > something like that.
> >
> > My point being, it doesn't matter what OS you are
> > using - if you put a slow application on it, it'll
> > still run slow no matter what you do to it.
> >
> > Paul
> >
> > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> >> So, I am curious why all of us are putting up with
> >> this.
> >>
> >> For years we were all told that Linux was superior,
> >> that Linux was how
> >> a real web server should function...
> >>
> >> Now, we have CCM 5 and CCM 6... and the web server
> >> performance, is,
> >> er, well, how do we put this...
> >>
> >> Here we go. It is far worse than anything Microsoft
> >> has ever released.
> >> I think Microsoft BOB would be a better web server
> >> than whatever crap
> >> is on the backend of this RedHat monstrosity.
> >>
> >> In CallManger 4.3, using Microsoft WINDOWS(!!!!!!)
> >> IIS and Cisco
> >> Tomcat, I can scream through all of the web pages;
> >> do DB reads in
> >> nanoseconds; do DB inserts almost as quick, and I
> >> have never seen a
> >> message on my web browser saying, 'Loading, Please
> >> Wait'
> >>
> >> I mean, seriously, this is godawful slow.
> >>
> >> I just installed CCM5.1.3 on an HP 7845, 4GB of RAM,
> >> dual dual core
> >> Xeons... and all I see at every mouse click is
> >> 'Loading, Please
> >> Wait'..
> >>
> >> Is Cisco punishing us for something? Did we offend
> >> them in some way? I
> >> am at a point where I am wondering if I should tell
> >> customers about
> >> these performance issues prior to picking a version
> >> ('if you want to
> >> be able to administer this and not slash your
> >> wrists, waiting for the
> >> home page to load, you may want to go with 4.3...')
> >> because I am tired
> >> of hearing about it afterwards...
> >>
> >> Seriously guys, I mean, this pretty much kills
> >> whatever argument you
> >> Linux freaks had that this was better than
> >> Windows... add the whole
> >> SFTP requirement, and you can see why the Microsoft
> >> side won this
> >> argument...
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jonvoip at gmail.com  Fri Oct  5 13:18:37 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 5 Oct 2007 12:18:37 -0500
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <470654A1.5050906@psu.edu>
References: <538320.58722.qm@web57202.mail.re3.yahoo.com>
	<470654A1.5050906@psu.edu>
Message-ID: <5d093f9a0710051018k2696ccefndfd3aacfb9b941f1@mail.gmail.com>

We upgrade cuz the customer hears 'linux' and thinks it is better...
that is also what the sales people are selling...

To be honest, the only kick-ass features for CCM5 are BAT and the LDAP
integration.



Jonathan

On 10/5/07, Bill Simon <bills at psu.edu> wrote:
> Yeah, just a WAG but I'd suspect the DB, not the web server.
>
> CCM 4.x remains, in my opinion, the best out there.  Why are you all
> moving to the higher versions, anyway?  Just for SIP endpoints?
>
>
> Paul Choi wrote:
> > Linux is just an OS. What people put on top of it is
> > largely their choice. If Cisco installed wine and ran
> > IIS on top of that for their web server, then that
> > would likely run slowly.
> >
> > I am not informed on the Linux-based CCM boxen but I'd
> > certainly hope that Cisco is using apache to serve its
> > webpages. I've heard that the DB is Informix or
> > something like that.
> >
> > My point being, it doesn't matter what OS you are
> > using - if you put a slow application on it, it'll
> > still run slow no matter what you do to it.
> >
> > Paul
> >
> > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> >> So, I am curious why all of us are putting up with
> >> this.
> >>
> >> For years we were all told that Linux was superior,
> >> that Linux was how
> >> a real web server should function...
> >>
> >> Now, we have CCM 5 and CCM 6... and the web server
> >> performance, is,
> >> er, well, how do we put this...
> >>
> >> Here we go. It is far worse than anything Microsoft
> >> has ever released.
> >> I think Microsoft BOB would be a better web server
> >> than whatever crap
> >> is on the backend of this RedHat monstrosity.
> >>
> >> In CallManger 4.3, using Microsoft WINDOWS(!!!!!!)
> >> IIS and Cisco
> >> Tomcat, I can scream through all of the web pages;
> >> do DB reads in
> >> nanoseconds; do DB inserts almost as quick, and I
> >> have never seen a
> >> message on my web browser saying, 'Loading, Please
> >> Wait'
> >>
> >> I mean, seriously, this is godawful slow.
> >>
> >> I just installed CCM5.1.3 on an HP 7845, 4GB of RAM,
> >> dual dual core
> >> Xeons... and all I see at every mouse click is
> >> 'Loading, Please
> >> Wait'..
> >>
> >> Is Cisco punishing us for something? Did we offend
> >> them in some way? I
> >> am at a point where I am wondering if I should tell
> >> customers about
> >> these performance issues prior to picking a version
> >> ('if you want to
> >> be able to administer this and not slash your
> >> wrists, waiting for the
> >> home page to load, you may want to go with 4.3...')
> >> because I am tired
> >> of hearing about it afterwards...
> >>
> >> Seriously guys, I mean, this pretty much kills
> >> whatever argument you
> >> Linux freaks had that this was better than
> >> Windows... add the whole
> >> SFTP requirement, and you can see why the Microsoft
> >> side won this
> >> argument...
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From dustin.s.fowler at gmail.com  Fri Oct  5 13:49:05 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Fri, 5 Oct 2007 13:49:05 -0400
Subject: [cisco-voip] IPCC Queue Question
Message-ID: <4706791f.2806240a.686f.ffff8907@mx.google.com>

All,

 

If a call is presented to an agent and the agent does not pick up, will the
call me given a number in queue. Basically starting at the end of the line?

 

Dustin

 

 

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From svoll.voip at gmail.com  Fri Oct  5 13:54:58 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 5 Oct 2007 10:54:58 -0700
Subject: [cisco-voip] upgrade Exchange in a Unity enviroment
Message-ID: <f84a38d30710051054q48eb6ddl9069c33ee839f56a@mail.gmail.com>

I will be upgrading Unity from 4.0.5 to 5.x in the next couple months.

is there any problem if the server guys bring up a new exchange 2007 before
hand so long as the UM subscribers are not on the exchange 2007 box? or does
that screw up everything due to schema, etc.

If I'm running two exchange 2003 servers and add a 2007 I just don't want it
to bring everything else down as far as UM is considered.

Thanks

scott
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From svoll.voip at gmail.com  Fri Oct  5 13:58:42 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 5 Oct 2007 10:58:42 -0700
Subject: [cisco-voip] IPCC Queue Question
In-Reply-To: <4706791f.2806240a.686f.ffff8907@mx.google.com>
References: <4706791f.2806240a.686f.ffff8907@mx.google.com>
Message-ID: <f84a38d30710051058t6c68e162s7a5d233b8ab56a7b@mail.gmail.com>

"should" go to the next agent. and will kick the agent that didn't answer
it, into "not ready".  make sure agents know that so somebody is always in a
ready state.

scott


On 10/5/07, Dustin S Fowler <dustin.s.fowler at gmail.com> wrote:
>
>  All,
>
>
>
> If a call is presented to an agent and the agent does not pick up, will
> the call me given a number in queue. Basically starting at the end of the
> line?
>
>
>
> Dustin
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Brian.VanBenschoten at inacom.com  Fri Oct  5 14:00:13 2007
From: Brian.VanBenschoten at inacom.com (Brian Van Benschoten)
Date: Fri, 5 Oct 2007 13:00:13 -0500
Subject: [cisco-voip] IPCC Queue Question
In-Reply-To: <f84a38d30710051058t6c68e162s7a5d233b8ab56a7b@mail.gmail.com>
References: <4706791f.2806240a.686f.ffff8907@mx.google.com>
	<f84a38d30710051058t6c68e162s7a5d233b8ab56a7b@mail.gmail.com>
Message-ID: <153D0FAADC2C90459D202D16D24A94AE0A4625A6@exch1.inacom-msn.com>

In IPCC 4.0 and forward there is a system wide setting to change that
missed call behavior to keep the agent ready.. Not recommended but
sometime it helps when agents are not managed well and no one is ready
all the time.

 

________________________________________

Brian Van Benschoten 

Practice Leader - Unified Communications

Consulting Engineer - CCIE # 5421

Inacom Information Systems

3001 West Beltline Highway

Madison, WI  53713  USA

608-661-7780 (IP Phone)

608-661-7701 (FAX)

608-661-7777 (Customer Service)

brian.vanbenschoten at inacom.com

http://www.inacom.com <http://www.inacom.com/> 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Friday, October 05, 2007 12:59 PM
To: Dustin S Fowler
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC Queue Question

 

"should" go to the next agent. and will kick the agent that didn't
answer it, into "not ready".  make sure agents know that so somebody is
always in a ready state.

 

scott

 

On 10/5/07, Dustin S Fowler <dustin.s.fowler at gmail.com> wrote: 

All,

 

If a call is presented to an agent and the agent does not pick up, will
the call me given a number in queue. Basically starting at the end of
the line?

 

Dustin

 

 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From SCASPER at mtb.com  Fri Oct  5 15:55:43 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Fri, 05 Oct 2007 15:55:43 -0400
Subject: [cisco-voip] SIP Addressing Change
In-Reply-To: <1191585387.6476.33.camel@mda>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
	<4704BB73.9874.0039.0@mtb.com> <1191585387.6476.33.camel@mda>
Message-ID: <47065E81.9874.0039.0@mtb.com>

Thanks for the responses. Found what I was looking for - there is a "localhost" command under 
 
voice services voip
sip 
 
fyi - topology is:
 
 CCM 4.1.3 ----H323------IPIPGW-------SIP------Carrier

>>> "tele" <tele at plexialab.it> 10/5/2007 7:56 AM >>>
10.x.x.33 is the IP of your cisco?
I'm not sure but you can try to manually set the host, you should see
the hostname and not the IP.

ip host test.ipttrun.com 10.x.x.33

if this not work you can do it with custom TCL script:

http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/tclappa.htm#wp1007665 

:tele

On Thu, 2007-10-04 at 10:07 -0400, STEVEN CASPER wrote:
> I am trying to implement a SIP trunking test with a carrier. I am
> sending them 
>  
>  From: "Cisco Test One" <sip:5422137 at 10.x.x.33>. 
>  
> They want to see
>  
>  From: "Cisco Test One" <sip:5422137 at test.ipttrunk.com> 
>  
> Is there a way to make this change on the router or Call Manager? The
> gateway is set up in Call Manager as a H323 gateway.
>  
>  
>  
>  
> Thanks,
> 
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026 
> 
> ************************************
> This email may contain privileged and/or confidential information that
> is intended solely for the use of the addressee. If you are not the
> intended recipient or entity, you are strictly prohibited from
> disclosing, copying, distributing or using any of the information
> contained in the transmission. If you received this communication in
> error, please contact the sender immediately and destroy the material
> in its entirety, whether electronic or hard copy. This communication
> may contain nonpublic personal information about consumers subject to
> the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley
> Act. You may not directly or indirectly reuse or disclose such
> information for any purpose other than to provide the services for
> which you are receiving the information.
> There are risks associated with the use of electronic transmission.
> The sender of this information does not control the method of
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> transmission.
> ************************************
> 
> 
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip 

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************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee.  If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission.  If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy.  This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
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From ano-1 at worldonline.dk  Fri Oct  5 15:42:14 2007
From: ano-1 at worldonline.dk (Katrine og Michael)
Date: Fri, 5 Oct 2007 21:42:14 +0200
Subject: [cisco-voip] Script for AutoAttendant Unity Express
Message-ID: <001501c80787$d45c0940$0c000a0a@ano>

Hello

Can somebody help me?

I need a script for unity express. The script will be used when a phone i busy. When the phone is busy the calling person will hear a voice saying: press 1 to callcontact, press 2 for try agin and press 3 for end call. When 1 is pressed the calling person will be transferred to callcontact. When 2 is pressed the calling person will put in cue whith music and the it will try to contact the called person again. When 3 is pressed the script will terminate the call.

Can sombody please help me? I?ve attatched my script, but it is not working.

your sincerely

Michael from Denmark :-)

-- 
Jeg beskyttes af den gratis SPAMfighter til privatbrugere.
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From cheesevoice at gmail.com  Fri Oct  5 17:02:44 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Fri, 5 Oct 2007 16:02:44 -0500
Subject: [cisco-voip] A little Fun
Message-ID: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>

Thought you guys might want to take a break and help me out on a "related"
issue...

We are interviewing new Network Engineer Applicants next week.

While I was writing up my list of questions (the boss needs 10) I thought I
would throw this out to you guys.
If you could write an interview question for your next Network Engineer
(let's say Professional Level certification in Networking, Security or
Voice)
what would you ask them?  (try to keep it technical, nothing about Packer's
tickets or Beer...lol)
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From jackson.tim at gmail.com  Fri Oct  5 17:22:44 2007
From: jackson.tim at gmail.com (Tim Jackson)
Date: Fri, 5 Oct 2007 16:22:44 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
Message-ID: <4407932e0710051422i14e86904jf9e429b875b680c@mail.gmail.com>

Hand him some configuration information (Put this IP on this interface, use
vlans XXX, etc) a few routers and some other gear w/ access to Google and
say "Configure this... When it's done, use the phone to call me and let me
know that you're done"

Works pretty well for cutting through the BS... Lets you know if they
actually know what they're doing or not, and with Google they should be able
to do just about anything...

--
Tim

On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
>
> Thought you guys might want to take a break and help me out on a "related"
> issue...
>
> We are interviewing new Network Engineer Applicants next week.
>
> While I was writing up my list of questions (the boss needs 10) I thought
> I would throw this out to you guys.
> If you could write an interview question for your next Network Engineer
> (let's say Professional Level certification in Networking, Security or
> Voice)
> what would you ask them?  (try to keep it technical, nothing about
> Packer's tickets or Beer...lol)
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jason.aarons at us.didata.com  Fri Oct  5 17:43:44 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 5 Oct 2007 17:43:44 -0400
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <5d093f9a0710051018k2696ccefndfd3aacfb9b941f1@mail.gmail.com>
References: <538320.58722.qm@web57202.mail.re3.yahoo.com><470654A1.5050906@psu.edu>
	<5d093f9a0710051018k2696ccefndfd3aacfb9b941f1@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F007020AF1@USNAEXCH.na.didata.local>

Customers do complain the "Loading" is slow, especially when on a Phone
DN you hit Line Partition and "Loading" changes focus.  You would think
Cisco could speed it up.  I do miss the SQL Enterprise Manager for
queries versus the SOAP AXL or command line I've seen TAC do. TAC isn't
happy when they have to type long select statements.

Informix isn't the database of choice by DBAs.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Friday, October 05, 2007 1:19 PM
To: Bill Simon
Cc: cisco voip list
Subject: Re: [cisco-voip] CCM Web Page performance issues, aka
'Loading,Please Wait' (a rant)

We upgrade cuz the customer hears 'linux' and thinks it is better...
that is also what the sales people are selling...

To be honest, the only kick-ass features for CCM5 are BAT and the LDAP
integration.



Jonathan

On 10/5/07, Bill Simon <bills at psu.edu> wrote:
> Yeah, just a WAG but I'd suspect the DB, not the web server.
>
> CCM 4.x remains, in my opinion, the best out there.  Why are you all
> moving to the higher versions, anyway?  Just for SIP endpoints?
>
>
> Paul Choi wrote:
> > Linux is just an OS. What people put on top of it is
> > largely their choice. If Cisco installed wine and ran
> > IIS on top of that for their web server, then that
> > would likely run slowly.
> >
> > I am not informed on the Linux-based CCM boxen but I'd
> > certainly hope that Cisco is using apache to serve its
> > webpages. I've heard that the DB is Informix or
> > something like that.
> >
> > My point being, it doesn't matter what OS you are
> > using - if you put a slow application on it, it'll
> > still run slow no matter what you do to it.
> >
> > Paul
> >
> > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> >> So, I am curious why all of us are putting up with
> >> this.
> >>
> >> For years we were all told that Linux was superior,
> >> that Linux was how
> >> a real web server should function...
> >>
> >> Now, we have CCM 5 and CCM 6... and the web server
> >> performance, is,
> >> er, well, how do we put this...
> >>
> >> Here we go. It is far worse than anything Microsoft
> >> has ever released.
> >> I think Microsoft BOB would be a better web server
> >> than whatever crap
> >> is on the backend of this RedHat monstrosity.
> >>
> >> In CallManger 4.3, using Microsoft WINDOWS(!!!!!!)
> >> IIS and Cisco
> >> Tomcat, I can scream through all of the web pages;
> >> do DB reads in
> >> nanoseconds; do DB inserts almost as quick, and I
> >> have never seen a
> >> message on my web browser saying, 'Loading, Please
> >> Wait'
> >>
> >> I mean, seriously, this is godawful slow.
> >>
> >> I just installed CCM5.1.3 on an HP 7845, 4GB of RAM,
> >> dual dual core
> >> Xeons... and all I see at every mouse click is
> >> 'Loading, Please
> >> Wait'..
> >>
> >> Is Cisco punishing us for something? Did we offend
> >> them in some way? I
> >> am at a point where I am wondering if I should tell
> >> customers about
> >> these performance issues prior to picking a version
> >> ('if you want to
> >> be able to administer this and not slash your
> >> wrists, waiting for the
> >> home page to load, you may want to go with 4.3...')
> >> because I am tired
> >> of hearing about it afterwards...
> >>
> >> Seriously guys, I mean, this pretty much kills
> >> whatever argument you
> >> Linux freaks had that this was better than
> >> Windows... add the whole
> >> SFTP requirement, and you can see why the Microsoft
> >> side won this
> >> argument...
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From asobihoudai at yahoo.com  Fri Oct  5 17:44:34 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 14:44:34 -0700 (PDT)
Subject: [cisco-voip] A little Fun
In-Reply-To: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
Message-ID: <960111.19800.qm@web57204.mail.re3.yahoo.com>

That depends on what level of engineer for which
you're searching. If you're looking for someone that
is a little green behind the ears, I'd recommend
interviewing upon their communication skills, honesty,
and at least their ability to configure the very
basics of a Cisco box.

Paul


--- Kevin Dunn <cheesevoice at gmail.com> wrote:

> Thought you guys might want to take a break and help
> me out on a "related"
> issue...
> 
> We are interviewing new Network Engineer Applicants
> next week.
> 
> While I was writing up my list of questions (the
> boss needs 10) I thought I
> would throw this out to you guys.
> If you could write an interview question for your
> next Network Engineer
> (let's say Professional Level certification in
> Networking, Security or
> Voice)
> what would you ask them?  (try to keep it technical,
> nothing about Packer's
> tickets or Beer...lol)
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
____________________________________________________________________________________
Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games.
http://sims.yahoo.com/  

From asobihoudai at yahoo.com  Fri Oct  5 17:47:33 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 14:47:33 -0700 (PDT)
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F007020AF1@USNAEXCH.na.didata.local>
Message-ID: <724538.58409.qm@web57211.mail.re3.yahoo.com>

Just be glad they didn't pick something like, Oracle.
Oh the humanity!

IANADBA (I'm not a DBA) so I don't know the difference
in how SQL Server 2005 compares to say, PostgreSQL. I
hope that Cisco finds a faster DBMS if it really is as
slow as you folks make it sound.

--- "Jason Aarons (US)" <jason.aarons at us.didata.com>
wrote:

> Customers do complain the "Loading" is slow,
> especially when on a Phone
> DN you hit Line Partition and "Loading" changes
> focus.  You would think
> Cisco could speed it up.  I do miss the SQL
> Enterprise Manager for
> queries versus the SOAP AXL or command line I've
> seen TAC do. TAC isn't
> happy when they have to type long select statements.
> 
> Informix isn't the database of choice by DBAs.
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Jonathan
> Charles
> Sent: Friday, October 05, 2007 1:19 PM
> To: Bill Simon
> Cc: cisco voip list
> Subject: Re: [cisco-voip] CCM Web Page performance
> issues, aka
> 'Loading,Please Wait' (a rant)
> 
> We upgrade cuz the customer hears 'linux' and thinks
> it is better...
> that is also what the sales people are selling...
> 
> To be honest, the only kick-ass features for CCM5
> are BAT and the LDAP
> integration.
> 
> 
> 
> Jonathan
> 
> On 10/5/07, Bill Simon <bills at psu.edu> wrote:
> > Yeah, just a WAG but I'd suspect the DB, not the
> web server.
> >
> > CCM 4.x remains, in my opinion, the best out
> there.  Why are you all
> > moving to the higher versions, anyway?  Just for
> SIP endpoints?
> >
> >
> > Paul Choi wrote:
> > > Linux is just an OS. What people put on top of
> it is
> > > largely their choice. If Cisco installed wine
> and ran
> > > IIS on top of that for their web server, then
> that
> > > would likely run slowly.
> > >
> > > I am not informed on the Linux-based CCM boxen
> but I'd
> > > certainly hope that Cisco is using apache to
> serve its
> > > webpages. I've heard that the DB is Informix or
> > > something like that.
> > >
> > > My point being, it doesn't matter what OS you
> are
> > > using - if you put a slow application on it,
> it'll
> > > still run slow no matter what you do to it.
> > >
> > > Paul
> > >
> > > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > >> So, I am curious why all of us are putting up
> with
> > >> this.
> > >>
> > >> For years we were all told that Linux was
> superior,
> > >> that Linux was how
> > >> a real web server should function...
> > >>
> > >> Now, we have CCM 5 and CCM 6... and the web
> server
> > >> performance, is,
> > >> er, well, how do we put this...
> > >>
> > >> Here we go. It is far worse than anything
> Microsoft
> > >> has ever released.
> > >> I think Microsoft BOB would be a better web
> server
> > >> than whatever crap
> > >> is on the backend of this RedHat monstrosity.
> > >>
> > >> In CallManger 4.3, using Microsoft
> WINDOWS(!!!!!!)
> > >> IIS and Cisco
> > >> Tomcat, I can scream through all of the web
> pages;
> > >> do DB reads in
> > >> nanoseconds; do DB inserts almost as quick, and
> I
> > >> have never seen a
> > >> message on my web browser saying, 'Loading,
> Please
> > >> Wait'
> > >>
> > >> I mean, seriously, this is godawful slow.
> > >>
> > >> I just installed CCM5.1.3 on an HP 7845, 4GB of
> RAM,
> > >> dual dual core
> > >> Xeons... and all I see at every mouse click is
> > >> 'Loading, Please
> > >> Wait'..
> > >>
> > >> Is Cisco punishing us for something? Did we
> offend
> > >> them in some way? I
> > >> am at a point where I am wondering if I should
> tell
> > >> customers about
> > >> these performance issues prior to picking a
> version
> > >> ('if you want to
> > >> be able to administer this and not slash your
> > >> wrists, waiting for the
> > >> home page to load, you may want to go with
> 4.3...')
> > >> because I am tired
> > >> of hearing about it afterwards...
> > >>
> > >> Seriously guys, I mean, this pretty much kills
> > >> whatever argument you
> > >> Linux freaks had that this was better than
> > >> Windows... add the whole
> > >> SFTP requirement, and you can see why the
> Microsoft
> > >> side won this
> > >> argument...
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> -----------------------------------------
> Disclaimer:
> 
> This e-mail communication and any attachments may
> contain
> confidential and privileged information and is for
> use by the
> designated addressee(s) named above only.  If you
> are not the
> intended addressee, you are hereby notified that you
> have received
> this communication in error and that any use or
> reproduction of
> this email or its contents is strictly prohibited
> and may be
> unlawful.  If you have received this communication
> in error, please
> notify us immediately by replying to this message
> and deleting it
> from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
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From asobihoudai at yahoo.com  Fri Oct  5 17:49:07 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 14:49:07 -0700 (PDT)
Subject: [cisco-voip] Script for AutoAttendant Unity Express
In-Reply-To: <001501c80787$d45c0940$0c000a0a@ano>
Message-ID: <942069.22286.qm@web57204.mail.re3.yahoo.com>

What version of CUE is this?

Paul
--- Katrine og Michael <ano-1 at worldonline.dk> wrote:

> Hello
> 
> Can somebody help me?
> 
> I need a script for unity express. The script will
> be used when a phone i busy. When the phone is busy
> the calling person will hear a voice saying: press 1
> to callcontact, press 2 for try agin and press 3 for
> end call. When 1 is pressed the calling person will
> be transferred to callcontact. When 2 is pressed the
> calling person will put in cue whith music and the
> it will try to contact the called person again. When
> 3 is pressed the script will terminate the call.
> 
> Can sombody please help me? I?ve attatched my
> script, but it is not working.
> 
> your sincerely
> 
> Michael from Denmark :-)
> 
> --
> Jeg beskyttes af den gratis SPAMfighter til
> privatbrugere.
> Den har indtil videre sparet mig for at f? 690
> spam-mails.
> Betalende brugere f?r ikke denne besked i deres
> e-mails.
> Hent gratis SPAMfighter her:
> http://www.spamfighter.com/lda
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
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From jonvoip at gmail.com  Fri Oct  5 17:51:57 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 5 Oct 2007 16:51:57 -0500
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <724538.58409.qm@web57211.mail.re3.yahoo.com>
References: <C1FE15183DA37645BC0633BC604E44F007020AF1@USNAEXCH.na.didata.local>
	<724538.58409.qm@web57211.mail.re3.yahoo.com>
Message-ID: <5d093f9a0710051451n42fa4f58sacad3a176fd72fe2@mail.gmail.com>

You have no idea...

The only way to show you is to have you set up a CCM 4.3 box on
Windows 2003 and CCM6.0 box...

Put CCM4.3 on a VMWare with too little ram (say 1GB).

Put CCM6.0 on a 7845.

The 4.3 box will beat the 6.0 box by seconds on each page load or query.

I have timed it. The average time to load an empty page (go to
device-phone, how long does it take for that page to load?): for
CCM6.0, about 8 seconds, for 4.3, about 1.5.

For queries, 4.3 nukes the crap out of 6 (or 5)... it can be 20
seconds for a CCM6 query, and about 3 seconds for 4.3...




Jonathan

On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> Just be glad they didn't pick something like, Oracle.
> Oh the humanity!
>
> IANADBA (I'm not a DBA) so I don't know the difference
> in how SQL Server 2005 compares to say, PostgreSQL. I
> hope that Cisco finds a faster DBMS if it really is as
> slow as you folks make it sound.
>
> --- "Jason Aarons (US)" <jason.aarons at us.didata.com>
> wrote:
>
> > Customers do complain the "Loading" is slow,
> > especially when on a Phone
> > DN you hit Line Partition and "Loading" changes
> > focus.  You would think
> > Cisco could speed it up.  I do miss the SQL
> > Enterprise Manager for
> > queries versus the SOAP AXL or command line I've
> > seen TAC do. TAC isn't
> > happy when they have to type long select statements.
> >
> > Informix isn't the database of choice by DBAs.
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Jonathan
> > Charles
> > Sent: Friday, October 05, 2007 1:19 PM
> > To: Bill Simon
> > Cc: cisco voip list
> > Subject: Re: [cisco-voip] CCM Web Page performance
> > issues, aka
> > 'Loading,Please Wait' (a rant)
> >
> > We upgrade cuz the customer hears 'linux' and thinks
> > it is better...
> > that is also what the sales people are selling...
> >
> > To be honest, the only kick-ass features for CCM5
> > are BAT and the LDAP
> > integration.
> >
> >
> >
> > Jonathan
> >
> > On 10/5/07, Bill Simon <bills at psu.edu> wrote:
> > > Yeah, just a WAG but I'd suspect the DB, not the
> > web server.
> > >
> > > CCM 4.x remains, in my opinion, the best out
> > there.  Why are you all
> > > moving to the higher versions, anyway?  Just for
> > SIP endpoints?
> > >
> > >
> > > Paul Choi wrote:
> > > > Linux is just an OS. What people put on top of
> > it is
> > > > largely their choice. If Cisco installed wine
> > and ran
> > > > IIS on top of that for their web server, then
> > that
> > > > would likely run slowly.
> > > >
> > > > I am not informed on the Linux-based CCM boxen
> > but I'd
> > > > certainly hope that Cisco is using apache to
> > serve its
> > > > webpages. I've heard that the DB is Informix or
> > > > something like that.
> > > >
> > > > My point being, it doesn't matter what OS you
> > are
> > > > using - if you put a slow application on it,
> > it'll
> > > > still run slow no matter what you do to it.
> > > >
> > > > Paul
> > > >
> > > > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> > > >
> > > >> So, I am curious why all of us are putting up
> > with
> > > >> this.
> > > >>
> > > >> For years we were all told that Linux was
> > superior,
> > > >> that Linux was how
> > > >> a real web server should function...
> > > >>
> > > >> Now, we have CCM 5 and CCM 6... and the web
> > server
> > > >> performance, is,
> > > >> er, well, how do we put this...
> > > >>
> > > >> Here we go. It is far worse than anything
> > Microsoft
> > > >> has ever released.
> > > >> I think Microsoft BOB would be a better web
> > server
> > > >> than whatever crap
> > > >> is on the backend of this RedHat monstrosity.
> > > >>
> > > >> In CallManger 4.3, using Microsoft
> > WINDOWS(!!!!!!)
> > > >> IIS and Cisco
> > > >> Tomcat, I can scream through all of the web
> > pages;
> > > >> do DB reads in
> > > >> nanoseconds; do DB inserts almost as quick, and
> > I
> > > >> have never seen a
> > > >> message on my web browser saying, 'Loading,
> > Please
> > > >> Wait'
> > > >>
> > > >> I mean, seriously, this is godawful slow.
> > > >>
> > > >> I just installed CCM5.1.3 on an HP 7845, 4GB of
> > RAM,
> > > >> dual dual core
> > > >> Xeons... and all I see at every mouse click is
> > > >> 'Loading, Please
> > > >> Wait'..
> > > >>
> > > >> Is Cisco punishing us for something? Did we
> > offend
> > > >> them in some way? I
> > > >> am at a point where I am wondering if I should
> > tell
> > > >> customers about
> > > >> these performance issues prior to picking a
> > version
> > > >> ('if you want to
> > > >> be able to administer this and not slash your
> > > >> wrists, waiting for the
> > > >> home page to load, you may want to go with
> > 4.3...')
> > > >> because I am tired
> > > >> of hearing about it afterwards...
> > > >>
> > > >> Seriously guys, I mean, this pretty much kills
> > > >> whatever argument you
> > > >> Linux freaks had that this was better than
> > > >> Windows... add the whole
> > > >> SFTP requirement, and you can see why the
> > Microsoft
> > > >> side won this
> > > >> argument...
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may
> > contain
> > confidential and privileged information and is for
> > use by the
> > designated addressee(s) named above only.  If you
> > are not the
> > intended addressee, you are hereby notified that you
> > have received
> > this communication in error and that any use or
> > reproduction of
> > this email or its contents is strictly prohibited
> > and may be
> > unlawful.  If you have received this communication
> > in error, please
> > notify us immediately by replying to this message
> > and deleting it
> > from your computer. Thank you.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
>
> ____________________________________________________________________________________
> Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.
> http://answers.yahoo.com/dir/?link=list&sid=396545469
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From asobihoudai at yahoo.com  Fri Oct  5 17:54:57 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 14:54:57 -0700 (PDT)
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <5d093f9a0710051451n42fa4f58sacad3a176fd72fe2@mail.gmail.com>
Message-ID: <778413.23734.qm@web57204.mail.re3.yahoo.com>

This is not encouraging news... 

We're about to embark on a CCM6 install on Monday. . .

:(


--- Jonathan Charles <jonvoip at gmail.com> wrote:

> You have no idea...
> 
> The only way to show you is to have you set up a CCM
> 4.3 box on
> Windows 2003 and CCM6.0 box...
> 
> Put CCM4.3 on a VMWare with too little ram (say
> 1GB).
> 
> Put CCM6.0 on a 7845.
> 
> The 4.3 box will beat the 6.0 box by seconds on each
> page load or query.
> 
> I have timed it. The average time to load an empty
> page (go to
> device-phone, how long does it take for that page to
> load?): for
> CCM6.0, about 8 seconds, for 4.3, about 1.5.
> 
> For queries, 4.3 nukes the crap out of 6 (or 5)...
> it can be 20
> seconds for a CCM6 query, and about 3 seconds for
> 4.3...
> 
> 
> 
> 
> Jonathan
> 
> On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> > Just be glad they didn't pick something like,
> Oracle.
> > Oh the humanity!
> >
> > IANADBA (I'm not a DBA) so I don't know the
> difference
> > in how SQL Server 2005 compares to say,
> PostgreSQL. I
> > hope that Cisco finds a faster DBMS if it really
> is as
> > slow as you folks make it sound.
> >
> > --- "Jason Aarons (US)"
> <jason.aarons at us.didata.com>
> > wrote:
> >
> > > Customers do complain the "Loading" is slow,
> > > especially when on a Phone
> > > DN you hit Line Partition and "Loading" changes
> > > focus.  You would think
> > > Cisco could speed it up.  I do miss the SQL
> > > Enterprise Manager for
> > > queries versus the SOAP AXL or command line I've
> > > seen TAC do. TAC isn't
> > > happy when they have to type long select
> statements.
> > >
> > > Informix isn't the database of choice by DBAs.
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > > Behalf Of Jonathan
> > > Charles
> > > Sent: Friday, October 05, 2007 1:19 PM
> > > To: Bill Simon
> > > Cc: cisco voip list
> > > Subject: Re: [cisco-voip] CCM Web Page
> performance
> > > issues, aka
> > > 'Loading,Please Wait' (a rant)
> > >
> > > We upgrade cuz the customer hears 'linux' and
> thinks
> > > it is better...
> > > that is also what the sales people are
> selling...
> > >
> > > To be honest, the only kick-ass features for
> CCM5
> > > are BAT and the LDAP
> > > integration.
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On 10/5/07, Bill Simon <bills at psu.edu> wrote:
> > > > Yeah, just a WAG but I'd suspect the DB, not
> the
> > > web server.
> > > >
> > > > CCM 4.x remains, in my opinion, the best out
> > > there.  Why are you all
> > > > moving to the higher versions, anyway?  Just
> for
> > > SIP endpoints?
> > > >
> > > >
> > > > Paul Choi wrote:
> > > > > Linux is just an OS. What people put on top
> of
> > > it is
> > > > > largely their choice. If Cisco installed
> wine
> > > and ran
> > > > > IIS on top of that for their web server,
> then
> > > that
> > > > > would likely run slowly.
> > > > >
> > > > > I am not informed on the Linux-based CCM
> boxen
> > > but I'd
> > > > > certainly hope that Cisco is using apache to
> > > serve its
> > > > > webpages. I've heard that the DB is Informix
> or
> > > > > something like that.
> > > > >
> > > > > My point being, it doesn't matter what OS
> you
> > > are
> > > > > using - if you put a slow application on it,
> > > it'll
> > > > > still run slow no matter what you do to it.
> > > > >
> > > > > Paul
> > > > >
> > > > > --- Jonathan Charles <jonvoip at gmail.com>
> wrote:
> > > > >
> > > > >> So, I am curious why all of us are putting
> up
> > > with
> > > > >> this.
> > > > >>
> > > > >> For years we were all told that Linux was
> > > superior,
> > > > >> that Linux was how
> > > > >> a real web server should function...
> > > > >>
> > > > >> Now, we have CCM 5 and CCM 6... and the web
> > > server
> > > > >> performance, is,
> > > > >> er, well, how do we put this...
> > > > >>
> > > > >> Here we go. It is far worse than anything
> > > Microsoft
> > > > >> has ever released.
> > > > >> I think Microsoft BOB would be a better web
> > > server
> > > > >> than whatever crap
> > > > >> is on the backend of this RedHat
> monstrosity.
> > > > >>
> > > > >> In CallManger 4.3, using Microsoft
> > > WINDOWS(!!!!!!)
> > > > >> IIS and Cisco
> > > > >> Tomcat, I can scream through all of the web
> > > pages;
> > > > >> do DB reads in
> > > > >> nanoseconds; do DB inserts almost as quick,
> and
> > > I
> > > > >> have never seen a
> > > > >> message on my web browser saying, 'Loading,
> > > Please
> > > > >> Wait'
> > > > >>
> > > > >> I mean, seriously, this is godawful slow.
> > > > >>
> > > > >> I just installed CCM5.1.3 on an HP 7845,
> 4GB of
> > > RAM,
> > > > >> dual dual core
> > > > >> Xeons... and all I see at every mouse click
> is
> > > > >> 'Loading, Please
> > > > >> Wait'..
> > > > >>
> > > > >> Is Cisco punishing us for something? Did we
> > > offend
> > > > >> them in some way? I
> > > > >> am at a point where I am wondering if I
> should
> > > tell
> > > > >> customers about
> > > > >> these performance issues prior to picking a
> > > version
> > > > >> ('if you want to
> > > > >> be able to administer this and not slash
> your
> > > > >> wrists, waiting for the
> > > > >> home page to load, you may want to go with
> > > 4.3...')
> > > > >> because I am tired
> > > > >> of hearing about it afterwards...
> > > > >>
> > > > >> Seriously guys, I mean, this pretty much
> kills
> > > > >> whatever argument you
> > > > >> Linux freaks had that this was better than
> > > > >> Windows... add the whole
> > > > >> SFTP requirement, and you can see why the
> > > Microsoft
> > > > >> side won this
> > > > >> argument...
> > > >
> _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > -----------------------------------------
> > > Disclaimer:
> > >
> > > This e-mail communication and any attachments
> may
> > > contain
> > > confidential and privileged information and is
> for
> > > use by the
> > > designated addressee(s) named above only.  If
> you
> > > are not the
> > > intended addressee, you are hereby notified that
> you
> > > have received
> > > this communication in error and that any use or
> > > reproduction of
> > > this email or its contents is strictly
> prohibited
> > > and may be
> > > unlawful.  If you have received this
> communication
> > > in error, please
> > > notify us immediately by replying to this
> message
> > > and deleting it
> > > from your computer. Thank you.
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> >
> >
> >
>
____________________________________________________________________________________
> > Be a better Globetrotter. Get better travel
> answers from someone who knows. Yahoo! Answers -
> Check it out.
> >
>
http://answers.yahoo.com/dir/?link=list&sid=396545469
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 



       
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From rspeed at gmail.com  Fri Oct  5 18:08:45 2007
From: rspeed at gmail.com (ryan speed)
Date: Fri, 5 Oct 2007 15:08:45 -0700
Subject: [cisco-voip] A little Fun
In-Reply-To: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
Message-ID: <a6e20ec60710051508l3eb792adsb8d15b80b1af523c@mail.gmail.com>

never hurts to throw a scenario or two in.

you're tasked with upgrading the ios (pre-definined) on a voice
gateway for a remote customer, what steps would you take and what
precautions would you deem necessary for a trouble free upgrade with a
solid backout plan in the event something goes wrong.

take that answer and move onto another scenario (which if they
answered the first one correctly should result in a pass or fail
situation)

you've upgraded to the new ios and rebooted the router, 30 minutes
have passed and you're unable to reach the router, how will you
proceed to troubleshoot/rectify the situation so the customer isn't
stuck with a dead router with 12 hours till business opens again.

On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> Thought you guys might want to take a break and help me out on a "related"
> issue...
>
> We are interviewing new Network Engineer Applicants next week.
>
> While I was writing up my list of questions (the boss needs 10) I thought I
> would throw this out to you guys.
> If you could write an interview question for your next Network Engineer
> (let's say Professional Level certification in Networking, Security or
> Voice)
> what would you ask them?  (try to keep it technical, nothing about Packer's
> tickets or Beer...lol)
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From asobihoudai at yahoo.com  Fri Oct  5 18:10:17 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 15:10:17 -0700 (PDT)
Subject: [cisco-voip] CCME 4 + CUE 3 networking
In-Reply-To: <960111.19800.qm@web57204.mail.re3.yahoo.com>
Message-ID: <998653.76362.qm@web57201.mail.re3.yahoo.com>

Is it possible to have remote CCME 4.0 users use a CUE
3.0.1 module on a remote box across the WAN? It sounds
slightly complex since MWI signals must be configured
as well as DSPs for transcoding G.711 -> G.729. I
can't find in the documents whether this is possible
or not.

Paul


       
____________________________________________________________________________________
Yahoo! oneSearch: Finally, mobile search 
that gives answers, not web links. 
http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC

From svoll.voip at gmail.com  Fri Oct  5 18:36:24 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 5 Oct 2007 15:36:24 -0700
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <778413.23734.qm@web57204.mail.re3.yahoo.com>
References: <5d093f9a0710051451n42fa4f58sacad3a176fd72fe2@mail.gmail.com>
	<778413.23734.qm@web57204.mail.re3.yahoo.com>
Message-ID: <f84a38d30710051536m16be400cv5f996c1d5ea4d1db@mail.gmail.com>

but the good news is theres no more java issues ;-)

On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> This is not encouraging news...
>
> We're about to embark on a CCM6 install on Monday. . .
>
> :(
>
>
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
>
> > You have no idea...
> >
> > The only way to show you is to have you set up a CCM
> > 4.3 box on
> > Windows 2003 and CCM6.0 box...
> >
> > Put CCM4.3 on a VMWare with too little ram (say
> > 1GB).
> >
> > Put CCM6.0 on a 7845.
> >
> > The 4.3 box will beat the 6.0 box by seconds on each
> > page load or query.
> >
> > I have timed it. The average time to load an empty
> > page (go to
> > device-phone, how long does it take for that page to
> > load?): for
> > CCM6.0, about 8 seconds, for 4.3, about 1.5.
> >
> > For queries, 4.3 nukes the crap out of 6 (or 5)...
> > it can be 20
> > seconds for a CCM6 query, and about 3 seconds for
> > 4.3...
> >
> >
> >
> >
> > Jonathan
> >
> > On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> > > Just be glad they didn't pick something like,
> > Oracle.
> > > Oh the humanity!
> > >
> > > IANADBA (I'm not a DBA) so I don't know the
> > difference
> > > in how SQL Server 2005 compares to say,
> > PostgreSQL. I
> > > hope that Cisco finds a faster DBMS if it really
> > is as
> > > slow as you folks make it sound.
> > >
> > > --- "Jason Aarons (US)"
> > <jason.aarons at us.didata.com>
> > > wrote:
> > >
> > > > Customers do complain the "Loading" is slow,
> > > > especially when on a Phone
> > > > DN you hit Line Partition and "Loading" changes
> > > > focus.  You would think
> > > > Cisco could speed it up.  I do miss the SQL
> > > > Enterprise Manager for
> > > > queries versus the SOAP AXL or command line I've
> > > > seen TAC do. TAC isn't
> > > > happy when they have to type long select
> > statements.
> > > >
> > > > Informix isn't the database of choice by DBAs.
> > > >
> > > > -----Original Message-----
> > > > From: cisco-voip-bounces at puck.nether.net
> > > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > > > Behalf Of Jonathan
> > > > Charles
> > > > Sent: Friday, October 05, 2007 1:19 PM
> > > > To: Bill Simon
> > > > Cc: cisco voip list
> > > > Subject: Re: [cisco-voip] CCM Web Page
> > performance
> > > > issues, aka
> > > > 'Loading,Please Wait' (a rant)
> > > >
> > > > We upgrade cuz the customer hears 'linux' and
> > thinks
> > > > it is better...
> > > > that is also what the sales people are
> > selling...
> > > >
> > > > To be honest, the only kick-ass features for
> > CCM5
> > > > are BAT and the LDAP
> > > > integration.
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On 10/5/07, Bill Simon <bills at psu.edu> wrote:
> > > > > Yeah, just a WAG but I'd suspect the DB, not
> > the
> > > > web server.
> > > > >
> > > > > CCM 4.x remains, in my opinion, the best out
> > > > there.  Why are you all
> > > > > moving to the higher versions, anyway?  Just
> > for
> > > > SIP endpoints?
> > > > >
> > > > >
> > > > > Paul Choi wrote:
> > > > > > Linux is just an OS. What people put on top
> > of
> > > > it is
> > > > > > largely their choice. If Cisco installed
> > wine
> > > > and ran
> > > > > > IIS on top of that for their web server,
> > then
> > > > that
> > > > > > would likely run slowly.
> > > > > >
> > > > > > I am not informed on the Linux-based CCM
> > boxen
> > > > but I'd
> > > > > > certainly hope that Cisco is using apache to
> > > > serve its
> > > > > > webpages. I've heard that the DB is Informix
> > or
> > > > > > something like that.
> > > > > >
> > > > > > My point being, it doesn't matter what OS
> > you
> > > > are
> > > > > > using - if you put a slow application on it,
> > > > it'll
> > > > > > still run slow no matter what you do to it.
> > > > > >
> > > > > > Paul
> > > > > >
> > > > > > --- Jonathan Charles <jonvoip at gmail.com>
> > wrote:
> > > > > >
> > > > > >> So, I am curious why all of us are putting
> > up
> > > > with
> > > > > >> this.
> > > > > >>
> > > > > >> For years we were all told that Linux was
> > > > superior,
> > > > > >> that Linux was how
> > > > > >> a real web server should function...
> > > > > >>
> > > > > >> Now, we have CCM 5 and CCM 6... and the web
> > > > server
> > > > > >> performance, is,
> > > > > >> er, well, how do we put this...
> > > > > >>
> > > > > >> Here we go. It is far worse than anything
> > > > Microsoft
> > > > > >> has ever released.
> > > > > >> I think Microsoft BOB would be a better web
> > > > server
> > > > > >> than whatever crap
> > > > > >> is on the backend of this RedHat
> > monstrosity.
> > > > > >>
> > > > > >> In CallManger 4.3, using Microsoft
> > > > WINDOWS(!!!!!!)
> > > > > >> IIS and Cisco
> > > > > >> Tomcat, I can scream through all of the web
> > > > pages;
> > > > > >> do DB reads in
> > > > > >> nanoseconds; do DB inserts almost as quick,
> > and
> > > > I
> > > > > >> have never seen a
> > > > > >> message on my web browser saying, 'Loading,
> > > > Please
> > > > > >> Wait'
> > > > > >>
> > > > > >> I mean, seriously, this is godawful slow.
> > > > > >>
> > > > > >> I just installed CCM5.1.3 on an HP 7845,
> > 4GB of
> > > > RAM,
> > > > > >> dual dual core
> > > > > >> Xeons... and all I see at every mouse click
> > is
> > > > > >> 'Loading, Please
> > > > > >> Wait'..
> > > > > >>
> > > > > >> Is Cisco punishing us for something? Did we
> > > > offend
> > > > > >> them in some way? I
> > > > > >> am at a point where I am wondering if I
> > should
> > > > tell
> > > > > >> customers about
> > > > > >> these performance issues prior to picking a
> > > > version
> > > > > >> ('if you want to
> > > > > >> be able to administer this and not slash
> > your
> > > > > >> wrists, waiting for the
> > > > > >> home page to load, you may want to go with
> > > > 4.3...')
> > > > > >> because I am tired
> > > > > >> of hearing about it afterwards...
> > > > > >>
> > > > > >> Seriously guys, I mean, this pretty much
> > kills
> > > > > >> whatever argument you
> > > > > >> Linux freaks had that this was better than
> > > > > >> Windows... add the whole
> > > > > >> SFTP requirement, and you can see why the
> > > > Microsoft
> > > > > >> side won this
> > > > > >> argument...
> > > > >
> > _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > >
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > > -----------------------------------------
> > > > Disclaimer:
> > > >
> > > > This e-mail communication and any attachments
> > may
> > > > contain
> > > > confidential and privileged information and is
> > for
> > > > use by the
> > > > designated addressee(s) named above only.  If
> > you
> > > > are not the
> > > > intended addressee, you are hereby notified that
> > you
> > > > have received
> > > > this communication in error and that any use or
> > > > reproduction of
> > > > this email or its contents is strictly
> > prohibited
> > > > and may be
> > > > unlawful.  If you have received this
> > communication
> > > > in error, please
> > > > notify us immediately by replying to this
> > message
> > > > and deleting it
> > > > from your computer. Thank you.
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > >
> > >
> > >
> > >
> >
>
> ____________________________________________________________________________________
> > > Be a better Globetrotter. Get better travel
> > answers from someone who knows. Yahoo! Answers -
> > Check it out.
> > >
> >
> http://answers.yahoo.com/dir/?link=list&sid=396545469
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
>
>
>
>
>
> ____________________________________________________________________________________
> Building a website is a piece of cake. Yahoo! Small Business gives you all
> the tools to get online.
> http://smallbusiness.yahoo.com/webhosting
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From asobihoudai at yahoo.com  Fri Oct  5 18:49:40 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 15:49:40 -0700 (PDT)
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <f84a38d30710051536m16be400cv5f996c1d5ea4d1db@mail.gmail.com>
Message-ID: <449016.72441.qm@web57207.mail.re3.yahoo.com>

and having to install antivirus crap...


--- Scott Voll <svoll.voip at gmail.com> wrote:

> but the good news is theres no more java issues ;-)
> 
> On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> >
> > This is not encouraging news...
> >
> > We're about to embark on a CCM6 install on Monday.
> . .
> >
> > :(
> >
> >
> > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > > You have no idea...
> > >
> > > The only way to show you is to have you set up a
> CCM
> > > 4.3 box on
> > > Windows 2003 and CCM6.0 box...
> > >
> > > Put CCM4.3 on a VMWare with too little ram (say
> > > 1GB).
> > >
> > > Put CCM6.0 on a 7845.
> > >
> > > The 4.3 box will beat the 6.0 box by seconds on
> each
> > > page load or query.
> > >
> > > I have timed it. The average time to load an
> empty
> > > page (go to
> > > device-phone, how long does it take for that
> page to
> > > load?): for
> > > CCM6.0, about 8 seconds, for 4.3, about 1.5.
> > >
> > > For queries, 4.3 nukes the crap out of 6 (or
> 5)...
> > > it can be 20
> > > seconds for a CCM6 query, and about 3 seconds
> for
> > > 4.3...
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On 10/5/07, Paul Choi <asobihoudai at yahoo.com>
> wrote:
> > > > Just be glad they didn't pick something like,
> > > Oracle.
> > > > Oh the humanity!
> > > >
> > > > IANADBA (I'm not a DBA) so I don't know the
> > > difference
> > > > in how SQL Server 2005 compares to say,
> > > PostgreSQL. I
> > > > hope that Cisco finds a faster DBMS if it
> really
> > > is as
> > > > slow as you folks make it sound.
> > > >
> > > > --- "Jason Aarons (US)"
> > > <jason.aarons at us.didata.com>
> > > > wrote:
> > > >
> > > > > Customers do complain the "Loading" is slow,
> > > > > especially when on a Phone
> > > > > DN you hit Line Partition and "Loading"
> changes
> > > > > focus.  You would think
> > > > > Cisco could speed it up.  I do miss the SQL
> > > > > Enterprise Manager for
> > > > > queries versus the SOAP AXL or command line
> I've
> > > > > seen TAC do. TAC isn't
> > > > > happy when they have to type long select
> > > statements.
> > > > >
> > > > > Informix isn't the database of choice by
> DBAs.
> > > > >
> > > > > -----Original Message-----
> > > > > From: cisco-voip-bounces at puck.nether.net
> > > > > [mailto:cisco-voip-bounces at puck.nether.net]
> On
> > > > > Behalf Of Jonathan
> > > > > Charles
> > > > > Sent: Friday, October 05, 2007 1:19 PM
> > > > > To: Bill Simon
> > > > > Cc: cisco voip list
> > > > > Subject: Re: [cisco-voip] CCM Web Page
> > > performance
> > > > > issues, aka
> > > > > 'Loading,Please Wait' (a rant)
> > > > >
> > > > > We upgrade cuz the customer hears 'linux'
> and
> > > thinks
> > > > > it is better...
> > > > > that is also what the sales people are
> > > selling...
> > > > >
> > > > > To be honest, the only kick-ass features for
> > > CCM5
> > > > > are BAT and the LDAP
> > > > > integration.
> > > > >
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > > On 10/5/07, Bill Simon <bills at psu.edu>
> wrote:
> > > > > > Yeah, just a WAG but I'd suspect the DB,
> not
> > > the
> > > > > web server.
> > > > > >
> > > > > > CCM 4.x remains, in my opinion, the best
> out
> > > > > there.  Why are you all
> > > > > > moving to the higher versions, anyway? 
> Just
> > > for
> > > > > SIP endpoints?
> > > > > >
> > > > > >
> > > > > > Paul Choi wrote:
> > > > > > > Linux is just an OS. What people put on
> top
> > > of
> > > > > it is
> > > > > > > largely their choice. If Cisco installed
> > > wine
> > > > > and ran
> > > > > > > IIS on top of that for their web server,
> > > then
> > > > > that
> > > > > > > would likely run slowly.
> > > > > > >
> > > > > > > I am not informed on the Linux-based CCM
> > > boxen
> > > > > but I'd
> > > > > > > certainly hope that Cisco is using
> apache to
> > > > > serve its
> > > > > > > webpages. I've heard that the DB is
> Informix
> > > or
> > > > > > > something like that.
> > > > > > >
> > > > > > > My point being, it doesn't matter what
> OS
> > > you
> > > > > are
> > > > > > > using - if you put a slow application on
> it,
> > > > > it'll
> > > > > > > still run slow no matter what you do to
> it.
> > > > > > >
> > > > > > > Paul
> > > > > > >
> > > > > > > --- Jonathan Charles <jonvoip at gmail.com>
> > > wrote:
> > > > > > >
> > > > > > >> So, I am curious why all of us are
> putting
> > > up
> > > > > with
> > > > > > >> this.
> > > > > > >>
> > > > > > >> For years we were all told that Linux
> was
> > > > > superior,
> > > > > > >> that Linux was how
> > > > > > >> a real web server should function...
> > > > > > >>
> > > > > > >> Now, we have CCM 5 and CCM 6... and the
> web
> > > > > server
> > > > > > >> performance, is,
> > > > > > >> er, well, how do we put this...
> > > > > > >>
> > > > > > >> Here we go. It is far worse than
> anything
> > > > > Microsoft
> > > > > > >> has ever released.
> > > > > > >> I think Microsoft BOB would be a better
> web
> > > > > server
> > > > > > >> than whatever crap
> > > > > > >> is on the backend of this RedHat
> > > monstrosity.
> > > > > > >>
> > > > > > >> In CallManger 4.3, using Microsoft
> > > > > WINDOWS(!!!!!!)
> > > > > > >> IIS and Cisco
> > > > > > >> Tomcat, I can scream through all of the
> web
> > > > > pages;
> > > > > > >> do DB reads in
> > > > > > >> nanoseconds; do DB inserts almost as
> quick,
> > > and
> > > > > I
> > > > > > >> have never seen a
> > > > > > >> message on my web browser saying,
> 'Loading,
> > > > > Please
> > > > > > >> Wait'
> > > > > > >>
> > > > > > >> I mean, seriously, this is godawful
> slow.
> > > > > > >>
> > > > > > >> I just installed CCM5.1.3 on an HP
> 7845,
> > > 4GB of
> > > > > RAM,
> > > > > > >> dual dual core
> > > > > > >> Xeons... and all I see at every mouse
> click
> > > is
> > > > > > >> 'Loading, Please
> > > > > > >> Wait'..
> > > > > > >>
> > > > > > >> Is Cisco punishing us for something?
> Did we
> > > > > offend
> > > > > > >> them in some way? I
> > > > > > >> am at a point where I am wondering if I
> > > should
> > > > > tell
> > > > > > >> customers about
> > > > > > >> these performance issues prior to
> picking a
> > > > > version
> > > > > > >> ('if you want to
> > > > > > >> be able to administer this and not
> slash
> > > your
> > > > > > >> wrists, waiting for the
> > > > > > >> home page to load, you may want to go
> with
> > > > > 4.3...')
> > > > > > >> because I am tired
> > > > > > >> of hearing about it afterwards...
> > > > > > >>
> > > > > > >> Seriously guys, I mean, this pretty
> much
> > > kills
> > > > > > >> whatever argument you
> > > > > > >> Linux freaks had that this was better
> than
> > > > > > >> Windows... add the whole
> > > > > > >> SFTP requirement, and you can see why
> the
> > > > > Microsoft
> > > > > > >> side won this
> > > > > > >> argument...
> > > > > >
> > > _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > >
> > > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > >
> _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > > > -----------------------------------------
> > > > > Disclaimer:
> > > > >
> > > > > This e-mail communication and any
> attachments
> > > may
> > > > > contain
> > > > > confidential and privileged information and
> is
> > > for
> > > > > use by the
> > > > > designated addressee(s) named above only. 
> If
> > > you
> > > > > are not the
> > > > > intended addressee, you are hereby notified
> that
> > > you
> > > > > have received
> > > > > this communication in error and that any use
> or
> > > > > reproduction of
> > > > > this email or its contents is strictly
> > > prohibited
> > > > > and may be
> > > > > unlawful.  If you have received this
> > > communication
> > > > > in error, please
> > > > > notify us immediately by replying to this
> > > message
> > > > > and deleting it
> > > > > from your computer. Thank you.
> > > > >
> _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > >
> >
> >
>
____________________________________________________________________________________
> > > > Be a better Globetrotter. Get better travel
> > > answers from someone who knows. Yahoo! Answers -
> > > Check it out.
> > > >
> > >
> >
>
http://answers.yahoo.com/dir/?link=list&sid=396545469
> > > >
> _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> >
> >
> >
> >
> >
> >
>
____________________________________________________________________________________
> > Building a website is a piece of cake. Yahoo!
> Small Business gives you all
> > the tools to get online.
> > http://smallbusiness.yahoo.com/webhosting
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 



       
____________________________________________________________________________________
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From jonvoip at gmail.com  Fri Oct  5 18:52:26 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 5 Oct 2007 17:52:26 -0500
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <449016.72441.qm@web57207.mail.re3.yahoo.com>
References: <f84a38d30710051536m16be400cv5f996c1d5ea4d1db@mail.gmail.com>
	<449016.72441.qm@web57207.mail.re3.yahoo.com>
Message-ID: <5d093f9a0710051552y3404733ale2c135e424aaae6d@mail.gmail.com>

Yeah, cuz being annoyed for 15 seconds while you have to type in your
credentials into the java console, and having to install mcafee is
much worse than the 500 hours each year you will waste while waiting
for a ccm web page to load...

Stick with CCM 4.3...

The weird thing is that ccm 6 is easily twice as bad as ccm 5... if
this progression continues, I figure by ccm 10 it will take two hours
to search for a phone.


Jonathan

On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> and having to install antivirus crap...
>
>
> --- Scott Voll <svoll.voip at gmail.com> wrote:
>
> > but the good news is theres no more java issues ;-)
> >
> > On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> > >
> > > This is not encouraging news...
> > >
> > > We're about to embark on a CCM6 install on Monday.
> > . .
> > >
> > > :(
> > >
> > >
> > > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > > > You have no idea...
> > > >
> > > > The only way to show you is to have you set up a
> > CCM
> > > > 4.3 box on
> > > > Windows 2003 and CCM6.0 box...
> > > >
> > > > Put CCM4.3 on a VMWare with too little ram (say
> > > > 1GB).
> > > >
> > > > Put CCM6.0 on a 7845.
> > > >
> > > > The 4.3 box will beat the 6.0 box by seconds on
> > each
> > > > page load or query.
> > > >
> > > > I have timed it. The average time to load an
> > empty
> > > > page (go to
> > > > device-phone, how long does it take for that
> > page to
> > > > load?): for
> > > > CCM6.0, about 8 seconds, for 4.3, about 1.5.
> > > >
> > > > For queries, 4.3 nukes the crap out of 6 (or
> > 5)...
> > > > it can be 20
> > > > seconds for a CCM6 query, and about 3 seconds
> > for
> > > > 4.3...
> > > >
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On 10/5/07, Paul Choi <asobihoudai at yahoo.com>
> > wrote:
> > > > > Just be glad they didn't pick something like,
> > > > Oracle.
> > > > > Oh the humanity!
> > > > >
> > > > > IANADBA (I'm not a DBA) so I don't know the
> > > > difference
> > > > > in how SQL Server 2005 compares to say,
> > > > PostgreSQL. I
> > > > > hope that Cisco finds a faster DBMS if it
> > really
> > > > is as
> > > > > slow as you folks make it sound.
> > > > >
> > > > > --- "Jason Aarons (US)"
> > > > <jason.aarons at us.didata.com>
> > > > > wrote:
> > > > >
> > > > > > Customers do complain the "Loading" is slow,
> > > > > > especially when on a Phone
> > > > > > DN you hit Line Partition and "Loading"
> > changes
> > > > > > focus.  You would think
> > > > > > Cisco could speed it up.  I do miss the SQL
> > > > > > Enterprise Manager for
> > > > > > queries versus the SOAP AXL or command line
> > I've
> > > > > > seen TAC do. TAC isn't
> > > > > > happy when they have to type long select
> > > > statements.
> > > > > >
> > > > > > Informix isn't the database of choice by
> > DBAs.
> > > > > >
> > > > > > -----Original Message-----
> > > > > > From: cisco-voip-bounces at puck.nether.net
> > > > > > [mailto:cisco-voip-bounces at puck.nether.net]
> > On
> > > > > > Behalf Of Jonathan
> > > > > > Charles
> > > > > > Sent: Friday, October 05, 2007 1:19 PM
> > > > > > To: Bill Simon
> > > > > > Cc: cisco voip list
> > > > > > Subject: Re: [cisco-voip] CCM Web Page
> > > > performance
> > > > > > issues, aka
> > > > > > 'Loading,Please Wait' (a rant)
> > > > > >
> > > > > > We upgrade cuz the customer hears 'linux'
> > and
> > > > thinks
> > > > > > it is better...
> > > > > > that is also what the sales people are
> > > > selling...
> > > > > >
> > > > > > To be honest, the only kick-ass features for
> > > > CCM5
> > > > > > are BAT and the LDAP
> > > > > > integration.
> > > > > >
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > >
> > > > > > On 10/5/07, Bill Simon <bills at psu.edu>
> > wrote:
> > > > > > > Yeah, just a WAG but I'd suspect the DB,
> > not
> > > > the
> > > > > > web server.
> > > > > > >
> > > > > > > CCM 4.x remains, in my opinion, the best
> > out
> > > > > > there.  Why are you all
> > > > > > > moving to the higher versions, anyway?
> > Just
> > > > for
> > > > > > SIP endpoints?
> > > > > > >
> > > > > > >
> > > > > > > Paul Choi wrote:
> > > > > > > > Linux is just an OS. What people put on
> > top
> > > > of
> > > > > > it is
> > > > > > > > largely their choice. If Cisco installed
> > > > wine
> > > > > > and ran
> > > > > > > > IIS on top of that for their web server,
> > > > then
> > > > > > that
> > > > > > > > would likely run slowly.
> > > > > > > >
> > > > > > > > I am not informed on the Linux-based CCM
> > > > boxen
> > > > > > but I'd
> > > > > > > > certainly hope that Cisco is using
> > apache to
> > > > > > serve its
> > > > > > > > webpages. I've heard that the DB is
> > Informix
> > > > or
> > > > > > > > something like that.
> > > > > > > >
> > > > > > > > My point being, it doesn't matter what
> > OS
> > > > you
> > > > > > are
> > > > > > > > using - if you put a slow application on
> > it,
> > > > > > it'll
> > > > > > > > still run slow no matter what you do to
> > it.
> > > > > > > >
> > > > > > > > Paul
> > > > > > > >
> > > > > > > > --- Jonathan Charles <jonvoip at gmail.com>
> > > > wrote:
> > > > > > > >
> > > > > > > >> So, I am curious why all of us are
> > putting
> > > > up
> > > > > > with
> > > > > > > >> this.
> > > > > > > >>
> > > > > > > >> For years we were all told that Linux
> > was
> > > > > > superior,
> > > > > > > >> that Linux was how
> > > > > > > >> a real web server should function...
> > > > > > > >>
> > > > > > > >> Now, we have CCM 5 and CCM 6... and the
> > web
> > > > > > server
> > > > > > > >> performance, is,
> > > > > > > >> er, well, how do we put this...
> > > > > > > >>
> > > > > > > >> Here we go. It is far worse than
> > anything
> > > > > > Microsoft
> > > > > > > >> has ever released.
> > > > > > > >> I think Microsoft BOB would be a better
> > web
> > > > > > server
> > > > > > > >> than whatever crap
> > > > > > > >> is on the backend of this RedHat
> > > > monstrosity.
> > > > > > > >>
> > > > > > > >> In CallManger 4.3, using Microsoft
> > > > > > WINDOWS(!!!!!!)
> > > > > > > >> IIS and Cisco
> > > > > > > >> Tomcat, I can scream through all of the
> > web
> > > > > > pages;
> > > > > > > >> do DB reads in
> > > > > > > >> nanoseconds; do DB inserts almost as
> > quick,
> > > > and
> > > > > > I
> > > > > > > >> have never seen a
> > > > > > > >> message on my web browser saying,
> > 'Loading,
> > > > > > Please
> > > > > > > >> Wait'
> > > > > > > >>
> > > > > > > >> I mean, seriously, this is godawful
> > slow.
> > > > > > > >>
> > > > > > > >> I just installed CCM5.1.3 on an HP
> > 7845,
> > > > 4GB of
> > > > > > RAM,
> > > > > > > >> dual dual core
> > > > > > > >> Xeons... and all I see at every mouse
> > click
> > > > is
> > > > > > > >> 'Loading, Please
> > > > > > > >> Wait'..
> > > > > > > >>
> > > > > > > >> Is Cisco punishing us for something?
> > Did we
> > > > > > offend
> > > > > > > >> them in some way? I
> > > > > > > >> am at a point where I am wondering if I
> > > > should
> > > > > > tell
> > > > > > > >> customers about
> > > > > > > >> these performance issues prior to
> > picking a
> > > > > > version
> > > > > > > >> ('if you want to
> > > > > > > >> be able to administer this and not
> > slash
> > > > your
> > > > > > > >> wrists, waiting for the
> > > > > > > >> home page to load, you may want to go
> > with
> > > > > > 4.3...')
> > > > > > > >> because I am tired
> > > > > > > >> of hearing about it afterwards...
> > > > > > > >>
> > > > > > > >> Seriously guys, I mean, this pretty
> > much
> > > > kills
> > > > > > > >> whatever argument you
> > > > > > > >> Linux freaks had that this was better
> > than
> > > > > > > >> Windows... add the whole
> > > > > > > >> SFTP requirement, and you can see why
> > the
> > > > > > Microsoft
> > > > > > > >> side won this
> > > > > > > >> argument...
> > > > > > >
> > > > _______________________________________________
> > > > > > > cisco-voip mailing list
> > > > > > > cisco-voip at puck.nether.net
> > > > > > >
> > > > > >
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > >
> > > > > >
> > _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > >
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > > > -----------------------------------------
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> > > > > >
> > > > > > This e-mail communication and any
> > attachments
> > > > may
> > > > > > contain
> > > > > > confidential and privileged information and
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> > > > for
> > > > > > use by the
> > > > > > designated addressee(s) named above only.
> > If
> > > > you
> > > > > > are not the
> > > > > > intended addressee, you are hereby notified
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> > or
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> > > > > > this email or its contents is strictly
> > > > prohibited
> > > > > > and may be
> > > > > > unlawful.  If you have received this
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> > > > > > in error, please
> > > > > > notify us immediately by replying to this
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> > > > > > and deleting it
> > > > > > from your computer. Thank you.
> > > > > >
> > _______________________________________________
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> > > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > >
> > >
> > >
> >
> ____________________________________________________________________________________
> > > > > Be a better Globetrotter. Get better travel
> > > > answers from someone who knows. Yahoo! Answers -
> > > > Check it out.
> > > > >
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> > >
> >
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> > > > >
> > _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > >
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > >
> > >
> > >
> > >
> > >
> > >
> > >
> >
> ____________________________________________________________________________________
> > > Building a website is a piece of cake. Yahoo!
> > Small Business gives you all
> > > the tools to get online.
> > > http://smallbusiness.yahoo.com/webhosting
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> > >
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> >
>
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From asobihoudai at yahoo.com  Fri Oct  5 19:21:20 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 5 Oct 2007 16:21:20 -0700 (PDT)
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <5d093f9a0710051552y3404733ale2c135e424aaae6d@mail.gmail.com>
Message-ID: <848628.37979.qm@web57208.mail.re3.yahoo.com>

I'd argue that not having to worry about viruses is
huge. I have a customer whose phone system was taken
out for two weeks during Slammer. There are still
rootkits but those are far fewer than viruses.

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> Yeah, cuz being annoyed for 15 seconds while you
> have to type in your
> credentials into the java console, and having to
> install mcafee is
> much worse than the 500 hours each year you will
> waste while waiting
> for a ccm web page to load...
> 
> Stick with CCM 4.3...
> 
> The weird thing is that ccm 6 is easily twice as bad
> as ccm 5... if
> this progression continues, I figure by ccm 10 it
> will take two hours
> to search for a phone.
> 
> 
> Jonathan
> 
> On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> > and having to install antivirus crap...
> >
> >
> > --- Scott Voll <svoll.voip at gmail.com> wrote:
> >
> > > but the good news is theres no more java issues
> ;-)
> > >
> > > On 10/5/07, Paul Choi <asobihoudai at yahoo.com>
> wrote:
> > > >
> > > > This is not encouraging news...
> > > >
> > > > We're about to embark on a CCM6 install on
> Monday.
> > > . .
> > > >
> > > > :(
> > > >
> > > >
> > > > --- Jonathan Charles <jonvoip at gmail.com>
> wrote:
> > > >
> > > > > You have no idea...
> > > > >
> > > > > The only way to show you is to have you set
> up a
> > > CCM
> > > > > 4.3 box on
> > > > > Windows 2003 and CCM6.0 box...
> > > > >
> > > > > Put CCM4.3 on a VMWare with too little ram
> (say
> > > > > 1GB).
> > > > >
> > > > > Put CCM6.0 on a 7845.
> > > > >
> > > > > The 4.3 box will beat the 6.0 box by seconds
> on
> > > each
> > > > > page load or query.
> > > > >
> > > > > I have timed it. The average time to load an
> > > empty
> > > > > page (go to
> > > > > device-phone, how long does it take for that
> > > page to
> > > > > load?): for
> > > > > CCM6.0, about 8 seconds, for 4.3, about 1.5.
> > > > >
> > > > > For queries, 4.3 nukes the crap out of 6 (or
> > > 5)...
> > > > > it can be 20
> > > > > seconds for a CCM6 query, and about 3
> seconds
> > > for
> > > > > 4.3...
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > > On 10/5/07, Paul Choi
> <asobihoudai at yahoo.com>
> > > wrote:
> > > > > > Just be glad they didn't pick something
> like,
> > > > > Oracle.
> > > > > > Oh the humanity!
> > > > > >
> > > > > > IANADBA (I'm not a DBA) so I don't know
> the
> > > > > difference
> > > > > > in how SQL Server 2005 compares to say,
> > > > > PostgreSQL. I
> > > > > > hope that Cisco finds a faster DBMS if it
> > > really
> > > > > is as
> > > > > > slow as you folks make it sound.
> > > > > >
> > > > > > --- "Jason Aarons (US)"
> > > > > <jason.aarons at us.didata.com>
> > > > > > wrote:
> > > > > >
> > > > > > > Customers do complain the "Loading" is
> slow,
> > > > > > > especially when on a Phone
> > > > > > > DN you hit Line Partition and "Loading"
> > > changes
> > > > > > > focus.  You would think
> > > > > > > Cisco could speed it up.  I do miss the
> SQL
> > > > > > > Enterprise Manager for
> > > > > > > queries versus the SOAP AXL or command
> line
> > > I've
> > > > > > > seen TAC do. TAC isn't
> > > > > > > happy when they have to type long select
> > > > > statements.
> > > > > > >
> > > > > > > Informix isn't the database of choice by
> > > DBAs.
> > > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: cisco-voip-bounces at puck.nether.net
> > > > > > >
> [mailto:cisco-voip-bounces at puck.nether.net]
> > > On
> > > > > > > Behalf Of Jonathan
> > > > > > > Charles
> > > > > > > Sent: Friday, October 05, 2007 1:19 PM
> > > > > > > To: Bill Simon
> > > > > > > Cc: cisco voip list
> > > > > > > Subject: Re: [cisco-voip] CCM Web Page
> > > > > performance
> > > > > > > issues, aka
> > > > > > > 'Loading,Please Wait' (a rant)
> > > > > > >
> > > > > > > We upgrade cuz the customer hears
> 'linux'
> > > and
> > > > > thinks
> > > > > > > it is better...
> > > > > > > that is also what the sales people are
> > > > > selling...
> > > > > > >
> > > > > > > To be honest, the only kick-ass features
> for
> > > > > CCM5
> > > > > > > are BAT and the LDAP
> > > > > > > integration.
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > Jonathan
> > > > > > >
> > > > > > > On 10/5/07, Bill Simon <bills at psu.edu>
> > > wrote:
> > > > > > > > Yeah, just a WAG but I'd suspect the
> DB,
> > > not
> > > > > the
> > > > > > > web server.
> > > > > > > >
> > > > > > > > CCM 4.x remains, in my opinion, the
> best
> > > out
> > > > > > > there.  Why are you all
> > > > > > > > moving to the higher versions, anyway?
> > > Just
> > > > > for
> > > > > > > SIP endpoints?
> > > > > > > >
> > > > > > > >
> > > > > > > > Paul Choi wrote:
> > > > > > > > > Linux is just an OS. What people put
> on
> > > top
> > > > > of
> > > > > > > it is
> > > > > > > > > largely their choice. If Cisco
> installed
> > > > > wine
> > > > > > > and ran
> > > > > > > > > IIS on top of that for their web
> server,
> > > > > then
> > > > > > > that
> > > > > > > > > would likely run slowly.
> > > > > > > > >
> > > > > > > > > I am not informed on the Linux-based
> CCM
> > > > > boxen
> > > > > > > but I'd
> > > > > > > > > certainly hope that Cisco is using
> > > apache to
> > > > > > > serve its
> > > > > > > > > webpages. I've heard that the DB is
> > > Informix
> > > > > or
> > > > > > > > > something like that.
> > > > > > > > >
> > > > > > > > > My point being, it doesn't matter
> what
> > > OS
> > > > > you
> > > > > > > are
> > > > > > > > > using - if you put a slow
> application on
> > > it,
> > > > > > > it'll
> > > > > > > > > still run slow no matter what you do
> to
> > > it.
> > > > > > > > >
> > > > > > > > > Paul
> > > > > > > > >
> > > > > > > > > --- Jonathan Charles
> <jonvoip at gmail.com>
> > > > > wrote:
> > > > > > > > >
> > > > > > > > >> So, I am curious why all of us are
> > > putting
> > > > > up
> > > > > > > with
> > > > > > > > >> this.
> > > > > > > > >>
> > > > > > > > >> For years we were all told that
> Linux
> > > was
> > > > > > > superior,
> > > > > > > > >> that Linux was how
> > > > > > > > >> a real web server should
> function...
> > > > > > > > >>
> > > > > > > > >> Now, we have CCM 5 and CCM 6... and
> the
> > > web
> > > > > > > server
> > > > > > > > >> performance, is,
> > > > > > > > >> er, well, how do we put this...
> > > > > > > > >>
> > > > > > > > >> Here we go. It is far worse than
> > > anything
> > > > > > > Microsoft
> > > > > > > > >> has ever released.
> > > > > > > > >> I think Microsoft BOB would be a
> better
> > > web
> > > > > > > server
> > > > > > > > >> than whatever crap
> > > > > > > > >> is on the backend of this RedHat
> > > > > monstrosity.
> > > > > > > > >>
> > > > > > > > >> In CallManger 4.3, using Microsoft
> > > > > > > WINDOWS(!!!!!!)
> > > > > > > > >> IIS and Cisco
> > > > > > > > >> Tomcat, I can scream through all of
> the
> > > web
> > > > > > > pages;
> > > > > > > > >> do DB reads in
> > > > > > > > >> nanoseconds; do DB inserts almost
> as
> > > quick,
> > > > > and
> > > > > > > I
> > > > > > > > >> have never seen a
> > > > > > > > >> message on my web browser saying,
> > > 'Loading,
> > > > > > > Please
> > > > > > > > >> Wait'
> > > > > > > > >>
> > > > > > > > >> I mean, seriously, this is godawful
> > > slow.
> > > > > > > > >>
> > > > > > > > >> I just installed CCM5.1.3 on an HP
> > > 7845,
> > > > > 4GB of
> > > > > > > RAM,
> > > > > > > > >> dual dual core
> > > > > > > > >> Xeons... and all I see at every
> mouse
> > > click
> > > > > is
> > > > > > > > >> 'Loading, Please
> > > > > > > > >> Wait'..
> > > > > > > > >>
> > > > > > > > >> Is Cisco punishing us for
> something?
> > > Did we
> > > > > > > offend
> > > > > > > > >> them in some way? I
> > > > > > > > >> am at a point where I am wondering
> if I
> > > > > should
> > > > > > > tell
> > > > > > > > >> customers about
> > > > > > > > >> these performance issues prior to
> > > picking a
> > > > > > > version
> > > > > > > > >> ('if you want to
> > > > > > > > >> be able to administer this and not
> > > slash
> > > > > your
> > > > > > > > >> wrists, waiting for the
> > > > > > > > >> home page to load, you may want to
> go
> > > with
> > > > > > > 4.3...')
> > > > > > > > >> because I am tired
> > > > > > > > >> of hearing about it afterwards...
> > > > > > > > >>
> > > > > > > > >> Seriously guys, I mean, this pretty
> > > much
> > > > > kills
> > > > > > > > >> whatever argument you
> > > > > > > > >> Linux freaks had that this was
> better
> > > than
> > > > > > > > >> Windows... add the whole
> > > > > > > > >> SFTP requirement, and you can see
> why
> > > the
> > > > > > > Microsoft
> > > > > > > > >> side won this
> > > > > > > > >> argument...
> > > > > > > >
> > > > >
> _______________________________________________
> > > > > > > > cisco-voip mailing list
> > > > > > > > cisco-voip at puck.nether.net
> > > > > > > >
> > > > > > >
> > > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > > >
> > > > > > >
> > > _______________________________________________
> > > > > > > cisco-voip mailing list
> > > > > > > cisco-voip at puck.nether.net
> > > > > > >
> > > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > >
> > > > > > >
> -----------------------------------------
> > > > > > > Disclaimer:
> > > > > > >
> > > > > > > This e-mail communication and any
> > > attachments
> > > > > may
> > > > > > > contain
> > > > > > > confidential and privileged information
> and
> > > is
> > > > > for
> > > > > > > use by the
> > > > > > > designated addressee(s) named above
> only.
> > > If
> > > > > you
> > > > > > > are not the
> > > > > > > intended addressee, you are hereby
> notified
> > > that
> > > > > you
> > > > > > > have received
> > > > > > > this communication in error and that any
> use
> > > or
> > > > > > > reproduction of
> > > > > > > this email or its contents is strictly
> > > > > prohibited
> > > > > > > and may be
> > > > > > > unlawful.  If you have received this
> > > > > communication
> > > > > > > in error, please
> > > > > > > notify us immediately by replying to
> this
> > > > > message
> > > > > > > and deleting it
> > > > > > > from your computer. Thank you.
> > > > > > >
> > > _______________________________________________
> > > > > > > cisco-voip mailing list
> > > > > > > cisco-voip at puck.nether.net
> > > > > > >
> > > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > >
> > > >
> > > >
> > >
> >
>
____________________________________________________________________________________
> > > > > > Be a better Globetrotter. Get better
> travel
> > > > > answers from someone who knows. Yahoo!
> Answers -
> > > > > Check it out.
> > > > > >
> > > > >
> > > >
> > >
> >
>
http://answers.yahoo.com/dir/?link=list&sid=396545469
> > > > > >
> > > _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > >
> > > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > >
> >
>
____________________________________________________________________________________
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> > > >
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> > > >
> > >
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> > > >
> > >
> >
> >
> >
> >
> >
>
____________________________________________________________________________________
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> and hotels with Yahoo! FareChase.
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      ____________________________________________________________________________________
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From jonvoip at gmail.com  Fri Oct  5 22:29:36 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 5 Oct 2007 21:29:36 -0500
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <848628.37979.qm@web57208.mail.re3.yahoo.com>
References: <5d093f9a0710051552y3404733ale2c135e424aaae6d@mail.gmail.com>
	<848628.37979.qm@web57208.mail.re3.yahoo.com>
Message-ID: <5d093f9a0710051929j24ac721el929018090d207322@mail.gmail.com>

In my four years of doing this, I have not directly experienced a
single customer with a virus on CallManager.

I have had customers with Virii on their Unity servers however.

Mostly it is because the CCM doesn't have the same default gateway and
is on a different subnet than the PCs, and they generally don't have
internet access.

Yes, it is a big deal if/when it happens, however, you are far more
likely to lose a publisher from bad hardware than from a virus.

Also, I am pretty sure I could rebuild their Publisher and all their
Subscribers in a day or so (if they bothered backing it up, which was
easy in CCM4.X).

To be really honest, I think CSA on CCM kills most viruses before they
can do anything.


Jonathan

On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> I'd argue that not having to worry about viruses is
> huge. I have a customer whose phone system was taken
> out for two weeks during Slammer. There are still
> rootkits but those are far fewer than viruses.
>
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
>
> > Yeah, cuz being annoyed for 15 seconds while you
> > have to type in your
> > credentials into the java console, and having to
> > install mcafee is
> > much worse than the 500 hours each year you will
> > waste while waiting
> > for a ccm web page to load...
> >
> > Stick with CCM 4.3...
> >
> > The weird thing is that ccm 6 is easily twice as bad
> > as ccm 5... if
> > this progression continues, I figure by ccm 10 it
> > will take two hours
> > to search for a phone.
> >
> >
> > Jonathan
> >
> > On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> > > and having to install antivirus crap...
> > >
> > >
> > > --- Scott Voll <svoll.voip at gmail.com> wrote:
> > >
> > > > but the good news is theres no more java issues
> > ;-)
> > > >
> > > > On 10/5/07, Paul Choi <asobihoudai at yahoo.com>
> > wrote:
> > > > >
> > > > > This is not encouraging news...
> > > > >
> > > > > We're about to embark on a CCM6 install on
> > Monday.
> > > > . .
> > > > >
> > > > > :(
> > > > >
> > > > >
> > > > > --- Jonathan Charles <jonvoip at gmail.com>
> > wrote:
> > > > >
> > > > > > You have no idea...
> > > > > >
> > > > > > The only way to show you is to have you set
> > up a
> > > > CCM
> > > > > > 4.3 box on
> > > > > > Windows 2003 and CCM6.0 box...
> > > > > >
> > > > > > Put CCM4.3 on a VMWare with too little ram
> > (say
> > > > > > 1GB).
> > > > > >
> > > > > > Put CCM6.0 on a 7845.
> > > > > >
> > > > > > The 4.3 box will beat the 6.0 box by seconds
> > on
> > > > each
> > > > > > page load or query.
> > > > > >
> > > > > > I have timed it. The average time to load an
> > > > empty
> > > > > > page (go to
> > > > > > device-phone, how long does it take for that
> > > > page to
> > > > > > load?): for
> > > > > > CCM6.0, about 8 seconds, for 4.3, about 1.5.
> > > > > >
> > > > > > For queries, 4.3 nukes the crap out of 6 (or
> > > > 5)...
> > > > > > it can be 20
> > > > > > seconds for a CCM6 query, and about 3
> > seconds
> > > > for
> > > > > > 4.3...
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > >
> > > > > > On 10/5/07, Paul Choi
> > <asobihoudai at yahoo.com>
> > > > wrote:
> > > > > > > Just be glad they didn't pick something
> > like,
> > > > > > Oracle.
> > > > > > > Oh the humanity!
> > > > > > >
> > > > > > > IANADBA (I'm not a DBA) so I don't know
> > the
> > > > > > difference
> > > > > > > in how SQL Server 2005 compares to say,
> > > > > > PostgreSQL. I
> > > > > > > hope that Cisco finds a faster DBMS if it
> > > > really
> > > > > > is as
> > > > > > > slow as you folks make it sound.
> > > > > > >
> > > > > > > --- "Jason Aarons (US)"
> > > > > > <jason.aarons at us.didata.com>
> > > > > > > wrote:
> > > > > > >
> > > > > > > > Customers do complain the "Loading" is
> > slow,
> > > > > > > > especially when on a Phone
> > > > > > > > DN you hit Line Partition and "Loading"
> > > > changes
> > > > > > > > focus.  You would think
> > > > > > > > Cisco could speed it up.  I do miss the
> > SQL
> > > > > > > > Enterprise Manager for
> > > > > > > > queries versus the SOAP AXL or command
> > line
> > > > I've
> > > > > > > > seen TAC do. TAC isn't
> > > > > > > > happy when they have to type long select
> > > > > > statements.
> > > > > > > >
> > > > > > > > Informix isn't the database of choice by
> > > > DBAs.
> > > > > > > >
> > > > > > > > -----Original Message-----
> > > > > > > > From: cisco-voip-bounces at puck.nether.net
> > > > > > > >
> > [mailto:cisco-voip-bounces at puck.nether.net]
> > > > On
> > > > > > > > Behalf Of Jonathan
> > > > > > > > Charles
> > > > > > > > Sent: Friday, October 05, 2007 1:19 PM
> > > > > > > > To: Bill Simon
> > > > > > > > Cc: cisco voip list
> > > > > > > > Subject: Re: [cisco-voip] CCM Web Page
> > > > > > performance
> > > > > > > > issues, aka
> > > > > > > > 'Loading,Please Wait' (a rant)
> > > > > > > >
> > > > > > > > We upgrade cuz the customer hears
> > 'linux'
> > > > and
> > > > > > thinks
> > > > > > > > it is better...
> > > > > > > > that is also what the sales people are
> > > > > > selling...
> > > > > > > >
> > > > > > > > To be honest, the only kick-ass features
> > for
> > > > > > CCM5
> > > > > > > > are BAT and the LDAP
> > > > > > > > integration.
> > > > > > > >
> > > > > > > >
> > > > > > > >
> > > > > > > > Jonathan
> > > > > > > >
> > > > > > > > On 10/5/07, Bill Simon <bills at psu.edu>
> > > > wrote:
> > > > > > > > > Yeah, just a WAG but I'd suspect the
> > DB,
> > > > not
> > > > > > the
> > > > > > > > web server.
> > > > > > > > >
> > > > > > > > > CCM 4.x remains, in my opinion, the
> > best
> > > > out
> > > > > > > > there.  Why are you all
> > > > > > > > > moving to the higher versions, anyway?
> > > > Just
> > > > > > for
> > > > > > > > SIP endpoints?
> > > > > > > > >
> > > > > > > > >
> > > > > > > > > Paul Choi wrote:
> > > > > > > > > > Linux is just an OS. What people put
> > on
> > > > top
> > > > > > of
> > > > > > > > it is
> > > > > > > > > > largely their choice. If Cisco
> > installed
> > > > > > wine
> > > > > > > > and ran
> > > > > > > > > > IIS on top of that for their web
> > server,
> > > > > > then
> > > > > > > > that
> > > > > > > > > > would likely run slowly.
> > > > > > > > > >
> > > > > > > > > > I am not informed on the Linux-based
> > CCM
> > > > > > boxen
> > > > > > > > but I'd
> > > > > > > > > > certainly hope that Cisco is using
> > > > apache to
> > > > > > > > serve its
> > > > > > > > > > webpages. I've heard that the DB is
> > > > Informix
> > > > > > or
> > > > > > > > > > something like that.
> > > > > > > > > >
> > > > > > > > > > My point being, it doesn't matter
> > what
> > > > OS
> > > > > > you
> > > > > > > > are
> > > > > > > > > > using - if you put a slow
> > application on
> > > > it,
> > > > > > > > it'll
> > > > > > > > > > still run slow no matter what you do
> > to
> > > > it.
> > > > > > > > > >
> > > > > > > > > > Paul
> > > > > > > > > >
> > > > > > > > > > --- Jonathan Charles
> > <jonvoip at gmail.com>
> > > > > > wrote:
> > > > > > > > > >
> > > > > > > > > >> So, I am curious why all of us are
> > > > putting
> > > > > > up
> > > > > > > > with
> > > > > > > > > >> this.
> > > > > > > > > >>
> > > > > > > > > >> For years we were all told that
> > Linux
> > > > was
> > > > > > > > superior,
> > > > > > > > > >> that Linux was how
> > > > > > > > > >> a real web server should
> > function...
> > > > > > > > > >>
> > > > > > > > > >> Now, we have CCM 5 and CCM 6... and
> > the
> > > > web
> > > > > > > > server
> > > > > > > > > >> performance, is,
> > > > > > > > > >> er, well, how do we put this...
> > > > > > > > > >>
> > > > > > > > > >> Here we go. It is far worse than
> > > > anything
> > > > > > > > Microsoft
> > > > > > > > > >> has ever released.
> > > > > > > > > >> I think Microsoft BOB would be a
> > better
> > > > web
> > > > > > > > server
> > > > > > > > > >> than whatever crap
> > > > > > > > > >> is on the backend of this RedHat
> > > > > > monstrosity.
> > > > > > > > > >>
> > > > > > > > > >> In CallManger 4.3, using Microsoft
> > > > > > > > WINDOWS(!!!!!!)
> > > > > > > > > >> IIS and Cisco
> > > > > > > > > >> Tomcat, I can scream through all of
> > the
> > > > web
> > > > > > > > pages;
> > > > > > > > > >> do DB reads in
> > > > > > > > > >> nanoseconds; do DB inserts almost
> > as
> > > > quick,
> > > > > > and
> > > > > > > > I
> > > > > > > > > >> have never seen a
> > > > > > > > > >> message on my web browser saying,
> > > > 'Loading,
> > > > > > > > Please
> > > > > > > > > >> Wait'
> > > > > > > > > >>
> > > > > > > > > >> I mean, seriously, this is godawful
> > > > slow.
> > > > > > > > > >>
> > > > > > > > > >> I just installed CCM5.1.3 on an HP
> > > > 7845,
> > > > > > 4GB of
> > > > > > > > RAM,
> > > > > > > > > >> dual dual core
> > > > > > > > > >> Xeons... and all I see at every
> > mouse
> > > > click
> > > > > > is
> > > > > > > > > >> 'Loading, Please
> > > > > > > > > >> Wait'..
> > > > > > > > > >>
> > > > > > > > > >> Is Cisco punishing us for
> > something?
> > > > Did we
> > > > > > > > offend
> > > > > > > > > >> them in some way? I
> > > > > > > > > >> am at a point where I am wondering
> > if I
> > > > > > should
> > > > > > > > tell
> > > > > > > > > >> customers about
> > > > > > > > > >> these performance issues prior to
> > > > picking a
> > > > > > > > version
> > > > > > > > > >> ('if you want to
> > > > > > > > > >> be able to administer this and not
> > > > slash
> > > > > > your
> > > > > > > > > >> wrists, waiting for the
> > > > > > > > > >> home page to load, you may want to
> > go
> > > > with
> > > > > > > > 4.3...')
> > > > > > > > > >> because I am tired
> > > > > > > > > >> of hearing about it afterwards...
> > > > > > > > > >>
> > > > > > > > > >> Seriously guys, I mean, this pretty
> > > > much
> > > > > > kills
> > > > > > > > > >> whatever argument you
> > > > > > > > > >> Linux freaks had that this was
> > better
> > > > than
> > > > > > > > > >> Windows... add the whole
> > > > > > > > > >> SFTP requirement, and you can see
> > why
> > > > the
> > > > > > > > Microsoft
> > > > > > > > > >> side won this
> > > > > > > > > >> argument...
> > > > > > > > >
> > > > > >
> > _______________________________________________
> > > > > > > > > cisco-voip mailing list
> > > > > > > > > cisco-voip at puck.nether.net
> > > > > > > > >
> > > > > > > >
> > > > > >
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > > > >
> > > > > > > >
> > > > _______________________________________________
> > > > > > > > cisco-voip mailing list
> > > > > > > > cisco-voip at puck.nether.net
> > > > > > > >
> > > > > >
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > > >
> > > > > > > >
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From jonvoip at gmail.com  Sat Oct  6 00:39:33 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 5 Oct 2007 23:39:33 -0500
Subject: [cisco-voip] CCME 4 + CUE 3 networking
In-Reply-To: <998653.76362.qm@web57201.mail.re3.yahoo.com>
References: <960111.19800.qm@web57204.mail.re3.yahoo.com>
	<998653.76362.qm@web57201.mail.re3.yahoo.com>
Message-ID: <5d093f9a0710052139l36e87189l42919aafefcca7cd@mail.gmail.com>

On your remote CME, create an H.323 Dial Peer for Vmail to go to the
remote CUE... then the SIP dial peer on that router will take care of
it.

For MWI, you may be screwed, cuz you can't have multiple MWI ons and offs...



Jonathan

On 10/5/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> Is it possible to have remote CCME 4.0 users use a CUE
> 3.0.1 module on a remote box across the WAN? It sounds
> slightly complex since MWI signals must be configured
> as well as DSPs for transcoding G.711 -> G.729. I
> can't find in the documents whether this is possible
> or not.
>
> Paul
>
>
>
> ____________________________________________________________________________________
> Yahoo! oneSearch: Finally, mobile search
> that gives answers, not web links.
> http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Sat Oct  6 00:56:27 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 5 Oct 2007 23:56:27 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
Message-ID: <5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com>

Here's a few....


What IP Protocol number is UDP?

What command turns RIP into a quasi-link state protocol?

What command turns a layer-2 switched interface into a layer-3
routable interface?

What is the advantage of a separate voice vlan?

What is the purpose of the command 'ebgp multihop'?

How do you prevent MGCP from using a loopback as its source interface?

Identify a version of IOS that is compatible with CME.

What is the minimum bandwidth for intracluster communication between
CallManagers?

The command 'passive interface fa0/0' does not prevent routing updates
from being received for which routing protocol(s)?

What could be the side effects of lowering the ARP timeout to 3
minutes? What is the default?

What are the potential problems while doing mutual redistribution of
two routing protocols? How do you mitigate this problem?

What configuration register setting on a Cisco router will bypass the
startup configuration file on bootup?

What is the recommended DSCP for VoIP traffic? [make sure they mention
CS3 or AF31 for control]

What tool can be used to correct CallManager 4.X database replication
problems? Do you have this file?

What version of CallManager supports IPCC 4.5?

What is the recommended Antivirus application for CallManager 5.X?

What keypad entry resets a Cisco 7940 IP Phone?

How would you find out the IP address assigned to an ATA 186? [looking
for the #80 thing]

What is MDIX?

What happens if you fail to enter the 'login' command under 'line vty
0 4' on a router?

What debug would you use to see inbound and outbound call info on a PRI?

Why would you choose CatIOS over IOS for a Cisco 6500 series switch?
[looking for auto qos and support for the 6608]

Which protocol, MGCP or H.323 would you choose for a gateway, and why?
[see if they know about limits of each, benefits of each]

You have an IP phone that says, 'configuring IP' but will not register, why?

You have an IP phone that says, 'error DBConfig' and will not register, why?

You have an IP phone that says, 'temp Fail', what does this mean?


Work on those...


Jonathan





On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> Thought you guys might want to take a break and help me out on a "related"
> issue...
>
> We are interviewing new Network Engineer Applicants next week.
>
> While I was writing up my list of questions (the boss needs 10) I thought I
> would throw this out to you guys.
> If you could write an interview question for your next Network Engineer
> (let's say Professional Level certification in Networking, Security or
> Voice)
> what would you ask them?  (try to keep it technical, nothing about Packer's
> tickets or Beer...lol)
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Sat Oct  6 00:58:10 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 5 Oct 2007 23:58:10 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <a6e20ec60710051508l3eb792adsb8d15b80b1af523c@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
	<a6e20ec60710051508l3eb792adsb8d15b80b1af523c@mail.gmail.com>
Message-ID: <5d093f9a0710052158x7dd1219aiea4f718c7dc7a583@mail.gmail.com>

I keep telling the newbies about 'reload in 15' command, but they
always forget to do it when they need to...



Jonathan

On 10/5/07, ryan speed <rspeed at gmail.com> wrote:
> never hurts to throw a scenario or two in.
>
> you're tasked with upgrading the ios (pre-definined) on a voice
> gateway for a remote customer, what steps would you take and what
> precautions would you deem necessary for a trouble free upgrade with a
> solid backout plan in the event something goes wrong.
>
> take that answer and move onto another scenario (which if they
> answered the first one correctly should result in a pass or fail
> situation)
>
> you've upgraded to the new ios and rebooted the router, 30 minutes
> have passed and you're unable to reach the router, how will you
> proceed to troubleshoot/rectify the situation so the customer isn't
> stuck with a dead router with 12 hours till business opens again.
>
> On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > Thought you guys might want to take a break and help me out on a "related"
> > issue...
> >
> > We are interviewing new Network Engineer Applicants next week.
> >
> > While I was writing up my list of questions (the boss needs 10) I thought I
> > would throw this out to you guys.
> > If you could write an interview question for your next Network Engineer
> > (let's say Professional Level certification in Networking, Security or
> > Voice)
> > what would you ask them?  (try to keep it technical, nothing about Packer's
> > tickets or Beer...lol)
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From ano-1 at worldonline.dk  Sat Oct  6 10:37:20 2007
From: ano-1 at worldonline.dk (Katrine og Michael)
Date: Sat, 6 Oct 2007 16:37:20 +0200
Subject: [cisco-voip] Script for AutoAttendant Unity Express
References: <942069.22286.qm@web57204.mail.re3.yahoo.com>
Message-ID: <001601c80826$67170340$0c000a0a@ano>

It?s 3.01
----- Original Message ----- 
From: "Paul Choi" <asobihoudai at yahoo.com>
To: "Katrine og Michael" <ano-1 at worldonline.dk>; 
<cisco-voip at puck.nether.net>
Sent: Friday, October 05, 2007 11:49 PM
Subject: Re: [cisco-voip] Script for AutoAttendant Unity Express


> What version of CUE is this?
>
> Paul
> --- Katrine og Michael <ano-1 at worldonline.dk> wrote:
>
>> Hello
>>
>> Can somebody help me?
>>
>> I need a script for unity express. The script will
>> be used when a phone i busy. When the phone is busy
>> the calling person will hear a voice saying: press 1
>> to callcontact, press 2 for try agin and press 3 for
>> end call. When 1 is pressed the calling person will
>> be transferred to callcontact. When 2 is pressed the
>> calling person will put in cue whith music and the
>> it will try to contact the called person again. When
>> 3 is pressed the script will terminate the call.
>>
>> Can sombody please help me? I?ve attatched my
>> script, but it is not working.
>>
>> your sincerely
>>
>> Michael from Denmark :-)
>>
>> --
>> Jeg beskyttes af den gratis SPAMfighter til
>> privatbrugere.
>> Den har indtil videre sparet mig for at f? 690
>> spam-mails.
>> Betalende brugere f?r ikke denne besked i deres
>> e-mails.
>> Hent gratis SPAMfighter her:
>> http://www.spamfighter.com/lda
>> > _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ____________________________________________________________________________________
> Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, 
> news, photos & more.
> http://mobile.yahoo.com/go?refer=1GNXIC
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.488 / Virus Database: 269.14.1/1051 - Release Date: 
> 05-10-2007 12:27
>
> 


-- 
Jeg beskyttes af den gratis SPAMfighter til privatbrugere.
Den har indtil videre sparet mig for at f? 690 spam-mails.
Betalende brugere f?r ikke denne besked i deres e-mails.
Hent gratis SPAMfighter her: http://www.spamfighter.com/lda



From jozjan at cnl.tuke.sk  Sun Oct  7 06:59:07 2007
From: jozjan at cnl.tuke.sk (Jozef Janitor)
Date: Sun, 7 Oct 2007 13:59:07 +0300
Subject: [cisco-voip] A little Fun
In-Reply-To: <5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
	<5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com>
Message-ID: <004f01c808d1$1608ed50$421ac7f0$@tuke.sk>

Nice list of questions :)
Could you please also provide answers to your questions?

Thank you.

All the best,
   Jozef Janitor


> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Jonathan Charles
> Sent: Saturday, October 06, 2007 7:56 AM
> To: Kevin Dunn
> Cc: Cisco Voice
> Subject: Re: [cisco-voip] A little Fun
> 
> Here's a few....
> 
> 
> What IP Protocol number is UDP?
> 
> What command turns RIP into a quasi-link state protocol?
> 
> What command turns a layer-2 switched interface into a layer-3
> routable interface?
> 
> What is the advantage of a separate voice vlan?
> 
> What is the purpose of the command 'ebgp multihop'?
> 
> How do you prevent MGCP from using a loopback as its source interface?
> 
> Identify a version of IOS that is compatible with CME.
> 
> What is the minimum bandwidth for intracluster communication between
> CallManagers?
> 
> The command 'passive interface fa0/0' does not prevent routing updates
> from being received for which routing protocol(s)?
> 
> What could be the side effects of lowering the ARP timeout to 3
> minutes? What is the default?
> 
> What are the potential problems while doing mutual redistribution of
> two routing protocols? How do you mitigate this problem?
> 
> What configuration register setting on a Cisco router will bypass the
> startup configuration file on bootup?
> 
> What is the recommended DSCP for VoIP traffic? [make sure they mention
> CS3 or AF31 for control]
> 
> What tool can be used to correct CallManager 4.X database replication
> problems? Do you have this file?
> 
> What version of CallManager supports IPCC 4.5?
> 
> What is the recommended Antivirus application for CallManager 5.X?
> 
> What keypad entry resets a Cisco 7940 IP Phone?
> 
> How would you find out the IP address assigned to an ATA 186? [looking
> for the #80 thing]
> 
> What is MDIX?
> 
> What happens if you fail to enter the 'login' command under 'line vty
> 0 4' on a router?
> 
> What debug would you use to see inbound and outbound call info on a
> PRI?
> 
> Why would you choose CatIOS over IOS for a Cisco 6500 series switch?
> [looking for auto qos and support for the 6608]
> 
> Which protocol, MGCP or H.323 would you choose for a gateway, and why?
> [see if they know about limits of each, benefits of each]
> 
> You have an IP phone that says, 'configuring IP' but will not register,
> why?
> 
> You have an IP phone that says, 'error DBConfig' and will not register,
> why?
> 
> You have an IP phone that says, 'temp Fail', what does this mean?
> 
> 
> Work on those...
> 
> 
> Jonathan
> 
> 
> 
> 
> 
> On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > Thought you guys might want to take a break and help me out on a
> "related"
> > issue...
> >
> > We are interviewing new Network Engineer Applicants next week.
> >
> > While I was writing up my list of questions (the boss needs 10) I
> thought I
> > would throw this out to you guys.
> > If you could write an interview question for your next Network
> Engineer
> > (let's say Professional Level certification in Networking, Security
> or
> > Voice)
> > what would you ask them?  (try to keep it technical, nothing about
> Packer's
> > tickets or Beer...lol)
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jonvoip at gmail.com  Sun Oct  7 11:28:05 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 7 Oct 2007 10:28:05 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
	<5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com>
Message-ID: <5d093f9a0710070828w3af81036qb2fee53165b5fdec@mail.gmail.com>

On 10/5/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> Here's a few....
>
>
> What IP Protocol number is UDP?
>

17

> What command turns RIP into a quasi-link state protocol?
>

On the interface:

ip rip triggered

> What command turns a layer-2 switched interface into a layer-3
> routable interface?
>

no switchport

> What is the advantage of a separate voice vlan?
>

To establish separate QoS for voice traffic and prevent data traffic
from using those queues, to segment traffic

> What is the purpose of the command 'ebgp multihop'?
>

If the IP on a BGP neighbor is more than one hop away, you need this command.

> How do you prevent MGCP from using a loopback as its source interface?
>

no mgcp bind loopback 0

> Identify a version of IOS that is compatible with CME.
>

12.4.4XC, 12.4.11XJ, 12.4.15T

> What is the minimum bandwidth for intracluster communication between
> CallManagers?
>

900k

> The command 'passive interface fa0/0' does not prevent routing updates
> from being received for which routing protocol(s)?
>

RIP

> What could be the side effects of lowering the ARP timeout to 3
> minutes? What is the default?
>

The default is 4 hours. Lowering this causes intermittant network
failures as devices re-ARP for destinations and the first packets
fail.

> What are the potential problems while doing mutual redistribution of
> two routing protocols? How do you mitigate this problem?
>

Routing loops from hell, use distribute lists to prevent your
advertised routes from being sent back at you.

> What configuration register setting on a Cisco router will bypass the
> startup configuration file on bootup?
>

0x2142

> What is the recommended DSCP for VoIP traffic? [make sure they mention
> CS3 or AF31 for control]
>

EF for payload, CS and AF31 for control

> What tool can be used to correct CallManager 4.X database replication
> problems? Do you have this file?
>

DBLHelper

> What version of CallManager supports IPCC 4.5?
>

5.x and 6.x

> What is the recommended Antivirus application for CallManager 5.X?
>

Macafee v8... etc.

> What keypad entry resets a Cisco 7940 IP Phone?
>

**#**

> How would you find out the IP address assigned to an ATA 186? [looking
> for the #80 thing]
>

hook up an analog phone, press button on ATA, when the British chick
says, "Configuration Menu" type # 8 0, the system will tell you the IP
Address.

> What is MDIX?
>

It is a switch feature that autosenses the cable (straight thru or
crossover) and changes the pinouts of the port to match.

> What happens if you fail to enter the 'login' command under 'line vty
> 0 4' on a router?
>

You do not get prompted for a login when you telnet to the router/switch.

> What debug would you use to see inbound and outbound call info on a PRI?
>

debug isdn q931

> Why would you choose CatIOS over IOS for a Cisco 6500 series switch?
> [looking for auto qos and support for the 6608]
>

Auto QoS and the 6608 blade are not supported in IOS

> Which protocol, MGCP or H.323 would you choose for a gateway, and why?
> [see if they know about limits of each, benefits of each]
>

MGCP provides for advanced PRI features, H.323 is more reliable for
low bandwidth links and provides for call survivability.

> You have an IP phone that says, 'configuring IP' but will not register, why?
>

Can't get an IP address... check DHCP server, vlan, etc.

> You have an IP phone that says, 'error DBConfig' and will not register, why?
>

Phone not configured on CCM and auto-registration is disabled.

> You have an IP phone that says, 'temp Fail', what does this mean?
>

Wrong Codec.

>
> Work on those...
>
>
> Jonathan
>
>
>
>
>
> On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > Thought you guys might want to take a break and help me out on a "related"
> > issue...
> >
> > We are interviewing new Network Engineer Applicants next week.
> >
> > While I was writing up my list of questions (the boss needs 10) I thought I
> > would throw this out to you guys.
> > If you could write an interview question for your next Network Engineer
> > (let's say Professional Level certification in Networking, Security or
> > Voice)
> > what would you ask them?  (try to keep it technical, nothing about Packer's
> > tickets or Beer...lol)
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From jozjan at cnl.tuke.sk  Sun Oct  7 11:41:36 2007
From: jozjan at cnl.tuke.sk (Jozef Janitor)
Date: Sun, 7 Oct 2007 18:41:36 +0300
Subject: [cisco-voip] A little Fun
In-Reply-To: <5d093f9a0710070828w3af81036qb2fee53165b5fdec@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>	
	<5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com>
	<5d093f9a0710070828w3af81036qb2fee53165b5fdec@mail.gmail.com>
Message-ID: <00b701c808f8$8d1b3020$a7519060$@tuke.sk>

Thank you.

As far as I know the **#** combination doesn't restart the 7940 with SIP
firmware.
You have to press the " * + 6 + Settings " combination at the same time to
restart it.

Anyway, thanks :)

All the best,
   Jozef Janitor


> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Sunday, October 07, 2007 6:28 PM
> To: Jozef Janitor
> Cc: Cisco Voice
> Subject: Re: [cisco-voip] A little Fun
> 
> On 10/5/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > Here's a few....
> >
> >
> > What IP Protocol number is UDP?
> >
> 
> 17
> 
> > What command turns RIP into a quasi-link state protocol?
> >
> 
> On the interface:
> 
> ip rip triggered
> 
> > What command turns a layer-2 switched interface into a layer-3
> > routable interface?
> >
> 
> no switchport
> 
> > What is the advantage of a separate voice vlan?
> >
> 
> To establish separate QoS for voice traffic and prevent data traffic
> from using those queues, to segment traffic
> 
> > What is the purpose of the command 'ebgp multihop'?
> >
> 
> If the IP on a BGP neighbor is more than one hop away, you need this
> command.
> 
> > How do you prevent MGCP from using a loopback as its source
> interface?
> >
> 
> no mgcp bind loopback 0
> 
> > Identify a version of IOS that is compatible with CME.
> >
> 
> 12.4.4XC, 12.4.11XJ, 12.4.15T
> 
> > What is the minimum bandwidth for intracluster communication between
> > CallManagers?
> >
> 
> 900k
> 
> > The command 'passive interface fa0/0' does not prevent routing
> updates
> > from being received for which routing protocol(s)?
> >
> 
> RIP
> 
> > What could be the side effects of lowering the ARP timeout to 3
> > minutes? What is the default?
> >
> 
> The default is 4 hours. Lowering this causes intermittant network
> failures as devices re-ARP for destinations and the first packets
> fail.
> 
> > What are the potential problems while doing mutual redistribution of
> > two routing protocols? How do you mitigate this problem?
> >
> 
> Routing loops from hell, use distribute lists to prevent your
> advertised routes from being sent back at you.
> 
> > What configuration register setting on a Cisco router will bypass the
> > startup configuration file on bootup?
> >
> 
> 0x2142
> 
> > What is the recommended DSCP for VoIP traffic? [make sure they
> mention
> > CS3 or AF31 for control]
> >
> 
> EF for payload, CS and AF31 for control
> 
> > What tool can be used to correct CallManager 4.X database replication
> > problems? Do you have this file?
> >
> 
> DBLHelper
> 
> > What version of CallManager supports IPCC 4.5?
> >
> 
> 5.x and 6.x
> 
> > What is the recommended Antivirus application for CallManager 5.X?
> >
> 
> Macafee v8... etc.
> 
> > What keypad entry resets a Cisco 7940 IP Phone?
> >
> 
> **#**
> 
> > How would you find out the IP address assigned to an ATA 186?
> [looking
> > for the #80 thing]
> >
> 
> hook up an analog phone, press button on ATA, when the British chick
> says, "Configuration Menu" type # 8 0, the system will tell you the IP
> Address.
> 
> > What is MDIX?
> >
> 
> It is a switch feature that autosenses the cable (straight thru or
> crossover) and changes the pinouts of the port to match.
> 
> > What happens if you fail to enter the 'login' command under 'line vty
> > 0 4' on a router?
> >
> 
> You do not get prompted for a login when you telnet to the
> router/switch.
> 
> > What debug would you use to see inbound and outbound call info on a
> PRI?
> >
> 
> debug isdn q931
> 
> > Why would you choose CatIOS over IOS for a Cisco 6500 series switch?
> > [looking for auto qos and support for the 6608]
> >
> 
> Auto QoS and the 6608 blade are not supported in IOS
> 
> > Which protocol, MGCP or H.323 would you choose for a gateway, and
> why?
> > [see if they know about limits of each, benefits of each]
> >
> 
> MGCP provides for advanced PRI features, H.323 is more reliable for
> low bandwidth links and provides for call survivability.
> 
> > You have an IP phone that says, 'configuring IP' but will not
> register, why?
> >
> 
> Can't get an IP address... check DHCP server, vlan, etc.
> 
> > You have an IP phone that says, 'error DBConfig' and will not
> register, why?
> >
> 
> Phone not configured on CCM and auto-registration is disabled.
> 
> > You have an IP phone that says, 'temp Fail', what does this mean?
> >
> 
> Wrong Codec.
> 
> >
> > Work on those...
> >
> >
> > Jonathan
> >
> >
> >
> >
> >
> > On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > > Thought you guys might want to take a break and help me out on a
> "related"
> > > issue...
> > >
> > > We are interviewing new Network Engineer Applicants next week.
> > >
> > > While I was writing up my list of questions (the boss needs 10) I
> thought I
> > > would throw this out to you guys.
> > > If you could write an interview question for your next Network
> Engineer
> > > (let's say Professional Level certification in Networking, Security
> or
> > > Voice)
> > > what would you ask them?  (try to keep it technical, nothing about
> Packer's
> > > tickets or Beer...lol)
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >


From jonvoip at gmail.com  Sun Oct  7 19:41:26 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 7 Oct 2007 18:41:26 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <00b701c808f8$8d1b3020$a7519060$@tuke.sk>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
	<5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com>
	<5d093f9a0710070828w3af81036qb2fee53165b5fdec@mail.gmail.com>
	<00b701c808f8$8d1b3020$a7519060$@tuke.sk>
Message-ID: <5d093f9a0710071641s6c292867jec5d0140726a845b@mail.gmail.com>

SIP is dumb.

But I will concede the correction, you do need to be in Settings mode
for the **#** to work...



Jonathan

On 10/7/07, Jozef Janitor <jozjan at cnl.tuke.sk> wrote:
> Thank you.
>
> As far as I know the **#** combination doesn't restart the 7940 with SIP
> firmware.
> You have to press the " * + 6 + Settings " combination at the same time to
> restart it.
>
> Anyway, thanks :)
>
> All the best,
>    Jozef Janitor
>
>
> > -----Original Message-----
> > From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > Sent: Sunday, October 07, 2007 6:28 PM
> > To: Jozef Janitor
> > Cc: Cisco Voice
> > Subject: Re: [cisco-voip] A little Fun
> >
> > On 10/5/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > Here's a few....
> > >
> > >
> > > What IP Protocol number is UDP?
> > >
> >
> > 17
> >
> > > What command turns RIP into a quasi-link state protocol?
> > >
> >
> > On the interface:
> >
> > ip rip triggered
> >
> > > What command turns a layer-2 switched interface into a layer-3
> > > routable interface?
> > >
> >
> > no switchport
> >
> > > What is the advantage of a separate voice vlan?
> > >
> >
> > To establish separate QoS for voice traffic and prevent data traffic
> > from using those queues, to segment traffic
> >
> > > What is the purpose of the command 'ebgp multihop'?
> > >
> >
> > If the IP on a BGP neighbor is more than one hop away, you need this
> > command.
> >
> > > How do you prevent MGCP from using a loopback as its source
> > interface?
> > >
> >
> > no mgcp bind loopback 0
> >
> > > Identify a version of IOS that is compatible with CME.
> > >
> >
> > 12.4.4XC, 12.4.11XJ, 12.4.15T
> >
> > > What is the minimum bandwidth for intracluster communication between
> > > CallManagers?
> > >
> >
> > 900k
> >
> > > The command 'passive interface fa0/0' does not prevent routing
> > updates
> > > from being received for which routing protocol(s)?
> > >
> >
> > RIP
> >
> > > What could be the side effects of lowering the ARP timeout to 3
> > > minutes? What is the default?
> > >
> >
> > The default is 4 hours. Lowering this causes intermittant network
> > failures as devices re-ARP for destinations and the first packets
> > fail.
> >
> > > What are the potential problems while doing mutual redistribution of
> > > two routing protocols? How do you mitigate this problem?
> > >
> >
> > Routing loops from hell, use distribute lists to prevent your
> > advertised routes from being sent back at you.
> >
> > > What configuration register setting on a Cisco router will bypass the
> > > startup configuration file on bootup?
> > >
> >
> > 0x2142
> >
> > > What is the recommended DSCP for VoIP traffic? [make sure they
> > mention
> > > CS3 or AF31 for control]
> > >
> >
> > EF for payload, CS and AF31 for control
> >
> > > What tool can be used to correct CallManager 4.X database replication
> > > problems? Do you have this file?
> > >
> >
> > DBLHelper
> >
> > > What version of CallManager supports IPCC 4.5?
> > >
> >
> > 5.x and 6.x
> >
> > > What is the recommended Antivirus application for CallManager 5.X?
> > >
> >
> > Macafee v8... etc.
> >
> > > What keypad entry resets a Cisco 7940 IP Phone?
> > >
> >
> > **#**
> >
> > > How would you find out the IP address assigned to an ATA 186?
> > [looking
> > > for the #80 thing]
> > >
> >
> > hook up an analog phone, press button on ATA, when the British chick
> > says, "Configuration Menu" type # 8 0, the system will tell you the IP
> > Address.
> >
> > > What is MDIX?
> > >
> >
> > It is a switch feature that autosenses the cable (straight thru or
> > crossover) and changes the pinouts of the port to match.
> >
> > > What happens if you fail to enter the 'login' command under 'line vty
> > > 0 4' on a router?
> > >
> >
> > You do not get prompted for a login when you telnet to the
> > router/switch.
> >
> > > What debug would you use to see inbound and outbound call info on a
> > PRI?
> > >
> >
> > debug isdn q931
> >
> > > Why would you choose CatIOS over IOS for a Cisco 6500 series switch?
> > > [looking for auto qos and support for the 6608]
> > >
> >
> > Auto QoS and the 6608 blade are not supported in IOS
> >
> > > Which protocol, MGCP or H.323 would you choose for a gateway, and
> > why?
> > > [see if they know about limits of each, benefits of each]
> > >
> >
> > MGCP provides for advanced PRI features, H.323 is more reliable for
> > low bandwidth links and provides for call survivability.
> >
> > > You have an IP phone that says, 'configuring IP' but will not
> > register, why?
> > >
> >
> > Can't get an IP address... check DHCP server, vlan, etc.
> >
> > > You have an IP phone that says, 'error DBConfig' and will not
> > register, why?
> > >
> >
> > Phone not configured on CCM and auto-registration is disabled.
> >
> > > You have an IP phone that says, 'temp Fail', what does this mean?
> > >
> >
> > Wrong Codec.
> >
> > >
> > > Work on those...
> > >
> > >
> > > Jonathan
> > >
> > >
> > >
> > >
> > >
> > > On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > > > Thought you guys might want to take a break and help me out on a
> > "related"
> > > > issue...
> > > >
> > > > We are interviewing new Network Engineer Applicants next week.
> > > >
> > > > While I was writing up my list of questions (the boss needs 10) I
> > thought I
> > > > would throw this out to you guys.
> > > > If you could write an interview question for your next Network
> > Engineer
> > > > (let's say Professional Level certification in Networking, Security
> > or
> > > > Voice)
> > > > what would you ask them?  (try to keep it technical, nothing about
> > Packer's
> > > > tickets or Beer...lol)
> > > >
> > > >
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
>
>

From Khalid_Khursheed at hotmail.com  Sun Oct  7 22:05:01 2007
From: Khalid_Khursheed at hotmail.com (Syed Khalid Ali)
Date: Mon, 8 Oct 2007 07:05:01 +0500
Subject: [cisco-voip] A little Fun
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
	<a6e20ec60710051508l3eb792adsb8d15b80b1af523c@mail.gmail.com>
Message-ID: <BAY113-DAV639A17BCDF9766ECEE3F8E8A40@phx.gbl>

hi,
 here r a few:

what is unicast, broadcast and multicast
how a switch learns a mac address
what does snap do at osi layer 2
what is mdix
difference between routing and switching
how to bridging avoid loop in switched network
how to avoid loop in distance vector protocols
types of routing protocols
what protocol and port number does bgp uses.
what types of area does ospf support
different between types of lsa in ospf
why need redistribution
why need route summarization
what is cidr
why do we need hierachy in ospf or is-is. can we do so in eigrp as well.
is is-is independent of ip protocol. if yes what is the advantage
what does a pasive interface do
difference between ccm, ccm business edition and ccme
what voice signaling protocol does ccm use
in how many ways we can implement call control (centralized or distributed) 
in a  voice enabled network
when to use mgcp/h323 and sip
what voip signaling protocol does ccm use
how to calculate bandwidth for viop enabled network
difference between the call control methods.
what is the recomended anti virus for ccm
why need a sperate voice vlan
what is the qos value for voice and voice control traffic
what is difference between congestion management and avoidance
techniques available in cisco ios to do congestion management and avoidance 
(eg: fair queue,cwfq, llq, red and wred)
what is 802.1p
phone get the ip and vlan but unable to register with ccme/ccm  why?
what is v3 vpn
what are the phases in establishing a vpn.
why is the difference between site-to-site and access vpn.
what is the command to see vpn peers
why use gre tunnels with ipsec vpns?





----- Original Message ----- 
From: "ryan speed" <rspeed at gmail.com>
To: <cisco-voip at puck.nether.net>
Sent: Saturday, October 06, 2007 3:08 AM
Subject: Re: [cisco-voip] A little Fun


> never hurts to throw a scenario or two in.
>
> you're tasked with upgrading the ios (pre-definined) on a voice
> gateway for a remote customer, what steps would you take and what
> precautions would you deem necessary for a trouble free upgrade with a
> solid backout plan in the event something goes wrong.
>
> take that answer and move onto another scenario (which if they
> answered the first one correctly should result in a pass or fail
> situation)
>
> you've upgraded to the new ios and rebooted the router, 30 minutes
> have passed and you're unable to reach the router, how will you
> proceed to troubleshoot/rectify the situation so the customer isn't
> stuck with a dead router with 12 hours till business opens again.
>
> On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
>> Thought you guys might want to take a break and help me out on a 
>> "related"
>> issue...
>>
>> We are interviewing new Network Engineer Applicants next week.
>>
>> While I was writing up my list of questions (the boss needs 10) I thought 
>> I
>> would throw this out to you guys.
>> If you could write an interview question for your next Network Engineer
>> (let's say Professional Level certification in Networking, Security or
>> Voice)
>> what would you ask them?  (try to keep it technical, nothing about 
>> Packer's
>> tickets or Beer...lol)
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From Timothy_Frazee at adp.com  Sun Oct  7 23:01:11 2007
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Sun, 7 Oct 2007 22:01:11 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <5d093f9a0710071641s6c292867jec5d0140726a845b@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com><5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com><5d093f9a0710070828w3af81036qb2fee53165b5fdec@mail.gmail.com><00b701c808f8$8d1b3020$a7519060$@tuke.sk>
	<5d093f9a0710071641s6c292867jec5d0140726a845b@mail.gmail.com>
Message-ID: <723FA17C6AA4C4428CDF572DCAEE08D4052ED61C@EXCHANGEDS01.ds.ad.adp.com>

That depends on the firmware load.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Sunday, October 07, 2007 6:41 PM
To: Jozef Janitor
Cc: Cisco Voice
Subject: Re: [cisco-voip] A little Fun

SIP is dumb.

But I will concede the correction, you do need to be in Settings mode
for the **#** to work...



Jonathan

On 10/7/07, Jozef Janitor <jozjan at cnl.tuke.sk> wrote:
> Thank you.
>
> As far as I know the **#** combination doesn't restart the 7940 with
SIP
> firmware.
> You have to press the " * + 6 + Settings " combination at the same
time to
> restart it.
>
> Anyway, thanks :)
>
> All the best,
>    Jozef Janitor
>
>
> > -----Original Message-----
> > From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > Sent: Sunday, October 07, 2007 6:28 PM
> > To: Jozef Janitor
> > Cc: Cisco Voice
> > Subject: Re: [cisco-voip] A little Fun
> >
> > On 10/5/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > Here's a few....
> > >
> > >
> > > What IP Protocol number is UDP?
> > >
> >
> > 17
> >
> > > What command turns RIP into a quasi-link state protocol?
> > >
> >
> > On the interface:
> >
> > ip rip triggered
> >
> > > What command turns a layer-2 switched interface into a layer-3
> > > routable interface?
> > >
> >
> > no switchport
> >
> > > What is the advantage of a separate voice vlan?
> > >
> >
> > To establish separate QoS for voice traffic and prevent data traffic
> > from using those queues, to segment traffic
> >
> > > What is the purpose of the command 'ebgp multihop'?
> > >
> >
> > If the IP on a BGP neighbor is more than one hop away, you need this
> > command.
> >
> > > How do you prevent MGCP from using a loopback as its source
> > interface?
> > >
> >
> > no mgcp bind loopback 0
> >
> > > Identify a version of IOS that is compatible with CME.
> > >
> >
> > 12.4.4XC, 12.4.11XJ, 12.4.15T
> >
> > > What is the minimum bandwidth for intracluster communication
between
> > > CallManagers?
> > >
> >
> > 900k
> >
> > > The command 'passive interface fa0/0' does not prevent routing
> > updates
> > > from being received for which routing protocol(s)?
> > >
> >
> > RIP
> >
> > > What could be the side effects of lowering the ARP timeout to 3
> > > minutes? What is the default?
> > >
> >
> > The default is 4 hours. Lowering this causes intermittant network
> > failures as devices re-ARP for destinations and the first packets
> > fail.
> >
> > > What are the potential problems while doing mutual redistribution
of
> > > two routing protocols? How do you mitigate this problem?
> > >
> >
> > Routing loops from hell, use distribute lists to prevent your
> > advertised routes from being sent back at you.
> >
> > > What configuration register setting on a Cisco router will bypass
the
> > > startup configuration file on bootup?
> > >
> >
> > 0x2142
> >
> > > What is the recommended DSCP for VoIP traffic? [make sure they
> > mention
> > > CS3 or AF31 for control]
> > >
> >
> > EF for payload, CS and AF31 for control
> >
> > > What tool can be used to correct CallManager 4.X database
replication
> > > problems? Do you have this file?
> > >
> >
> > DBLHelper
> >
> > > What version of CallManager supports IPCC 4.5?
> > >
> >
> > 5.x and 6.x
> >
> > > What is the recommended Antivirus application for CallManager 5.X?
> > >
> >
> > Macafee v8... etc.
> >
> > > What keypad entry resets a Cisco 7940 IP Phone?
> > >
> >
> > **#**
> >
> > > How would you find out the IP address assigned to an ATA 186?
> > [looking
> > > for the #80 thing]
> > >
> >
> > hook up an analog phone, press button on ATA, when the British chick
> > says, "Configuration Menu" type # 8 0, the system will tell you the
IP
> > Address.
> >
> > > What is MDIX?
> > >
> >
> > It is a switch feature that autosenses the cable (straight thru or
> > crossover) and changes the pinouts of the port to match.
> >
> > > What happens if you fail to enter the 'login' command under 'line
vty
> > > 0 4' on a router?
> > >
> >
> > You do not get prompted for a login when you telnet to the
> > router/switch.
> >
> > > What debug would you use to see inbound and outbound call info on
a
> > PRI?
> > >
> >
> > debug isdn q931
> >
> > > Why would you choose CatIOS over IOS for a Cisco 6500 series
switch?
> > > [looking for auto qos and support for the 6608]
> > >
> >
> > Auto QoS and the 6608 blade are not supported in IOS
> >
> > > Which protocol, MGCP or H.323 would you choose for a gateway, and
> > why?
> > > [see if they know about limits of each, benefits of each]
> > >
> >
> > MGCP provides for advanced PRI features, H.323 is more reliable for
> > low bandwidth links and provides for call survivability.
> >
> > > You have an IP phone that says, 'configuring IP' but will not
> > register, why?
> > >
> >
> > Can't get an IP address... check DHCP server, vlan, etc.
> >
> > > You have an IP phone that says, 'error DBConfig' and will not
> > register, why?
> > >
> >
> > Phone not configured on CCM and auto-registration is disabled.
> >
> > > You have an IP phone that says, 'temp Fail', what does this mean?
> > >
> >
> > Wrong Codec.
> >
> > >
> > > Work on those...
> > >
> > >
> > > Jonathan
> > >
> > >
> > >
> > >
> > >
> > > On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > > > Thought you guys might want to take a break and help me out on a
> > "related"
> > > > issue...
> > > >
> > > > We are interviewing new Network Engineer Applicants next week.
> > > >
> > > > While I was writing up my list of questions (the boss needs 10)
I
> > thought I
> > > > would throw this out to you guys.
> > > > If you could write an interview question for your next Network
> > Engineer
> > > > (let's say Professional Level certification in Networking,
Security
> > or
> > > > Voice)
> > > > what would you ask them?  (try to keep it technical, nothing
about
> > Packer's
> > > > tickets or Beer...lol)
> > > >
> > > >
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.

From jonvoip at gmail.com  Mon Oct  8 00:36:28 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 7 Oct 2007 23:36:28 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <723FA17C6AA4C4428CDF572DCAEE08D4052ED61C@EXCHANGEDS01.ds.ad.adp.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
	<5d093f9a0710052156x23d624e8kc1e928a3f32ab780@mail.gmail.com>
	<5d093f9a0710070828w3af81036qb2fee53165b5fdec@mail.gmail.com>
	<00b701c808f8$8d1b3020$a7519060$@tuke.sk>
	<5d093f9a0710071641s6c292867jec5d0140726a845b@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D4052ED61C@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <5d093f9a0710072136i77256895w4af0addf866f9ed3@mail.gmail.com>

Yes, but SIP for Cisco IP Phones is really dumb.

I heard it straight from the Cisco developers themselves... they said
they will never be able to achieve feature parity between SIP and
SCCP.

So, why on earth would anyone run a Cisco 7971 on CallManager with
SIP? I can only presume because they hate the touch screen on their
phone...



Jonathan

On 10/7/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote:
> That depends on the firmware load.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Sunday, October 07, 2007 6:41 PM
> To: Jozef Janitor
> Cc: Cisco Voice
> Subject: Re: [cisco-voip] A little Fun
>
> SIP is dumb.
>
> But I will concede the correction, you do need to be in Settings mode
> for the **#** to work...
>
>
>
> Jonathan
>
> On 10/7/07, Jozef Janitor <jozjan at cnl.tuke.sk> wrote:
> > Thank you.
> >
> > As far as I know the **#** combination doesn't restart the 7940 with
> SIP
> > firmware.
> > You have to press the " * + 6 + Settings " combination at the same
> time to
> > restart it.
> >
> > Anyway, thanks :)
> >
> > All the best,
> >    Jozef Janitor
> >
> >
> > > -----Original Message-----
> > > From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > > Sent: Sunday, October 07, 2007 6:28 PM
> > > To: Jozef Janitor
> > > Cc: Cisco Voice
> > > Subject: Re: [cisco-voip] A little Fun
> > >
> > > On 10/5/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > > Here's a few....
> > > >
> > > >
> > > > What IP Protocol number is UDP?
> > > >
> > >
> > > 17
> > >
> > > > What command turns RIP into a quasi-link state protocol?
> > > >
> > >
> > > On the interface:
> > >
> > > ip rip triggered
> > >
> > > > What command turns a layer-2 switched interface into a layer-3
> > > > routable interface?
> > > >
> > >
> > > no switchport
> > >
> > > > What is the advantage of a separate voice vlan?
> > > >
> > >
> > > To establish separate QoS for voice traffic and prevent data traffic
> > > from using those queues, to segment traffic
> > >
> > > > What is the purpose of the command 'ebgp multihop'?
> > > >
> > >
> > > If the IP on a BGP neighbor is more than one hop away, you need this
> > > command.
> > >
> > > > How do you prevent MGCP from using a loopback as its source
> > > interface?
> > > >
> > >
> > > no mgcp bind loopback 0
> > >
> > > > Identify a version of IOS that is compatible with CME.
> > > >
> > >
> > > 12.4.4XC, 12.4.11XJ, 12.4.15T
> > >
> > > > What is the minimum bandwidth for intracluster communication
> between
> > > > CallManagers?
> > > >
> > >
> > > 900k
> > >
> > > > The command 'passive interface fa0/0' does not prevent routing
> > > updates
> > > > from being received for which routing protocol(s)?
> > > >
> > >
> > > RIP
> > >
> > > > What could be the side effects of lowering the ARP timeout to 3
> > > > minutes? What is the default?
> > > >
> > >
> > > The default is 4 hours. Lowering this causes intermittant network
> > > failures as devices re-ARP for destinations and the first packets
> > > fail.
> > >
> > > > What are the potential problems while doing mutual redistribution
> of
> > > > two routing protocols? How do you mitigate this problem?
> > > >
> > >
> > > Routing loops from hell, use distribute lists to prevent your
> > > advertised routes from being sent back at you.
> > >
> > > > What configuration register setting on a Cisco router will bypass
> the
> > > > startup configuration file on bootup?
> > > >
> > >
> > > 0x2142
> > >
> > > > What is the recommended DSCP for VoIP traffic? [make sure they
> > > mention
> > > > CS3 or AF31 for control]
> > > >
> > >
> > > EF for payload, CS and AF31 for control
> > >
> > > > What tool can be used to correct CallManager 4.X database
> replication
> > > > problems? Do you have this file?
> > > >
> > >
> > > DBLHelper
> > >
> > > > What version of CallManager supports IPCC 4.5?
> > > >
> > >
> > > 5.x and 6.x
> > >
> > > > What is the recommended Antivirus application for CallManager 5.X?
> > > >
> > >
> > > Macafee v8... etc.
> > >
> > > > What keypad entry resets a Cisco 7940 IP Phone?
> > > >
> > >
> > > **#**
> > >
> > > > How would you find out the IP address assigned to an ATA 186?
> > > [looking
> > > > for the #80 thing]
> > > >
> > >
> > > hook up an analog phone, press button on ATA, when the British chick
> > > says, "Configuration Menu" type # 8 0, the system will tell you the
> IP
> > > Address.
> > >
> > > > What is MDIX?
> > > >
> > >
> > > It is a switch feature that autosenses the cable (straight thru or
> > > crossover) and changes the pinouts of the port to match.
> > >
> > > > What happens if you fail to enter the 'login' command under 'line
> vty
> > > > 0 4' on a router?
> > > >
> > >
> > > You do not get prompted for a login when you telnet to the
> > > router/switch.
> > >
> > > > What debug would you use to see inbound and outbound call info on
> a
> > > PRI?
> > > >
> > >
> > > debug isdn q931
> > >
> > > > Why would you choose CatIOS over IOS for a Cisco 6500 series
> switch?
> > > > [looking for auto qos and support for the 6608]
> > > >
> > >
> > > Auto QoS and the 6608 blade are not supported in IOS
> > >
> > > > Which protocol, MGCP or H.323 would you choose for a gateway, and
> > > why?
> > > > [see if they know about limits of each, benefits of each]
> > > >
> > >
> > > MGCP provides for advanced PRI features, H.323 is more reliable for
> > > low bandwidth links and provides for call survivability.
> > >
> > > > You have an IP phone that says, 'configuring IP' but will not
> > > register, why?
> > > >
> > >
> > > Can't get an IP address... check DHCP server, vlan, etc.
> > >
> > > > You have an IP phone that says, 'error DBConfig' and will not
> > > register, why?
> > > >
> > >
> > > Phone not configured on CCM and auto-registration is disabled.
> > >
> > > > You have an IP phone that says, 'temp Fail', what does this mean?
> > > >
> > >
> > > Wrong Codec.
> > >
> > > >
> > > > Work on those...
> > > >
> > > >
> > > > Jonathan
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > > > > Thought you guys might want to take a break and help me out on a
> > > "related"
> > > > > issue...
> > > > >
> > > > > We are interviewing new Network Engineer Applicants next week.
> > > > >
> > > > > While I was writing up my list of questions (the boss needs 10)
> I
> > > thought I
> > > > > would throw this out to you guys.
> > > > > If you could write an interview question for your next Network
> > > Engineer
> > > > > (let's say Professional Level certification in Networking,
> Security
> > > or
> > > > > Voice)
> > > > > what would you ask them?  (try to keep it technical, nothing
> about
> > > Packer's
> > > > > tickets or Beer...lol)
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > >
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
>

From cisco-voip at ibh.net  Mon Oct  8 04:47:14 2007
From: cisco-voip at ibh.net (Andre Beck)
Date: Mon, 8 Oct 2007 10:47:14 +0200
Subject: [cisco-voip] 5.1 active disk filling?
Message-ID: <20071008084714.GA26790@ibh.de>

Hi,

I didn't have significant problems with 5.1 so far (other than those with
localization that got fixed) and was running 5.1.1.3000-5 until some
weeks ago. Then I upgraded to 5.1.2.3000-2 without problems and the box
ran flawless until yesterday, when CCM suddenly dropped all H.323 gateway
connections. Restarting the CCM service helped.

This morning, the box stopped answering HTTP connections (as tracked
by our Nagios monitoring) and had to be rebooted, after this it again
resumed normal operation.

What I noticed in the OS Admin Web interface is that the "Disk/active"
filesystems are rather full (my 5.1.1 is at 93%, the 5.1.2 was at 95%)
and they are *constantly* *growing*.

I switched versions back to 5.1.1 to verify this, it's the same there:

                        Total            Free            Used

Disk/active         12317944K         928084K       10764132K (93%)
[...]
Disk/active         12317944K         928068K       10764148K (93%)
[...]
Disk/active         12317944K         928064K       10764152K (93%)

without the number ever getting smaller again.

What would write to the software partions instead of to the common or
log partitions? Is this normal and maybe garbage collected every some
hours/days or might this be a reason for an installation suddenly failing
after weeks of runtime?

I'm probably going to upgrade to 5.1.3, which will let me lose my 5.1.1
fallback installation. Hopefully this will be stable again.

BTW, the absence of an "expert" login to actually search for the reason
of the growing diskspace is an ENTIRELY SILLY IDEA. I'm feeling like a
lobotomized ex-hacker at this "admin" shell...

Thanks,
Andre.
-- 
    .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From rratliff at cisco.com  Mon Oct  8 09:57:33 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 8 Oct 2007 09:57:33 -0400
Subject: [cisco-voip] 5.1 active disk filling?
In-Reply-To: <20071008084714.GA26790@ibh.de>
References: <20071008084714.GA26790@ibh.de>
Message-ID: <4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com>

Until you start getting RTMT alerts about the active partition being  
full don't worry about it.  93% utilization is pretty normal for a CM  
5 box.

There were however some bugs that caused the active partition to fill  
up.  CSCsk18010 is one of them, but only affects IBM servers and is  
fixed in 5.1(3).

I'd encourage you to open a case with TAC to troubleshoot Tomcat (the  
web server) and your h.323 issue.  They definitely sound like  
problems but disk utilization is not the cause.

-Ryan

On Oct 8, 2007, at 4:47 AM, Andre Beck wrote:

Hi,

I didn't have significant problems with 5.1 so far (other than those  
with
localization that got fixed) and was running 5.1.1.3000-5 until some
weeks ago. Then I upgraded to 5.1.2.3000-2 without problems and the box
ran flawless until yesterday, when CCM suddenly dropped all H.323  
gateway
connections. Restarting the CCM service helped.

This morning, the box stopped answering HTTP connections (as tracked
by our Nagios monitoring) and had to be rebooted, after this it again
resumed normal operation.

What I noticed in the OS Admin Web interface is that the "Disk/active"
filesystems are rather full (my 5.1.1 is at 93%, the 5.1.2 was at 95%)
and they are *constantly* *growing*.

I switched versions back to 5.1.1 to verify this, it's the same there:

                         Total            Free            Used

Disk/active         12317944K         928084K       10764132K (93%)
[...]
Disk/active         12317944K         928068K       10764148K (93%)
[...]
Disk/active         12317944K         928064K       10764152K (93%)

without the number ever getting smaller again.

What would write to the software partions instead of to the common or
log partitions? Is this normal and maybe garbage collected every some
hours/days or might this be a reason for an installation suddenly  
failing
after weeks of runtime?

I'm probably going to upgrade to 5.1.3, which will let me lose my 5.1.1
fallback installation. Hopefully this will be stable again.

BTW, the absence of an "expert" login to actually search for the reason
of the growing diskspace is an ENTIRELY SILLY IDEA. I'm feeling like a
lobotomized ex-hacker at this "admin" shell...

Thanks,
Andre.
-- 
     .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From brian.gibson at stifel.com  Mon Oct  8 12:38:54 2007
From: brian.gibson at stifel.com (Gibson, Brian (Jersey City))
Date: Mon, 8 Oct 2007 11:38:54 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <BAY113-DAV639A17BCDF9766ECEE3F8E8A40@phx.gbl>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com><a6e20ec60710051508l3eb792adsb8d15b80b1af523c@mail.gmail.com>
	<BAY113-DAV639A17BCDF9766ECEE3F8E8A40@phx.gbl>
Message-ID: <5CC6FF682D32164BA5F2E830ABDF1460018D31AD@SNEXCHANGE02.stifelnet.stifel.local>

Personally I try to stay away from engineers specific configuration
questions because being able to answer those questions doesn't speak to
their overall networking knowledge but rather their configuration
experience.  You can teach anyone to configure a router or switch.  You
can't teach them how IP operates.

Anyway here are some basic questions I ask mid-level people I interview.

1. What is the purpose of ARP?

2. How does traceroute operate?

3. If you did a ping test and the response you received was "TTL expired
in transit" what does that most likely indicate?

4. Why is UDP used for multimedia services rather than TCP?  

5. What does IGMP do?  What does PIM do?  Extra credit for being able to
differentiate between sparse and dense.

Don't ask questions that are specific to your environment since the
prospective employee doesn't know your environment.  And don't try to
catch the interviewee on gotchas.  The purpose of the test is to get an
idea of how good the engineer is, not make him or her feel dumb because
they forgot some relatively minor point.  Also broader questions allow
the interviewee to speak so you can get a better idea of their technical
communications skills.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Syed Khalid Ali
Sent: Sunday, October 07, 2007 10:05 PM
To: ryan speed
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] A little Fun

hi,
 here r a few:

what is unicast, broadcast and multicast
how a switch learns a mac address
what does snap do at osi layer 2
what is mdix
difference between routing and switching
how to bridging avoid loop in switched network
how to avoid loop in distance vector protocols
types of routing protocols
what protocol and port number does bgp uses.
what types of area does ospf support
different between types of lsa in ospf
why need redistribution
why need route summarization
what is cidr
why do we need hierachy in ospf or is-is. can we do so in eigrp as well.
is is-is independent of ip protocol. if yes what is the advantage
what does a pasive interface do
difference between ccm, ccm business edition and ccme
what voice signaling protocol does ccm use
in how many ways we can implement call control (centralized or
distributed) 
in a  voice enabled network
when to use mgcp/h323 and sip
what voip signaling protocol does ccm use
how to calculate bandwidth for viop enabled network
difference between the call control methods.
what is the recomended anti virus for ccm
why need a sperate voice vlan
what is the qos value for voice and voice control traffic
what is difference between congestion management and avoidance
techniques available in cisco ios to do congestion management and
avoidance 
(eg: fair queue,cwfq, llq, red and wred)
what is 802.1p
phone get the ip and vlan but unable to register with ccme/ccm  why?
what is v3 vpn
what are the phases in establishing a vpn.
why is the difference between site-to-site and access vpn.
what is the command to see vpn peers
why use gre tunnels with ipsec vpns?





----- Original Message ----- 
From: "ryan speed" <rspeed at gmail.com>
To: <cisco-voip at puck.nether.net>
Sent: Saturday, October 06, 2007 3:08 AM
Subject: Re: [cisco-voip] A little Fun


> never hurts to throw a scenario or two in.
>
> you're tasked with upgrading the ios (pre-definined) on a voice
> gateway for a remote customer, what steps would you take and what
> precautions would you deem necessary for a trouble free upgrade with a
> solid backout plan in the event something goes wrong.
>
> take that answer and move onto another scenario (which if they
> answered the first one correctly should result in a pass or fail
> situation)
>
> you've upgraded to the new ios and rebooted the router, 30 minutes
> have passed and you're unable to reach the router, how will you
> proceed to troubleshoot/rectify the situation so the customer isn't
> stuck with a dead router with 12 hours till business opens again.
>
> On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
>> Thought you guys might want to take a break and help me out on a 
>> "related"
>> issue...
>>
>> We are interviewing new Network Engineer Applicants next week.
>>
>> While I was writing up my list of questions (the boss needs 10) I
thought 
>> I
>> would throw this out to you guys.
>> If you could write an interview question for your next Network
Engineer
>> (let's say Professional Level certification in Networking, Security
or
>> Voice)
>> what would you ask them?  (try to keep it technical, nothing about 
>> Packer's
>> tickets or Beer...lol)
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

********************************************************************************
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IMPORTANT: Please do not use e-mail to request or authorize the
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********************************************************************************


From jason.aarons at us.didata.com  Mon Oct  8 14:04:35 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 8 Oct 2007 14:04:35 -0400
Subject: [cisco-voip] 79x0 vs 79x1 (BLF?)
Message-ID: <C1FE15183DA37645BC0633BC604E44F00707E811@USNAEXCH.na.didata.local>

Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to
have "auto dial and appearance", they have been using the 7940 a week,
along with CCMUser and don't think AbbrDial/FastDials/Personal Directory
meets their needs. Their old phone had lots of buttons, etc.

What is the CallManager 5.1 feature called where a button can be both a
Shared Line and Speed Dial?  I'm trying to find the URL if it requires a
79x1 or would it work with a 7960.

What are the caveats? Can they setup/manage via CCMUser? 

I've been stuck on 4.1 a long time and recall seeing something about
this in 5.x, but can't find the details.  I know 4.1 you could have
either a shared line or a speed dial but not both.

 -jason

 




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From jonvoip at gmail.com  Mon Oct  8 14:53:16 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 8 Oct 2007 13:53:16 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <5CC6FF682D32164BA5F2E830ABDF1460018D31AD@SNEXCHANGE02.stifelnet.stifel.local>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
	<a6e20ec60710051508l3eb792adsb8d15b80b1af523c@mail.gmail.com>
	<BAY113-DAV639A17BCDF9766ECEE3F8E8A40@phx.gbl>
	<5CC6FF682D32164BA5F2E830ABDF1460018D31AD@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <5d093f9a0710081153p737af0casd8a14f9e320ecb60@mail.gmail.com>

But then you miss the depth of their skill...

Personally, if a CCNA doesn't know what ARP is or does, he might as
well use that cert to wipe his butt with for all it's worth...

A better one would be:

Explain how hold-down works for a distance-vector routing protocol...

Explain the IP Phone registration process...

etc...

I am not a fan of the factoid questions... I prefer the scenarios...
cuz if they can't shoot trouble, it is unlikely they can do
anything...



Jonathan



On 10/8/07, Gibson, Brian (Jersey City) <brian.gibson at stifel.com> wrote:
> Personally I try to stay away from engineers specific configuration
> questions because being able to answer those questions doesn't speak to
> their overall networking knowledge but rather their configuration
> experience.  You can teach anyone to configure a router or switch.  You
> can't teach them how IP operates.
>
> Anyway here are some basic questions I ask mid-level people I interview.
>
> 1. What is the purpose of ARP?
>
> 2. How does traceroute operate?
>
> 3. If you did a ping test and the response you received was "TTL expired
> in transit" what does that most likely indicate?
>
> 4. Why is UDP used for multimedia services rather than TCP?
>
> 5. What does IGMP do?  What does PIM do?  Extra credit for being able to
> differentiate between sparse and dense.
>
> Don't ask questions that are specific to your environment since the
> prospective employee doesn't know your environment.  And don't try to
> catch the interviewee on gotchas.  The purpose of the test is to get an
> idea of how good the engineer is, not make him or her feel dumb because
> they forgot some relatively minor point.  Also broader questions allow
> the interviewee to speak so you can get a better idea of their technical
> communications skills.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Syed Khalid Ali
> Sent: Sunday, October 07, 2007 10:05 PM
> To: ryan speed
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] A little Fun
>
> hi,
>  here r a few:
>
> what is unicast, broadcast and multicast
> how a switch learns a mac address
> what does snap do at osi layer 2
> what is mdix
> difference between routing and switching
> how to bridging avoid loop in switched network
> how to avoid loop in distance vector protocols
> types of routing protocols
> what protocol and port number does bgp uses.
> what types of area does ospf support
> different between types of lsa in ospf
> why need redistribution
> why need route summarization
> what is cidr
> why do we need hierachy in ospf or is-is. can we do so in eigrp as well.
> is is-is independent of ip protocol. if yes what is the advantage
> what does a pasive interface do
> difference between ccm, ccm business edition and ccme
> what voice signaling protocol does ccm use
> in how many ways we can implement call control (centralized or
> distributed)
> in a  voice enabled network
> when to use mgcp/h323 and sip
> what voip signaling protocol does ccm use
> how to calculate bandwidth for viop enabled network
> difference between the call control methods.
> what is the recomended anti virus for ccm
> why need a sperate voice vlan
> what is the qos value for voice and voice control traffic
> what is difference between congestion management and avoidance
> techniques available in cisco ios to do congestion management and
> avoidance
> (eg: fair queue,cwfq, llq, red and wred)
> what is 802.1p
> phone get the ip and vlan but unable to register with ccme/ccm  why?
> what is v3 vpn
> what are the phases in establishing a vpn.
> why is the difference between site-to-site and access vpn.
> what is the command to see vpn peers
> why use gre tunnels with ipsec vpns?
>
>
>
>
>
> ----- Original Message -----
> From: "ryan speed" <rspeed at gmail.com>
> To: <cisco-voip at puck.nether.net>
> Sent: Saturday, October 06, 2007 3:08 AM
> Subject: Re: [cisco-voip] A little Fun
>
>
> > never hurts to throw a scenario or two in.
> >
> > you're tasked with upgrading the ios (pre-definined) on a voice
> > gateway for a remote customer, what steps would you take and what
> > precautions would you deem necessary for a trouble free upgrade with a
> > solid backout plan in the event something goes wrong.
> >
> > take that answer and move onto another scenario (which if they
> > answered the first one correctly should result in a pass or fail
> > situation)
> >
> > you've upgraded to the new ios and rebooted the router, 30 minutes
> > have passed and you're unable to reach the router, how will you
> > proceed to troubleshoot/rectify the situation so the customer isn't
> > stuck with a dead router with 12 hours till business opens again.
> >
> > On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> >> Thought you guys might want to take a break and help me out on a
> >> "related"
> >> issue...
> >>
> >> We are interviewing new Network Engineer Applicants next week.
> >>
> >> While I was writing up my list of questions (the boss needs 10) I
> thought
> >> I
> >> would throw this out to you guys.
> >> If you could write an interview question for your next Network
> Engineer
> >> (let's say Professional Level certification in Networking, Security
> or
> >> Voice)
> >> what would you ask them?  (try to keep it technical, nothing about
> >> Packer's
> >> tickets or Beer...lol)
> >>
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> ********************************************************************************
> All electronic messages sent and received by Stifel Nicolaus
> Associates are subject to review by Stifel Nicolaus.  Stifel Nicolaus
> may retain and reproduce electronic messages for state, federal, or
> other regulatory agencies as required by applicable law.
> IMPORTANT: Please do not use e-mail to request or authorize the
> purchase or sale of any security or commodity, send fund transfer
> instructions, or otherwise conduct any securities transactions.  Any
> requests, orders, instructions, or time-sensitive messages sent by
> e-mail cannot be accepted or processed by Stifel Nicolaus.  The
> accuracy of any information sent by Stifel Nicolaus through e-mail
> cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
>
> Stifel, Nicolaus & Company, Incorporated
> Member NYSE & SIPC
> Headquarters:  501 N. Broadway, St. Louis, MO   63102
> 314-342-2000
> ********************************************************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From Dave.Dockendorf at ecolab.com  Mon Oct  8 14:59:47 2007
From: Dave.Dockendorf at ecolab.com (Dockendorf, Dave)
Date: Mon, 8 Oct 2007 13:59:47 -0500
Subject: [cisco-voip] ATA
In-Reply-To: <228dc9bd0710041013p73fcc61ch4cf2f524013a69f@mail.gmail.com>
Message-ID: <12844D65263E09419912D6EE22FB30E2D152D3@USEAGAN1505P.GLOBAL.ECOLAB.CORP>

Here is the full list of voice menu codes
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administr
ation_guide_chapter09186a00800c4d20.html#61645

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Michael Sy
Sent: Thursday, October 04, 2007 12:13 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ATA


ATA new install not coming up at all, red blinkies.  any assistance pls.
ccm 4.2.3 es44

-- 
Mike Sy . 

CONFIDENTIALITY NOTICE: This e-mail communication and any attachments may contain proprietary and privileged information for the use of the designated recipients named above. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the original message.

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From brian.gibson at stifel.com  Mon Oct  8 15:03:42 2007
From: brian.gibson at stifel.com (Gibson, Brian (Jersey City))
Date: Mon, 8 Oct 2007 14:03:42 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <5d093f9a0710081153p737af0casd8a14f9e320ecb60@mail.gmail.com>
References: <84b74aec0710051402u5349644dqdfc7d8f5f43ba3ce@mail.gmail.com>
	<a6e20ec60710051508l3eb792adsb8d15b80b1af523c@mail.gmail.com>
	<BAY113-DAV639A17BCDF9766ECEE3F8E8A40@phx.gbl>
	<5CC6FF682D32164BA5F2E830ABDF1460018D31AD@SNEXCHANGE02.stifelnet.stifel.local>
	<5d093f9a0710081153p737af0casd8a14f9e320ecb60@mail.gmail.com>
Message-ID: <5CC6FF682D32164BA5F2E830ABDF146001A5AB66@SNEXCHANGE02.stifelnet.stifel.local>

I was always surprised by how many engineers, particularly mid-level
engineers, didn't understand what ARP really did.  Sure they could
recite the definition but they really didn't have a clear understanding
of the OSI model so didn't really understand what ARP really did.

The problem with scenarios is that are many possible solutions where you
may only be looking for one.

Someone earlier had offered the scenario "You've upgraded the IOS on a
router and now you can't get to it.  What do you do?"  Well there are
any number of different options depending on the circumstances and
environment you are working in.  

The Hold-down question is pretty good.  The IP Phone registration
question is a little too broad, IMO. 

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Monday, October 08, 2007 2:53 PM
To: Gibson, Brian (Jersey City)
Cc: Syed Khalid Ali; ryan speed; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] A little Fun

But then you miss the depth of their skill...

Personally, if a CCNA doesn't know what ARP is or does, he might as
well use that cert to wipe his butt with for all it's worth...

A better one would be:

Explain how hold-down works for a distance-vector routing protocol...

Explain the IP Phone registration process...

etc...

I am not a fan of the factoid questions... I prefer the scenarios...
cuz if they can't shoot trouble, it is unlikely they can do
anything...



Jonathan



On 10/8/07, Gibson, Brian (Jersey City) <brian.gibson at stifel.com> wrote:
> Personally I try to stay away from engineers specific configuration
> questions because being able to answer those questions doesn't speak
to
> their overall networking knowledge but rather their configuration
> experience.  You can teach anyone to configure a router or switch.
You
> can't teach them how IP operates.
>
> Anyway here are some basic questions I ask mid-level people I
interview.
>
> 1. What is the purpose of ARP?
>
> 2. How does traceroute operate?
>
> 3. If you did a ping test and the response you received was "TTL
expired
> in transit" what does that most likely indicate?
>
> 4. Why is UDP used for multimedia services rather than TCP?
>
> 5. What does IGMP do?  What does PIM do?  Extra credit for being able
to
> differentiate between sparse and dense.
>
> Don't ask questions that are specific to your environment since the
> prospective employee doesn't know your environment.  And don't try to
> catch the interviewee on gotchas.  The purpose of the test is to get
an
> idea of how good the engineer is, not make him or her feel dumb
because
> they forgot some relatively minor point.  Also broader questions allow
> the interviewee to speak so you can get a better idea of their
technical
> communications skills.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Syed Khalid
Ali
> Sent: Sunday, October 07, 2007 10:05 PM
> To: ryan speed
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] A little Fun
>
> hi,
>  here r a few:
>
> what is unicast, broadcast and multicast
> how a switch learns a mac address
> what does snap do at osi layer 2
> what is mdix
> difference between routing and switching
> how to bridging avoid loop in switched network
> how to avoid loop in distance vector protocols
> types of routing protocols
> what protocol and port number does bgp uses.
> what types of area does ospf support
> different between types of lsa in ospf
> why need redistribution
> why need route summarization
> what is cidr
> why do we need hierachy in ospf or is-is. can we do so in eigrp as
well.
> is is-is independent of ip protocol. if yes what is the advantage
> what does a pasive interface do
> difference between ccm, ccm business edition and ccme
> what voice signaling protocol does ccm use
> in how many ways we can implement call control (centralized or
> distributed)
> in a  voice enabled network
> when to use mgcp/h323 and sip
> what voip signaling protocol does ccm use
> how to calculate bandwidth for viop enabled network
> difference between the call control methods.
> what is the recomended anti virus for ccm
> why need a sperate voice vlan
> what is the qos value for voice and voice control traffic
> what is difference between congestion management and avoidance
> techniques available in cisco ios to do congestion management and
> avoidance
> (eg: fair queue,cwfq, llq, red and wred)
> what is 802.1p
> phone get the ip and vlan but unable to register with ccme/ccm  why?
> what is v3 vpn
> what are the phases in establishing a vpn.
> why is the difference between site-to-site and access vpn.
> what is the command to see vpn peers
> why use gre tunnels with ipsec vpns?
>
>
>
>
>
> ----- Original Message -----
> From: "ryan speed" <rspeed at gmail.com>
> To: <cisco-voip at puck.nether.net>
> Sent: Saturday, October 06, 2007 3:08 AM
> Subject: Re: [cisco-voip] A little Fun
>
>
> > never hurts to throw a scenario or two in.
> >
> > you're tasked with upgrading the ios (pre-definined) on a voice
> > gateway for a remote customer, what steps would you take and what
> > precautions would you deem necessary for a trouble free upgrade with
a
> > solid backout plan in the event something goes wrong.
> >
> > take that answer and move onto another scenario (which if they
> > answered the first one correctly should result in a pass or fail
> > situation)
> >
> > you've upgraded to the new ios and rebooted the router, 30 minutes
> > have passed and you're unable to reach the router, how will you
> > proceed to troubleshoot/rectify the situation so the customer isn't
> > stuck with a dead router with 12 hours till business opens again.
> >
> > On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> >> Thought you guys might want to take a break and help me out on a
> >> "related"
> >> issue...
> >>
> >> We are interviewing new Network Engineer Applicants next week.
> >>
> >> While I was writing up my list of questions (the boss needs 10) I
> thought
> >> I
> >> would throw this out to you guys.
> >> If you could write an interview question for your next Network
> Engineer
> >> (let's say Professional Level certification in Networking, Security
> or
> >> Voice)
> >> what would you ask them?  (try to keep it technical, nothing about
> >> Packer's
> >> tickets or Beer...lol)
> >>
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
************************************************************************
********
> All electronic messages sent and received by Stifel Nicolaus
> Associates are subject to review by Stifel Nicolaus.  Stifel Nicolaus
> may retain and reproduce electronic messages for state, federal, or
> other regulatory agencies as required by applicable law.
> IMPORTANT: Please do not use e-mail to request or authorize the
> purchase or sale of any security or commodity, send fund transfer
> instructions, or otherwise conduct any securities transactions.  Any
> requests, orders, instructions, or time-sensitive messages sent by
> e-mail cannot be accepted or processed by Stifel Nicolaus.  The
> accuracy of any information sent by Stifel Nicolaus through e-mail
> cannot be warranted or guaranteed by Stifel Nicolaus or its
affiliates.
>
> Stifel, Nicolaus & Company, Incorporated
> Member NYSE & SIPC
> Headquarters:  501 N. Broadway, St. Louis, MO   63102
> 314-342-2000
>
************************************************************************
********
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From matt at saskin.net  Mon Oct  8 15:13:31 2007
From: matt at saskin.net (Matthew Saskin)
Date: Mon, 08 Oct 2007 15:13:31 -0400
Subject: [cisco-voip] IPCC Express Reporting access
In-Reply-To: <592843.1986.qm@web44803.mail.sp1.yahoo.com>
References: <592843.1986.qm@web44803.mail.sp1.yahoo.com>
Message-ID: <470A815B.6090600@saskin.net>

One possible way (although labor intensive on your end) would be to 
provide direct ODBC access to the database, however only allow 
permissions to run selects, etc. on views that you create.  This would 
effectively let you filter the data before the client can even query it.

As I said, time consuming on your end...

-matt

Cucumber Green wrote:
> I think you should take a look into the db_cra database. Specifically 
> the table ContactCallDetail.
> If the data of your contact centers is stored in the same database, in 
> the same table, there would be a security concern by giving one of your 
> users access through ODBC, as he/she would be able to look at records of 
> other contact centers.
>  
> I have done custom reports using Crystal Reports (which rely on SQL 
> Server stored procedures), never used ODBC to extract the data directly.
> 
> */Ed Leatherman <ealeatherman at gmail.com>/* wrote:
> 
>     Curious how people handle reporting with IPCC Express.. we run a
>     number of smaller call centers off of our IPCC Express system. Up
>     till now we've just been generating reports using the historical
>     reporting tool, but there are some units now that would like the
>     reports to either be in a different format or totally different
>     reports altogether than what we can get from the canned tool.
> 
>     Do most people use Crystal to accomplish this? Anyone provided an
>     ODBC connection to more database savvy users? My database knowledge
>     is pretty limited and I know zilch about crystal, trying to figure
>     out the best way (security and performance wise) to handle it and
>     still provide the info these folks are asking for. They already
>     understand they will be doing anything special with reporting
>     after-hours.
> 
>     -- 
>     Ed Leatherman
>     Senior Voice Engineer
>     West Virginia University
>     Telecommunications and Network Operations
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> ------------------------------------------------------------------------
> Luggage? GPS? Comic books?
> Check out fitting gifts for grads 
> <http://us.rd.yahoo.com/evt=48249/*http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz> 
> at Yahoo! Search.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jonvoip at gmail.com  Mon Oct  8 15:40:12 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 8 Oct 2007 14:40:12 -0500
Subject: [cisco-voip] A little Fun
In-Reply-To: <EAC75A12BE260246A8145B73D2613087F482FE@exchange00.semaphore.lan>
References: <5CC6FF682D32164BA5F2E830ABDF146001A5AB66@SNEXCHANGE02.stifelnet.stifel.local>
	<EAC75A12BE260246A8145B73D2613087F482FE@exchange00.semaphore.lan>
Message-ID: <5d093f9a0710081240n7a3d1133yedc471a30214965e@mail.gmail.com>

Would you penalize them for saying that their personal information and
a picture of them in a g-string will be planted all over the interwebs
and they will receive directed advertising?

I like the question, especially if you include DNS, and a
determination if the destination is local or remote based on subnet
mask, etc...

One good thing, ARP-wise to know is how the source and destination
MACs (or whatever datalink addresses are affixed...) change between
each routed node...

Here's a good one.

You look at a CAM table on a switch, and you see a lot of MAC
addresses all connected via one port... why?


Jonathan

On 10/8/07, Dylan Vanderhoof <DylanV at semaphore.com> wrote:
> We've found that questions requiring knowledge of the full range of the networking stack can provide a lot of insight.
>
> A particularly fascinating one is: "Describe in as much detail as you can what happens when you type 'www.google.com' into your browser"  Obviously, you can lead them from there, but it serves to indicate what they know about both layer 2 and 3, as well as some of the application layer stuff.
>
> The real point of an interview question from my perspective is to give a starting point from which to more specifically probe their knowledge of areas that could be a problem.  We've particularly found that few of the engineers we've interviewed have a particularly good grasp of layer 2.
>
> -D
>
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> > Gibson, Brian (Jersey City)
> > Sent: Monday, October 08, 2007 12:04 PM
> > To: Jonathan Charles
> > Cc: Syed Khalid Ali; cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] A little Fun
> >
> >
> > I was always surprised by how many engineers, particularly mid-level
> > engineers, didn't understand what ARP really did.  Sure they could
> > recite the definition but they really didn't have a clear
> > understanding
> > of the OSI model so didn't really understand what ARP really did.
> >
> > The problem with scenarios is that are many possible
> > solutions where you
> > may only be looking for one.
> >
> > Someone earlier had offered the scenario "You've upgraded the IOS on a
> > router and now you can't get to it.  What do you do?"  Well there are
> > any number of different options depending on the circumstances and
> > environment you are working in.
> >
> > The Hold-down question is pretty good.  The IP Phone registration
> > question is a little too broad, IMO.
> >
> > -----Original Message-----
> > From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > Sent: Monday, October 08, 2007 2:53 PM
> > To: Gibson, Brian (Jersey City)
> > Cc: Syed Khalid Ali; ryan speed; cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] A little Fun
> >
> > But then you miss the depth of their skill...
> >
> > Personally, if a CCNA doesn't know what ARP is or does, he might as
> > well use that cert to wipe his butt with for all it's worth...
> >
> > A better one would be:
> >
> > Explain how hold-down works for a distance-vector routing protocol...
> >
> > Explain the IP Phone registration process...
> >
> > etc...
> >
> > I am not a fan of the factoid questions... I prefer the scenarios...
> > cuz if they can't shoot trouble, it is unlikely they can do
> > anything...
> >
> >
> >
> > Jonathan
> >
> >
> >
> > On 10/8/07, Gibson, Brian (Jersey City)
> > <brian.gibson at stifel.com> wrote:
> > > Personally I try to stay away from engineers specific configuration
> > > questions because being able to answer those questions doesn't speak
> > to
> > > their overall networking knowledge but rather their configuration
> > > experience.  You can teach anyone to configure a router or switch.
> > You
> > > can't teach them how IP operates.
> > >
> > > Anyway here are some basic questions I ask mid-level people I
> > interview.
> > >
> > > 1. What is the purpose of ARP?
> > >
> > > 2. How does traceroute operate?
> > >
> > > 3. If you did a ping test and the response you received was "TTL
> > expired
> > > in transit" what does that most likely indicate?
> > >
> > > 4. Why is UDP used for multimedia services rather than TCP?
> > >
> > > 5. What does IGMP do?  What does PIM do?  Extra credit for
> > being able
> > to
> > > differentiate between sparse and dense.
> > >
> > > Don't ask questions that are specific to your environment since the
> > > prospective employee doesn't know your environment.  And
> > don't try to
> > > catch the interviewee on gotchas.  The purpose of the test is to get
> > an
> > > idea of how good the engineer is, not make him or her feel dumb
> > because
> > > they forgot some relatively minor point.  Also broader
> > questions allow
> > > the interviewee to speak so you can get a better idea of their
> > technical
> > > communications skills.
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Syed Khalid
> > Ali
> > > Sent: Sunday, October 07, 2007 10:05 PM
> > > To: ryan speed
> > > Cc: cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] A little Fun
> > >
> > > hi,
> > >  here r a few:
> > >
> > > what is unicast, broadcast and multicast
> > > how a switch learns a mac address
> > > what does snap do at osi layer 2
> > > what is mdix
> > > difference between routing and switching
> > > how to bridging avoid loop in switched network
> > > how to avoid loop in distance vector protocols
> > > types of routing protocols
> > > what protocol and port number does bgp uses.
> > > what types of area does ospf support
> > > different between types of lsa in ospf
> > > why need redistribution
> > > why need route summarization
> > > what is cidr
> > > why do we need hierachy in ospf or is-is. can we do so in eigrp as
> > well.
> > > is is-is independent of ip protocol. if yes what is the advantage
> > > what does a pasive interface do
> > > difference between ccm, ccm business edition and ccme
> > > what voice signaling protocol does ccm use
> > > in how many ways we can implement call control (centralized or
> > > distributed)
> > > in a  voice enabled network
> > > when to use mgcp/h323 and sip
> > > what voip signaling protocol does ccm use
> > > how to calculate bandwidth for viop enabled network
> > > difference between the call control methods.
> > > what is the recomended anti virus for ccm
> > > why need a sperate voice vlan
> > > what is the qos value for voice and voice control traffic
> > > what is difference between congestion management and avoidance
> > > techniques available in cisco ios to do congestion management and
> > > avoidance
> > > (eg: fair queue,cwfq, llq, red and wred)
> > > what is 802.1p
> > > phone get the ip and vlan but unable to register with ccme/ccm  why?
> > > what is v3 vpn
> > > what are the phases in establishing a vpn.
> > > why is the difference between site-to-site and access vpn.
> > > what is the command to see vpn peers
> > > why use gre tunnels with ipsec vpns?
> > >
> > >
> > >
> > >
> > >
> > > ----- Original Message -----
> > > From: "ryan speed" <rspeed at gmail.com>
> > > To: <cisco-voip at puck.nether.net>
> > > Sent: Saturday, October 06, 2007 3:08 AM
> > > Subject: Re: [cisco-voip] A little Fun
> > >
> > >
> > > > never hurts to throw a scenario or two in.
> > > >
> > > > you're tasked with upgrading the ios (pre-definined) on a voice
> > > > gateway for a remote customer, what steps would you take and what
> > > > precautions would you deem necessary for a trouble free
> > upgrade with
> > a
> > > > solid backout plan in the event something goes wrong.
> > > >
> > > > take that answer and move onto another scenario (which if they
> > > > answered the first one correctly should result in a pass or fail
> > > > situation)
> > > >
> > > > you've upgraded to the new ios and rebooted the router, 30 minutes
> > > > have passed and you're unable to reach the router, how will you
> > > > proceed to troubleshoot/rectify the situation so the
> > customer isn't
> > > > stuck with a dead router with 12 hours till business opens again.
> > > >
> > > > On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > > >> Thought you guys might want to take a break and help me out on a
> > > >> "related"
> > > >> issue...
> > > >>
> > > >> We are interviewing new Network Engineer Applicants next week.
> > > >>
> > > >> While I was writing up my list of questions (the boss needs 10) I
> > > thought
> > > >> I
> > > >> would throw this out to you guys.
> > > >> If you could write an interview question for your next Network
> > > Engineer
> > > >> (let's say Professional Level certification in
> > Networking, Security
> > > or
> > > >> Voice)
> > > >> what would you ask them?  (try to keep it technical,
> > nothing about
> > > >> Packer's
> > > >> tickets or Beer...lol)
> > > >>
> > > >>
> > > >> _______________________________________________
> > > >> cisco-voip mailing list
> > > >> cisco-voip at puck.nether.net
> > > >> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >>
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > **************************************************************
> > **********
> > ********
> > > All electronic messages sent and received by Stifel Nicolaus
> > > Associates are subject to review by Stifel Nicolaus.
> > Stifel Nicolaus
> > > may retain and reproduce electronic messages for state, federal, or
> > > other regulatory agencies as required by applicable law.
> > > IMPORTANT: Please do not use e-mail to request or authorize the
> > > purchase or sale of any security or commodity, send fund transfer
> > > instructions, or otherwise conduct any securities transactions.  Any
> > > requests, orders, instructions, or time-sensitive messages sent by
> > > e-mail cannot be accepted or processed by Stifel Nicolaus.  The
> > > accuracy of any information sent by Stifel Nicolaus through e-mail
> > > cannot be warranted or guaranteed by Stifel Nicolaus or its
> > affiliates.
> > >
> > > Stifel, Nicolaus & Company, Incorporated
> > > Member NYSE & SIPC
> > > Headquarters:  501 N. Broadway, St. Louis, MO   63102
> > > 314-342-2000
> > >
> > **************************************************************
> > **********
> > ********
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From DylanV at semaphore.com  Mon Oct  8 15:12:24 2007
From: DylanV at semaphore.com (Dylan Vanderhoof)
Date: Mon, 8 Oct 2007 12:12:24 -0700
Subject: [cisco-voip] A little Fun
In-Reply-To: <5CC6FF682D32164BA5F2E830ABDF146001A5AB66@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <EAC75A12BE260246A8145B73D2613087F482FE@exchange00.semaphore.lan>

We've found that questions requiring knowledge of the full range of the networking stack can provide a lot of insight.

A particularly fascinating one is: "Describe in as much detail as you can what happens when you type 'www.google.com' into your browser"  Obviously, you can lead them from there, but it serves to indicate what they know about both layer 2 and 3, as well as some of the application layer stuff.

The real point of an interview question from my perspective is to give a starting point from which to more specifically probe their knowledge of areas that could be a problem.  We've particularly found that few of the engineers we've interviewed have a particularly good grasp of layer 2.

-D

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Gibson, Brian (Jersey City)
> Sent: Monday, October 08, 2007 12:04 PM
> To: Jonathan Charles
> Cc: Syed Khalid Ali; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] A little Fun
>
>
> I was always surprised by how many engineers, particularly mid-level
> engineers, didn't understand what ARP really did.  Sure they could
> recite the definition but they really didn't have a clear
> understanding
> of the OSI model so didn't really understand what ARP really did.
>
> The problem with scenarios is that are many possible
> solutions where you
> may only be looking for one.
>
> Someone earlier had offered the scenario "You've upgraded the IOS on a
> router and now you can't get to it.  What do you do?"  Well there are
> any number of different options depending on the circumstances and
> environment you are working in.
>
> The Hold-down question is pretty good.  The IP Phone registration
> question is a little too broad, IMO.
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Monday, October 08, 2007 2:53 PM
> To: Gibson, Brian (Jersey City)
> Cc: Syed Khalid Ali; ryan speed; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] A little Fun
>
> But then you miss the depth of their skill...
>
> Personally, if a CCNA doesn't know what ARP is or does, he might as
> well use that cert to wipe his butt with for all it's worth...
>
> A better one would be:
>
> Explain how hold-down works for a distance-vector routing protocol...
>
> Explain the IP Phone registration process...
>
> etc...
>
> I am not a fan of the factoid questions... I prefer the scenarios...
> cuz if they can't shoot trouble, it is unlikely they can do
> anything...
>
>
>
> Jonathan
>
>
>
> On 10/8/07, Gibson, Brian (Jersey City)
> <brian.gibson at stifel.com> wrote:
> > Personally I try to stay away from engineers specific configuration
> > questions because being able to answer those questions doesn't speak
> to
> > their overall networking knowledge but rather their configuration
> > experience.  You can teach anyone to configure a router or switch.
> You
> > can't teach them how IP operates.
> >
> > Anyway here are some basic questions I ask mid-level people I
> interview.
> >
> > 1. What is the purpose of ARP?
> >
> > 2. How does traceroute operate?
> >
> > 3. If you did a ping test and the response you received was "TTL
> expired
> > in transit" what does that most likely indicate?
> >
> > 4. Why is UDP used for multimedia services rather than TCP?
> >
> > 5. What does IGMP do?  What does PIM do?  Extra credit for
> being able
> to
> > differentiate between sparse and dense.
> >
> > Don't ask questions that are specific to your environment since the
> > prospective employee doesn't know your environment.  And
> don't try to
> > catch the interviewee on gotchas.  The purpose of the test is to get
> an
> > idea of how good the engineer is, not make him or her feel dumb
> because
> > they forgot some relatively minor point.  Also broader
> questions allow
> > the interviewee to speak so you can get a better idea of their
> technical
> > communications skills.
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Syed Khalid
> Ali
> > Sent: Sunday, October 07, 2007 10:05 PM
> > To: ryan speed
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] A little Fun
> >
> > hi,
> >  here r a few:
> >
> > what is unicast, broadcast and multicast
> > how a switch learns a mac address
> > what does snap do at osi layer 2
> > what is mdix
> > difference between routing and switching
> > how to bridging avoid loop in switched network
> > how to avoid loop in distance vector protocols
> > types of routing protocols
> > what protocol and port number does bgp uses.
> > what types of area does ospf support
> > different between types of lsa in ospf
> > why need redistribution
> > why need route summarization
> > what is cidr
> > why do we need hierachy in ospf or is-is. can we do so in eigrp as
> well.
> > is is-is independent of ip protocol. if yes what is the advantage
> > what does a pasive interface do
> > difference between ccm, ccm business edition and ccme
> > what voice signaling protocol does ccm use
> > in how many ways we can implement call control (centralized or
> > distributed)
> > in a  voice enabled network
> > when to use mgcp/h323 and sip
> > what voip signaling protocol does ccm use
> > how to calculate bandwidth for viop enabled network
> > difference between the call control methods.
> > what is the recomended anti virus for ccm
> > why need a sperate voice vlan
> > what is the qos value for voice and voice control traffic
> > what is difference between congestion management and avoidance
> > techniques available in cisco ios to do congestion management and
> > avoidance
> > (eg: fair queue,cwfq, llq, red and wred)
> > what is 802.1p
> > phone get the ip and vlan but unable to register with ccme/ccm  why?
> > what is v3 vpn
> > what are the phases in establishing a vpn.
> > why is the difference between site-to-site and access vpn.
> > what is the command to see vpn peers
> > why use gre tunnels with ipsec vpns?
> >
> >
> >
> >
> >
> > ----- Original Message -----
> > From: "ryan speed" <rspeed at gmail.com>
> > To: <cisco-voip at puck.nether.net>
> > Sent: Saturday, October 06, 2007 3:08 AM
> > Subject: Re: [cisco-voip] A little Fun
> >
> >
> > > never hurts to throw a scenario or two in.
> > >
> > > you're tasked with upgrading the ios (pre-definined) on a voice
> > > gateway for a remote customer, what steps would you take and what
> > > precautions would you deem necessary for a trouble free
> upgrade with
> a
> > > solid backout plan in the event something goes wrong.
> > >
> > > take that answer and move onto another scenario (which if they
> > > answered the first one correctly should result in a pass or fail
> > > situation)
> > >
> > > you've upgraded to the new ios and rebooted the router, 30 minutes
> > > have passed and you're unable to reach the router, how will you
> > > proceed to troubleshoot/rectify the situation so the
> customer isn't
> > > stuck with a dead router with 12 hours till business opens again.
> > >
> > > On 10/5/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > >> Thought you guys might want to take a break and help me out on a
> > >> "related"
> > >> issue...
> > >>
> > >> We are interviewing new Network Engineer Applicants next week.
> > >>
> > >> While I was writing up my list of questions (the boss needs 10) I
> > thought
> > >> I
> > >> would throw this out to you guys.
> > >> If you could write an interview question for your next Network
> > Engineer
> > >> (let's say Professional Level certification in
> Networking, Security
> > or
> > >> Voice)
> > >> what would you ask them?  (try to keep it technical,
> nothing about
> > >> Packer's
> > >> tickets or Beer...lol)
> > >>
> > >>
> > >> _______________________________________________
> > >> cisco-voip mailing list
> > >> cisco-voip at puck.nether.net
> > >> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
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> >
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> >
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jason.aarons at us.didata.com  Mon Oct  8 15:53:55 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 8 Oct 2007 15:53:55 -0400
Subject: [cisco-voip] 5.1 Speed Dial Presence
Message-ID: <C1FE15183DA37645BC0633BC604E44F00707EA18@USNAEXCH.na.didata.local>

I think what they want is DSS/BLF, does it require a dedicated presence
server or can CCM  5.1 do this natively?

Cisco Unified CallManager Speed Dial Presence 

Cisco Unified CallManager supports the ability for a speed dial to have
presence capabilities via a busy lamp field (BLF) speed dial. BLF speed
dials work as both a speed dial and a presence indicator. Only the
system administrator can configure a BLF speed dial; a system user is
not allowed to configure a BLF speed dial. 

States are: 

Idle 

Busy 

Unknown 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementa
tion_design_guide_chapter09186a008070bc02.html#wp1070507
<javascript:newWin('http://www.cisco.com/en/US/products/sw/voicesw/ps556
/products_implementation_design_guide_chapter09186a008070bc02.html#wp107
0507')> 

 

From: Jason Aarons (US) 
Sent: Monday, October 08, 2007 2:05 PM
To: cisco-voip at puck.nether.net
Subject: 79x0 vs 79x1 (BLF?)

 

Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to
have "auto dial and appearance", they have been using the 7940 a week,
along with CCMUser and don't think AbbrDial/FastDials/Personal Directory
meets their needs. Their old phone had lots of buttons, etc.

What is the CallManager 5.1 feature called where a button can be both a
Shared Line and Speed Dial?  I'm trying to find the URL if it requires a
79x1 or would it work with a 7960.

What are the caveats? Can they setup/manage via CCMUser? 

I've been stuck on 4.1 a long time and recall seeing something about
this in 5.x, but can't find the details.  I know 4.1 you could have
either a shared line or a speed dial but not both.

 -jason

 




-----------------------------------------
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From wsisk at cisco.com  Mon Oct  8 16:01:12 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 08 Oct 2007 16:01:12 -0400
Subject: [cisco-voip] 5.1 Speed Dial Presence
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707EA18@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00707EA18@USNAEXCH.na.didata.local>
Message-ID: <470A8C88.30703@cisco.com>

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From jason.aarons at us.didata.com  Mon Oct  8 16:08:57 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 8 Oct 2007 16:08:57 -0400
Subject: [cisco-voip] 5.1 Speed Dial Presence
In-Reply-To: <470A8C88.30703@cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F00707EA18@USNAEXCH.na.didata.local>
	<470A8C88.30703@cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00707EA77@USNAEXCH.na.didata.local>

Any benefits if they used 79x1 over 7960 for Speed Dial Presence? They
are about to order more 7960s over 7961s due to costs. I wish there was
some motivating  one-reason issue to get customers to buy 79x1s. I know
the benefits of 79x1, and would only buy this model myself, but costs
are sometimes the winner with customers over logic. Can't we end of life
the 7960s and make the 7961 same price!

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Monday, October 08, 2007 4:01 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence

 

native.  no CUPS required. /wes

Jason Aarons (US) wrote: 

I think what they want is DSS/BLF, does it require a dedicated presence
server or can CCM  5.1 do this natively?

Cisco Unified CallManager Speed Dial Presence 

Cisco Unified CallManager supports the ability for a speed dial to have
presence capabilities via a busy lamp field (BLF) speed dial. BLF speed
dials work as both a speed dial and a presence indicator. Only the
system administrator can configure a BLF speed dial; a system user is
not allowed to configure a BLF speed dial. 

States are: 

Idle 

Busy 

Unknown 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementa
tion_design_guide_chapter09186a008070bc02.html#wp1070507
<javascript:newWin('http://www.cisco.com/en/US/products/sw/voicesw/ps556
/products_implementation_design_guide_chapter09186a008070bc02.html#wp107
0507')> 

 

From: Jason Aarons (US) 
Sent: Monday, October 08, 2007 2:05 PM
To: cisco-voip at puck.nether.net
Subject: 79x0 vs 79x1 (BLF?)

 

Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to
have "auto dial and appearance", they have been using the 7940 a week,
along with CCMUser and don't think AbbrDial/FastDials/Personal Directory
meets their needs. Their old phone had lots of buttons, etc.

What is the CallManager 5.1 feature called where a button can be both a
Shared Line and Speed Dial?  I'm trying to find the URL if it requires a
79x1 or would it work with a 7960.

What are the caveats? Can they setup/manage via CCMUser? 

I've been stuck on 4.1 a long time and recall seeing something about
this in 5.x, but can't find the details.  I know 4.1 you could have
either a shared line or a speed dial but not both.

 -jason

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 

 


________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



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From jsteinberg at gmail.com  Mon Oct  8 16:45:45 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Mon, 8 Oct 2007 16:45:45 -0400
Subject: [cisco-voip] 5.1 Speed Dial Presence
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707EA77@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00707EA18@USNAEXCH.na.didata.local>
	<470A8C88.30703@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00707EA77@USNAEXCH.na.didata.local>
Message-ID: <f1b65f880710081345p51b98bdcq4cce173c8e18734c@mail.gmail.com>

i would highly recommend the 79x1 if they are complaining about the
lack of BLF status.  The 79x1 light up the line button with BLF info.
The 79x0 dont.

If they really miss their buttons they can use the 7931 phone with
CM6.  I haven't used this phone yet but i think the price is much less
than the 7961.

On 10/8/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> Any benefits if they used 79x1 over 7960 for Speed Dial Presence? They are
> about to order more 7960s over 7961s due to costs. I wish there was some
> motivating  one-reason issue to get customers to buy 79x1s. I know the
> benefits of 79x1, and would only buy this model myself, but costs are
> sometimes the winner with customers over logic. Can't we end of life the
> 7960s and make the 7961 same price!
>
>
>
>
>
> From: Wes Sisk [mailto:wsisk at cisco.com]
>  Sent: Monday, October 08, 2007 4:01 PM
>  To: Jason Aarons (US)
>  Cc: cisco-voip at puck.nether.net
>  Subject: Re: [cisco-voip] 5.1 Speed Dial Presence
>
>
>
>
> native.  no CUPS required. /wes
>
>  Jason Aarons (US) wrote:
>
> I think what they want is DSS/BLF, does it require a dedicated presence
> server or can CCM  5.1 do this natively?
>
> Cisco Unified CallManager Speed Dial Presence
>
>  Cisco Unified CallManager supports the ability for a speed dial to have
> presence capabilities via a busy lamp field (BLF) speed dial. BLF speed
> dials work as both a speed dial and a presence indicator. Only the system
> administrator can configure a BLF speed dial; a system user is not allowed
> to configure a BLF speed dial.
>
>  States are:
>
>  Idle
>
>  Busy
>
>  Unknown
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_chapter09186a008070bc02.html#wp1070507
>
>
>
>
>
> From: Jason Aarons (US)
>  Sent: Monday, October 08, 2007 2:05 PM
>  To: cisco-voip at puck.nether.net
>  Subject: 79x0 vs 79x1 (BLF?)
>
>
>
> Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to have
> "auto dial and appearance", they have been using the 7940 a week, along with
> CCMUser and don't think AbbrDial/FastDials/Personal Directory meets their
> needs. Their old phone had lots of buttons, etc.
>
> What is the CallManager 5.1 feature called where a button can be both a
> Shared Line and Speed Dial?  I'm trying to find the URL if it requires a
> 79x1 or would it work with a 7960.
>
> What are the caveats? Can they setup/manage via CCMUser?
>
> I've been stuck on 4.1 a long time and recall seeing something about this in
> 5.x, but can't find the details.  I know 4.1 you could have either a shared
> line or a speed dial but not both.
>
>  -jason
>
>
>  ________________________________
>
>
> Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
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>
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>
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>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
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>

From wsisk at cisco.com  Mon Oct  8 17:24:22 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 08 Oct 2007 17:24:22 -0400
Subject: [cisco-voip] 5.1 Speed Dial Presence
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707EA77@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00707EA18@USNAEXCH.na.didata.local>
	<470A8C88.30703@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00707EA77@USNAEXCH.na.didata.local>
Message-ID: <470AA006.4080007@cisco.com>

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From Nick.Kassel at Charles-Stanley.co.uk  Tue Oct  9 03:59:16 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 9 Oct 2007 08:59:16 +0100
Subject: [cisco-voip] 5.1 Speed Dial Presence
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>

Wes

Does that include no more development for 7970

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: 08 October 2007 22:24
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence

 

x1 allows you to see BLF status of calling party in
missed/placed/received directories.  Personally, this is one of my
favorite features.

x1 - will be around much longer and get new features including midlets.
x0 - no more feature development.

/Wes

Jason Aarons (US) wrote: 

Any benefits if they used 79x1 over 7960 for Speed Dial Presence? They
are about to order more 7960s over 7961s due to costs. I wish there was
some motivating  one-reason issue to get customers to buy 79x1s. I know
the benefits of 79x1, and would only buy this model myself, but costs
are sometimes the winner with customers over logic. Can't we end of life
the 7960s and make the 7961 same price!

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Monday, October 08, 2007 4:01 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence

 

native.  no CUPS required. /wes

Jason Aarons (US) wrote: 

I think what they want is DSS/BLF, does it require a dedicated presence
server or can CCM  5.1 do this natively?

Cisco Unified CallManager Speed Dial Presence 

Cisco Unified CallManager supports the ability for a speed dial to have
presence capabilities via a busy lamp field (BLF) speed dial. BLF speed
dials work as both a speed dial and a presence indicator. Only the
system administrator can configure a BLF speed dial; a system user is
not allowed to configure a BLF speed dial. 

States are: 

Idle 

Busy 

Unknown 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementa
tion_design_guide_chapter09186a008070bc02.html#wp1070507

 

From: Jason Aarons (US) 
Sent: Monday, October 08, 2007 2:05 PM
To: cisco-voip at puck.nether.net
Subject: 79x0 vs 79x1 (BLF?)

 

Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to
have "auto dial and appearance", they have been using the 7940 a week,
along with CCMUser and don't think AbbrDial/FastDials/Personal Directory
meets their needs. Their old phone had lots of buttons, etc.

What is the CallManager 5.1 feature called where a button can be both a
Shared Line and Speed Dial?  I'm trying to find the URL if it requires a
79x1 or would it work with a 7960.

What are the caveats? Can they setup/manage via CCMUser? 

I've been stuck on 4.1 a long time and recall seeing something about
this in 5.x, but can't find the details.  I know 4.1 you could have
either a shared line or a speed dial but not both.

 -jason

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 

 
 
 



________________________________



 
 
 
 
  
 
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cisco-voip at puck.nether.net
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From cisco-voip at ibh.net  Tue Oct  9 05:50:18 2007
From: cisco-voip at ibh.net (Andre Beck)
Date: Tue, 9 Oct 2007 11:50:18 +0200
Subject: [cisco-voip] 5.1 active disk filling?
In-Reply-To: <4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com>
References: <20071008084714.GA26790@ibh.de>
	<4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com>
Message-ID: <20071009095018.GB1620@ibh.de>

Hi Ryan,

On Mon, Oct 08, 2007 at 09:57:33AM -0400, Ryan Ratliff wrote:
> Until you start getting RTMT alerts about the active partition being  
> full don't worry about it.  93% utilization is pretty normal for a CM  
> 5 box.

Ok ;)

Meanwhile we also see occasions where the used space actually dropped
down a bit and the partition with 5.1(3) is currently at 91% only. So
I'm relaxed about that, it just seemed odd.
 
> There were however some bugs that caused the active partition to fill  
> up.  CSCsk18010 is one of them, but only affects IBM servers and is  
> fixed in 5.1(3).

This one is a HP DL380 with 146GB disks and we applied the patch to
bump up to 5.1(3) yesterday. So far all looks Ok, but it did so for
more than a week after bumping to 5.1(2) as well...
 
> I'd encourage you to open a case with TAC to troubleshoot Tomcat (the  
> web server) and your h.323 issue.  They definitely sound like  
> problems but disk utilization is not the cause.

I'll have to open a case should the same situation ever arise again
after the upgrade, as there will be no other potential way out then.
Of course I hope 5.1(3) might have fixed whatever it was ;)

Thanks again,
Andre.
-- 
    .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From cisco-voip at ibh.net  Tue Oct  9 06:03:13 2007
From: cisco-voip at ibh.net (Andre Beck)
Date: Tue, 9 Oct 2007 12:03:13 +0200
Subject: [cisco-voip] 79x0 vs 79x1 (BLF?)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707E811@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00707E811@USNAEXCH.na.didata.local>
Message-ID: <20071009100313.GC1620@ibh.de>

Jason,

On Mon, Oct 08, 2007 at 02:04:35PM -0400, Jason Aarons (US) wrote:
> Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to
> have "auto dial and appearance", they have been using the 7940 a week,
> along with CCMUser and don't think AbbrDial/FastDials/Personal Directory
> meets their needs. Their old phone had lots of buttons, etc.

You can add more (LEDed) buttons to 7960s using a sidecar (7914).
In principle you can do this to a 7940 as well, but AFAIK there is a CCM
blind spot with this combo (no problem with CCME - or was it the other way
around?)

> What is the CallManager 5.1 feature called where a button can be both a
> Shared Line and Speed Dial?

BLF, but it's *not* a shared line. It's a monitor only (and it's good
this way).

>  I'm trying to find the URL if it requires a
> 79x1 or would it work with a 7960.

It works with 7940/60 as well but it is hard to see the feedback as those
models don't have LEDs in their buttons. It's just the symbol in the LCD
that changes for them, and as their LCDs aren't good either...

Go with the x1 models IMO. Better LCD as well, and LEDs.

> What are the caveats? Can they setup/manage via CCMUser? 

AFAIR no - BLFs in 5.1 are CCMAdmin only. But I hope this will change some
day. It's probably rather "not done yet" instead of a privacy feature...

The most significant caveat of all this IMO is that you can monitor a
line, but not a phone. If you monitor a DN that is line 1 on phone A
and phone A has other lines, the user on phone A might be busy talking
on line 2 but the BLF still shows the DN as beeing free. Even with
a busy limit of 1 in all lines. This essentially kills all usage of
secondary lines when you want to have reliable BLF...

HTH,
Andre.
-- 
    .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From cisco-voip at ibh.net  Tue Oct  9 06:45:45 2007
From: cisco-voip at ibh.net (Andre Beck)
Date: Tue, 9 Oct 2007 12:45:45 +0200
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <5d093f9a0710050527u3608da18gdbebca41544fc0de@mail.gmail.com>
References: <5d093f9a0710050527u3608da18gdbebca41544fc0de@mail.gmail.com>
Message-ID: <20071009104545.GD1620@ibh.de>

Hi,

On Fri, Oct 05, 2007 at 07:27:47AM -0500, Jonathan Charles wrote:
> So, I am curious why all of us are putting up with this.

Because there is no choice? 4.x is severly dated and we want BLF and
other features...
 
> For years we were all told that Linux was superior, that Linux was how
> a real web server should function...

And it is. The problem here is, we are not talking about a web server.
 
> Now, we have CCM 5 and CCM 6... and the web server performance, is,
> er, well, how do we put this...
> 
> Here we go. It is far worse than anything Microsoft has ever released.

This is missing the target.

Do you really *know* what the pure web server performance in this setup
is? Or are you just concluding it from the web UI in your browser that
feels like supergiants beeing sucked through nanotubes?

> I think Microsoft BOB would be a better web server than whatever crap
> is on the backend of this RedHat monstrosity.

Bait. But I bite. Yes, there is a monstrosity. No doubt. Seeing how
this rather obsolete OS base (RHEL 3 aka Linux 2.4 - this is as old
as W2k) is flattened by an insanely obese application server is no
fun. But this is not in much way caused by the underlying OS (it might
be in some way - availability of midlayer software Cisco wants to use
like a mostly MSSQL compatible DBMS). The infrastructure on top of the
OS would flatten Windows as well. (BTW, wasn't 6.0 supposed to run on
both OS again? Anyone who can compare to Win2k3?)

And when it comes to the web UI, I actually fear that most of the
sluggishness is happening at the *browser* side, with reliance on
JavaScript and even Java Applet code. Something is popping up these
silly wait splash screens, and it costs more performance than just
plain dumping the pages IMO.
 
> In CallManger 4.3, using Microsoft WINDOWS(!!!!!!)  IIS and Cisco
> Tomcat, I can scream through all of the web pages; do DB reads in
> nanoseconds; do DB inserts almost as quick, and I have never seen a
> message on my web browser saying, 'Loading, Please Wait'

You've never seen this strange Java Proxy Applet that blocks your page
access for seconds? Strange. I cannot scream through 4.x pages either,
with this artificial roadblock in the way.
 
> I mean, seriously, this is godawful slow.

Yes, it is.

I was plainly shocked to see how long it takes for the RHEL stuff
to just boot to the prompt (my Debian servers with half the resources
boot way faster), but *then* to see it take a *full* *other* *nine*
*MINUTES* for the application stuff (DB, CCM et all) to come up. It
is indeed pathetic.

> I just installed CCM5.1.3 on an HP 7845, 4GB of RAM, dual dual core
> Xeons...

Wasting a Gig ;)

> and all I see at every mouse click is 'Loading, Please Wait'..

Yep. Mostly you just wait for the JavaScript code to display this
message beeing downloaded, beeing executed, messing with the CSS to
display an inpage Ad^WSplash, messing again with the CSS to undisplay
it again, and then actually showing the results.

> Is Cisco punishing us for something? Did we offend them in some way? I
> am at a point where I am wondering if I should tell customers about
> these performance issues prior to picking a version ('if you want to
> be able to administer this and not slash your wrists, waiting for the
> home page to load, you may want to go with 4.3...') because I am tired
> of hearing about it afterwards...

I'm telling customers that  each new version bloats badly compared
to the older ones (which, we don't want to forget, were enourmous bloat
in the first place, my first 3.3 CCM on W2k was equipped with at-that-time
plenty 1.5GiB and trying to run CRS flattened it badly - this silly thing
alone allocated 450MiB of RAM).
 
> Seriously guys, I mean, this pretty much kills whatever argument you
> Linux freaks had that this was better than Windows...

It doesn't. Sadly, it kills any welcoming expectations we Linux "freaks"
had for CCM on a free and open platform. It got worse.

> add the whole  SFTP requirement,

It's just an entirely silly decision by *Cisco* to close the platform
into "Appliance mode" instead of allowing people at least the same
freedom of tinkering with the OS that we had on Windows. All these
complications would have been unnecessary. But it was crippled
intentionally, and now I can't even debug problems myself anymore but
have to input convoluted commands at the lobotomized admin "shell".

> and you can see why the Microsoft side won this argument...

Which argument did MS win? Not the one in Europe. SCNR ;)

Andre.
-- 
    .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From jeff at ocjtech.us  Tue Oct  9 08:36:43 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Tue, 09 Oct 2007 07:36:43 -0500
Subject: [cisco-voip] CCM Web Page performance issues, aka 'Loading,
	Please Wait' (a rant)
In-Reply-To: <20071009104545.GD1620@ibh.de>
References: <5d093f9a0710050527u3608da18gdbebca41544fc0de@mail.gmail.com>
	<20071009104545.GD1620@ibh.de>
Message-ID: <1191933403.3234.13.camel@lt21223.campus.dmacc.edu>

On Tue, 2007-10-09 at 12:45 +0200, Andre Beck wrote:
>
> And when it comes to the web UI, I actually fear that most of the
> sluggishness is happening at the *browser* side, with reliance on
> JavaScript and even Java Applet code. Something is popping up these
> silly wait splash screens, and it costs more performance than just
> plain dumping the pages IMO.

I'd have to agree here.  There's a noticeable difference in performance
of the CCM web pages when I switch back and forth between my laptop
(1.8GHz Pentium M) and my desktop (3.2GHz Pentium 4).

> > In CallManger 4.3, using Microsoft WINDOWS(!!!!!!)  IIS and Cisco
> > Tomcat, I can scream through all of the web pages; do DB reads in
> > nanoseconds; do DB inserts almost as quick, and I have never seen a
> > message on my web browser saying, 'Loading, Please Wait'
> 
> You've never seen this strange Java Proxy Applet that blocks your page
> access for seconds? Strange. I cannot scream through 4.x pages either,
> with this artificial roadblock in the way.

Yeah, if I did *anything* in the CCM UI before the proxy applet finished
loading something would crash and I'd have to fully exit IE and restart
it.

> It's just an entirely silly decision by *Cisco* to close the platform
> into "Appliance mode" instead of allowing people at least the same
> freedom of tinkering with the OS that we had on Windows. All these
> complications would have been unnecessary. But it was crippled
> intentionally, and now I can't even debug problems myself anymore but
> have to input convoluted commands at the lobotomized admin "shell".

Cisco has always treated CallManagers as appliances where you weren't
supposed to tinker with the internals.  It's just that with CCM5/Linux
they were able to finally lock things down enough so that end-users
couldn't mess around under the hood.  With CCM4/Windows they
couldn't/didn't lock things down so tightly.  Personally, I don't mind
too terribly - at least now I don't need to reboot on a monthly basis to
install the Microsoft patches du jour.

And from Cisco's perspective I'm sure that it makes TAC's job easier if
end-users aren't fiddling around with the OS in unsupported ways.

Jeff

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From pwalenta at wi.rr.com  Tue Oct  9 08:52:03 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 9 Oct 2007 07:52:03 -0500
Subject: [cisco-voip] QIP anyone?
In-Reply-To: <253144.45095.qm@web57201.mail.re3.yahoo.com>
References: <f84a38d30710050811x7d9ca40er1b7887808f7c816a@mail.gmail.com>
	<253144.45095.qm@web57201.mail.re3.yahoo.com>
Message-ID: <016c01c80a73$315770b0$0232a8c0@walentawk1>

QIP is a DHCP/Dynamic DNS/IP Address Management platform.  I believe Avaya
owns it now.  It's been around for many years. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Friday, October 05, 2007 10:46 AM
To: Scott Voll; Wes Sisk
Cc: cbomba at s4nets.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] QIP anyone?

Since I'm stoopider than Scott, I'd like to know too.

Paul

--- Scott Voll <svoll.voip at gmail.com> wrote:

> excuse my stupid question......... But what is QIP?
> 
> Thanks
> 
> Scott
> 
> 
> On 10/5/07, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > seen a few and it seems to work.  the
> installations i've worked on set
> > some interesting dhcp options like decreasing arp
> timeout which
> > sometimes cause issues.  generally don't set any
> arp timeout or set it
> > hight like 1 hour.  as with all it depends on how
> qip is configured.
> >
> > /Wes
> >
> > Christopher M. Bomba wrote:
> > > Anyone using QIP for DHCP?  I have a client that
> will be implementing
> > Cisco IPT and is using QIP for their DHPC
> application.  Wanted to know if
> > there were any gotchas or anything to look out for
> before we get started.
> > >
> > > Thanks,
> > >
> > > Chris
> > >
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
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From pwalenta at wi.rr.com  Tue Oct  9 08:52:49 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 9 Oct 2007 07:52:49 -0500
Subject: [cisco-voip] CIPTUG
Message-ID: <016d01c80a73$4c658ea0$0232a8c0@walentawk1>

It's almost that time of year again.
 
Anyone planning on being there?
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From Kathy.Tarkington at AGG.com  Tue Oct  9 08:55:49 2007
From: Kathy.Tarkington at AGG.com (Tarkington, Kathy L.)
Date: Tue, 9 Oct 2007 08:55:49 -0400
Subject: [cisco-voip] CIPTUG
Message-ID: <2683B0F497088C489578F16771529BAA036B3CE5@ATLEX01.agg.local>

Wish I could go this year; however my son graduates from basic training at the same time. From the lineup I've seen, looks like it's going to be a great conference!

----- Original Message -----
From: cisco-voip-bounces at puck.nether.net <cisco-voip-bounces at puck.nether.net>
To: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>
Sent: Tue Oct 09 08:52:49 2007
Subject: [cisco-voip] CIPTUG

It's almost that time of year again.
 
Anyone planning on being there?

Note:
This message and any attachments from the law firm Arnall Golden Gregory LLP may contain CONFIDENTIAL and legally protected information.  If you are not the addressee and an intended recipient, please do not read, copy, use or disclose this communication to others; also, please notify the sender by replying to this message, and then delete it from your system.  Thank you.
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From jason.aarons at us.didata.com  Tue Oct  9 09:02:48 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 9 Oct 2007 09:02:48 -0400
Subject: [cisco-voip] 5.1 Speed Dial Presence
In-Reply-To: <470AA006.4080007@cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F00707EA18@USNAEXCH.na.didata.local>
	<470A8C88.30703@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00707EA77@USNAEXCH.na.didata.local>
	<470AA006.4080007@cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00707EEF9@USNAEXCH.na.didata.local>

I setup this DSS/BLF (Speed Dial Presence), she is the receptionist for
a manager, now when you call the manager her phone doesn't ring like a
shared line did.

Should it? I don't see a ring/flash/disable setting for this.

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Monday, October 08, 2007 5:24 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence

 

x1 allows you to see BLF status of calling party in
missed/placed/received directories.  Personally, this is one of my
favorite features.

x1 - will be around much longer and get new features including midlets.
x0 - no more feature development.

/Wes

Jason Aarons (US) wrote: 

Any benefits if they used 79x1 over 7960 for Speed Dial Presence? They
are about to order more 7960s over 7961s due to costs. I wish there was
some motivating  one-reason issue to get customers to buy 79x1s. I know
the benefits of 79x1, and would only buy this model myself, but costs
are sometimes the winner with customers over logic. Can't we end of life
the 7960s and make the 7961 same price!

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Monday, October 08, 2007 4:01 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence

 

native.  no CUPS required. /wes

Jason Aarons (US) wrote: 

I think what they want is DSS/BLF, does it require a dedicated presence
server or can CCM  5.1 do this natively?

Cisco Unified CallManager Speed Dial Presence 

Cisco Unified CallManager supports the ability for a speed dial to have
presence capabilities via a busy lamp field (BLF) speed dial. BLF speed
dials work as both a speed dial and a presence indicator. Only the
system administrator can configure a BLF speed dial; a system user is
not allowed to configure a BLF speed dial. 

States are: 

Idle 

Busy 

Unknown 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementa
tion_design_guide_chapter09186a008070bc02.html#wp1070507
<javascript:newWin('http://www.cisco.com/en/US/products/sw/voicesw/ps556
/products_implementation_design_guide_chapter09186a008070bc02.html#wp107
0507')> 

 

From: Jason Aarons (US) 
Sent: Monday, October 08, 2007 2:05 PM
To: cisco-voip at puck.nether.net
Subject: 79x0 vs 79x1 (BLF?)

 

Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to
have "auto dial and appearance", they have been using the 7940 a week,
along with CCMUser and don't think AbbrDial/FastDials/Personal Directory
meets their needs. Their old phone had lots of buttons, etc.

What is the CallManager 5.1 feature called where a button can be both a
Shared Line and Speed Dial?  I'm trying to find the URL if it requires a
79x1 or would it work with a 7960.

What are the caveats? Can they setup/manage via CCMUser? 

I've been stuck on 4.1 a long time and recall seeing something about
this in 5.x, but can't find the details.  I know 4.1 you could have
either a shared line or a speed dial but not both.

 -jason

 

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From bills at psu.edu  Tue Oct  9 09:24:34 2007
From: bills at psu.edu (Bill Simon)
Date: Tue, 09 Oct 2007 09:24:34 -0400
Subject: [cisco-voip] CIPTUG
In-Reply-To: <016d01c80a73$4c658ea0$0232a8c0@walentawk1>
References: <016d01c80a73$4c658ea0$0232a8c0@walentawk1>
Message-ID: <470B8112.2050900@psu.edu>

I'll be there with four or five others from Penn State University. 
We'll be wearing our blue & white.

Philip Walenta wrote:
> It's almost that time of year again.
>  
> Anyone planning on being there?

From c3voip at nc.rr.com  Tue Oct  9 09:46:28 2007
From: c3voip at nc.rr.com (c3voip)
Date: Tue, 9 Oct 2007 09:46:28 -0400
Subject: [cisco-voip] Hide IP Services from CCMUser page
Message-ID: <000001c80a7a$cae56be0$60b043a0$@rr.com>

 

Is there a way to hide certain IP Services on the CCMUser page to keep users
from being able to add them to their phones?

 

Thanks,

-C

 

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From dustin.s.fowler at gmail.com  Tue Oct  9 10:08:49 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Tue, 9 Oct 2007 10:08:49 -0400
Subject: [cisco-voip] CUCM 6.0 Attendant Consle
Message-ID: <470b8b99.132b400a.56e1.ffff9bba@mx.google.com>

 

All,

 

Has anyone seen an issue with the attendant console reporting:

 

Unable to connect to Server

 

 

It was previously working and stopped working this morning.

 

Dustin

 

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From jason.aarons at us.didata.com  Tue Oct  9 10:39:01 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 9 Oct 2007 10:39:01 -0400
Subject: [cisco-voip] freeFTPd stopping? (sftp server for CCM 5.1 backup)
Message-ID: <C1FE15183DA37645BC0633BC604E44F00707F186@USNAEXCH.na.didata.local>

 freeFTPd runs as a service on a Windows 2000 server, but I notice after
a couple days or a reboot it shows stopped, I hit start, apply and save,
anyone else having issues with freeFTPd or can attest to its value?

 




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From jason.aarons at us.didata.com  Tue Oct  9 10:45:27 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 9 Oct 2007 10:45:27 -0400
Subject: [cisco-voip] Reason why CP7914 doesn't work with CP7940?
Message-ID: <C1FE15183DA37645BC0633BC604E44F00707F1B6@USNAEXCH.na.didata.local>

I take everything Cisco says and repeat it to customers without
question.

 

Customers challenge everything; today I was asked why doesn't the
CP-7914 support the CP-7940? Anyone have a technical answer?

 




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From svoll.voip at gmail.com  Tue Oct  9 11:01:12 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 9 Oct 2007 08:01:12 -0700
Subject: [cisco-voip] CIPTUG
In-Reply-To: <470B8112.2050900@psu.edu>
References: <016d01c80a73$4c658ea0$0232a8c0@walentawk1>
	<470B8112.2050900@psu.edu>
Message-ID: <f84a38d30710090801g451484f5vce75a79eebcc7ffc@mail.gmail.com>

I'll be there.  Anyone seen then final schedule including times for each
workshop?

when is everyone going down?  Going to the SIGs? if so, which ones?  I know
I'll be at the education vertical but I'm not sure about the technical one.

Scott


On 10/9/07, Bill Simon <bills at psu.edu> wrote:
>
> I'll be there with four or five others from Penn State University.
> We'll be wearing our blue & white.
>
> Philip Walenta wrote:
> > It's almost that time of year again.
> >
> > Anyone planning on being there?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Ruben.Montes at eu.didata.com  Tue Oct  9 11:54:57 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Tue, 9 Oct 2007 17:54:57 +0200
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
References: <20071008084714.GA26790@ibh.de><4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com>
	<20071009095018.GB1620@ibh.de>
Message-ID: <FA233791F398B848B978464507739E8FD6BFF1@EUBEBRUSVEX1.eu.didata.local>

Hello,

I'm approximating to the max number of partitions in a CCM 5.x(75
partitions), and I have tried to delete some of them I'm not using. 

Despite the Dependency Records shows nothing, I'm not able to delete
them, because the CCM says that it is being used...

Any idea how can I troubleshoot this?

Thanks in advance,

Ruben

From dpriebe at theriverbank.com  Tue Oct  9 11:31:51 2007
From: dpriebe at theriverbank.com (Duane Priebe)
Date: Tue, 9 Oct 2007 10:31:51 -0500
Subject: [cisco-voip] Shutting down a switch port based on time of day ...
Message-ID: <43D799CBE4331D44999AB7C9CD67F30203AFA05F@NTEXCHANGE.riverbank.com>

Sorry, I know this might be off topic for this group but could anyone
give me info on a way that I could have a switch port (IOS 12.4)
shutdown based on time of day?
 
Duane
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From pwalenta at wi.rr.com  Tue Oct  9 12:18:45 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 9 Oct 2007 11:18:45 -0500
Subject: [cisco-voip] CIPTUG
In-Reply-To: <f84a38d30710090801g451484f5vce75a79eebcc7ffc@mail.gmail.com>
References: <016d01c80a73$4c658ea0$0232a8c0@walentawk1>
	<470B8112.2050900@psu.edu>
	<f84a38d30710090801g451484f5vce75a79eebcc7ffc@mail.gmail.com>
Message-ID: <01d201c80a90$1146cc40$0232a8c0@walentawk1>

I'll be there from Sunday-Thursday night.  I'll be at the Polycom booth most
of the time.

  _____  

From: Scott Voll [mailto:svoll.voip at gmail.com] 
Sent: Tuesday, October 09, 2007 10:01 AM
To: Bill Simon
Cc: Philip Walenta; Cisco Voip
Subject: Re: [cisco-voip] CIPTUG


I'll be there.  Anyone seen then final schedule including times for each
workshop?
 
when is everyone going down?  Going to the SIGs? if so, which ones?  I know
I'll be at the education vertical but I'm not sure about the technical one.
 
Scott

 
On 10/9/07, Bill Simon <bills at psu.edu> wrote: 

I'll be there with four or five others from Penn State University.
We'll be wearing our blue & white. 

Philip Walenta wrote:
> It's almost that time of year again.
>
> Anyone planning on being there?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



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From brian.gibson at stifel.com  Tue Oct  9 12:19:27 2007
From: brian.gibson at stifel.com (Gibson, Brian (Jersey City))
Date: Tue, 9 Oct 2007 11:19:27 -0500
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
In-Reply-To: <FA233791F398B848B978464507739E8FD6BFF1@EUBEBRUSVEX1.eu.didata.local>
References: <20071008084714.GA26790@ibh.de><4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com><20071009095018.GB1620@ibh.de>
	<FA233791F398B848B978464507739E8FD6BFF1@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <5CC6FF682D32164BA5F2E830ABDF146001A5AC8F@SNEXCHANGE02.stifelnet.stifel.local>

Where did you read that you can only have 75 partitions?  We have nearly
300.  

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruben Montes
(Europe)
Sent: Tuesday, October 09, 2007 11:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty

Hello,

I'm approximating to the max number of partitions in a CCM 5.x(75
partitions), and I have tried to delete some of them I'm not using. 

Despite the Dependency Records shows nothing, I'm not able to delete
them, because the CCM says that it is being used...

Any idea how can I troubleshoot this?

Thanks in advance,

Ruben
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

********************************************************************************
All electronic messages sent and received by Stifel Nicolaus
Associates are subject to review by Stifel Nicolaus.  Stifel Nicolaus
may retain and reproduce electronic messages for state, federal, or
other regulatory agencies as required by applicable law.  
IMPORTANT: Please do not use e-mail to request or authorize the
purchase or sale of any security or commodity, send fund transfer
instructions, or otherwise conduct any securities transactions.  Any 
requests, orders, instructions, or time-sensitive messages sent by
e-mail cannot be accepted or processed by Stifel Nicolaus.  The 
accuracy of any information sent by Stifel Nicolaus through e-mail 
cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
 
Stifel, Nicolaus & Company, Incorporated
Member NYSE & SIPC
Headquarters:  501 N. Broadway, St. Louis, MO   63102
314-342-2000
********************************************************************************


From james.dust at charles-stanley.co.uk  Tue Oct  9 12:24:23 2007
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Tue, 9 Oct 2007 17:24:23 +0100
Subject: [cisco-voip] Translation Pattern/Calling Party Transform Mask
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE02B202F3@CSEXCH01.charles-stanley.co.uk>

I have a problem where the '0' is not being sent on inbound calls.

 

What I would like to do is somehow add a '9' and a '0' so that when a
user gets a missed call in their missed call list it will show '9' then
'0 xxx xxx xxxx'

 

Can I use the Calling Party Transform Mask within the Translation
Pattern to accomplish this?

 

Thanks

 

James

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From Ruben.Montes at eu.didata.com  Tue Oct  9 12:25:01 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Tue, 9 Oct 2007 18:25:01 +0200
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
References: <20071008084714.GA26790@ibh.de><4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com><20071009095018.GB1620@ibh.de>
	<FA233791F398B848B978464507739E8FD6BFF1@EUBEBRUSVEX1.eu.didata.local>
	<5CC6FF682D32164BA5F2E830ABDF146001A5AC8F@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <FA233791F398B848B978464507739E8FD6C01C@EUBEBRUSVEX1.eu.didata.local>

Hi,

When you create a new Partition, if you read the lines above, it says
you can only create 75. Maybe am I wrong? I'm running CCM 5.1(2)

Regards,

Ruben

-----Mensaje original-----
De: Gibson, Brian (Jersey City) [mailto:brian.gibson at stifel.com] 
Enviado el: martes, 09 de octubre de 2007 18:19
Para: Ruben Montes (Europe); cisco-voip at puck.nether.net
Asunto: RE: [cisco-voip] Cannot delete Partition, Dependency Records
empty

Where did you read that you can only have 75 partitions?  We have nearly
300.  

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruben Montes
(Europe)
Sent: Tuesday, October 09, 2007 11:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty

Hello,

I'm approximating to the max number of partitions in a CCM 5.x(75
partitions), and I have tried to delete some of them I'm not using. 

Despite the Dependency Records shows nothing, I'm not able to delete
them, because the CCM says that it is being used...

Any idea how can I troubleshoot this?

Thanks in advance,

Ruben
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

************************************************************************
********
All electronic messages sent and received by Stifel Nicolaus
Associates are subject to review by Stifel Nicolaus.  Stifel Nicolaus
may retain and reproduce electronic messages for state, federal, or
other regulatory agencies as required by applicable law.  
IMPORTANT: Please do not use e-mail to request or authorize the
purchase or sale of any security or commodity, send fund transfer
instructions, or otherwise conduct any securities transactions.  Any 
requests, orders, instructions, or time-sensitive messages sent by
e-mail cannot be accepted or processed by Stifel Nicolaus.  The 
accuracy of any information sent by Stifel Nicolaus through e-mail 
cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
 
Stifel, Nicolaus & Company, Incorporated
Member NYSE & SIPC
Headquarters:  501 N. Broadway, St. Louis, MO   63102
314-342-2000
************************************************************************
********


From jeff at ocjtech.us  Tue Oct  9 12:30:53 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Tue, 09 Oct 2007 11:30:53 -0500
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
In-Reply-To: <FA233791F398B848B978464507739E8FD6C01C@EUBEBRUSVEX1.eu.didata.local>
References: <20071008084714.GA26790@ibh.de>
	<4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com>
	<20071009095018.GB1620@ibh.de>
	<FA233791F398B848B978464507739E8FD6BFF1@EUBEBRUSVEX1.eu.didata.local>
	<5CC6FF682D32164BA5F2E830ABDF146001A5AC8F@SNEXCHANGE02.stifelnet.stifel.local>
	<FA233791F398B848B978464507739E8FD6C01C@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <1191947453.3234.17.camel@lt21223.campus.dmacc.edu>

On Tue, 2007-10-09 at 18:25 +0200, Ruben Montes (Europe) wrote:
> 
> When you create a new Partition, if you read the lines above, it says
> you can only create 75. Maybe am I wrong? I'm running CCM 5.1(2)

I believe that message refers to the fact that you can only create 75
partitions *at a time* using the web interface.

Jeff

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From Ruben.Montes at eu.didata.com  Tue Oct  9 12:33:00 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Tue, 9 Oct 2007 18:33:00 +0200
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
References: <20071008084714.GA26790@ibh.de>
	<4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com>
	<20071009095018.GB1620@ibh.de>
	<FA233791F398B848B978464507739E8FD6BFF1@EUBEBRUSVEX1.eu.didata.local>
	<5CC6FF682D32164BA5F2E830ABDF146001A5AC8F@SNEXCHANGE02.stifelnet.stifel.local>
	<FA233791F398B848B978464507739E8FD6C01C@EUBEBRUSVEX1.eu.didata.local>
	<1191947453.3234.17.camel@lt21223.campus.dmacc.edu>
Message-ID: <FA233791F398B848B978464507739E8FD6C024@EUBEBRUSVEX1.eu.didata.local>

Hi,

Fine then, anyway, I'm still having the same problem because I have some
'locked' partitions...

Thanks,

Ruben

-----Mensaje original-----
De: Jeffrey C. Ollie [mailto:jeff at ocjtech.us] 
Enviado el: martes, 09 de octubre de 2007 18:31
Para: Ruben Montes (Europe)
CC: Gibson, Brian (Jersey City); cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Cannot delete Partition, Dependency Records
empty

On Tue, 2007-10-09 at 18:25 +0200, Ruben Montes (Europe) wrote:
> 
> When you create a new Partition, if you read the lines above, it says
> you can only create 75. Maybe am I wrong? I'm running CCM 5.1(2)

I believe that message refers to the fact that you can only create 75
partitions *at a time* using the web interface.

Jeff


From Edward.Countryman at provena.org  Tue Oct  9 12:39:38 2007
From: Edward.Countryman at provena.org (Countryman, Edward)
Date: Tue, 9 Oct 2007 11:39:38 -0500
Subject: [cisco-voip] Help with Paging interface
In-Reply-To: <1191933403.3234.13.camel@lt21223.campus.dmacc.edu>
Message-ID: <898237D7FE163A40A3257E385562F27E0160B34F@AEXMV02.phnet.phroot.local>


I've installed an FXO port in my router and believe I have it configured
correctly in CCM but cannot get it to open when connected to a VALCOM
V2000A.  

When dialing the route pattern for this port; it appears to sit for
several seconds then go fast busy.  It is almost like it is waiting for
something from the valcom (dial tone detect?)

I presently have the battery feed switch on the valcom set to ON. 

Any ideas?

From asobihoudai at yahoo.com  Tue Oct  9 13:05:45 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 9 Oct 2007 10:05:45 -0700 (PDT)
Subject: [cisco-voip] CIPTUG
In-Reply-To: <2683B0F497088C489578F16771529BAA036B3CE5@ATLEX01.agg.local>
Message-ID: <784076.2872.qm@web57214.mail.re3.yahoo.com>

If you can, please take notes and show us them later
since I can't attend this year.

Paul
--- "Tarkington, Kathy L." <Kathy.Tarkington at AGG.com>
wrote:

> Wish I could go this year; however my son graduates
> from basic training at the same time. From the
> lineup I've seen, looks like it's going to be a
> great conference!
> 
> ----- Original Message -----
> From: cisco-voip-bounces at puck.nether.net
> <cisco-voip-bounces at puck.nether.net>
> To: cisco-voip at puck.nether.net
> <cisco-voip at puck.nether.net>
> Sent: Tue Oct 09 08:52:49 2007
> Subject: [cisco-voip] CIPTUG
> 
> It's almost that time of year again.
>  
> Anyone planning on being there?
> 
> Note:
> This message and any attachments from the law firm
> Arnall Golden Gregory LLP may contain CONFIDENTIAL
> and legally protected information.  If you are not
> the addressee and an intended recipient, please do
> not read, copy, use or disclose this communication
> to others; also, please notify the sender by
> replying to this message, and then delete it from
> your system.  Thank you.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



      ____________________________________________________________________________________
Check out the hottest 2008 models today at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html

From asobihoudai at yahoo.com  Tue Oct  9 13:10:55 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 9 Oct 2007 10:10:55 -0700 (PDT)
Subject: [cisco-voip] Reason why CP7914 doesn't work with CP7940?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707F1B6@USNAEXCH.na.didata.local>
Message-ID: <399548.34168.qm@web57212.mail.re3.yahoo.com>

Tell them it wasn't designed to work with anything
less than a 7960.

Paul
--- "Jason Aarons (US)" <jason.aarons at us.didata.com>
wrote:

> I take everything Cisco says and repeat it to
> customers without
> question.
> 
>  
> 
> Customers challenge everything; today I was asked
> why doesn't the
> CP-7914 support the CP-7940? Anyone have a technical
> answer?
> 
>  
> 
> 
> 
> 
> -----------------------------------------
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> 
> This e-mail communication and any attachments may
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_______________________________________________
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From greencucumber2007 at yahoo.com  Tue Oct  9 13:12:40 2007
From: greencucumber2007 at yahoo.com (Cucumber Green)
Date: Tue, 9 Oct 2007 10:12:40 -0700 (PDT)
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
In-Reply-To: <FA233791F398B848B978464507739E8FD6C024@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <464405.10379.qm@web44803.mail.sp1.yahoo.com>

I think there is an actual limit of the number of partitions, the number depends on the names you use for the partitions, but its surely larger than 75.
   
  I experienced this partition deletion problem ( but in CCM 4.2), if I remember correctly, it was a DN not being used but belonging to the partition, I went into RoutePlan->RoutePlanReport looked for the DN and deleted it.

"Ruben Montes (Europe)" <Ruben.Montes at eu.didata.com> wrote:
  Hi,

Fine then, anyway, I'm still having the same problem because I have some
'locked' partitions...

Thanks,

Ruben

-----Mensaje original-----
De: Jeffrey C. Ollie [mailto:jeff at ocjtech.us] 
Enviado el: martes, 09 de octubre de 2007 18:31
Para: Ruben Montes (Europe)
CC: Gibson, Brian (Jersey City); cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Cannot delete Partition, Dependency Records
empty

On Tue, 2007-10-09 at 18:25 +0200, Ruben Montes (Europe) wrote:
> 
> When you create a new Partition, if you read the lines above, it says
> you can only create 75. Maybe am I wrong? I'm running CCM 5.1(2)

I believe that message refers to the fact that you can only create 75
partitions *at a time* using the web interface.

Jeff

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


       
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From ealeatherman at gmail.com  Tue Oct  9 13:12:59 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 9 Oct 2007 13:12:59 -0400
Subject: [cisco-voip] CIPTUG
In-Reply-To: <016d01c80a73$4c658ea0$0232a8c0@walentawk1>
References: <016d01c80a73$4c658ea0$0232a8c0@walentawk1>
Message-ID: <94a1afde0710091012t17bf8fa2r48fb882ff4fe714@mail.gmail.com>

Wish I could make it. I'm not personally going but we're sending two others.
One of them is doing a break-out session on end-user training actually.

On 10/9/07, Philip Walenta <pwalenta at wi.rr.com> wrote:
>
>  It's almost that time of year again.
>
> Anyone planning on being there?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From pwalenta at wi.rr.com  Tue Oct  9 13:22:58 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 9 Oct 2007 12:22:58 -0500
Subject: [cisco-voip] CIPTUG
In-Reply-To: <f84a38d30710090926l2d184e32p667b4b87c84ce200@mail.gmail.com>
References: <016d01c80a73$4c658ea0$0232a8c0@walentawk1>
	<470B8112.2050900@psu.edu>
	<f84a38d30710090801g451484f5vce75a79eebcc7ffc@mail.gmail.com>
	<01d201c80a90$1146cc40$0232a8c0@walentawk1>
	<f84a38d30710090926l2d184e32p667b4b87c84ce200@mail.gmail.com>
Message-ID: <01f801c80a99$09b1cb20$0232a8c0@walentawk1>

Yes, I work for Polycom as a Consulting SE and in product development. 
 
I wrote a phone app for them that controls video endpoints from 7970's.
You'll see it at the show.
 
I did some Informacast support when at Berbee but not much.

  _____  

From: Scott Voll [mailto:svoll.voip at gmail.com] 
Sent: Tuesday, October 09, 2007 11:27 AM
To: Philip Walenta
Subject: Re: [cisco-voip] CIPTUG


so are you working for Polycom now?  If so, what are you doing? sales?  You
used to support InformaCast right?
 
Scott

 
On 10/9/07, Philip Walenta <pwalenta at wi.rr.com> wrote: 

I'll be there from Sunday-Thursday night.  I'll be at the Polycom booth most
of the time.

  _____  

From: Scott Voll [mailto:svoll.voip at gmail.com] 
Sent: Tuesday, October 09, 2007 10:01 AM 
To: Bill Simon
Cc: Philip Walenta; Cisco Voip
Subject: Re: [cisco-voip] CIPTUG

 

I'll be there.  Anyone seen then final schedule including times for each
workshop?
 
when is everyone going down?  Going to the SIGs? if so, which ones?  I know
I'll be at the education vertical but I'm not sure about the technical one.
 
Scott

 
On 10/9/07, Bill Simon <bills at psu.edu> wrote: 

I'll be there with four or five others from Penn State University.
We'll be wearing our blue & white. 

Philip Walenta wrote:
> It's almost that time of year again.
>
> Anyone planning on being there?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 




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From asobihoudai at yahoo.com  Tue Oct  9 13:30:47 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 9 Oct 2007 10:30:47 -0700 (PDT)
Subject: [cisco-voip] CCME CNFA timeout reset bug
In-Reply-To: <01f801c80a99$09b1cb20$0232a8c0@walentawk1>
Message-ID: <657059.19989.qm@web57215.mail.re3.yahoo.com>

Has anyone else experienced a CCME bug where the CNFA
timer is automatically reset to 10s? I have had this
happen even with ephone-dn-templates applied! This is
very irritating and I'd like to know if I'm the only
one who has had this happen.

Paul


       
____________________________________________________________________________________
Yahoo! oneSearch: Finally, mobile search 
that gives answers, not web links. 
http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC

From svoll.voip at gmail.com  Tue Oct  9 13:32:06 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 9 Oct 2007 10:32:06 -0700
Subject: [cisco-voip] Shutting down a switch port based on time of day
	...
In-Reply-To: <43D799CBE4331D44999AB7C9CD67F30203AFA05F@NTEXCHANGE.riverbank.com>
References: <43D799CBE4331D44999AB7C9CD67F30203AFA05F@NTEXCHANGE.riverbank.com>
Message-ID: <f84a38d30710091032l5ce5ab3du1c31c68cc42236d@mail.gmail.com>

Kiwi Cat tools could be used to script it to do that ;-)

scott


On 10/9/07, Duane Priebe <dpriebe at theriverbank.com> wrote:
>
>  Sorry, I know this might be off topic for this group but could anyone
> give me info on a way that I could have a switch port (IOS 12.4) shutdown
> based on time of day?
>
> Duane
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From pwalenta at wi.rr.com  Tue Oct  9 13:36:45 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 9 Oct 2007 12:36:45 -0500
Subject: [cisco-voip] CCME CNFA timeout reset bug
In-Reply-To: <657059.19989.qm@web57215.mail.re3.yahoo.com>
References: <01f801c80a99$09b1cb20$0232a8c0@walentawk1>
	<657059.19989.qm@web57215.mail.re3.yahoo.com>
Message-ID: <020601c80a9a$f6d6e920$0232a8c0@walentawk1>

Are you using the web admin tool, or setting it via CLI? 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Tuesday, October 09, 2007 12:31 PM
To: 'Cisco Voip'
Subject: [cisco-voip] CCME CNFA timeout reset bug

Has anyone else experienced a CCME bug where the CNFA timer is automatically
reset to 10s? I have had this happen even with ephone-dn-templates applied!
This is very irritating and I'd like to know if I'm the only one who has had
this happen.

Paul


       
____________________________________________________________________________
________
Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. 
http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jason.aarons at us.didata.com  Tue Oct  9 13:45:51 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 9 Oct 2007 13:45:51 -0400
Subject: [cisco-voip] Unity Connection >System Settings> Message Store Quota
	12000000 Bytes
Message-ID: <C1FE15183DA37645BC0633BC604E44F00707F545@USNAEXCH.na.didata.local>

Warning Quote defaults to 12000000 Bytes

 

How many minutes is this for g711 messages? For a MCS-7825-H3 with 500
users I'm wondering how much higher I should increase it, users are
complaining it's too small.

 




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From asobihoudai at yahoo.com  Tue Oct  9 13:48:24 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 9 Oct 2007 10:48:24 -0700 (PDT)
Subject: [cisco-voip] CCME CNFA timeout reset bug
In-Reply-To: <020601c80a9a$f6d6e920$0232a8c0@walentawk1>
Message-ID: <836902.40741.qm@web57207.mail.re3.yahoo.com>

CLI

--- Philip Walenta <pwalenta at wi.rr.com> wrote:

> Are you using the web admin tool, or setting it via
> CLI? 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Paul Choi
> Sent: Tuesday, October 09, 2007 12:31 PM
> To: 'Cisco Voip'
> Subject: [cisco-voip] CCME CNFA timeout reset bug
> 
> Has anyone else experienced a CCME bug where the
> CNFA timer is automatically
> reset to 10s? I have had this happen even with
> ephone-dn-templates applied!
> This is very irritating and I'd like to know if I'm
> the only one who has had
> this happen.
> 
> Paul
> 
> 
>        
>
____________________________________________________________________________
> ________
> Yahoo! oneSearch: Finally, mobile search that gives
> answers, not web links. 
>
http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 



      ____________________________________________________________________________________
Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more!
http://tv.yahoo.com/collections/3658 

From bills at psu.edu  Tue Oct  9 13:52:56 2007
From: bills at psu.edu (Bill Simon)
Date: Tue, 09 Oct 2007 13:52:56 -0400
Subject: [cisco-voip] Unity Connection >System Settings> Message Store
 Quota	12000000 Bytes
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707F545@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00707F545@USNAEXCH.na.didata.local>
Message-ID: <470BBFF8.5090509@psu.edu>

I estimate 10KB / sec needed for storage.  That's 64 Kbit/sec (8 
KByte/sec) for the G711 codec and another 2KB/sec to be generous for 
overhead and filesystem rounding-up.

12000000 Bytes seems to come out to 1172 seconds, or 19.5 minutes.

Jason Aarons (US) wrote:
> Warning Quote defaults to 12000000 Bytes
> 
>  
> 
> How many minutes is this for g711 messages? For a MCS-7825-H3 with 500 
> users I?m wondering how much higher I should increase it, users are 
> complaining it?s too small.

From toddnh65 at gmail.com  Tue Oct  9 14:02:25 2007
From: toddnh65 at gmail.com (Todd Franklin)
Date: Tue, 9 Oct 2007 13:02:25 -0500
Subject: [cisco-voip] MeetMe
Message-ID: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>

I need to setup a conference call.  We will have 5 or 6 people calling in
from outside.
So I decided maybe MeetMe was the best way.  I setup some MeetMe numbers.
One is 4099.
But when someone dials in from the outside they get "I don't recognize that
as a valid extension".
Am I misusing MeetMe??
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From svoll.voip at gmail.com  Tue Oct  9 14:03:12 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 9 Oct 2007 11:03:12 -0700
Subject: [cisco-voip] CIPTUG
In-Reply-To: <784076.2872.qm@web57214.mail.re3.yahoo.com>
References: <2683B0F497088C489578F16771529BAA036B3CE5@ATLEX01.agg.local>
	<784076.2872.qm@web57214.mail.re3.yahoo.com>
Message-ID: <f84a38d30710091103o24b0fd81g747891ad4ff9ab32@mail.gmail.com>

They are going Green this year.  they are putting all presos on a usb key
for each attendee.  ( I only know this because all presos had to be in last
month)

Scott


On 10/9/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> If you can, please take notes and show us them later
> since I can't attend this year.
>
> Paul
> --- "Tarkington, Kathy L." <Kathy.Tarkington at AGG.com>
> wrote:
>
> > Wish I could go this year; however my son graduates
> > from basic training at the same time. From the
> > lineup I've seen, looks like it's going to be a
> > great conference!
> >
> > ----- Original Message -----
> > From: cisco-voip-bounces at puck.nether.net
> > <cisco-voip-bounces at puck.nether.net>
> > To: cisco-voip at puck.nether.net
> > <cisco-voip at puck.nether.net>
> > Sent: Tue Oct 09 08:52:49 2007
> > Subject: [cisco-voip] CIPTUG
> >
> > It's almost that time of year again.
> >
> > Anyone planning on being there?
> >
> > Note:
> > This message and any attachments from the law firm
> > Arnall Golden Gregory LLP may contain CONFIDENTIAL
> > and legally protected information.  If you are not
> > the addressee and an intended recipient, please do
> > not read, copy, use or disclose this communication
> > to others; also, please notify the sender by
> > replying to this message, and then delete it from
> > your system.  Thank you.
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ____________________________________________________________________________________
> Check out the hottest 2008 models today at Yahoo! Autos.
> http://autos.yahoo.com/new_cars.html
> _______________________________________________
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From svoll.voip at gmail.com  Tue Oct  9 14:07:49 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 9 Oct 2007 11:07:49 -0700
Subject: [cisco-voip] MeetMe
In-Reply-To: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>
References: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>
Message-ID: <f84a38d30710091107p45ef35d2g90174994a625f623@mail.gmail.com>

the MeetMe conference has to be started from an IP phone.

IE>  IP Phone --> press Meetme and then 4099

then outside callers dial XXX-XXX-4099 (so long as this is a DID) and then
everyone is conference together.

if ouside caller dials first then they get that error message.

Scott


On 10/9/07, Todd Franklin <toddnh65 at gmail.com> wrote:
>
> I need to setup a conference call.  We will have 5 or 6 people calling in
> from outside.
> So I decided maybe MeetMe was the best way.  I setup some MeetMe numbers.
> One is 4099.
> But when someone dials in from the outside they get "I don't recognize
> that as a valid extension".
> Am I misusing MeetMe??
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Brandon.Ogden at amersports.com  Tue Oct  9 14:10:41 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Tue, 9 Oct 2007 11:10:41 -0700
Subject: [cisco-voip] MeetMe
In-Reply-To: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00C72EDD@seas7633.am.amersports.int>

Setup a translation to point an external extension to your meet me
extension.  

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Tuesday, October 09, 2007 12:02 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MeetMe

 

I need to setup a conference call.  We will have 5 or 6 people calling
in from outside.
So I decided maybe MeetMe was the best way.  I setup some MeetMe
numbers.  One is 4099.
But when someone dials in from the outside they get "I don't recognize
that as a valid extension". 
Am I misusing MeetMe??




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From SCASPER at mtb.com  Tue Oct  9 14:20:14 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Tue, 09 Oct 2007 14:20:14 -0400
Subject: [cisco-voip]  Intercom History
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
Message-ID: <470B8E1E.9874.0039.0@mtb.com>

Just upgraded to 4.1.3sr5d -  users now see Intercom History under their Directory button on 7941 and 7961 telephones. In fact, this is now above the Corporate Directory so these users now have to scroll down to get to the Corporate Directory. This of course irritates most users and they want Intercom History removed.
 
What is this for, is it functional? I thought intercom was a Rls 6.x feature. 
 
 
 
Thanks,

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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From nick.jon.griffin at gmail.com  Tue Oct  9 14:33:52 2007
From: nick.jon.griffin at gmail.com (Nick Griffin)
Date: Tue, 9 Oct 2007 14:33:52 -0400
Subject: [cisco-voip] CCM 4.X and Bogen UTI1 Integration
Message-ID: <a2b352c40710091133t2d016617k97f7de76a8816cd6@mail.gmail.com>

I've got paging setup using the station port on a UTI1 successfully. The
next task is to have calls to a main line ring overhead. I thought I could
perhaps use the Night Ring Leads on the unit to accomplish this. I've tried
this with an FXS port configured for Ground Start. I am able to call and
hear the night ring, the problem is that at this point the call is connected
and the caller will hear dead air as the night bell continues to chime :)
Perhaps I just need a different type of port, and that I am not sure of. The
Documentation for the Bogen mentions contact closure, I'm not sure how to
accomplish that:

The UTI1 night ringer signaling feature is designed to alert personnel to
incoming
calls after normal business hours.The feature is activated by a contact
closure from
the PBX.
The night ringer normally sounds a simulated ring tone, but can be
programmed
to sound a chime tone.
To physically connect the night ringer wiring:
     Make sure that the power is disconnected.
1.
2.   Wire the contact closure used for night ring to the NTR (+5V) and C
     terminals on the UTI1. Maximum contact resistance for contact closure
     activation is 1000 ohms. Open collector type outputs for controlling a
page
     may also be used.

Anyone have any suggestions?

TIA,

Nick Griffin
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From keith.klevenski at rig.net  Tue Oct  9 14:14:29 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Tue, 9 Oct 2007 13:14:29 -0500
Subject: [cisco-voip] MeetMe
In-Reply-To: <7AC408DB7A31614EA81CD71F351097312E3F7C83@HOUEXH02.rignet.org>
References: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>
	<7AC408DB7A31614EA81CD71F351097312E3F7C83@HOUEXH02.rignet.org>
Message-ID: <E0F48CE151B5D440B3649903D7221DA263906D@HOUEXH02.rignet.org>

This was assuming as Scott just said that you have started the MeetMe
from an IP phone and the call was still failing from the outside...
basically making sure this extension was able to be called from the
PSTN.  

 

Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077         

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net <http://www.rig.net/> 

 

"Extending the power of your knowledge"

 

From: Keith Klevenski 
Sent: Tuesday, October 09, 2007 1:09 PM
To: Todd Franklin; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MeetMe

 

Can you put that extension on a phone and call it?

 

Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077         

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net <http://www.rig.net/> 

 

"Extending the power of your knowledge"

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Tuesday, October 09, 2007 1:02 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MeetMe

 

I need to setup a conference call.  We will have 5 or 6 people calling
in from outside.
So I decided maybe MeetMe was the best way.  I setup some MeetMe
numbers.  One is 4099.
But when someone dials in from the outside they get "I don't recognize
that as a valid extension". 
Am I misusing MeetMe??



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From keith.klevenski at rig.net  Tue Oct  9 14:08:35 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Tue, 9 Oct 2007 13:08:35 -0500
Subject: [cisco-voip] MeetMe
In-Reply-To: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>
References: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>
Message-ID: <E0F48CE151B5D440B3649903D7221DA263906C@HOUEXH02.rignet.org>

Can you put that extension on a phone and call it?

 

Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077         

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net <http://www.rig.net/> 

 

"Extending the power of your knowledge"

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Tuesday, October 09, 2007 1:02 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MeetMe

 

I need to setup a conference call.  We will have 5 or 6 people calling
in from outside.
So I decided maybe MeetMe was the best way.  I setup some MeetMe
numbers.  One is 4099.
But when someone dials in from the outside they get "I don't recognize
that as a valid extension". 
Am I misusing MeetMe??




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From cisco1.charles at gmail.com  Tue Oct  9 14:43:05 2007
From: cisco1.charles at gmail.com (charles george)
Date: Tue, 9 Oct 2007 14:43:05 -0400
Subject: [cisco-voip] SQL database remove DC
Message-ID: <9552cdf0710091143s651ee1a3ga4e7768d7d8745a9@mail.gmail.com>

how would I use SQL so that I could remove the DC that Unity was pointing to
when the DCGC tool will not work.
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From wsisk at cisco.com  Tue Oct  9 15:11:05 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 09 Oct 2007 15:11:05 -0400
Subject: [cisco-voip] Hide IP Services from CCMUser page
In-Reply-To: <000001c80a7a$cae56be0$60b043a0$@rr.com>
References: <000001c80a7a$cae56be0$60b043a0$@rr.com>
Message-ID: <470BD249.9010605@cisco.com>

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From wsisk at cisco.com  Tue Oct  9 15:12:51 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 09 Oct 2007 15:12:51 -0400
Subject: [cisco-voip] 5.1 Speed Dial Presence
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
Message-ID: <470BD2B3.5000808@cisco.com>

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From wsisk at cisco.com  Tue Oct  9 15:13:42 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 09 Oct 2007 15:13:42 -0400
Subject: [cisco-voip] 5.1 Speed Dial Presence
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707EEF9@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00707EA18@USNAEXCH.na.didata.local>
	<470A8C88.30703@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00707EA77@USNAEXCH.na.didata.local>
	<470AA006.4080007@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00707EEF9@USNAEXCH.na.didata.local>
Message-ID: <470BD2E6.8070206@cisco.com>

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From wsisk at cisco.com  Tue Oct  9 15:15:35 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 09 Oct 2007 15:15:35 -0400
Subject: [cisco-voip] Intercom History
In-Reply-To: <470B8E1E.9874.0039.0@mtb.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
	<470B8E1E.9874.0039.0@mtb.com>
Message-ID: <470BD357.9040404@cisco.com>

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From Ruben.Montes at eu.didata.com  Tue Oct  9 15:19:32 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Tue, 9 Oct 2007 21:19:32 +0200
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
References: <464405.10379.qm@web44803.mail.sp1.yahoo.com>
Message-ID: <FA233791F398B848B978464507739E8F3E43C3@EUBEBRUSVEX1.eu.didata.local>

Hi,
 
in the unassigned dn report it doesn't either appear...

________________________________

De: cisco-voip-bounces at puck.nether.net en nombre de Cucumber Green
Enviado el: mar 09/10/2007 19:12
Para: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Cannot delete Partition, Dependency Records empty


I think there is an actual limit of the number of partitions, the number depends on the names you use for the partitions, but its surely larger than 75.
 
I experienced this partition deletion problem ( but in CCM 4.2), if I remember correctly, it was a DN not being used but belonging to the partition, I went into RoutePlan->RoutePlanReport looked for the DN and deleted it.

"Ruben Montes (Europe)" <Ruben.Montes at eu.didata.com> wrote:

	Hi,
	
	Fine then, anyway, I'm still having the same problem because I have some
	'locked' partitions...
	
	Thanks,
	
	Ruben
	
	-----Mensaje original-----
	De: Jeffrey C. Ollie [mailto:jeff at ocjtech.us] 
	Enviado el: martes, 09 de octubre de 2007 18:31
	Para: Ruben Montes (Europe)
	CC: Gibson, Brian (Jersey City); cisco-voip at puck.nether.net
	Asunto: Re: [cisco-voip] Cannot delete Partition, Dependency Records
	empty
	
	On Tue, 2007-10-09 at 18:25 +0200, Ruben Montes (Europe) wrote:
	> 
	> When you create a new Partition, if you read the lines above, it says
	> you can only create 75. Maybe am I wrong? I'm running CCM 5.1(2)
	
	I believe that message refers to the fact that you can only create 75
	partitions *at a time* using the web interface.
	
	Jeff
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	


________________________________

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From wsisk at cisco.com  Tue Oct  9 15:34:28 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 09 Oct 2007 15:34:28 -0400
Subject: [cisco-voip] 5.1 active disk filling?
In-Reply-To: <20071009095018.GB1620@ibh.de>
References: <20071008084714.GA26790@ibh.de>	<4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com>
	<20071009095018.GB1620@ibh.de>
Message-ID: <470BD7C4.3050907@cisco.com>

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From mjdsy9 at gmail.com  Tue Oct  9 15:42:59 2007
From: mjdsy9 at gmail.com (Michael Sy)
Date: Tue, 9 Oct 2007 12:42:59 -0700
Subject: [cisco-voip] background image
Message-ID: <228dc9bd0710091242w790b1eeaqd0afff9dc531be67@mail.gmail.com>

assistance needed pls in creating and uploading the background image for
7940 and 7970 phones..ccm 4.2..thanks in advance

-- 
Mike
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From Scott.Voll at wesd.org  Tue Oct  9 15:53:49 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 9 Oct 2007 12:53:49 -0700
Subject: [cisco-voip]  5.1 active disk filling?
Message-ID: <407055A92CECCB499C922A2D35FC19A6099EDE12@Apollo.wesd.org>

CSCsk61129 MOH file upload fails due to lack of disk space
workaround: use smaller MoH files

 

That's nice........ we don't care that you have recorded this great
audio ad for your business...... CHOP IT DOWN ;-)

 

Scott

 

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From brian.gibson at stifel.com  Tue Oct  9 14:53:24 2007
From: brian.gibson at stifel.com (Gibson, Brian (Jersey City))
Date: Tue, 9 Oct 2007 13:53:24 -0500
Subject: [cisco-voip] MeetMe
In-Reply-To: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>
References: <6101bda50710091102u2f2b1d54p4e46fcb9dd010a2e@mail.gmail.com>
Message-ID: <5CC6FF682D32164BA5F2E830ABDF146001A5AD28@SNEXCHANGE02.stifelnet.stifel.local>

Is 4099 a valid PSTN DN?  

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Tuesday, October 09, 2007 2:02 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MeetMe

 

I need to setup a conference call.  We will have 5 or 6 people calling
in from outside.
So I decided maybe MeetMe was the best way.  I setup some MeetMe
numbers.  One is 4099.
But when someone dials in from the outside they get "I don't recognize
that as a valid extension". 
Am I misusing MeetMe??





********************************************************************************
All electronic messages sent and received by Stifel Nicolaus
Associates are subject to review by Stifel Nicolaus.  Stifel Nicolaus
may retain and reproduce electronic messages for state, federal, or
other regulatory agencies as required by applicable law.  
IMPORTANT: Please do not use e-mail to request or authorize the
purchase or sale of any security or commodity, send fund transfer
instructions, or otherwise conduct any securities transactions.  Any 
requests, orders, instructions, or time-sensitive messages sent by
e-mail cannot be accepted or processed by Stifel Nicolaus.  The 
accuracy of any information sent by Stifel Nicolaus through e-mail 
cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
 
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********************************************************************************

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From wsisk at cisco.com  Tue Oct  9 16:07:37 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 09 Oct 2007 16:07:37 -0400
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
In-Reply-To: <FA233791F398B848B978464507739E8F3E43C3@EUBEBRUSVEX1.eu.didata.local>
References: <464405.10379.qm@web44803.mail.sp1.yahoo.com>
	<FA233791F398B848B978464507739E8F3E43C3@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <470BDF89.60705@cisco.com>

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From wsisk at cisco.com  Tue Oct  9 16:17:50 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 09 Oct 2007 16:17:50 -0400
Subject: [cisco-voip] 5.1 active disk filling?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A6099EDE12@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A6099EDE12@Apollo.wesd.org>
Message-ID: <470BE1EE.9030201@cisco.com>

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From db7td at gmx.de  Tue Oct  9 16:18:05 2007
From: db7td at gmx.de (Dietmar)
Date: Tue, 9 Oct 2007 22:18:05 +0200
Subject: [cisco-voip] Access restrictions AXL/SOAP CCM4.2
Message-ID: <200710092218.05838.db7td@gmx.de>

Hi all,

while playing a little with the AXL/SOAP interface of CCM 4.2, I realized that 
every authenticated user is able to read and modify (!) ALL settings, even 
these of phones/lines that are not assigned to that user.

I would have expected that a user can only modify his own settings with the 
AXL/SOAP interface. Hopefully, this is true and someone has an idea what is 
wrong with my CCM settings ;-). Any hints?


Thanks,
  Dietmar

From asobihoudai at yahoo.com  Tue Oct  9 15:47:20 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 9 Oct 2007 12:47:20 -0700 (PDT)
Subject: [cisco-voip] background image
In-Reply-To: <228dc9bd0710091242w790b1eeaqd0afff9dc531be67@mail.gmail.com>
Message-ID: <873342.9351.qm@web57213.mail.re3.yahoo.com>

Background images aren't supported on 40s/60s...must
have 41s/61s

--- Michael Sy <mjdsy9 at gmail.com> wrote:

> assistance needed pls in creating and uploading the
> background image for
> 7940 and 7970 phones..ccm 4.2..thanks in advance
> 
> -- 
> Mike
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



      ____________________________________________________________________________________
Don't let your dream ride pass you by. Make it a reality with Yahoo! Autos.
http://autos.yahoo.com/index.html
 



From VLoschiavo at data-corporation.com  Tue Oct  9 16:56:13 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Tue, 9 Oct 2007 16:56:13 -0400
Subject: [cisco-voip] background image
In-Reply-To: <873342.9351.qm@web57213.mail.re3.yahoo.com>
References: <228dc9bd0710091242w790b1eeaqd0afff9dc531be67@mail.gmail.com>
	<873342.9351.qm@web57213.mail.re3.yahoo.com>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF57136B9861@MAIL02.hq.data-corporation.com>

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Tuesday, October 09, 2007 3:47 PM
To: Michael Sy; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background image

Background images aren't supported on 40s/60s...must
have 41s/61s

--- Michael Sy <mjdsy9 at gmail.com> wrote:

> assistance needed pls in creating and uploading the
> background image for
> 7940 and 7970 phones..ccm 4.2..thanks in advance
> 
> -- 
> Mike
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



     That is correct.

You need a 7941, 7942, 7945, 7961, 7962, 7965, 7970, or 7971  (aka an
XML capable phone).

However,
For the 7940's and 7960's it is possible to use the IDLE URL feature to
display a low quality image.

Check the IP Phone Services SDK from Crisco for the application to
convert the images to XML format.

Or check out 
http://www.ccmstuff.net/
or more how-to's and examples.

-Vince



From greencucumber2007 at yahoo.com  Tue Oct  9 16:35:14 2007
From: greencucumber2007 at yahoo.com (Cucumber Green)
Date: Tue, 9 Oct 2007 13:35:14 -0700 (PDT)
Subject: [cisco-voip] MeetMe
In-Reply-To: <5CC6FF682D32164BA5F2E830ABDF146001A5AD28@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <999640.86013.qm@web44805.mail.sp1.yahoo.com>

I think one person must start the meet-me conference by picking up the phone, and selecting the meet-me soft-button and then entering 4099
   
  After that, other people can join in by calling the 4099 number

"Gibson, Brian (Jersey City)" <brian.gibson at stifel.com> wrote:
                Is 4099 a valid PSTN DN?  
   
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Tuesday, October 09, 2007 2:02 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MeetMe

   
  I need to setup a conference call.  We will have 5 or 6 people calling in from outside.
So I decided maybe MeetMe was the best way.  I setup some MeetMe numbers.  One is 4099.
But when someone dials in from the outside they get "I don't recognize that as a valid extension". 
Am I misusing MeetMe??




  ********************************************************************************
  All electronic messages sent and received by Stifel Nicolaus
  Associates are subject to review by Stifel Nicolaus. Stifel Nicolaus
  may retain and reproduce electronic messages for state, federal, or
  other regulatory agencies as required by applicable law. 
  IMPORTANT: Please do not use e-mail to request or authorize the
  purchase or sale of any security or commodity, send fund transfer
  instructions, or otherwise conduct any securities transactions. Any 
  requests, orders, instructions, or time-sensitive messages sent by
  e-mail cannot be accepted or processed by Stifel Nicolaus. The 
  accuracy of any information sent by Stifel Nicolaus through e-mail 
  cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
  
  Stifel, Nicolaus & Company, Incorporated
  Member NYSE & SIPC
  Headquarters: 501 N. Broadway, St. Louis, MO 63102
  314-342-2000
  ********************************************************************************
   
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
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From VLoschiavo at data-corporation.com  Tue Oct  9 17:33:09 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Tue, 9 Oct 2007 17:33:09 -0400
Subject: [cisco-voip] MeetMe
In-Reply-To: <999640.86013.qm@web44805.mail.sp1.yahoo.com>
References: <5CC6FF682D32164BA5F2E830ABDF146001A5AD28@SNEXCHANGE02.stifelnet.stifel.local>
	<999640.86013.qm@web44805.mail.sp1.yahoo.com>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF57136B987D@MAIL02.hq.data-corporation.com>

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cucumber Green
Sent: Tuesday, October 09, 2007 4:35 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MeetMe

 

I think one person must start the meet-me conference by picking up the
phone, and selecting the meet-me soft-button and then entering 4099

 

After that, other people can join in by calling the 4099 number

"Gibson, Brian (Jersey City)" <brian.gibson at stifel.com> wrote:

	Is 4099 a valid PSTN DN?  

	 

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
	Sent: Tuesday, October 09, 2007 2:02 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] MeetMe

	 

	I need to setup a conference call.  We will have 5 or 6 people
calling in from outside.
	So I decided maybe MeetMe was the best way.  I setup some MeetMe
numbers.  One is 4099.
	But when someone dials in from the outside they get "I don't
recognize that as a valid extension". 
	Am I misusing MeetMe??
	
	

	
************************************************************************
********

	All electronic messages sent and received by Stifel Nicolaus

	Associates are subject to review by Stifel Nicolaus. Stifel
Nicolaus

	may retain and reproduce electronic messages for state, federal,
or

	other regulatory agencies as required by applicable law. 

	IMPORTANT: Please do not use e-mail to request or authorize the

	purchase or sale of any security or commodity, send fund
transfer

	instructions, or otherwise conduct any securities transactions.
Any 

	requests, orders, instructions, or time-sensitive messages sent
by

	e-mail cannot be accepted or processed by Stifel Nicolaus. The 

	accuracy of any information sent by Stifel Nicolaus through
e-mail 

	cannot be warranted or guaranteed by Stifel Nicolaus or its
affiliates.

	Stifel, Nicolaus & Company, Incorporated

	Member NYSE & SIPC

	Headquarters: 501 N. Broadway, St. Louis, MO 63102

	314-342-2000

	
************************************************************************
********

	 

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

 

  That is absolutely correct.....the Conference must be started from the
MeetMe Softkey.

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From mjdsy9 at gmail.com  Tue Oct  9 17:56:34 2007
From: mjdsy9 at gmail.com (Michael Sy)
Date: Tue, 9 Oct 2007 14:56:34 -0700
Subject: [cisco-voip] background image
In-Reply-To: <CB4BC8C24EF6E149AED9FDB9B9AF57136B9861@MAIL02.hq.data-corporation.com>
References: <228dc9bd0710091242w790b1eeaqd0afff9dc531be67@mail.gmail.com>
	<873342.9351.qm@web57213.mail.re3.yahoo.com>
	<CB4BC8C24EF6E149AED9FDB9B9AF57136B9861@MAIL02.hq.data-corporation.com>
Message-ID: <228dc9bd0710091456u7a066bcfh1c7a9d9d15ce694@mail.gmail.com>

Thanks for your reply but this is for a 7941 and 7970. In the SDk it seems
its only for the 7912 . pls advise.

On 10/9/07, Vince Loschiavo <VLoschiavo at data-corporation.com> wrote:
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
> Sent: Tuesday, October 09, 2007 3:47 PM
> To: Michael Sy; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] background image
>
> Background images aren't supported on 40s/60s...must
> have 41s/61s
>
> --- Michael Sy <mjdsy9 at gmail.com> wrote:
>
> > assistance needed pls in creating and uploading the
> > background image for
> > 7940 and 7970 phones..ccm 4.2..thanks in advance
> >
> > --
> > Mike
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>     That is correct.
>
> You need a 7941, 7942, 7945, 7961, 7962, 7965, 7970, or 7971  (aka an
> XML capable phone).
>
> However,
> For the 7940's and 7960's it is possible to use the IDLE URL feature to
> display a low quality image.
>
> Check the IP Phone Services SDK from Crisco for the application to
> convert the images to XML format.
>
> Or check out
> http://www.ccmstuff.net/
> or more how-to's and examples.
>
> -Vince
>
>
>


-- 
Mike Sy
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From SCASPER at mtb.com  Tue Oct  9 19:08:42 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Tue, 09 Oct 2007 19:08:42 -0400
Subject: [cisco-voip] Intercom History
In-Reply-To: <470BD357.9040404@cisco.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
	<470B8E1E.9874.0039.0@mtb.com> <470BD357.9040404@cisco.com>
Message-ID: <470BD1B9.9874.0039.0@mtb.com>

Thanks I will do so. Is Intercom a feature in 4.1.3 with the addition of the service release 5d? Can not find any documentation on it.

>>> "Wes Sisk" <wsisk at cisco.com> 10/9/2007 3:15 PM >>>
phone loads are backward compatible.  Intercom history, like missed/received/placed calls, is stored in the phone and therefore sorted before directories URL features.  Let your account team know this is undesirable please.

/wes

STEVEN CASPER wrote: 

Just upgraded to 4.1.3sr5d -  users now see Intercom History under their Directory button on 7941 and 7961 telephones. In fact, this is now above the Corporate Directory so these users now have to scroll down to get to the Corporate Directory. This of course irritates most users and they want Intercom History removed.
 
What is this for, is it functional? I thought intercom was a Rls 6.x feature. 
 
 
 
Thanks,

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026 
************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
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************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee.  If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission.  If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy.  This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
There are risks associated with the use of electronic transmission.  The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
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From asobihoudai at yahoo.com  Tue Oct  9 18:26:05 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 9 Oct 2007 15:26:05 -0700 (PDT)
Subject: [cisco-voip] background image
In-Reply-To: <228dc9bd0710091456u7a066bcfh1c7a9d9d15ce694@mail.gmail.com>
Message-ID: <164545.21436.qm@web57210.mail.re3.yahoo.com>

You'll need to configure List.xml. Here's the info I
gleaned from the Googlemaster.

Creating Custom Background Images
You can provide users with a choice of background
images for the LCD screen on
their phones. Users can select a background image by
choosing Settings > User 
Preferences > Background Images on the phone.
The image choices that users see come from PNG images
and an XML file (called 
List.xml) that are stored on the TFTP server used by
the phone. By storing your 
own PNG files and editing the XML file on the TFTP
server, you can designate 
the background images from which users can choose. In
this way, you can provide 
custom images, such as your company logo.Page 5

6-5
Cisco Unified IP Phone Administration Guide for Cisco
Unified CallManager 4.2, Cisco Unified IP Phones
OL-8611-01
Chapter 6 Customizing the Cisco Unified IP Phone
Creating Custom Background Images
The following sections describe how you can customize
the background images 
that are available at your site by creating your own
PNG files and editing the 
List.xml file:
?
List.xml File Format Requirements, page 6-5.
?
PNG File Requirements for Custom Background Images,
page 6-6.
?
Configuring a Custom Background Image, page 6-7
List.xml File Format Requirements
The List.xml file defines an XML object that contains
a list of background images. 
The List.xml file is stored in the following folder on
the TFTP server:
C:\Program Files\Cisco\TFTPPath\Desktops\320x212x12
Tip
If you are manually creating the directory structure
and the List.xml file, you must
ensure that the directories and files can be accessed
by the user\CCMService, 
which is used by the TFTP service.
The List.xml file can include up to 50 background
images. The images are in the
order that they appear in the Background Images menu
on the phone. For each 
image, the List.xml file contains one element type,
called ImageItem. The 
ImageItem element includes these two attributes:
?
Image?Uniform resource identifier (URI) that specifies
where the phone 
obtains the thumbnail image that will appear on the
Background Images menu
on a Phone.
?
URL?URI that specifies where the phone obtains the
full size image.
The following example shows a List.xml file that
defines two images. The 
required Image and URL attributes must be included for
each image. The TFTP 
URI that is shown in the example is the only supported
method for linking to full
size and thumbnail images. HTTP URL support is not
provided.Page 6

Chapter 6
Customizing the Cisco Unified IP Phone
Creating Custom Background Images
6-6
Cisco Unified IP Phone Administration Guide for Cisco
Unified CallManager 4.2, Cisco Unified IP Phones 
OL-8611-01
List.xml Example
<CiscoIPPhoneImageList>
<ImageItem
Image=?TFTP:Desktops/320x212x12/TN-Fountain.png? 
URL=?TFTP:Desktops/320x212x12/Fountain.png?/> 
<ImageItem
Image=?TFTP:Desktops/320x212x12/TN-FullMoon.png? 
URL=?TFTP:Desktops/320x212x12/FullMoon.png?/> 
</CiscoIPPhoneImageList>
The Cisco Unified IP Phone firmware includes a default
background image. This 
image is not defined in the List.xml file. The default
image is always the first
image that appears in the Background Images menu on
the phone.
PNG File Requirements for Custom Background Images
Each background image requires two PNG files:
?
Full size image?Version that appears on the on the
phone.
?
Thumbnail image?Version that appears on the Background
Images screen 
from which users can select an image. Must be 25% of
the size of the full size 
image.
Tip
Many graphics programs provide a feature that will
resize a graphic. An easy way 
to create a thumbnail image is to first create and
save the full size image, then use 
the sizing feature in the graphics program to create a
version of that image that is 
25% of the original size. Save the thumbnail version
using a different name.
The PNG files for background images must meet the
following requirements for
proper display on the Cisco Unified IP Phone:
?
Full size image?320 pixels (width) X 212 pixels
(height). 
?
Thumbnail image?80 pixels (width) X 53 pixels
(height).
?
Color palette?Includes up to 12-bit color (4096
colors). You can use more 
than 12-bit color, but the phone will reduce the color
palette to12-bit before 
displaying the image. For best results, reduce the
color palette of an image to
12-bit when you create a PNG file.Page 7

6-7
Cisco Unified IP Phone Administration Guide for Cisco
Unified CallManager 4.2, Cisco Unified IP Phones
OL-8611-01
Chapter 6 Customizing the Cisco Unified IP Phone
Creating Custom Background Images
Tip
If you are using a graphics program that supports a
posterize feature for 
specifying the number of tonal levels per color
channel, set the number of
tonal levels per channel to 16 (16 red X 16 green X 16
blue = 4096 colors).
Configuring a Custom Background Image
To create custom background images for the Cisco
Unified IP Phone, follow these 
steps:
Procedure
Step 1
Create two PNG files for each image (a full size
version and a thumbnail version). 
Ensure the PNG files comply with the format guidelines
that are listed in the 
?PNG File Requirements for Custom Background Images?
section on page 6-6.
Step 2
Place the new PNG files that you created in the
following folder on the TFTP 
server for each Cisco Unified CallManager in the
cluster:
C:\Program Files\Cisco\TFTPPath\Desktops\320x212x12
Note
Cisco recommends that you also store backup copies of
custom image 
files in another location. You can use these backup
copies if the 
customized files are overwritten when you upgrade 
Cisco Unified CallManager.
Step 3
Use a text editor to edit the List.xml file. See the
?List.xml File Format
Requirements? section on page 6-5 for the location of
this file, formatting 
requirements, and a sample file.
Step 4
Save your modifications and close the List.xml file.
Note
When you upgrade Cisco Unified CallManager, a default
List.xml file
will replace your customized List.xml file. After your
customize the 
List.xml file, make a copy of the file and store it in
another location. After 
upgrading Cisco Unified CallManager, replace the
default List.xml file
with your stored copy.Page 8

Chapter 6
Customizing the Cisco Unified IP Phone
Configuring the Idle Display
6-8
Cisco Unified IP Phone Administration Guide for Cisco
Unified CallManager 4.2, Cisco Unified IP Phones 
OL-8611-01
Step 5
To cache the new List.xml file, stop and start the
TFTP service by using 
Cisco Unified CallManager Serviceability or disable
and re-enable the Enable 
Caching of Constant and Bin Files at Startup TFTP
service parameter (located in
the Advanced Service Parameters

HTH,
Paul

--- Michael Sy <mjdsy9 at gmail.com> wrote:

> Thanks for your reply but this is for a 7941 and
> 7970. In the SDk it seems
> its only for the 7912 . pls advise.
> 
> On 10/9/07, Vince Loschiavo
> <VLoschiavo at data-corporation.com> wrote:
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Paul Choi
> > Sent: Tuesday, October 09, 2007 3:47 PM
> > To: Michael Sy; cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] background image
> >
> > Background images aren't supported on
> 40s/60s...must
> > have 41s/61s
> >
> > --- Michael Sy <mjdsy9 at gmail.com> wrote:
> >
> > > assistance needed pls in creating and uploading
> the
> > > background image for
> > > 7940 and 7970 phones..ccm 4.2..thanks in advance
> > >
> > > --
> > > Mike
> > > >
> _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >     That is correct.
> >
> > You need a 7941, 7942, 7945, 7961, 7962, 7965,
> 7970, or 7971  (aka an
> > XML capable phone).
> >
> > However,
> > For the 7940's and 7960's it is possible to use
> the IDLE URL feature to
> > display a low quality image.
> >
> > Check the IP Phone Services SDK from Crisco for
> the application to
> > convert the images to XML format.
> >
> > Or check out
> > http://www.ccmstuff.net/
> > or more how-to's and examples.
> >
> > -Vince
> >
> >
> >
> 
> 
> -- 
> Mike Sy
> 



       
____________________________________________________________________________________
Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. 
http://mobile.yahoo.com/go?refer=1GNXIC

From Edward.Countryman at provena.org  Tue Oct  9 19:41:38 2007
From: Edward.Countryman at provena.org (Countryman, Edward)
Date: Tue, 9 Oct 2007 18:41:38 -0500
Subject: [cisco-voip] Access Valcom Paging from FXO port
In-Reply-To: <898237D7FE163A40A3257E385562F27E0160B34F@AEXMV02.phnet.phroot.local>
Message-ID: <898237D7FE163A40A3257E385562F27E0160B35F@AEXMV02.phnet.phroot.local>


Second request on this to the group

When I dial the DAC for the paging port it waits a moment then fast
busy.  

Has anyone done this with a valcom from a cisco FXO port?


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Countryman,
Edward
Sent: Tuesday, October 09, 2007 11:40 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Help with Paging interface


I've installed an FXO port in my router and believe I have it configured
correctly in CCM but cannot get it to open when connected to a VALCOM
V2000A.  

When dialing the route pattern for this port; it appears to sit for
several seconds then go fast busy.  It is almost like it is waiting for
something from the valcom (dial tone detect?)

I presently have the battery feed switch on the valcom set to ON. 

Any ideas?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jsteinberg at gmail.com  Tue Oct  9 20:02:00 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 9 Oct 2007 20:02:00 -0400
Subject: [cisco-voip] Shutting down a switch port based on time of day
	...
In-Reply-To: <43D799CBE4331D44999AB7C9CD67F30203AFA05F@NTEXCHANGE.riverbank.com>
References: <43D799CBE4331D44999AB7C9CD67F30203AFA05F@NTEXCHANGE.riverbank.com>
Message-ID: <f1b65f880710091702o10a0728eud3793df9210054cc@mail.gmail.com>

depending on the number of ports you could use time based ACL's on a
layer 3 device and put each switchport in it's own vlan.  obviously,
this doesn't scale but if you need this on only a subset of ports this
might work.  there maybe a better native way to do this but i'm not
aware of it.

http://www.cisco.com/en/US/products/sw/secursw/ps1018/products_tech_note09186a00800a5b9a.shtml#timebasedtimerange


On 10/9/07, Duane Priebe <dpriebe at theriverbank.com> wrote:
>
>
> Sorry, I know this might be off topic for this group but could anyone give
> me info on a way that I could have a switch port (IOS 12.4) shutdown based
> on time of day?
>
> Duane
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jsteinberg at gmail.com  Tue Oct  9 20:04:16 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 9 Oct 2007 20:04:16 -0400
Subject: [cisco-voip] Unity Connection >System Settings> Message Store
	Quota 12000000 Bytes
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707F545@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00707F545@USNAEXCH.na.didata.local>
Message-ID: <f1b65f880710091704x2962c293ycc2f450f4236cfa7@mail.gmail.com>

deleted items folder is counted towards the quota.  another option is
to lower the number of days before messages are purged from the
deleted items folder i think by default it is 15.

On 10/9/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> Warning Quote defaults to 12000000 Bytes
>
>
>
> How many minutes is this for g711 messages? For a MCS-7825-H3 with 500 users
> I'm wondering how much higher I should increase it, users are complaining
> it's too small.
>
>
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From smgustafson at gmail.com  Tue Oct  9 21:51:42 2007
From: smgustafson at gmail.com (Steve G)
Date: Tue, 9 Oct 2007 18:51:42 -0700
Subject: [cisco-voip] Access Valcom Paging from FXO port
In-Reply-To: <898237D7FE163A40A3257E385562F27E0160B35F@AEXMV02.phnet.phroot.local>
References: <898237D7FE163A40A3257E385562F27E0160B34F@AEXMV02.phnet.phroot.local>
	<898237D7FE163A40A3257E385562F27E0160B35F@AEXMV02.phnet.phroot.local>
Message-ID: <8a1f34b50710091851scf8fa3amfb6522d46983e674@mail.gmail.com>

Are you using h323 or MGCP to control the fxo port?

Do you see the green light on your FXO port flash when you dial the DN?  If
you see it flash, then you probably don't have it connected to the Valcom
correctly.  Please paste your router config for the voice port and
dial-peer.


On 10/9/07, Countryman, Edward <Edward.Countryman at provena.org> wrote:
>
>
> Second request on this to the group
>
> When I dial the DAC for the paging port it waits a moment then fast
> busy.
>
> Has anyone done this with a valcom from a cisco FXO port?
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Countryman,
> Edward
> Sent: Tuesday, October 09, 2007 11:40 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Help with Paging interface
>
>
> I've installed an FXO port in my router and believe I have it configured
> correctly in CCM but cannot get it to open when connected to a VALCOM
> V2000A.
>
> When dialing the route pattern for this port; it appears to sit for
> several seconds then go fast busy.  It is almost like it is waiting for
> something from the valcom (dial tone detect?)
>
> I presently have the battery feed switch on the valcom set to ON.
>
> Any ideas?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From ciscojock2002 at yahoo.com  Tue Oct  9 21:22:13 2007
From: ciscojock2002 at yahoo.com (Charles Ragan, Jr.)
Date: Tue, 9 Oct 2007 18:22:13 -0700 (PDT)
Subject: [cisco-voip] Access restrictions AXL/SOAP CCM4.2
In-Reply-To: <200710092218.05838.db7td@gmx.de>
Message-ID: <779799.28413.qm@web50302.mail.re2.yahoo.com>

Nope - it's true.  With 4.2 and later you can restrict using MLA.
   
  With previous versions of CM, we chose to restrict what ip address space could access the soap/axl interface within IIS.
   
  Charles

Dietmar <db7td at gmx.de> wrote:
  Hi all,

while playing a little with the AXL/SOAP interface of CCM 4.2, I realized that 
every authenticated user is able to read and modify (!) ALL settings, even 
these of phones/lines that are not assigned to that user.

I would have expected that a user can only modify his own settings with the 
AXL/SOAP interface. Hopefully, this is true and someone has an idea what is 
wrong with my CCM settings ;-). Any hints?


Thanks,
Dietmar
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



Charles Ragan Jr., Technology Consultant
Principal Consultant - CCIE #1764
Cell Phone - 336-442-4361
http://www.geocities.com/ciscojock2002/
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From Ruben.Montes at eu.didata.com  Wed Oct 10 07:48:25 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Wed, 10 Oct 2007 13:48:25 +0200
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
References: <464405.10379.qm@web44803.mail.sp1.yahoo.com>
	<FA233791F398B848B978464507739E8F3E43C3@EUBEBRUSVEX1.eu.didata.local>
	<470BDF89.60705@cisco.com>
Message-ID: <FA233791F398B848B978464507739E8FD6C1DE@EUBEBRUSVEX1.eu.didata.local>

Hi,

 

I'll try to search in the BAT templates and if I cannot find anything
I'll search in the database...

 

In the case I find the PKID in the database, how am I supposed to delete
this field?

 

Regards,

 

Ruben

 

________________________________

De: Wes Sisk [mailto:wsisk at cisco.com] 
Enviado el: martes, 09 de octubre de 2007 22:08
Para: Ruben Montes (Europe)
CC: Cucumber Green; cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Cannot delete Partition, Dependency Records
empty

 

Reuben,

Other spots to check:
BAT Templates (CSCsi08406)
autoregister phone partition (system->server)

Otherwise, let's export your db to plain text and search.  Resource
intensive so do after hours. Access your server CLI by ssh. Login using
the platform administrator account and log your session. Issue command:
show tech database dump

retrieve the file and search for the PKID of the partition.  If don't
find it zip and unicast me the file.

Example:
admin:show tech database dump
************************************************************************
****************************
This command reads and writes the entire database and will take quite
some time...please be patient.
************************************************************************
****************************
 -------------------- show tech database  -------------------- 
Entire database
Output is in file cm/trace/dbl/showtech26157.out 
admin:file get activelog cm/trace/dbl/showtech26157.out

In the resulting file any reference to fkroutepartition is a pointer to
the partition table. fkroutepartition always points to
routepartition.pkid example:
 -----------------------------------------------------------
         Table RoutePartition
 -----------------------------------------------------------
pkid,name,description,dialplanwizardgenid,fktimeschedule,tktimezone,useo
riginatingdevicetimezone,resettoggle,tkreset
"e472970e-51cd-8030-ad65-da44e16b5dc9","prtRTP","RTP
Partition","22","t","f","2"

/Wes

Ruben Montes (Europe) wrote: 

Hi,

 

in the unassigned dn report it doesn't either appear...

 

________________________________

De: cisco-voip-bounces at puck.nether.net en nombre de Cucumber Green
Enviado el: mar 09/10/2007 19:12
Para: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Cannot delete Partition, Dependency Records
empty

I think there is an actual limit of the number of partitions, the number
depends on the names you use for the partitions, but its surely larger
than 75.

 

I experienced this partition deletion problem ( but in CCM 4.2), if I
remember correctly, it was a DN not being used but belonging to the
partition, I went into RoutePlan->RoutePlanReport looked for the DN and
deleted it.

"Ruben Montes (Europe)" <Ruben.Montes at eu.didata.com>
<mailto:Ruben.Montes at eu.didata.com>  wrote:

	Hi,
	
	Fine then, anyway, I'm still having the same problem because I
have some
	'locked' partitions...
	
	Thanks,
	
	Ruben
	
	-----Mensaje original-----
	De: Jeffrey C. Ollie [mailto:jeff at ocjtech.us] 
	Enviado el: martes, 09 de octubre de 2007 18:31
	Para: Ruben Montes (Europe)
	CC: Gibson, Brian (Jersey City); cisco-voip at puck.nether.net
	Asunto: Re: [cisco-voip] Cannot delete Partition, Dependency
Records
	empty
	
	On Tue, 2007-10-09 at 18:25 +0200, Ruben Montes (Europe) wrote:
	> 
	> When you create a new Partition, if you read the lines above,
it says
	> you can only create 75. Maybe am I wrong? I'm running CCM
5.1(2)
	
	I believe that message refers to the fact that you can only
create 75
	partitions *at a time* using the web interface.
	
	Jeff
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

 

________________________________

Moody friends. Drama queens. Your life? Nope! - their life, your story.
Play Sims Stories at Yahoo! Games.
<http://us.rd.yahoo.com/evt=48224/*http:/sims.yahoo.com/> 

 



________________________________



 
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https://puck.nether.net/mailman/listinfo/cisco-voip
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From benjamin.kappauf at googlemail.com  Wed Oct 10 07:37:40 2007
From: benjamin.kappauf at googlemail.com (Benjamin Kappauf)
Date: Wed, 10 Oct 2007 13:37:40 +0200
Subject: [cisco-voip] call-manager-fallback wrong phone dn
Message-ID: <feidjf$kkr$1@sea.gmane.org>

What is the problem if in case of a fallback all my phones are 
registered fine to the voice router. But all phones don't get the 
correct DN. Most of them get a 65xx DN. A few are with ext. mobility 
activ. And they all have got DNs out of 4440 - 4449

Btw. no 65xx number exist on the callmanager.

Thanks alot for any help!


----------------------------------
call-manager-fallback
  secondary-dialtone 9
  max-conferences 8 gain -6
  transfer-system full-consult
  ip source-address x.x.x.x port 2000
  max-ephones 40
  max-dn 80 dual-line
  system message primary SRST Fallback
  system message secondary SRST
  time-format 24
  date-format dd-mm-yy


From Roconnell at unislumin.com  Wed Oct 10 10:00:54 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Wed, 10 Oct 2007 10:00:54 -0400
Subject: [cisco-voip] CME 2 languages
Message-ID: <18082B35CF0EBD4C8D265874508A615C071B0D2A@hermes.unislumin.com>

Hi all,

 

A question regarding CME and phone registration issue.

 

A new CME 4.1 (12.4(15)T1) installation with French and English
configuration files where some people using French and some English
7940/60 soft buttons.

 

Initially, bulk ephone and ephone-dn import was done (with default CME
settings) and all of those devices have registered properly on-site, as
expected.

 

Now, newly added phones will not register to CME even if ephone is
entered in configuration, it is required to run "create cnf" command in
order to successfully register a phone and that command has to be run
every time there is a new phone. 

 

These are related changes since the initial setup:

 

telephony-service 

 cnf-file location flash:
 cnf-file perphone

 user-locale 1 FR
 network-locale 1 CA

 

and creating a new phone with French template:

 

ephone  33
 mac-address 0016.C81F.E3ED
 ephone-template 1
 type 7940

 

Do you think that cnf-file perphone command together with a non default
ephone template could cause this? to run cnf creation every time new
phone is created?

 

Have you seen this before or is it just a bug?

 

 


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Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.
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From jason.aarons at us.didata.com  Wed Oct 10 10:57:24 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 10 Oct 2007 10:57:24 -0400
Subject: [cisco-voip] VM Alternate Greeting number of days to use?
Message-ID: <C1FE15183DA37645BC0633BC604E44F0070BBDAB@USNAEXCH.na.didata.local>

Alternate Greeting number of days to use? 

 

Customer has a old Nortel voicemail that when you setup a alternate
greeting it would ask how many days you want it to run, can you do this
with Unity 5.x or Unity Connection 2.0?




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
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this communication in error and that any use or reproduction of
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unlawful.  If you have received this communication in error, please
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From svoll.voip at gmail.com  Wed Oct 10 11:02:03 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 10 Oct 2007 08:02:03 -0700
Subject: [cisco-voip] VM Alternate Greeting number of days to use?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0070BBDAB@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0070BBDAB@USNAEXCH.na.didata.local>
Message-ID: <f84a38d30710100802v5f4a607bo5a2a545d385dabdb@mail.gmail.com>

you can in Unity 4.0.5 so I assume you can in 5.x

Scott


On 10/10/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  Alternate Greeting number of days to use?
>
>
>
> Customer has a old Nortel voicemail that when you setup a alternate
> greeting it would ask how many days you want it to run, can you do this with
> Unity 5.x or Unity Connection 2.0?
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
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> _______________________________________________
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From ErasmuE4 at telkom.co.za  Wed Oct 10 11:19:08 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 10 Oct 2007 17:19:08 +0200
Subject: [cisco-voip] 5.1 Speed Dial Presence
In-Reply-To: <470AA006.4080007@cisco.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01A923E7@CNTRRA20-XCS01.telkom.co.za>


I also read somewhere that the x0 versions does not support the
intercom functionality like whisper but the x1 like 7941 / 7961 does
... in ccm 6

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Monday, October 08, 2007 11:24 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence

 

x1 allows you to see BLF status of calling party in
missed/placed/received directories.  Personally, this is one of my
favorite features.

x1 - will be around much longer and get new features including
midlets.
x0 - no more feature development.

/Wes

Jason Aarons (US) wrote: 

Any benefits if they used 79x1 over 7960 for Speed Dial Presence? They
are about to order more 7960s over 7961s due to costs. I wish there
was some motivating  one-reason issue to get customers to buy 79x1s. I
know the benefits of 79x1, and would only buy this model myself, but
costs are sometimes the winner with customers over logic. Can't we end
of life the 7960s and make the 7961 same price!

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Monday, October 08, 2007 4:01 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence

 

native.  no CUPS required. /wes

Jason Aarons (US) wrote: 

I think what they want is DSS/BLF, does it require a dedicated
presence server or can CCM  5.1 do this natively?

Cisco Unified CallManager Speed Dial Presence 

Cisco Unified CallManager supports the ability for a speed dial to
have presence capabilities via a busy lamp field (BLF) speed dial. BLF
speed dials work as both a speed dial and a presence indicator. Only
the system administrator can configure a BLF speed dial; a system user
is not allowed to configure a BLF speed dial. 

States are: 

Idle 

Busy 

Unknown 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implemen
tation_design_guide_chapter09186a008070bc02.html#wp1070507
<javascript:newWin('http://www.cisco.com/en/US/products/sw/voicesw/ps5
56/products_implementation_design_guide_chapter09186a008070bc02.html#w
p1070507')> 

 

From: Jason Aarons (US) 
Sent: Monday, October 08, 2007 2:05 PM
To: cisco-voip at puck.nether.net
Subject: 79x0 vs 79x1 (BLF?)

 

Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button
to have "auto dial and appearance", they have been using the 7940 a
week, along with CCMUser and don't think AbbrDial/FastDials/Personal
Directory meets their needs. Their old phone had lots of buttons, etc.

What is the CallManager 5.1 feature called where a button can be both
a Shared Line and Speed Dial?  I'm trying to find the URL if it
requires a 79x1 or would it work with a 7960.

What are the caveats? Can they setup/manage via CCMUser? 

I've been stuck on 4.1 a long time and recall seeing something about
this in 5.x, but can't find the details.  I know 4.1 you could have
either a shared line or a speed dial but not both.

 -jason

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the intended
addressee, you are hereby notified that you have received this
communication in error and that any use or reproduction of this email
or its contents is strictly prohibited and may be unlawful. If you
have received this communication in error, please notify us
immediately by replying to this message and deleting it from your
computer. Thank you. 

 
 



________________________________



 
 
  
 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck..nether.net/mailman/listinfo/cisco-voip
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________________________________


Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the intended
addressee, you are hereby notified that you have received this
communication in error and that any use or reproduction of this email
or its contents is strictly prohibited and may be unlawful. If you
have received this communication in error, please notify us
immediately by replying to this message and deleting it from your
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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From wsisk at cisco.com  Wed Oct 10 11:22:00 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 10 Oct 2007 11:22:00 -0400
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
In-Reply-To: <FA233791F398B848B978464507739E8FD6C1DE@EUBEBRUSVEX1.eu.didata.local>
References: <464405.10379.qm@web44803.mail.sp1.yahoo.com>
	<FA233791F398B848B978464507739E8F3E43C3@EUBEBRUSVEX1.eu.didata.local>
	<470BDF89.60705@cisco.com>
	<FA233791F398B848B978464507739E8FD6C1DE@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <470CEE18.5080707@cisco.com>

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From tednugent73 at gmail.com  Wed Oct 10 12:05:34 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Wed, 10 Oct 2007 12:05:34 -0400
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
Message-ID: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>

I upgraded a customer to 5.13 for various bug fixes and now the CFwdALL no
longer shows in the CCMUser page. I opened a TAC case and I'm being told
it's a WAD! CFwdAll has been in the CCMUser page for as long as I can
remember and now it's gone and I'm expected to tell my customers that Cisco
decided you no longer need that? I've got many customers that use this
religiously and if this is actually a WAD then they're stuck on 5.12 and
lower? That just doesn't add up?


Wes have you heard anything about this and/or can you shed some light on it?
I understand that having all the CFwd options in CCMUser is not the best
solution but CFwdAll is really the only reason any of my customers use it?
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From ano-1 at worldonline.dk  Wed Oct 10 12:14:00 2007
From: ano-1 at worldonline.dk (Katrine og Michael)
Date: Wed, 10 Oct 2007 18:14:00 +0200
Subject: [cisco-voip] Problems whit a SIP connection
Message-ID: <001601c80b58$91589b70$0b000a0a@ano>

Hey,

Can somebody please help me?

I have configured CCME on a 2801 router and I have it connected to my SIP provider.

I?am having a problem.
When is is registred up against the SIP-erver everything  works fine, but after a while i get this message:


Oct 10 15:47:31.101: //-1/xxxxxxxxxxxx/SIP/Info/udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060

And then I can not get call in, but I can ring out the ccme.

I don?t know if is has anything to do whit my SIP-UA, som here it is:

sip-ua
 authentication username voi49495 password 12341
 no remote-party-id
 retry invite 4
 retry response 3
 retry bye 2
 retry cancel 2
 retry register 5
 timers register 250
 registrar dns:sip.cybercity.dk expires 3600

Please help me, and if you need more info, then just read



Michael from Denmark :-)

-- 
Jeg beskyttes af den gratis SPAMfighter til privatbrugere.
Den har indtil videre sparet mig for at f? 691 spam-mails.
Betalende brugere f?r ikke denne besked i deres e-mails.
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From rkulagow at gmail.com  Wed Oct 10 12:17:56 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Wed, 10 Oct 2007 11:17:56 -0500
Subject: [cisco-voip] Deploying RSVP in a CM 4.1.3 environment to supplement
 locations-based CAC
Message-ID: <470CFB34.3010202@gmail.com>

I'm reading Cisco Voice GW and GK.

Most of my remote sites have a single WAN connection.  Those sites have 
a location CAC of 270k in CM (which is what our provider will guarantee 
in their real-time voice queue for QOS).  Remote sites all use H.323.

Only VM traffic goes over the WAN back to Unity; all other calls go over 
PSTN.

If the WAN is out, then the site goes into SRST.  Users hate that.

At some sites we're deploying a second router through a different 
provider.  No QOS on that link.  ACLs block voice traffic, because if I 
allow phones to register over this link, then they'll think that they 
can talk to Unity, and we don't want that.

I'm thinking that if I can setup RSVP, I can set the interfaces so that 
the primary router has a rsvp bandwidth equal to the RTVo queue.  And 
that the secondary router will have a RSVP bandwidth of 0, but will 
allow SCCP traffic to traverse.  Phones will stay registered.

1)  Is this feasible?
2)  Is it practical?
2a)  Is it solving my problem?
3)  I don't have to deploy gatekeepers, do I?
3a)  Locations sound an awful lot like zones - why can't my Polycom 
H.323 video unit register to CM 4.1.3?  (why isn't CM a H.323 gatekeeper?)
4)  Will I set my locations bandwidth to Infinity and let the VGW decide 
  what to do?
5)  Am I going to use AAR at all on CM?  Would I use a route pattern to 
intercept the pilot number of Unity so that users that press "Messages" 
get their call sent to the gateway instead of setting up a WAN/IP call?

From jsteinberg at gmail.com  Wed Oct 10 12:31:26 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 10 Oct 2007 12:31:26 -0400
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
References: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
Message-ID: <f1b65f880710100931q46dbeb0fg6e4b558c11321fdf@mail.gmail.com>

I can't believe this would be WAD, this must be a bug.

In fact, I logged into my 5.1.3 lab box and saw the call fwd settings
there.  I then went to the Ent Params and saw that the param 'Show
Call Forwarding' was set to false, which was weird because I just
logged into ccmuser and saw the fwd settings.  I changed this param to
'true' logged out of ccmuser and back in and now, like you, i dont see
the fwd settings.  I went back to ent params and set the param back to
false and still don't see fwd settings in ccmuser.





On 10/10/07, Ted Nugent <tednugent73 at gmail.com> wrote:
> I upgraded a customer to 5.13 for various bug fixes and now the CFwdALL no
> longer shows in the CCMUser page. I opened a TAC case and I'm being told
> it's a WAD! CFwdAll has been in the CCMUser page for as long as I can
> remember and now it's gone and I'm expected to tell my customers that Cisco
> decided you no longer need that? I've got many customers that use this
> religiously and if this is actually a WAD then they're stuck on 5.12 and
> lower? That just doesn't add up?
>
>
> Wes have you heard anything about this and/or can you shed some light on it?
> I understand that having all the CFwd options in CCMUser is not the best
> solution but CFwdAll is really the only reason any of my customers use it?
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From deadpool.slayer.telecomm at gmail.com  Wed Oct 10 12:38:32 2007
From: deadpool.slayer.telecomm at gmail.com (Andrew F. Wolfe)
Date: Wed, 10 Oct 2007 11:38:32 -0500
Subject: [cisco-voip] CME 4.1 and a locked up 7920
Message-ID: <913929690710100938q390c6604s8e3b0a0dd6f5eb8@mail.gmail.com>

Has anyone out there seen an issue on CME with the 7920 locking up when
there is a shared line?  I found a couple of bugs that point to issues but
not the phone locking up with the previous calls caller ID.  I have verified
the firmware and have no other issues with my CME system now.  Any
suggestions would be appreciated.
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From rkulagow at gmail.com  Wed Oct 10 13:20:00 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Wed, 10 Oct 2007 12:20:00 -0500
Subject: [cisco-voip] Deploying RSVP in a CM 4.1.3 environment to
 supplement locations-based CAC
In-Reply-To: <185520.41289.qm@web57203.mail.re3.yahoo.com>
References: <185520.41289.qm@web57203.mail.re3.yahoo.com>
Message-ID: <470D09C0.9040607@gmail.com>

Paul Choi wrote:
> One quick note Robert, Cisco routers acting as H.323
> gateways don't even register to CCM. That doesn't mean
> they're not working properly. In fact, after adding an
> H.323 gateway to CCM 4.1(3), the status will remain
> Unknown or something to that effect.

Yep, I know that.

I'm just wondering why I have to setup an entirely separate 
infrastructure of H.323 Gatekeepers and configure "zones" and all that 
jazz for the Polycom video units to register to.  The Polycom's employ 
H.323 gatekeeper discovery, so why can't the Call Manager's answer, 
perform IP <-> E.164 mapping, determine if the Polycom can use IP to 
connect to another Polycom over the LAN/WAN, etc, etc?  Seems like 99% 
of that functionality is already _in_ CallManager.

From baryonyx5 at gmail.com  Wed Oct 10 13:27:41 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Wed, 10 Oct 2007 12:27:41 -0500
Subject: [cisco-voip] Reason why CP7914 doesn't work with CP7940?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00707F1B6@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00707F1B6@USNAEXCH.na.didata.local>
Message-ID: <a749d0620710101027t40cec6adm67e403d4510eb8f4@mail.gmail.com>

I don't think there's a technical reason.  The 40's and 60's use the same
firmware.  Cisco just doesn't expose the settings in ccmadmin I assume
because they want you to buy a 796x if you need more than two line buttons.


I think you can use a sidecar on a 7940 in CME but I can't remember for
sure.

On 10/9/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  I take everything Cisco says and repeat it to customers without question.
>
>
>
> Customers challenge everything; today I was asked why doesn't the CP-7914
> support the CP-7940? Anyone have a technical answer?
>
>
>
> ------------------------------
>
> * Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
kris seraphine
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From tednugent73 at gmail.com  Wed Oct 10 13:40:22 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Wed, 10 Oct 2007 13:40:22 -0400
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <f1b65f880710100931q46dbeb0fg6e4b558c11321fdf@mail.gmail.com>
References: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
	<f1b65f880710100931q46dbeb0fg6e4b558c11321fdf@mail.gmail.com>
Message-ID: <22e6380b0710101040yd5b72bek3d640570c5781df@mail.gmail.com>

Thats exactly how it went down for us.. The customer is NOT happy!

TAC is telling me that this was done purposely and that if we want it back
we need to "talk to our account team and file a PER (Product Enhancement
Request). This is not going to fly, I don;t even know how to have that
conversation with my customers...


On 10/10/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> I can't believe this would be WAD, this must be a bug.
>
> In fact, I logged into my 5.1.3 lab box and saw the call fwd settings
> there.  I then went to the Ent Params and saw that the param 'Show
> Call Forwarding' was set to false, which was weird because I just
> logged into ccmuser and saw the fwd settings.  I changed this param to
> 'true' logged out of ccmuser and back in and now, like you, i dont see
> the fwd settings.  I went back to ent params and set the param back to
> false and still don't see fwd settings in ccmuser.
>
>
>
>
>
> On 10/10/07, Ted Nugent <tednugent73 at gmail.com> wrote:
> > I upgraded a customer to 5.13 for various bug fixes and now the CFwdALL
> no
> > longer shows in the CCMUser page. I opened a TAC case and I'm being told
> > it's a WAD! CFwdAll has been in the CCMUser page for as long as I can
> > remember and now it's gone and I'm expected to tell my customers that
> Cisco
> > decided you no longer need that? I've got many customers that use this
> > religiously and if this is actually a WAD then they're stuck on 5.12 and
> > lower? That just doesn't add up?
> >
> >
> > Wes have you heard anything about this and/or can you shed some light on
> it?
> > I understand that having all the CFwd options in CCMUser is not the best
> > solution but CFwdAll is really the only reason any of my customers use
> it?
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
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From asobihoudai at yahoo.com  Wed Oct 10 13:46:00 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 10 Oct 2007 10:46:00 -0700 (PDT)
Subject: [cisco-voip] Reason why CP7914 doesn't work with CP7940?
In-Reply-To: <a749d0620710101027t40cec6adm67e403d4510eb8f4@mail.gmail.com>
Message-ID: <841589.4248.qm@web57201.mail.re3.yahoo.com>

AFAICT, there's no parameter to add a 7914 to a 794x
phone under CCME so I'm going to guess they're not
compatible.

Paul

--- Kris Seraphine <baryonyx5 at gmail.com> wrote:

> I don't think there's a technical reason.  The 40's
> and 60's use the same
> firmware.  Cisco just doesn't expose the settings in
> ccmadmin I assume
> because they want you to buy a 796x if you need more
> than two line buttons.
> 
> 
> I think you can use a sidecar on a 7940 in CME but I
> can't remember for
> sure.
> 
> On 10/9/07, Jason Aarons (US)
> <jason.aarons at us.didata.com> wrote:
> >
> >  I take everything Cisco says and repeat it to
> customers without question.
> >
> >
> >
> > Customers challenge everything; today I was asked
> why doesn't the CP-7914
> > support the CP-7940? Anyone have a technical
> answer?
> >
> >
> >
> > ------------------------------
> >
> > * Disclaimer: This e-mail communication and any
> attachments may contain
> > confidential and privileged information and is for
> use by the designated
> > addressee(s) named above only. If you are not the
> intended addressee, you
> > are hereby notified that you have received this
> communication in error and
> > that any use or reproduction of this email or its
> contents is strictly
> > prohibited and may be unlawful. If you have
> received this communication in
> > error, please notify us immediately by replying to
> this message and deleting
> > it from your computer. Thank you. *
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 
> 
> 
> -- 
> kris seraphine
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



      ____________________________________________________________________________________
Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more!
http://tv.yahoo.com/collections/3658 

From asobihoudai at yahoo.com  Wed Oct 10 12:51:08 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 10 Oct 2007 09:51:08 -0700 (PDT)
Subject: [cisco-voip] Deploying RSVP in a CM 4.1.3 environment to
	supplement locations-based CAC
In-Reply-To: <470CFB34.3010202@gmail.com>
Message-ID: <185520.41289.qm@web57203.mail.re3.yahoo.com>

One quick note Robert, Cisco routers acting as H.323
gateways don't even register to CCM. That doesn't mean
they're not working properly. In fact, after adding an
H.323 gateway to CCM 4.1(3), the status will remain
Unknown or something to that effect.

Paul

--- Robert Kulagowski <rkulagow at gmail.com> wrote:

> I'm reading Cisco Voice GW and GK.
> 
> Most of my remote sites have a single WAN
> connection.  Those sites have 
> a location CAC of 270k in CM (which is what our
> provider will guarantee 
> in their real-time voice queue for QOS).  Remote
> sites all use H.323.
> 
> Only VM traffic goes over the WAN back to Unity; all
> other calls go over 
> PSTN.
> 
> If the WAN is out, then the site goes into SRST. 
> Users hate that.
> 
> At some sites we're deploying a second router
> through a different 
> provider.  No QOS on that link.  ACLs block voice
> traffic, because if I 
> allow phones to register over this link, then
> they'll think that they 
> can talk to Unity, and we don't want that.
> 
> I'm thinking that if I can setup RSVP, I can set the
> interfaces so that 
> the primary router has a rsvp bandwidth equal to the
> RTVo queue.  And 
> that the secondary router will have a RSVP bandwidth
> of 0, but will 
> allow SCCP traffic to traverse.  Phones will stay
> registered.
> 
> 1)  Is this feasible?
> 2)  Is it practical?
> 2a)  Is it solving my problem?
> 3)  I don't have to deploy gatekeepers, do I?
> 3a)  Locations sound an awful lot like zones - why
> can't my Polycom 
> H.323 video unit register to CM 4.1.3?  (why isn't
> CM a H.323 gatekeeper?)
> 4)  Will I set my locations bandwidth to Infinity
> and let the VGW decide 
>   what to do?
> 5)  Am I going to use AAR at all on CM?  Would I use
> a route pattern to 
> intercept the pilot number of Unity so that users
> that press "Messages" 
> get their call sent to the gateway instead of
> setting up a WAN/IP call?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________
Pinpoint customers who are looking for what you sell. 
http://searchmarketing.yahoo.com/

From cshaffer at gmail.com  Wed Oct 10 14:10:33 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Wed, 10 Oct 2007 14:10:33 -0400
Subject: [cisco-voip] SIP Integration
Message-ID: <001b01c80b68$dab88810$3900490a@acg.arlington.local>

I'm reading some docs on Cisco about what the minimum version of Call
Manager Express I will need to utilize SIP trunks. From what I am seeing it
states 3.4, 4.0, and 4.1. It states for SIP integration. Here is the page
I'm referring to:

http://www.cisco.com/en/US/docs/voice_ip_comm/unity/compatibility/matrix/cus
iptrunkmtx.html



My VAR is stating that 3.4 does not support SIP trunk functionality.
Basically I want to establish some SIP trunks for testing from our Call
Manger Express to SIP providers. Can anyone point to a definitive answer to
this question for me?

Thanks

Curt


From brian.gibson at stifel.com  Wed Oct 10 14:10:31 2007
From: brian.gibson at stifel.com (Gibson, Brian (Jersey City))
Date: Wed, 10 Oct 2007 13:10:31 -0500
Subject: [cisco-voip] Problems with web interface in 5.1.2
Message-ID: <5CC6FF682D32164BA5F2E830ABDF146001A5AF88@SNEXCHANGE02.stifelnet.stifel.local>

We are currently experiencing a lot of problems with accessing web pages
on our publisher.  We keep getting 500 internal server errors.

 

Is there a relatively simply of restarting the web server without
rebooting the box?

 

Sorry if this is a simple question.  We just started using  the 5.X
stuff. 


********************************************************************************
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From jason.aarons at us.didata.com  Wed Oct 10 14:27:51 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 10 Oct 2007 14:27:51 -0400
Subject: [cisco-voip] Blackbox JPMT152A with VG224
Message-ID: <C1FE15183DA37645BC0633BC604E44F0070BC2AD@USNAEXCH.na.didata.local>

I must have the wrong part number from Blackbox, I'm trying to connect a
VG224 to a 24-port (25-Pair Telco Amphenol) patch panel without punching
anything down, just plug and play.

 

The part number I bought was the JPMT152A which has 4 25-Pair Telco
connections, why the 3 extra Amphenol connectors on this part?

 

Anyone know a correct part number/vendor for a 25-Pair Amphenol 24-Port
Patch Panel from Blackbox? I'm hoping someone can suggest something they
used to avoid mis-ordering again.

 




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From jason.aarons at us.didata.com  Wed Oct 10 14:34:00 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 10 Oct 2007 14:34:00 -0400
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <22e6380b0710101040yd5b72bek3d640570c5781df@mail.gmail.com>
References: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com><f1b65f880710100931q46dbeb0fg6e4b558c11321fdf@mail.gmail.com>
	<22e6380b0710101040yd5b72bek3d640570c5781df@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0070BC2CB@USNAEXCH.na.didata.local>

If it's by design it would be in the release notes or documented
somewhere, ask the Cisco CSE to show where it's documented. If it's not
documented it's a bug.

 

 I would give your Cisco AM the case number and have them escalate, if
you don't have a Cisco AM ask to speak to the manager.  I'm with you, I
think it's a bug and they haven't removed CFwdAll from CCMUser.  

 

I've found sometimes it takes 2-3 tries to find the right person at TAC
that is informed; when you have thousands of employees (some outsourced)
you sometimes have a problem with knowledge dissemination.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
Sent: Wednesday, October 10, 2007 1:40 PM
To: Justin Steinberg
Cc: cisco voip list
Subject: Re: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?

 

Thats exactly how it went down for us.. The customer is NOT happy!

 

TAC is telling me that this was done purposely and that if we want it
back we need to "talk to our account team and file a PER (Product
Enhancement Request). This is not going to fly, I don;t even know how to
have that conversation with my customers... 

 

On 10/10/07, Justin Steinberg <jsteinberg at gmail.com> wrote: 

I can't believe this would be WAD, this must be a bug.

In fact, I logged into my 5.1.3 lab box and saw the call fwd settings 
there.  I then went to the Ent Params and saw that the param 'Show
Call Forwarding' was set to false, which was weird because I just
logged into ccmuser and saw the fwd settings.  I changed this param to 
'true' logged out of ccmuser and back in and now, like you, i dont see
the fwd settings.  I went back to ent params and set the param back to
false and still don't see fwd settings in ccmuser.





On 10/10/07, Ted Nugent <tednugent73 at gmail.com> wrote:
> I upgraded a customer to 5.13 for various bug fixes and now the
CFwdALL no
> longer shows in the CCMUser page. I opened a TAC case and I'm being
told 
> it's a WAD! CFwdAll has been in the CCMUser page for as long as I can
> remember and now it's gone and I'm expected to tell my customers that
Cisco
> decided you no longer need that... I've got many customers that use
this 
> religiously and if this is actually a WAD then they're stuck on 5.12
and
> lower? That just doesn't add up?
>
>
> Wes have you heard anything about this and/or can you shed some light
on it? 
> I understand that having all the CFwd options in CCMUser is not the
best
> solution but CFwdAll is really the only reason any of my customers use
it...
>
>
> _______________________________________________ 
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip 
>

 




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From Matt.Slaga at us.didata.com  Wed Oct 10 14:35:39 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 10 Oct 2007 14:35:39 -0400
Subject: [cisco-voip] Problems whit a SIP connection
In-Reply-To: <001601c80b58$91589b70$0b000a0a@ano>
References: <001601c80b58$91589b70$0b000a0a@ano>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0070BC2D6@USNAEXCH.na.didata.local>

Change this:

 

registrar dns:sip.cybercity.dk expires 3600

 

to

 

registrar dns:sip.cybercity.dk expires 360

 

 

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Katrine og Michael
Sent: Wednesday, October 10, 2007 12:14 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Problems whit a SIP connection

 

Hey,

 

Can somebody please help me?

 

I have configured CCME on a 2801 router and I have it connected to my SIP provider.

 

I?am having a problem.

When is is registred up against the SIP-erver everything  works fine, but after a while i get this message:

 


Oct 10 15:47:31.101: //-1/xxxxxxxxxxxx/SIP/Info/udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060

 

And then I can not get call in, but I can ring out the ccme.

 

I don?t know if is has anything to do whit my SIP-UA, som here it is:

 

sip-ua
 authentication username voi49495 password 12341

 no remote-party-id
 retry invite 4
 retry response 3
 retry bye 2
 retry cancel 2
 retry register 5
 timers register 250
 registrar dns:sip.cybercity.dk expires 3600

 

Please help me, and if you need more info, then just read

 

 

 

Michael from Denmark :-)

 

________________________________

Jeg beskyttes af den gratis SPAMfighter til privatbrugere.
Den har indtil videre sparet mig for at f? 691 spam-mails
Betalende brugere f?r ikke denne besked i deres e-mails.
Hent en gratis SPAMfighter her <http://www.spamfighter.com/lda> .




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From svoll.voip at gmail.com  Wed Oct 10 14:42:41 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 10 Oct 2007 11:42:41 -0700
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0070BC2CB@USNAEXCH.na.didata.local>
References: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
	<f1b65f880710100931q46dbeb0fg6e4b558c11321fdf@mail.gmail.com>
	<22e6380b0710101040yd5b72bek3d640570c5781df@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0070BC2CB@USNAEXCH.na.didata.local>
Message-ID: <f84a38d30710101142x273152e0u18047a66e6b054ec@mail.gmail.com>

you go........ I wish there was a way with Cisco tac that as a high
percentage of your tickets continue to be escalate your default TAC engineer
level would also go up.

<rant>

I'm tired of continuing to get the new guy and the tickets take forever
because the new guy trys and trys then transfers you to someone else then
they transfer you to someone who actually knows something.

</rant>

People who only call for the hard stuff should be able to get to a teir
three person automatically.

just my two cents.

Scott



On 10/10/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  If it's by design it would be in the release notes or documented
> somewhere, ask the Cisco CSE to show where it's documented. If it's not
> documented it's a bug.
>
>
>
>  I would give your Cisco AM the case number and have them escalate, if you
> don't have a Cisco AM ask to speak to the manager.  I'm with you, I think
> it's a bug and they haven't removed CFwdAll from CCMUser.
>
>
>
> I've found sometimes it takes 2-3 tries to find the right person at TAC
> that is informed; when you have thousands of employees (some outsourced) you
> sometimes have a problem with knowledge dissemination.
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ted Nugent
> *Sent:* Wednesday, October 10, 2007 1:40 PM
> *To:* Justin Steinberg
> *Cc:* cisco voip list
> *Subject:* Re: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
>
>
>
> Thats exactly how it went down for us.. The customer is NOT happy!
>
>
>
> TAC is telling me that this was done purposely and that if we want it back
> we need to "talk to our account team and file a PER (Product Enhancement
> Request). This is not going to fly, I don;t even know how to have that
> conversation with my customers...
>
>
>
> On 10/10/07, *Justin Steinberg* <jsteinberg at gmail.com> wrote:
>
> I can't believe this would be WAD, this must be a bug.
>
> In fact, I logged into my 5.1.3 lab box and saw the call fwd settings
> there.  I then went to the Ent Params and saw that the param 'Show
> Call Forwarding' was set to false, which was weird because I just
> logged into ccmuser and saw the fwd settings.  I changed this param to
> 'true' logged out of ccmuser and back in and now, like you, i dont see
> the fwd settings.  I went back to ent params and set the param back to
> false and still don't see fwd settings in ccmuser.
>
>
>
>
>
> On 10/10/07, Ted Nugent <tednugent73 at gmail.com> wrote:
> > I upgraded a customer to 5.13 for various bug fixes and now the CFwdALL
> no
> > longer shows in the CCMUser page. I opened a TAC case and I'm being told
>
> > it's a WAD! CFwdAll has been in the CCMUser page for as long as I can
> > remember and now it's gone and I'm expected to tell my customers that
> Cisco
> > decided you no longer need that? I've got many customers that use this
> > religiously and if this is actually a WAD then they're stuck on 5.12 and
> > lower? That just doesn't add up?
> >
> >
> > Wes have you heard anything about this and/or can you shed some light on
> it?
> > I understand that having all the CFwd options in CCMUser is not the best
> > solution but CFwdAll is really the only reason any of my customers use
> it?
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck..nether.net/mailman/listinfo/cisco-voip
> <https://puck.nether.net/mailman/listinfo/cisco-voip>
> >
>
>
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jsteinberg at gmail.com  Wed Oct 10 14:47:43 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 10 Oct 2007 14:47:43 -0400
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <f84a38d30710101142x273152e0u18047a66e6b054ec@mail.gmail.com>
References: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
	<f1b65f880710100931q46dbeb0fg6e4b558c11321fdf@mail.gmail.com>
	<22e6380b0710101040yd5b72bek3d640570c5781df@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0070BC2CB@USNAEXCH.na.didata.local>
	<f84a38d30710101142x273152e0u18047a66e6b054ec@mail.gmail.com>
Message-ID: <f1b65f880710101147g58fabec8x8b4f5ec02a96ef0e@mail.gmail.com>

pass your ccie ;)

On 10/10/07, Scott Voll <svoll.voip at gmail.com> wrote:
> you go........ I wish there was a way with Cisco tac that as a high
> percentage of your tickets continue to be escalate your default TAC engineer
> level would also go up.
>
> <rant>
>
> I'm tired of continuing to get the new guy and the tickets take forever
> because the new guy trys and trys then transfers you to someone else then
> they transfer you to someone who actually knows something.
>
> </rant>
>
> People who only call for the hard stuff should be able to get to a teir
> three person automatically.
>
> just my two cents.
>
> Scott
>
>
>
> On 10/10/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> >
> >
> >
> >
> >
> > If it's by design it would be in the release notes or documented
> somewhere, ask the Cisco CSE to show where it's documented. If it's not
> documented it's a bug.
> >
> >
> >
> >  I would give your Cisco AM the case number and have them escalate, if you
> don't have a Cisco AM ask to speak to the manager.  I'm with you, I think
> it's a bug and they haven't removed CFwdAll from CCMUser.
> >
> >
> >
> > I've found sometimes it takes 2-3 tries to find the right person at TAC
> that is informed; when you have thousands of employees (some outsourced) you
> sometimes have a problem with knowledge dissemination.
> >
> >
> >
> >
> > From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Ted Nugent
> > Sent: Wednesday, October 10, 2007 1:40 PM
> > To: Justin Steinberg
> > Cc: cisco voip list
> > Subject: Re: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
> >
> >
> >
> >
> >
> > Thats exactly how it went down for us.. The customer is NOT happy!
> >
> >
> >
> >
> >
> > TAC is telling me that this was done purposely and that if we want it back
> we need to "talk to our account team and file a PER (Product Enhancement
> Request). This is not going to fly, I don;t even know how to have that
> conversation with my customers...
> >
> >
> >
> >
> > On 10/10/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
> >
> > I can't believe this would be WAD, this must be a bug.
> >
> > In fact, I logged into my 5.1.3 lab box and saw the call fwd settings
> > there.  I then went to the Ent Params and saw that the param 'Show
> > Call Forwarding' was set to false, which was weird because I just
> > logged into ccmuser and saw the fwd settings.  I changed this param to
> > 'true' logged out of ccmuser and back in and now, like you, i dont see
> > the fwd settings.  I went back to ent params and set the param back to
> > false and still don't see fwd settings in ccmuser.
> >
> >
> >
> >
> >
> > On 10/10/07, Ted Nugent <tednugent73 at gmail.com > wrote:
> > > I upgraded a customer to 5.13 for various bug fixes and now the CFwdALL
> no
> > > longer shows in the CCMUser page. I opened a TAC case and I'm being told
> > > it's a WAD! CFwdAll has been in the CCMUser page for as long as I can
> > > remember and now it's gone and I'm expected to tell my customers that
> Cisco
> > > decided you no longer need that? I've got many customers that use this
> > > religiously and if this is actually a WAD then they're stuck on 5.12 and
> > > lower? That just doesn't add up?
> > >
> > >
> > > Wes have you heard anything about this and/or can you shed some light on
> it?
> > > I understand that having all the CFwd options in CCMUser is not the best
> > > solution but CFwdAll is really the only reason any of my customers use
> it?
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck..nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> >
> >
> > ________________________________
>
> >
> >
> >
> > Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jeff at ocjtech.us  Wed Oct 10 15:22:19 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Wed, 10 Oct 2007 14:22:19 -0500
Subject: [cisco-voip] Blackbox JPMT152A with VG224
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0070BC2AD@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0070BC2AD@USNAEXCH.na.didata.local>
Message-ID: <1192044139.3347.7.camel@lt21223.campus.dmacc.edu>

On Wed, 2007-10-10 at 14:27 -0400, Jason Aarons (US) wrote:
> I must have the wrong part number from Blackbox, I?m trying to connect
> a VG224 to a 24-port (25-Pair Telco Amphenol) patch panel without
> punching anything down, just plug and play.
>
> The part number I bought was the JPMT152A which has 4 25-Pair Telco
> connections, why the 3 extra Amphenol connectors on this part?

Because the JPMT152A is for 24 Cat 5E connections, which have 4 pairs
per RJ45 jack.

> Anyone know a correct part number/vendor for a 25-Pair Amphenol
> 24-Port Patch Panel from Blackbox? I?m hoping someone can suggest
> something they used to avoid mis-ordering again.
> 

I didn't see anything at Blackbox's site, but Panduit has this:

http://www.panduit.com/search/product_details.asp?N=5000001+940+3001514+&Ne=940&region=USA&recName=VP24382TV25Y&Nao=100

Jeff
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From tednugent73 at gmail.com  Wed Oct 10 15:22:37 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Wed, 10 Oct 2007 15:22:37 -0400
Subject: [cisco-voip] Blackbox JPMT152A with VG224
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0070BC2AD@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0070BC2AD@USNAEXCH.na.didata.local>
Message-ID: <22e6380b0710101222r758eeda9h9565b00f3cff68f6@mail.gmail.com>

I just went through this a few weeks ago. They don't have anything in thier
catalog thats rack mountable but they do have a custom PN of CBCC303808
(Telco patch panel, 24 portx2 wire(RJ11) to 50pin female Telco). Took about
3 weeks to get them and don't quote me on this but I think they cost about
$80 each.

I've made my own in the past but I'm no wiz with a punch tool so this time I
just had them ordered...




On 10/10/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  I must have the wrong part number from Blackbox, I'm trying to connect a
> VG224 to a 24-port (25-Pair Telco Amphenol) patch panel without punching
> anything down, just plug and play.
>
>
>
> The part number I bought was the JPMT152A which has 4 25-Pair Telco
> connections, why the 3 extra Amphenol connectors on this part?
>
>
>
> Anyone know a correct part number/vendor for a 25-Pair Amphenol 24-Port
> Patch Panel from Blackbox? I'm hoping someone can suggest something they
> used to avoid mis-ordering again.
>
>
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From db7td at gmx.de  Wed Oct 10 15:22:10 2007
From: db7td at gmx.de (Dietmar)
Date: Wed, 10 Oct 2007 21:22:10 +0200
Subject: [cisco-voip] Access restrictions AXL/SOAP CCM4.2
In-Reply-To: <779799.28413.qm@web50302.mail.re2.yahoo.com>
References: <779799.28413.qm@web50302.mail.re2.yahoo.com>
Message-ID: <200710102122.10342.db7td@gmx.de>

Solution:

This is a known bug (CSCsc59620) that has been fixed in 4.2(1). Setting the 
system parameter "Enable AXL Access Levels" to TRUE denies unauthorized 
access to AXL. The default is, for whatever reason, FALSE!


Dietmar



On Wednesday 10 October 2007 03:22:13 Charles Ragan, Jr. wrote:
> Nope - it's true.  With 4.2 and later you can restrict using MLA.
>
>   With previous versions of CM, we chose to restrict what ip address space
> could access the soap/axl interface within IIS.
>
>   Charles
>
> Dietmar <db7td at gmx.de> wrote:
>   Hi all,
>
> while playing a little with the AXL/SOAP interface of CCM 4.2, I realized
> that every authenticated user is able to read and modify (!) ALL settings,
> even these of phones/lines that are not assigned to that user.
>
> I would have expected that a user can only modify his own settings with the
> AXL/SOAP interface. Hopefully, this is true and someone has an idea what is
> wrong with my CCM settings ;-). Any hints?
>
>
> Thanks,
> Dietmar
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> Charles Ragan Jr., Technology Consultant
> Principal Consultant - CCIE #1764
> Cell Phone - 336-442-4361
> http://www.geocities.com/ciscojock2002/


From brian.gibson at stifel.com  Wed Oct 10 17:10:58 2007
From: brian.gibson at stifel.com (Gibson, Brian (Jersey City))
Date: Wed, 10 Oct 2007 16:10:58 -0500
Subject: [cisco-voip] Problems with web interface in 5.1.2
In-Reply-To: <5CC6FF682D32164BA5F2E830ABDF146001A5AF88@SNEXCHANGE02.stifelnet.stifel.local>
References: <5CC6FF682D32164BA5F2E830ABDF146001A5AF88@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <5CC6FF682D32164BA5F2E830ABDF146001A5B062@SNEXCHANGE02.stifelnet.stifel.local>

For anyone who is interested the way to restart Tomcat is to SSH into
the server and enter

 

"utils service restart Cisco Tomcat"

 

Apparently that is the only way to restart Tomcat.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gibson, Brian
(Jersey City)
Sent: Wednesday, October 10, 2007 2:11 PM
To: cisco voip list
Subject: [cisco-voip] Problems with web interface in 5.1.2

 

We are currently experiencing a lot of problems with accessing web pages
on our publisher.  We keep getting 500 internal server errors.

 

Is there a relatively simply of restarting the web server without
rebooting the box?

 

Sorry if this is a simple question.  We just started using  the 5.X
stuff. 

************************************************************************
********

All electronic messages sent and received by Stifel Nicolaus

Associates are subject to review by Stifel Nicolaus. Stifel Nicolaus

may retain and reproduce electronic messages for state, federal, or

other regulatory agencies as required by applicable law. 

IMPORTANT: Please do not use e-mail to request or authorize the

purchase or sale of any security or commodity, send fund transfer

instructions, or otherwise conduct any securities transactions. Any 

requests, orders, instructions, or time-sensitive messages sent by

e-mail cannot be accepted or processed by Stifel Nicolaus. The 

accuracy of any information sent by Stifel Nicolaus through e-mail 

cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.

Stifel, Nicolaus & Company, Incorporated

Member NYSE & SIPC

Headquarters: 501 N. Broadway, St. Louis, MO 63102

314-342-2000

************************************************************************
********

 

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From ciscojock2002 at yahoo.com  Wed Oct 10 16:57:35 2007
From: ciscojock2002 at yahoo.com (Charles Ragan, Jr.)
Date: Wed, 10 Oct 2007 13:57:35 -0700 (PDT)
Subject: [cisco-voip] Access restrictions AXL/SOAP CCM4.2
In-Reply-To: <200710102122.10342.db7td@gmx.de>
Message-ID: <618435.45146.qm@web50304.mail.re2.yahoo.com>

True - but you'll still need to integrate to MLA for your 'allowed' users of soap/axl.....
   
  Charles

Dietmar <db7td at gmx.de> wrote:
  Solution:

This is a known bug (CSCsc59620) that has been fixed in 4.2(1). Setting the 
system parameter "Enable AXL Access Levels" to TRUE denies unauthorized 
access to AXL. The default is, for whatever reason, FALSE!


Dietmar



On Wednesday 10 October 2007 03:22:13 Charles Ragan, Jr. wrote:
> Nope - it's true. With 4.2 and later you can restrict using MLA.
>
> With previous versions of CM, we chose to restrict what ip address space
> could access the soap/axl interface within IIS.
>
> Charles
>
> Dietmar wrote:
> Hi all,
>
> while playing a little with the AXL/SOAP interface of CCM 4.2, I realized
> that every authenticated user is able to read and modify (!) ALL settings,
> even these of phones/lines that are not assigned to that user.
>
> I would have expected that a user can only modify his own settings with the
> AXL/SOAP interface. Hopefully, this is true and someone has an idea what is
> wrong with my CCM settings ;-). Any hints?
>
>
> Thanks,
> Dietmar
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> Charles Ragan Jr., Technology Consultant
> Principal Consultant - CCIE #1764
> Cell Phone - 336-442-4361
> http://www.geocities.com/ciscojock2002/

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



Charles Ragan Jr., Technology Consultant
Principal Consultant - CCIE #1764
Cell Phone - 336-442-4361
http://www.geocities.com/ciscojock2002/
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From Kathy.Tarkington at AGG.com  Wed Oct 10 18:08:31 2007
From: Kathy.Tarkington at AGG.com (Tarkington, Kathy L.)
Date: Wed, 10 Oct 2007 18:08:31 -0400
Subject: [cisco-voip] Contact Center Solution for Wallboard
Message-ID: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>

I am looking for a solution that will give my Help Desk a wallboard
showing agents logged in, agent status, calls in queue, etc.
Recommendations?
 
Thanks!

Note:
This message and any attachments from the law firm Arnall Golden Gregory LLP may contain CONFIDENTIAL and legally protected information.  If you are not the addressee and an intended recipient, please do not read, copy, use or disclose this communication to others; also, please notify the sender by replying to this message, and then delete it from your system.  Thank you.
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From svoll.voip at gmail.com  Wed Oct 10 18:25:09 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 10 Oct 2007 15:25:09 -0700
Subject: [cisco-voip] Contact Center Solution for Wallboard
In-Reply-To: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>
References: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>
Message-ID: <f84a38d30710101525u3c74ccccu2bd8a09fc4db43d5@mail.gmail.com>

Real time monitoring would work so long as no one logged out.

Scott


On 10/10/07, Tarkington, Kathy L. <Kathy.Tarkington at agg.com> wrote:
>
>  I am looking for a solution that will give my Help Desk a wallboard
> showing agents logged in, agent status, calls in queue, etc.
> Recommendations?
>
> Thanks!
> ------------------------------
>  *Note:*
>
> This message and any attachments from the law firm Arnall Golden Gregory
> LLP may contain CONFIDENTIAL and legally protected information.  If you
> are not the addressee and an intended recipient, please do not read, copy,
> use or disclose this communication to others;  also, please notify the
> sender by replying to this message, and then delete it from your system.
> Thank you.
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From SCASPER at mtb.com  Wed Oct 10 18:38:48 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Wed, 10 Oct 2007 18:38:48 -0400
Subject: [cisco-voip]  Software MTP will not register
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0070BC2D6@USNAEXCH.na.didata.local>
References: <001601c80b58$91589b70$0b000a0a@ano>
	<C1FE15183DA37645BC0633BC604E44F0070BC2D6@USNAEXCH.na.didata.local>
Message-ID: <470D1C38.9874.0039.0@mtb.com>

Can not get an MTP to register with CCM  4.1.3sr5d.  Router IOS is 12.4.11T.  Need the MTP to provide Hold for phones using SIP trunking thru a IPIPGW. I have made sure the name in the router and CCM match. What am I missing? 
 
Getting these errors in debug:
 
*Oct 10 16:26:37.395: sym_xapp_xlate_rpm_device_type: Unsupported appl type 1for ccm verion 2
*Oct 10 16:26:37.395: sccp_send_register_msg: SCCP appl service 1 is xlated to an invalid device type 2
 
Show sccp 
 
Software MTP Oper State: ACTIVE_IN_PROGRESS - Cause Code: NOT_REGIS_WITH_CCM
Active Call Manager: 10.95.3.18, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 2
Reported Max Streams: 48, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period:
 
Here is a config snip
 
sccp local Loopback0
sccp ccm 10.95.3.18 identifier 1 
sccp ccm 10.91.4.10 identifier 2 
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 2 register IPTRUNK_MTP
 
!
dspfarm profile 2 mtp
 codec g711ulaw
 maximum sessions software 24
 associate application SCCP
 
Thanks for any advice!
 
 
 
 
 
 


Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee.  If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission.  If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy.  This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
There are risks associated with the use of electronic transmission.  The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
************************************
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From jason.aarons at us.didata.com  Wed Oct 10 18:50:12 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 10 Oct 2007 18:50:12 -0400
Subject: [cisco-voip] Contact Center Solution for Wallboard
In-Reply-To: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>
References: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0071023C7@USNAEXCH.na.didata.local>

I've been happy with Symon, not cheap but their Digital Messaging is
pretty slick, and the NetBrite can be read from afar, they also have
software that can run in the PC's System Notification Area (where you
clock is) and pop-up on thresholds (similar to Outlook).

 

http://www.symon.com/

 

The Real Time Reports font is too small for a 40" LCD TV display, you
can write your own reports/queries, search for wallboardUser on CCO.

 

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tarkington,
Kathy L.
Sent: Wednesday, October 10, 2007 6:09 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Contact Center Solution for Wallboard

 

I am looking for a solution that will give my Help Desk a wallboard
showing agents logged in, agent status, calls in queue, etc.
Recommendations?

 

Thanks!

________________________________

Note:

This message and any attachments from the law firm Arnall Golden Gregory
LLP may contain CONFIDENTIAL and legally protected information.  If you
are not the addressee and an intended recipient, please do not read,
copy, use or disclose this communication to others;  also, please notify
the sender by replying to this message, and then delete it from your
system.  Thank you. 

________________________________




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From asobihoudai at yahoo.com  Wed Oct 10 18:18:30 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 10 Oct 2007 15:18:30 -0700 (PDT)
Subject: [cisco-voip] Contact Center Solution for Wallboard
In-Reply-To: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>
Message-ID: <942373.60673.qm@web57205.mail.re3.yahoo.com>

Hi Kathy,

This might be a bazooka to kill a mosquito but this
might be the sort of thing you're looking for.

http://tinyurl.com/3c8ubh

Cheers,
Paul

--- "Tarkington, Kathy L." <Kathy.Tarkington at AGG.com>
wrote:

> I am looking for a solution that will give my Help
> Desk a wallboard
> showing agents logged in, agent status, calls in
> queue, etc.
> Recommendations?
>  
> Thanks!
> 
> Note:
> This message and any attachments from the law firm
> Arnall Golden Gregory LLP may contain CONFIDENTIAL
> and legally protected information.  If you are not
> the addressee and an intended recipient, please do
> not read, copy, use or disclose this communication
> to others; also, please notify the sender by
> replying to this message, and then delete it from
> your system.  Thank you.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
____________________________________________________________________________________
Need a vacation? Get great deals
to amazing places on Yahoo! Travel.
http://travel.yahoo.com/

From BurwellJ at firstcharter.com  Wed Oct 10 19:25:08 2007
From: BurwellJ at firstcharter.com (Jason Burwell)
Date: Wed, 10 Oct 2007 19:25:08 -0400
Subject: [cisco-voip] Contact Center Solution for Wallboard
References: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>
Message-ID: <697161D0107BAD40AFE9A86B06509AC9101C82F0@FCEXCLUSTER.FirstCharter.com>

I have used Spectrum in the past. They make a good product.
 
http://www.specorp.com/
 
Jason


________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Tarkington, Kathy L.
Sent: Wed 10/10/2007 6:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Contact Center Solution for Wallboard


I am looking for a solution that will give my Help Desk a wallboard showing agents logged in, agent status, calls in queue, etc.  Recommendations?
 
Thanks!
________________________________

Note:

This message and any attachments from the law firm Arnall Golden Gregory LLP may contain CONFIDENTIAL and legally protected information.  If you are not the addressee and an intended recipient, please do not read, copy, use or disclose this communication to others;  also, please notify the sender by replying to this message, and then delete it from your system.  Thank you. 


________________________________




-----------------------------------------
*******************************************************************
***
The information transmitted is intended only for the person or
entity to which it is
addressed and may contain confidential and/or privileged material.
Any review,
retransmission, dissemination or other use of, or taking of any
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From myciscovoip at gmail.com  Thu Oct 11 05:21:41 2007
From: myciscovoip at gmail.com (Patrick Fischer)
Date: Thu, 11 Oct 2007 11:21:41 +0200
Subject: [cisco-voip] Software MTP will not register
In-Reply-To: <470D1C38.9874.0039.0@mtb.com>
References: <001601c80b58$91589b70$0b000a0a@ano>
	<C1FE15183DA37645BC0633BC604E44F0070BC2D6@USNAEXCH.na.didata.local>
	<470D1C38.9874.0039.0@mtb.com>
Message-ID: <fb5f789e0710110221m3aa3e45dh5dbd9ccc93feccfa@mail.gmail.com>

Hi Steven,

try to use "sccp ccm  <http://10.95.3.18/>x.x.x.x identifier x version 4.1"
instead. Otherwise the gateway will use the default which is AFAIK CCM
3.1which did not now anything about software MTP.

Regards,
Patrick

Change your

2007/10/11, STEVEN CASPER <SCASPER at mtb.com>:
>
>  Can not get an MTP to register with CCM  4.1.3sr5d.  Router IOS is
> 12.4.11T.  Need the MTP to provide Hold for phones using SIP trunking thru
> a IPIPGW. I have made sure the name in the router and CCM match. What am I
> missing?
>
> Getting these errors in debug:
>
> *Oct 10 16:26:37.395: sym_xapp_xlate_rpm_device_type: Unsupported appl
> type 1for ccm verion 2
> *Oct 10 16:26:37.395: sccp_send_register_msg: SCCP appl service 1 is
> xlated to an invalid device type 2
>
> Show sccp
>
> Software MTP Oper State: ACTIVE_IN_PROGRESS - Cause Code:
> NOT_REGIS_WITH_CCM
> Active Call Manager: 10.95.3.18, Port Number: 2000
> TCP Link Status: CONNECTED, Profile Identifier: 2
> Reported Max Streams: 48, Reported Max OOS Streams: 0
> Supported Codec: g711ulaw, Maximum Packetization Period: 30
> Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
> Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
> Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
> Period:
>
> Here is a config snip
>
> sccp local Loopback0
> sccp ccm 10.95.3.18 identifier 1
> sccp ccm 10.91.4.10 identifier 2
> sccp
> !
> sccp ccm group 1
>  associate ccm 1 priority 1
>  associate ccm 2 priority 2
>  associate profile 2 register IPTRUNK_MTP
>
> !
> dspfarm profile 2 mtp
>  codec g711ulaw
>  maximum sessions software 24
>  associate application SCCP
>
> Thanks for any advice!
>
>
>
>
>
>
>
>
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
>
> ************************************
> This email may contain privileged and/or confidential information that is
> intended solely for the use of the addressee. If you are not the intended
> recipient or entity, you are strictly prohibited from disclosing, copying,
> distributing or using any of the information contained in the transmission.
> If you received this communication in error, please contact the sender
> immediately and destroy the material in its entirety, whether electronic or
> hard copy. This communication may contain nonpublic personal information
> about consumers subject to the restrictions of the Gramm-Leach-Bliley Act
> and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or
> disclose such information for any purpose other than to provide the services
> for which you are receiving the information.
> There are risks associated with the use of electronic transmission. The
> sender of this information does not control the method of transmittal or
> service providers and assumes no duty or obligation for the security,
> receipt, or third party interception of this transmission.
> ************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From mark.mills at baesystems.com  Thu Oct 11 06:39:15 2007
From: mark.mills at baesystems.com (MILLS, Mark)
Date: Thu, 11 Oct 2007 20:09:15 +0930
Subject: [cisco-voip] VG224 buzzing/humming on incoming calls only
Message-ID: <54DD8B39132EFA4293A5C0565004E54608B38D58@SBEMAIL1.au.baesystems.com>

Hi All,

I was hoping I might be able to get some help with the setup of a VG224 that I am doing.  It is connected to CCM6 cluster.  I can place calls FROM the analog phones connected to the ports, and it all works properly with two way audio.

When making calls TO the analog lines on the VG224, the calling party simply hears a loud buzzing/humming noise and nothing else.  The analog user can hear nothing.

This happens for incoming PSTN calls and calls to other extensions to the VG224 connected devices.

I have not setup a VG224 for incoming calls before, so there is obviously something I have missed.  Below is the relevant config parts on the device.  Any help is appreciated!



stcapp ccm-group 1
stcapp
!
voice-port 2/0
 ren 3
 idle-voltage low
 disc_pi_off
 input gain 10
 output attenuation 10
 playout-delay minimum low
 cptone AU
 timing digit 53
 bearer-cap Speech
 station-id number XXXXXXXXXX
 caller-id enable
!
sccp local FastEthernet0/0
sccp ccm 10.194.3.10 identifier 20 version 4.1 
sccp ccm 10.194.4.10 identifier 10 version 4.1 
sccp      
!
sccp ccm group 1
 associate ccm 10 priority 1
 associate ccm 20 priority 2
 switchback method graceful
!
dial-peer voice XXXXXXXXXX pots
 service stcapp
 port 2/23
!

Cheers,
   Mark
"Warning:
The information contained in this email and any attached files is
confidential to BAE Systems Australia. If you are not the intended
recipient, any use, disclosure or copying of this email or any
attachments is expressly prohibited.  If you have received this email
in error, please notify us immediately. VIRUS: Every care has been
taken to ensure this email and its attachments are virus free,
however, any loss or damage incurred in using this email is not the
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From tednugent73 at gmail.com  Thu Oct 11 08:34:40 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 11 Oct 2007 08:34:40 -0400
Subject: [cisco-voip] VG224 buzzing/humming on incoming calls only
In-Reply-To: <54DD8B39132EFA4293A5C0565004E54608B38D58@SBEMAIL1.au.baesystems.com>
References: <54DD8B39132EFA4293A5C0565004E54608B38D58@SBEMAIL1.au.baesystems.com>
Message-ID: <22e6380b0710110534r36d63966s8f400247079da867@mail.gmail.com>

Is the vg224 grounded?



On 10/11/07, MILLS, Mark <mark.mills at baesystems.com> wrote:
>
>  Hi All,
>
> I was hoping I might be able to get some help with the setup of a VG224
> that I am doing.  It is connected to CCM6 cluster.  I can place calls FROM
> the analog phones connected to the ports, and it all works properly with two
> way audio.
>
> When making calls TO the analog lines on the VG224, the calling party
> simply hears a loud buzzing/humming noise and nothing else.  The analog user
> can hear nothing.
>
> This happens for incoming PSTN calls and calls to other extensions to the
> VG224 connected devices.
>
> I have not setup a VG224 for incoming calls before, so there is obviously
> something I have missed.  Below is the relevant config parts on the device.
> Any help is appreciated!
>
>
>
> stcapp ccm-group 1
> stcapp
> !
> voice-port 2/0
>  ren 3
>  idle-voltage low
>  disc_pi_off
>  input gain 10
>  output attenuation 10
>  playout-delay minimum low
>  cptone AU
>  timing digit 53
>  bearer-cap Speech
>  station-id number XXXXXXXXXX
>  caller-id enable
> !
> sccp local FastEthernet0/0
> sccp ccm 10.194.3.10 identifier 20 version 4.1
> sccp ccm 10.194.4.10 identifier 10 version 4.1
> sccp
> !
> sccp ccm group 1
>  associate ccm 10 priority 1
>  associate ccm 20 priority 2
>  switchback method graceful
> !
> dial-peer voice XXXXXXXXXX pots
>  service stcapp
>  port 2/23
> !
>
> Cheers,
>    Mark
>
> "Warning:
> The information contained in this email and any attached files is
> confidential to BAE Systems Australia. If you are not the intended
> recipient, any use, disclosure or copying of this email or any
> attachments is expressly prohibited.  If you have received this email
> in error, please notify us immediately. VIRUS: Every care has been
> taken to ensure this email and its attachments are virus free,
> however, any loss or damage incurred in using this email is not the
> sender's responsibility.  It is your responsibility to ensure virus
> checks are completed before installing any data sent in this email to
> your computer."
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From cheesevoice at gmail.com  Thu Oct 11 09:01:44 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Thu, 11 Oct 2007 08:01:44 -0500
Subject: [cisco-voip] VG224 buzzing/humming on incoming calls only
In-Reply-To: <54DD8B39132EFA4293A5C0565004E54608B38D58@SBEMAIL1.au.baesystems.com>
References: <54DD8B39132EFA4293A5C0565004E54608B38D58@SBEMAIL1.au.baesystems.com>
Message-ID: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com>

I just had this happen with a VG248.  Same exact issue, turned out the
analog lines had been victims of some lightning, the Telco replaced teh
facilities for the analog lines and the crackling went away... Hope that
helps.

On 10/11/07, MILLS, Mark <mark.mills at baesystems.com> wrote:
>
>  Hi All,
>
> I was hoping I might be able to get some help with the setup of a VG224
> that I am doing.  It is connected to CCM6 cluster.  I can place calls FROM
> the analog phones connected to the ports, and it all works properly with two
> way audio.
>
> When making calls TO the analog lines on the VG224, the calling party
> simply hears a loud buzzing/humming noise and nothing else.  The analog user
> can hear nothing.
>
> This happens for incoming PSTN calls and calls to other extensions to the
> VG224 connected devices.
>
> I have not setup a VG224 for incoming calls before, so there is obviously
> something I have missed.  Below is the relevant config parts on the device.
> Any help is appreciated!
>
>
>
> stcapp ccm-group 1
> stcapp
> !
> voice-port 2/0
>  ren 3
>  idle-voltage low
>  disc_pi_off
>  input gain 10
>  output attenuation 10
>  playout-delay minimum low
>  cptone AU
>  timing digit 53
>  bearer-cap Speech
>  station-id number XXXXXXXXXX
>  caller-id enable
> !
> sccp local FastEthernet0/0
> sccp ccm 10.194.3.10 identifier 20 version 4.1
> sccp ccm 10.194.4.10 identifier 10 version 4.1
> sccp
> !
> sccp ccm group 1
>  associate ccm 10 priority 1
>  associate ccm 20 priority 2
>  switchback method graceful
> !
> dial-peer voice XXXXXXXXXX pots
>  service stcapp
>  port 2/23
> !
>
> Cheers,
>    Mark
>
> "Warning:
> The information contained in this email and any attached files is
> confidential to BAE Systems Australia. If you are not the intended
> recipient, any use, disclosure or copying of this email or any
> attachments is expressly prohibited.  If you have received this email
> in error, please notify us immediately. VIRUS: Every care has been
> taken to ensure this email and its attachments are virus free,
> however, any loss or damage incurred in using this email is not the
> sender's responsibility.  It is your responsibility to ensure virus
> checks are completed before installing any data sent in this email to
> your computer."
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Max.Caines at wlv.ac.uk  Thu Oct 11 09:09:17 2007
From: Max.Caines at wlv.ac.uk (Caines, Max)
Date: Thu, 11 Oct 2007 14:09:17 +0100
Subject: [cisco-voip] Problem bringing up L2
Message-ID: <D7C66F5492F56948BC9244768A3283660271F127@exchange03.unv.wlv.ac.uk>

Hi
 
I don't normally deal with VoIP, so my apologies if I make simple errors. We
have a 3725 router with a VWIC handling calls for one of our sites, talking
to CallManager 4.1. It stopped working the other day, and after getting in
the telco, who couldn't find anything wrong, and our Cisco support firm
(ditto) the latter sent me a couple of URLs, one of which was to an article
on this list. It pointed me to  temporary fix, but not a permanent one. 
 
The problem is that when the router boots or the circuit is brought up, the
L2 link from the router to the telco switch won't come up. Below is the
output from "show isdn status":
 
Global ISDN Switchtype = primary-net5
 
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not
apply
 
ISDN Serial0/0:15 interface
        dsl 0, interface ISDN Switchtype = primary-net5
        L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003 
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
    Layer 3 Status:
        0 Active Layer 3 Call(s)
    Active dsl 0 CCBs = 0
    The Free Channel Mask:  0xFFFF7FFF
    Number of L2 Discards = 0, L2 Session ID = 1
    Total Allocated ISDN CCBs = 0

What I found was that I could bring the link up by doing:
 
conf t
int ser 0/0:15
no isdn bind-l3 ccm-manager
shut
no shut
isdn bind-l3 ccm-manager
 
After that the status is:
 
Global ISDN Switchtype = primary-net5
ISDN Serial0/0:15 interface
        dsl 0, interface ISDN Switchtype = primary-net5
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
    Layer 3 Status:
        0 Active Layer 3 Call(s)
    Active dsl 0 CCBs = 0
    The Free Channel Mask:  0xFFFF7FFF
    Number of L2 Discards = 0, L2 Session ID = 2
    Total Allocated ISDN CCBs = 0

However, if I reboot the router it goes back to the previous state. This
clearly isn't a good situation. I've upgraded the router to 12.3(14)T7, the
latest in its train, but this makes no difference. Is this a Cisco bug, or
is there something else I should try? The config is attached, if it helps
 
Thanks
 
Max Caines
IT Services, University of Wolverhampton
Wolverhampton, West Midlands WV1 1SB
Tel: 01902 322245 Fax: 01902 322777
 
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From aman.chugh at gmail.com  Thu Oct 11 09:53:20 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Thu, 11 Oct 2007 19:23:20 +0530
Subject: [cisco-voip] Problem bringing up L2
In-Reply-To: <D7C66F5492F56948BC9244768A3283660271F127@exchange03.unv.wlv.ac.uk>
References: <D7C66F5492F56948BC9244768A3283660271F127@exchange03.unv.wlv.ac.uk>
Message-ID: <d4c697ad0710110653g57099e44s7a3cdfcfa234049d@mail.gmail.com>

Doing a no mgcp and mgcp would bring up layer 2 on your router, I have ran
across this a couple of times. Didnt get deep into it to find out a
permanent fix.

Aman


On 10/11/07, Caines, Max <Max.Caines at wlv.ac.uk> wrote:
>
>  Hi
>
> I don't normally deal with VoIP, so my apologies if I make simple errors.
> We have a 3725 router with a VWIC handling calls for one of our sites,
> talking to CallManager 4.1. It stopped working the other day, and after
> getting in the telco, who couldn't find anything wrong, and our Cisco
> support firm (ditto) the latter sent me a couple of URLs, one of which was
> to an article on this list. It pointed me to  temporary fix, but not a
> permanent one.
>
> The problem is that when the router boots or the circuit is brought up,
> the L2 link from the router to the telco switch won't come up. Below is the
> output from "show isdn status":
>
> Global ISDN Switchtype = primary-net5
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may
> not apply
>
> ISDN Serial0/0:15 interface
>         dsl 0, interface ISDN Switchtype = primary-net5
>         L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
>     Layer 1 Status:
>         ACTIVE
>     Layer 2 Status:
>         TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
>     Layer 3 Status:
>         0 Active Layer 3 Call(s)
>     Active dsl 0 CCBs = 0
>     The Free Channel Mask:  0xFFFF7FFF
>     Number of L2 Discards = 0, L2 Session ID = 1
>     Total Allocated ISDN CCBs = 0
> What I found was that I could bring the link up by doing:
>
>  conf t
> int ser 0/0:15
> no isdn bind-l3 ccm-manager
> shut
> no shut
> isdn bind-l3 ccm-manager
>
> After that the status is:
>
> Global ISDN Switchtype = primary-net5
> ISDN Serial0/0:15 interface
>         dsl 0, interface ISDN Switchtype = primary-net5
>     Layer 1 Status:
>         ACTIVE
>     Layer 2 Status:
>         TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
>     Layer 3 Status:
>         0 Active Layer 3 Call(s)
>     Active dsl 0 CCBs = 0
>     The Free Channel Mask:  0xFFFF7FFF
>     Number of L2 Discards = 0, L2 Session ID = 2
>     Total Allocated ISDN CCBs = 0
> However, if I reboot the router it goes back to the previous state. This
> clearly isn't a good situation. I've upgraded the router to 12.3(14)T7,
> the latest in its train, but this makes no difference. Is this a Cisco bug,
> or is there something else I should try? The config is attached, if it helps
>
> Thanks
>
> Max Caines
> IT Services, University of Wolverhampton
> Wolverhampton, West Midlands WV1 1SB
> Tel: 01902 322245 Fax: 01902 322777
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From rratliff at cisco.com  Thu Oct 11 10:02:18 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 11 Oct 2007 10:02:18 -0400
Subject: [cisco-voip] Intercom History
In-Reply-To: <470BD1B9.9874.0039.0@mtb.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
	<470B8E1E.9874.0039.0@mtb.com> <470BD357.9040404@cisco.com>
	<470BD1B9.9874.0039.0@mtb.com>
Message-ID: <185C6467-DC1D-469D-81E7-613466A7F453@cisco.com>

No the intercom is only in 6.x however the phone load included with  
SR5d added support for that feature.  The phone however isn't smart  
enough to know that the CM it's registering with supports the feature  
or not so it always shows the intercom history.

I'll echo Wes' request to let your account team know if you don't  
like this.  Marketing needs to hear from the field about things like  
this.

-Ryan

On Oct 9, 2007, at 7:08 PM, STEVEN CASPER wrote:

Thanks I will do so. Is Intercom a feature in 4.1.3 with the addition  
of the service release 5d? Can not find any documentation on it.

 >>> "Wes Sisk" <wsisk at cisco.com> 10/9/2007 3:15 PM >>>
phone loads are backward compatible.  Intercom history, like missed/ 
received/placed calls, is stored in the phone and therefore sorted  
before directories URL features.  Let your account team know this is  
undesirable please.

/wes

STEVEN CASPER wrote:
> Just upgraded to 4.1.3sr5d -  users now see Intercom History under  
> their Directory button on 7941 and 7961 telephones. In fact, this  
> is now above the Corporate Directory so these users now have to  
> scroll down to get to the Corporate Directory. This of course  
> irritates most users and they want Intercom History removed.
>
> What is this for, is it functional? I thought intercom was a Rls  
> 6.x feature.
>
>
>
> Thanks,
>
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
> ************************************
> This email may contain privileged and/or confidential information  
> that is intended solely for the use of the addressee. If you are  
> not the intended recipient or entity, you are strictly prohibited  
> from disclosing, copying, distributing or using any of the  
> information contained in the transmission. If you received this  
> communication in error, please contact the sender immediately and  
> destroy the material in its entirety, whether electronic or hard  
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> for the security, receipt, or third party interception of this  
> transmission.
> ************************************
>
> _______________________________________________ cisco-voip mailing  
> list cisco-voip at puck.nether.net https://puck.nether.net/mailman/ 
> listinfo/cisco-voip
************************************
This email may contain privileged and/or confidential information  
that is intended solely for the use of the addressee. If you are not  
the intended recipient or entity, you are strictly prohibited from  
disclosing, copying, distributing or using any of the information  
contained in the transmission. If you received this communication in  
error, please contact the sender immediately and destroy the material  
in its entirety, whether electronic or hard copy. This communication  
may contain nonpublic personal information about consumers subject to  
the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley  
Act. You may not directly or indirectly reuse or disclose such  
information for any purpose other than to provide the services for  
which you are receiving the information.
There are risks associated with the use of electronic transmission.  
The sender of this information does not control the method of  
transmittal or service providers and assumes no duty or obligation  
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transmission.
************************************

_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

From brian.gibson at stifel.com  Thu Oct 11 10:02:42 2007
From: brian.gibson at stifel.com (Gibson, Brian (Jersey City))
Date: Thu, 11 Oct 2007 09:02:42 -0500
Subject: [cisco-voip] CM/IPT monitoring tools
Message-ID: <5CC6FF682D32164BA5F2E830ABDF146001A5B0D1@SNEXCHANGE02.stifelnet.stifel.local>

Hi guys,

 

Can you tell me what monitoring tools you use to keep an eye on your
CM/IPT system?  RTMT is fine for broad brush stuff but I am looking for
tools that are a bit more granular.

 

I have tried the Cisco Operations Manager and find it to be somewhat
lacking.   As with most CiscoWorks tools it doesn't seem to have much in
the way of fine tuning capability.  

 

More specifically I am looking for tools that can give me specific
alerts on PRIs going down, MGCP gateways losing connection, specific
phones deregistering.  But I don't want tools that will give me 100
alerts an hour.

 

I have tried to get use my SNMP poller(SolarWinds) to check these
statuses but I haven't had much luck in finding the right MIBs.

 

Any suggestions would be greatly appreciated.


********************************************************************************
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From asobihoudai at yahoo.com  Thu Oct 11 10:06:02 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 11 Oct 2007 07:06:02 -0700 (PDT)
Subject: [cisco-voip] VG224 buzzing/humming on incoming calls only
In-Reply-To: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com>
Message-ID: <483125.82354.qm@web57205.mail.re3.yahoo.com>

Are tip and ring reversed?

Paul

--- Kevin Dunn <cheesevoice at gmail.com> wrote:

> I just had this happen with a VG248.  Same exact
> issue, turned out the
> analog lines had been victims of some lightning, the
> Telco replaced teh
> facilities for the analog lines and the crackling
> went away... Hope that
> helps.
> 
> On 10/11/07, MILLS, Mark <mark.mills at baesystems.com>
> wrote:
> >
> >  Hi All,
> >
> > I was hoping I might be able to get some help with
> the setup of a VG224
> > that I am doing.  It is connected to CCM6 cluster.
>  I can place calls FROM
> > the analog phones connected to the ports, and it
> all works properly with two
> > way audio.
> >
> > When making calls TO the analog lines on the
> VG224, the calling party
> > simply hears a loud buzzing/humming noise and
> nothing else.  The analog user
> > can hear nothing.
> >
> > This happens for incoming PSTN calls and calls to
> other extensions to the
> > VG224 connected devices.
> >
> > I have not setup a VG224 for incoming calls
> before, so there is obviously
> > something I have missed.  Below is the relevant
> config parts on the device.
> > Any help is appreciated!
> >
> >
> >
> > stcapp ccm-group 1
> > stcapp
> > !
> > voice-port 2/0
> >  ren 3
> >  idle-voltage low
> >  disc_pi_off
> >  input gain 10
> >  output attenuation 10
> >  playout-delay minimum low
> >  cptone AU
> >  timing digit 53
> >  bearer-cap Speech
> >  station-id number XXXXXXXXXX
> >  caller-id enable
> > !
> > sccp local FastEthernet0/0
> > sccp ccm 10.194.3.10 identifier 20 version 4.1
> > sccp ccm 10.194.4.10 identifier 10 version 4.1
> > sccp
> > !
> > sccp ccm group 1
> >  associate ccm 10 priority 1
> >  associate ccm 20 priority 2
> >  switchback method graceful
> > !
> > dial-peer voice XXXXXXXXXX pots
> >  service stcapp
> >  port 2/23
> > !
> >
> > Cheers,
> >    Mark
> >
> > "Warning:
> > The information contained in this email and any
> attached files is
> > confidential to BAE Systems Australia. If you are
> not the intended
> > recipient, any use, disclosure or copying of this
> email or any
> > attachments is expressly prohibited.  If you have
> received this email
> > in error, please notify us immediately. VIRUS:
> Every care has been
> > taken to ensure this email and its attachments are
> virus free,
> > however, any loss or damage incurred in using this
> email is not the
> > sender's responsibility.  It is your
> responsibility to ensure virus
> > checks are completed before installing any data
> sent in this email to
> > your computer."
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
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Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.
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From Hesham.Moussa at vanco.co.uk  Thu Oct 11 10:52:10 2007
From: Hesham.Moussa at vanco.co.uk (Hesham Moussa)
Date: Thu, 11 Oct 2007 15:52:10 +0100
Subject: [cisco-voip] CM/IPT monitoring tools
In-Reply-To: <5CC6FF682D32164BA5F2E830ABDF146001A5B0D1@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <876C8F9957792243B670CAEBDA19C6D101EF5C59@LON-INT-MSGVS01.internal.vancocorp.com>

We use Prognosis (http://www.prognosis.com/).

 

It is very configurable and you can really drill down to what alerts you
want to send and the thresholds for sending alerts.

 

Not sure about setting it up to show you when specific phones
de-register but it has a facility to show large number of phones going
off line which you set as well as when the MGCP port stops listening,
ICCP port being down, etc.

 

It has a "dashboard" of sorts which allows you to manipulate and see
whatever is present on the CM/cluster. This can be accessed via a client
or a web page. 

 

We have it integrated into our monitoring system (SMARTS) and Prognosis
sends traps to it so the NMC can see alerts and raise tickets on them
without loading it up.

 

One drawback... they have per phone licensing and there isn't an
"unlimited" bundle which would suit us fine but our AM has been quite
sympathetic and is open to discussion on bundle discounts with the more
licenses we buy.

 

I was going to actually test Cisco Operations Manager in the lab to see
how it compares but seeing that I haven't heard rave reviews about it,
this is going to be low down on my to do list.

 

Ps. If you are going to contact them, ensure that you mention the CM
version you are currently running as support for 4.x is more mature than
5.x which was integrated in release 9.0.

 

Cheers,

 

H

 

Hesham Moussa

IPT Specialist

Team IPT - UK

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gibson, Brian
(Jersey City)
Sent: 11 October 2007 15:03
To: cisco voip list
Subject: [cisco-voip] CM/IPT monitoring tools

 

Hi guys,

 

Can you tell me what monitoring tools you use to keep an eye on your
CM/IPT system?  RTMT is fine for broad brush stuff but I am looking for
tools that are a bit more granular.

 

I have tried the Cisco Operations Manager and find it to be somewhat
lacking.   As with most CiscoWorks tools it doesn't seem to have much in
the way of fine tuning capability.  

 

More specifically I am looking for tools that can give me specific
alerts on PRIs going down, MGCP gateways losing connection, specific
phones deregistering.  But I don't want tools that will give me 100
alerts an hour.

 

I have tried to get use my SNMP poller(SolarWinds) to check these
statuses but I haven't had much luck in finding the right MIBs.

 

Any suggestions would be greatly appreciated.

************************************************************************
********

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Associates are subject to review by Stifel Nicolaus. Stifel Nicolaus

may retain and reproduce electronic messages for state, federal, or

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IMPORTANT: Please do not use e-mail to request or authorize the

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accuracy of any information sent by Stifel Nicolaus through e-mail 

cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.

Stifel, Nicolaus & Company, Incorporated

Member NYSE & SIPC

Headquarters: 501 N. Broadway, St. Louis, MO 63102

314-342-2000

************************************************************************
********

 



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From deadpool.slayer.telecomm at gmail.com  Thu Oct 11 11:28:59 2007
From: deadpool.slayer.telecomm at gmail.com (Andrew F. Wolfe)
Date: Thu, 11 Oct 2007 10:28:59 -0500
Subject: [cisco-voip] FXO Goes from Ringing to FXOLS_GUARD_OUT to Ringing
	and never gets answered by a phone
Message-ID: <913929690710110828m5a108c7fo78d050160d991e55@mail.gmail.com>

I have an FXO port that goes from Ringing to Connect to FXOLS_GUARD_OUT then
back to Ringing.  All other ports on the card are working fine and they all
co to the same destination number.  This is an MGCP gateway.  Any thoughts?
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From ano-1 at worldonline.dk  Thu Oct 11 11:48:17 2007
From: ano-1 at worldonline.dk (Katrine og Michael)
Date: Thu, 11 Oct 2007 17:48:17 +0200
Subject: [cisco-voip] Problems whit a SIP connection
References: <001601c80b58$91589b70$0b000a0a@ano>
	<C1FE15183DA37645BC0633BC604E44F0070BC2D6@USNAEXCH.na.didata.local>
Message-ID: <002b01c80c1e$24b3bb50$0d000a0a@ano>

Hey,

Thanks, it?s working now :-D
  ----- Original Message ----- 
  From: Matt Slaga (US) 
  To: Katrine og Michael ; cisco-voip at puck.nether.net 
  Sent: Wednesday, October 10, 2007 8:35 PM
  Subject: RE: [cisco-voip] Problems whit a SIP connection


  Change this:

   

  registrar dns:sip.cybercity.dk expires 3600

   

  to

   

  registrar dns:sip.cybercity.dk expires 360

   

   

   

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Katrine og Michael
  Sent: Wednesday, October 10, 2007 12:14 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Problems whit a SIP connection

   

  Hey,

   

  Can somebody please help me?

   

  I have configured CCME on a 2801 router and I have it connected to my SIP provider.

   

  I?am having a problem.

  When is is registred up against the SIP-erver everything  works fine, but after a while i get this message:

   


  Oct 10 15:47:31.101: //-1/xxxxxxxxxxxx/SIP/Info/udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060

   

  And then I can not get call in, but I can ring out the ccme.

   

  I don?t know if is has anything to do whit my SIP-UA, som here it is:

   

  sip-ua
   authentication username voi49495 password 12341

   no remote-party-id
   retry invite 4
   retry response 3
   retry bye 2
   retry cancel 2
   retry register 5
   timers register 250
   registrar dns:sip.cybercity.dk expires 3600

   

  Please help me, and if you need more info, then just read

   

   

   

  Michael from Denmark :-)

   


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From ealeatherman at gmail.com  Thu Oct 11 12:00:55 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 11 Oct 2007 12:00:55 -0400
Subject: [cisco-voip] CM/IPT monitoring tools
In-Reply-To: <5CC6FF682D32164BA5F2E830ABDF146001A5B0D1@SNEXCHANGE02.stifelnet.stifel.local>
References: <5CC6FF682D32164BA5F2E830ABDF146001A5B0D1@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <94a1afde0710110900h5ceeb5a5h106f3ebd97f674e@mail.gmail.com>

NetIQ has a product out there also for managing cisco voip, might be worth
looking at and comparing too. We're planning on evaluating it probably this
spring. We're in the same boat that you are with monitoring and management
tools.

I evaluated Prognosis a couple years ago but I wasn't thrilled with the
interface and setup, also at the time it was a agent loaded on CCM which I
didnt like. I understand with CM5 they have by necessity a separate agent
off of call manager now. Pre-sales/support didn't really give me a warm and
fuzzy feeling during our eval either, but that was just my experience and
like I said it was a few years ago. Now that I have a better handle on what
I want, maybe it would look better to me :) Last time they contacted me they
were trying to use the VTech tragedy as a possible sales point, which REALLY
turned me off big time.

We're actually looking at NetQoS now from a performance monitoring
perspective enterprise wide (data network, applications group, etc), looks
like its very good a performance monitoring obviously, but not sure yet if
it will alarm on IP phone registrations and the like yet. Still need to look
at their VoIP potion of the product myself.

We kinda got burned by Ciscoworks initially (before it was ops manager), we
got it when we first implemented VoIP as part of the order, not knowing any
better and not having any environment to test it out in. I've heard that its
improved, but like Hesham said in his email  have never heard any rave
reviews of it so evaluating it isn't a priority right now. Also Cisco
doesn't seem to offer any software upgrade assurance with their maintenance
packages for ciscoworks.. our net operations is running into this with their
ciscoworks LMS, so i'm less inclined to buy the voice solution if i'm going
to have to buy it all over again later to get new features and improvements
to the product.

On 10/11/07, Gibson, Brian (Jersey City) <brian.gibson at stifel.com> wrote:
>
>  Hi guys,
>
>
>
> Can you tell me what monitoring tools you use to keep an eye on your
> CM/IPT system?  RTMT is fine for broad brush stuff but I am looking for
> tools that are a bit more granular.
>
>
>
> I have tried the Cisco Operations Manager and find it to be somewhat
> lacking.   As with most CiscoWorks tools it doesn't seem to have much in the
> way of fine tuning capability.
>
>
>
> More specifically I am looking for tools that can give me specific alerts
> on PRIs going down, MGCP gateways losing connection, specific phones
> deregistering.  But I don't want tools that will give me 100 alerts an hour.
>
>
>
> I have tried to get use my SNMP poller(SolarWinds) to check these statuses
> but I haven't had much luck in finding the right MIBs.
>
>
>
> Any suggestions would be greatly appreciated.
>
>
> ********************************************************************************
>
> All electronic messages sent and received by Stifel Nicolaus
>
> Associates are subject to review by Stifel Nicolaus. Stifel Nicolaus
>
> may retain and reproduce electronic messages for state, federal, or
>
> other regulatory agencies as required by applicable law.
>
> IMPORTANT: Please do not use e-mail to request or authorize the
>
> purchase or sale of any security or commodity, send fund transfer
>
> instructions, or otherwise conduct any securities transactions. Any
>
> requests, orders, instructions, or time-sensitive messages sent by
>
> e-mail cannot be accepted or processed by Stifel Nicolaus. The
>
> accuracy of any information sent by Stifel Nicolaus through e-mail
>
> cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
>
>  Stifel, Nicolaus & Company, Incorporated
>
> Member NYSE & SIPC
>
> Headquarters: 501 N. Broadway, St. Louis, MO 63102
>
> 314-342-2000
>
>
> ********************************************************************************
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Thu Oct 11 12:05:22 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 11 Oct 2007 12:05:22 -0400
Subject: [cisco-voip] Contact Center Solution for Wallboard
In-Reply-To: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>
References: <2683B0F497088C489578F16771529BAA036B3D0F@ATLEX01.agg.local>
Message-ID: <94a1afde0710110905l5236f9c8ncdde39bc01858883@mail.gmail.com>

We are looking at a solution from company called checkmate called RealTime
Online (RTOL) for our IPCC Express:
www.checkmate.com
seems targeted for smaller call centers.

Ed

On 10/10/07, Tarkington, Kathy L. <Kathy.Tarkington at agg.com> wrote:
>
>  I am looking for a solution that will give my Help Desk a wallboard
> showing agents logged in, agent status, calls in queue, etc.
> Recommendations?
>
> Thanks!
>  ------------------------------
>  *Note:*
>
> This message and any attachments from the law firm Arnall Golden Gregory
> LLP may contain CONFIDENTIAL and legally protected information.  If you
> are not the addressee and an intended recipient, please do not read, copy,
> use or disclose this communication to others;  also, please notify the
> sender by replying to this message, and then delete it from your system.
> Thank you.
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From svoll.voip at gmail.com  Thu Oct 11 12:11:18 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 11 Oct 2007 09:11:18 -0700
Subject: [cisco-voip] CM/IPT monitoring tools
In-Reply-To: <94a1afde0710110900h5ceeb5a5h106f3ebd97f674e@mail.gmail.com>
References: <5CC6FF682D32164BA5F2E830ABDF146001A5B0D1@SNEXCHANGE02.stifelnet.stifel.local>
	<94a1afde0710110900h5ceeb5a5h106f3ebd97f674e@mail.gmail.com>
Message-ID: <f84a38d30710110911t3ff5b9b8y329e84d4dd8f5ea5@mail.gmail.com>

The CIPTUG confernce is a great place to see some of this stuff in person.
meet all the vendors and see things side by side.

it's not to late to register.  www.ciptug.org

Scott


On 10/11/07, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> NetIQ has a product out there also for managing cisco voip, might be worth
> looking at and comparing too. We're planning on evaluating it probably this
> spring. We're in the same boat that you are with monitoring and management
> tools.
>
> I evaluated Prognosis a couple years ago but I wasn't thrilled with the
> interface and setup, also at the time it was a agent loaded on CCM which I
> didnt like. I understand with CM5 they have by necessity a separate agent
> off of call manager now. Pre-sales/support didn't really give me a warm and
> fuzzy feeling during our eval either, but that was just my experience and
> like I said it was a few years ago. Now that I have a better handle on what
> I want, maybe it would look better to me :) Last time they contacted me they
> were trying to use the VTech tragedy as a possible sales point, which REALLY
> turned me off big time.
>
> We're actually looking at NetQoS now from a performance monitoring
> perspective enterprise wide (data network, applications group, etc), looks
> like its very good a performance monitoring obviously, but not sure yet if
> it will alarm on IP phone registrations and the like yet. Still need to look
> at their VoIP potion of the product myself.
>
> We kinda got burned by Ciscoworks initially (before it was ops manager),
> we got it when we first implemented VoIP as part of the order, not knowing
> any better and not having any environment to test it out in. I've heard that
> its improved, but like Hesham said in his email  have never heard any rave
> reviews of it so evaluating it isn't a priority right now. Also Cisco
> doesn't seem to offer any software upgrade assurance with their maintenance
> packages for ciscoworks.. our net operations is running into this with their
> ciscoworks LMS, so i'm less inclined to buy the voice solution if i'm going
> to have to buy it all over again later to get new features and improvements
> to the product.
>
>  On 10/11/07, Gibson, Brian (Jersey City) <brian.gibson at stifel.com> wrote:
>
> >   Hi guys,
> >
> >
> >
> > Can you tell me what monitoring tools you use to keep an eye on your
> > CM/IPT system?  RTMT is fine for broad brush stuff but I am looking for
> > tools that are a bit more granular.
> >
> >
> >
> > I have tried the Cisco Operations Manager and find it to be somewhat
> > lacking.   As with most CiscoWorks tools it doesn't seem to have much in the
> > way of fine tuning capability.
> >
> >
> >
> > More specifically I am looking for tools that can give me specific
> > alerts on PRIs going down, MGCP gateways losing connection, specific phones
> > deregistering.  But I don't want tools that will give me 100 alerts an hour.
> >
> >
> >
> > I have tried to get use my SNMP poller(SolarWinds) to check these
> > statuses but I haven't had much luck in finding the right MIBs.
> >
> >
> >
> > Any suggestions would be greatly appreciated.
> >
> >
> > ********************************************************************************
> >
> > All electronic messages sent and received by Stifel Nicolaus
> >
> > Associates are subject to review by Stifel Nicolaus. Stifel Nicolaus
> >
> > may retain and reproduce electronic messages for state, federal, or
> >
> > other regulatory agencies as required by applicable law.
> >
> > IMPORTANT: Please do not use e-mail to request or authorize the
> >
> > purchase or sale of any security or commodity, send fund transfer
> >
> > instructions, or otherwise conduct any securities transactions. Any
> >
> > requests, orders, instructions, or time-sensitive messages sent by
> >
> > e-mail cannot be accepted or processed by Stifel Nicolaus. The
> >
> > accuracy of any information sent by Stifel Nicolaus through e-mail
> >
> > cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
> >
> > Stifel, Nicolaus & Company, Incorporated
> >
> > Member NYSE & SIPC
> >
> > Headquarters: 501 N. Broadway, St. Louis, MO 63102
> >
> > 314-342-2000
> >
> >
> > ********************************************************************************
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Christian.Heiss at t-systems.com  Thu Oct 11 11:04:43 2007
From: Christian.Heiss at t-systems.com (=?iso-8859-1?Q?Hei=DF=2C_Christian?=)
Date: Thu, 11 Oct 2007 17:04:43 +0200
Subject: [cisco-voip] Intercom History
In-Reply-To: <470B8E1E.9874.0039.0@mtb.com>
Message-ID: <71D98D68FE651547B7136635F7DE4A609FEC33@ex-s03-08.ads.tnetpro.de>

Hi Steve,
 
it?s a "feature" in the PhoneLoad 8.3.1 of the phones 7941/7961. To remove the Intercom History from the Directory button i installed the Phoneload 8.2.2.
 
Regards
Christian

________________________________

Von: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] Im Auftrag von STEVEN CASPER
Gesendet: Dienstag, 9. Oktober 2007 20:20
An: cisco-voip at puck.nether.net
Betreff: [cisco-voip] Intercom History


Just upgraded to 4.1.3sr5d -  users now see Intercom History under their Directory button on 7941 and 7961 telephones. In fact, this is now above the Corporate Directory so these users now have to scroll down to get to the Corporate Directory. This of course irritates most users and they want Intercom History removed.
 
What is this for, is it functional? I thought intercom was a Rls 6.x feature. 
 
 
 
Thanks,

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026 

************************************
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From matt at saskin.net  Thu Oct 11 12:54:56 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 11 Oct 2007 12:54:56 -0400
Subject: [cisco-voip] CM/IPT monitoring tools
In-Reply-To: <94a1afde0710110900h5ceeb5a5h106f3ebd97f674e@mail.gmail.com>
References: <5CC6FF682D32164BA5F2E830ABDF146001A5B0D1@SNEXCHANGE02.stifelnet.stifel.local>
	<94a1afde0710110900h5ceeb5a5h106f3ebd97f674e@mail.gmail.com>
Message-ID: <470E5560.1060404@saskin.net>

I've become more and more intrigued by the NetQoS product - truly one of 
the few out there that can give insight into voice quality *without* 
relying on CDR/CMR's.

Personally, we run Prognosis internally and for all of our clients.  The 
most recent versions (8.1 for CM 4x and 8 for CM5+) are great - I've yet 
to run into any issues with the agent installation as it is on the Cisco 
"whitelist" which is generally taken as gospel.

With prognosis, you can get as granular with alerting as you want - for 
example you can alert at anything down to a specific single phone 
unregistering - bad call quality from a single phone, etc.

Also for what it's worth, we chose prognosis after doing an extended 
bake-off between it, cisco ops manager, and NetIQ.

-matt

Ed Leatherman wrote:
> NetIQ has a product out there also for managing cisco voip, might be 
> worth looking at and comparing too. We're planning on evaluating it 
> probably this spring. We're in the same boat that you are with 
> monitoring and management tools.
> 
> I evaluated Prognosis a couple years ago but I wasn't thrilled with the 
> interface and setup, also at the time it was a agent loaded on CCM which 
> I didnt like. I understand with CM5 they have by necessity a separate 
> agent off of call manager now. Pre-sales/support didn't really give me a 
> warm and fuzzy feeling during our eval either, but that was just my 
> experience and like I said it was a few years ago. Now that I have a 
> better handle on what I want, maybe it would look better to me :) Last 
> time they contacted me they were trying to use the VTech tragedy as a 
> possible sales point, which REALLY turned me off big time.
> 
> We're actually looking at NetQoS now from a performance monitoring 
> perspective enterprise wide (data network, applications group, etc), 
> looks like its very good a performance monitoring obviously, but not 
> sure yet if it will alarm on IP phone registrations and the like yet. 
> Still need to look at their VoIP potion of the product myself.
> 
> We kinda got burned by Ciscoworks initially (before it was ops manager), 
> we got it when we first implemented VoIP as part of the order, not 
> knowing any better and not having any environment to test it out in. 
> I've heard that its improved, but like Hesham said in his email  have 
> never heard any rave reviews of it so evaluating it isn't a priority 
> right now. Also Cisco doesn't seem to offer any software upgrade 
> assurance with their maintenance packages for ciscoworks.. our net 
> operations is running into this with their ciscoworks LMS, so i'm less 
> inclined to buy the voice solution if i'm going to have to buy it all 
> over again later to get new features and improvements to the product.
> 
> On 10/11/07, *Gibson, Brian (Jersey City)* <brian.gibson at stifel.com 
> <mailto:brian.gibson at stifel.com>> wrote:
> 
>     Hi guys,
> 
>      
> 
>     Can you tell me what monitoring tools you use to keep an eye on your
>     CM/IPT system?  RTMT is fine for broad brush stuff but I am looking
>     for tools that are a bit more granular.
> 
>      
> 
>     I have tried the Cisco Operations Manager and find it to be somewhat
>     lacking.   As with most CiscoWorks tools it doesn't seem to have
>     much in the way of fine tuning capability. 
> 
>      
> 
>     More specifically I am looking for tools that can give me specific
>     alerts on PRIs going down, MGCP gateways losing connection, specific
>     phones deregistering.  But I don't want tools that will give me 100
>     alerts an hour.
> 
>      
> 
>     I have tried to get use my SNMP poller(SolarWinds) to check these
>     statuses but I haven't had much luck in finding the right MIBs.
> 
>      
> 
>     Any suggestions would be greatly appreciated.
> 
>     ********************************************************************************
> 
>     All electronic messages sent and received by Stifel Nicolaus
> 
>     Associates are subject to review by Stifel Nicolaus. Stifel Nicolaus
> 
>     may retain and reproduce electronic messages for state, federal, or
> 
>     other regulatory agencies as required by applicable law.
> 
>     IMPORTANT: Please do not use e-mail to request or authorize the
> 
>     purchase or sale of any security or commodity, send fund transfer
> 
>     instructions, or otherwise conduct any securities transactions. Any
> 
>     requests, orders, instructions, or time-sensitive messages sent by
> 
>     e-mail cannot be accepted or processed by Stifel Nicolaus. The
> 
>     accuracy of any information sent by Stifel Nicolaus through e-mail
> 
>     cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
> 
>     Stifel, Nicolaus & Company, Incorporated
> 
>     Member NYSE & SIPC
> 
>     Headquarters: 501 N. Broadway, St. Louis, MO 63102
> 
>     314-342-2000
> 
>     ********************************************************************************
> 
>      
> 
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> 
> -- 
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jonvoip at gmail.com  Thu Oct 11 13:53:07 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 11 Oct 2007 12:53:07 -0500
Subject: [cisco-voip] Cannot delete Partition, Dependency Records empty
In-Reply-To: <FA233791F398B848B978464507739E8FD6BFF1@EUBEBRUSVEX1.eu.didata.local>
References: <20071008084714.GA26790@ibh.de>
	<4B8038C9-6676-499B-86B6-981A58F30BA1@cisco.com>
	<20071009095018.GB1620@ibh.de>
	<FA233791F398B848B978464507739E8FD6BFF1@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <5d093f9a0710111053t361dac70ib73ce88e4f0f7ab@mail.gmail.com>

Delete your Unassigned DNs.


J

On 10/9/07, Ruben Montes (Europe) <Ruben.Montes at eu.didata.com> wrote:
> Hello,
>
> I'm approximating to the max number of partitions in a CCM 5.x(75
> partitions), and I have tried to delete some of them I'm not using.
>
> Despite the Dependency Records shows nothing, I'm not able to delete
> them, because the CCM says that it is being used...
>
> Any idea how can I troubleshoot this?
>
> Thanks in advance,
>
> Ruben
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From ealeatherman at gmail.com  Thu Oct 11 14:03:55 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 11 Oct 2007 14:03:55 -0400
Subject: [cisco-voip] IPCCX 5 JTAPI client
Message-ID: <94a1afde0710111103i299ac19ct831c29b498c52e88@mail.gmail.com>

Just installed IPCC Express 5 in the lab, and i'm getting an error that the
JTAPI client does not match call manager.. but when i run the resynch
process from the web page like it says, it just gives me a generic error
message saying it failed and to try again. which I do and doesnt help.. also
have tried manually installing after downloading it from CCM plugins page
which didn't work either. Anyone run into a problem like that before?

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From tednugent73 at gmail.com  Thu Oct 11 14:25:08 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 11 Oct 2007 14:25:08 -0400
Subject: [cisco-voip] IPCCX 5 JTAPI client
In-Reply-To: <94a1afde0710111103i299ac19ct831c29b498c52e88@mail.gmail.com>
References: <94a1afde0710111103i299ac19ct831c29b498c52e88@mail.gmail.com>
Message-ID: <22e6380b0710111125p1f041938s1b3212c07a9329ef@mail.gmail.com>

Theres only a couple of versions of CM/JTAPI supported for UCCX 5 check out
the CRS compatibilty matrix if you haven't allready.

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.pdf





On 10/11/07, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Just installed IPCC Express 5 in the lab, and i'm getting an error that
> the JTAPI client does not match call manager.. but when i run the resynch
> process from the web page like it says, it just gives me a generic error
> message saying it failed and to try again. which I do and doesnt help.. also
> have tried manually installing after downloading it from CCM plugins page
> which didn't work either. Anyone run into a problem like that before?
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From svoll.voip at gmail.com  Thu Oct 11 15:12:09 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 11 Oct 2007 12:12:09 -0700
Subject: [cisco-voip] H323 port list
Message-ID: <f84a38d30710111212w20d8042eia11913974b6f1545@mail.gmail.com>

How do you setup a H323 VGW that has 12 FXO into a list so you don't have to
make a dial peer for each port?

Thanks

Scott
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From CarlosOrtiz at bayviewfinancial.com  Thu Oct 11 15:18:32 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 11 Oct 2007 15:18:32 -0400
Subject: [cisco-voip] H323 port list
In-Reply-To: <f84a38d30710111212w20d8042eia11913974b6f1545@mail.gmail.com>
Message-ID: <OF615AE3AB.B0D97D40-ON85257371.0069F781-85257371.006A0FE0@bftg.com>

I have used trunk groups for PRI but not FXO.  I believe that others on 
the list have mentioned it can be done with FXO as well.  Did you check 
the archives?

Carlos



"Scott Voll" <svoll.voip at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
10/11/2007 03:12 PM

To
"cisco voip list" <cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] H323 port list






How do you setup a H323 VGW that has 12 FXO into a list so you don't have 
to make a dial peer for each port?
 
Thanks
 
Scott
 
 _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From svoll.voip at gmail.com  Thu Oct 11 15:28:00 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 11 Oct 2007 12:28:00 -0700
Subject: [cisco-voip] H323 port list
In-Reply-To: <OF615AE3AB.B0D97D40-ON85257371.0069F781-85257371.006A0FE0@bftg.com>
References: <f84a38d30710111212w20d8042eia11913974b6f1545@mail.gmail.com>
	<OF615AE3AB.B0D97D40-ON85257371.0069F781-85257371.006A0FE0@bftg.com>
Message-ID: <f84a38d30710111228k7f06bc4ds183462e0c27c5174@mail.gmail.com>

Thanks.... I was looking for the trunk group command.....

Scott


On 10/11/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote:
>
>
> I have used trunk groups for PRI but not FXO.  I believe that others on
> the list have mentioned it can be done with FXO as well.  Did you check the
> archives?
>
> Carlos
>
>
>   *"Scott Voll" <svoll.voip at gmail.com>*
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 10/11/2007 03:12 PM
>    To
> "cisco voip list" <cisco-voip at puck.nether.net>  cc
>   Subject
> [cisco-voip] H323 port list
>
>
>
>
> How do you setup a H323 VGW that has 12 FXO into a list so you don't have
> to make a dial peer for each port?
>
> Thanks
>
> Scott
>
>  _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From cheesevoice at gmail.com  Thu Oct 11 15:28:51 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Thu, 11 Oct 2007 14:28:51 -0500
Subject: [cisco-voip] CCVP Certification
Message-ID: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>

Well I am finally going to start my CCVP exams...

Helpful hints, studyguides preffered and any friendly advice appreciated.

Thought I would start with CVOICE and QOS first then IPT (4.x)...thoughts?
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From rkulagow at gmail.com  Thu Oct 11 15:40:01 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Thu, 11 Oct 2007 14:40:01 -0500
Subject: [cisco-voip] H323 port list
In-Reply-To: <f84a38d30710111228k7f06bc4ds183462e0c27c5174@mail.gmail.com>
References: <f84a38d30710111212w20d8042eia11913974b6f1545@mail.gmail.com>	<OF615AE3AB.B0D97D40-ON85257371.0069F781-85257371.006A0FE0@bftg.com>
	<f84a38d30710111228k7f06bc4ds183462e0c27c5174@mail.gmail.com>
Message-ID: <470E7C11.7070103@gmail.com>

>     How do you setup a H323 VGW that has 12 FXO into a list so you don't
>     have to make a dial peer for each port?

trunk group FXO
  max-retry 5
  voice-class cause-code 1
  hunt-scheme round-robin

voice class cause-code 1
  no-circuit

That way if you have a bad FXO in the group it'll get skipped.

From CarlosOrtiz at bayviewfinancial.com  Thu Oct 11 15:32:36 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 11 Oct 2007 15:32:36 -0400
Subject: [cisco-voip] H323 port list
In-Reply-To: <f84a38d30710111228k7f06bc4ds183462e0c27c5174@mail.gmail.com>
Message-ID: <OFB730DB5F.53E88889-ON85257371.006B4866-85257371.006B59C2@bftg.com>

Check out this archived post.   I think Chris M Bomba has a good 
example......

http://www.gossamer-threads.com/lists/cisco/voip/61154?search_string=fxo%20trunk%20group;#61154





"Scott Voll" <svoll.voip at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
10/11/2007 03:28 PM

To
"CarlosOrtiz at bayviewfinancial.com" <CarlosOrtiz at bayviewfinancial.com>
cc
cisco-voip-bounces at puck.nether.net, cisco voip list 
<cisco-voip at puck.nether.net>
Subject
Re: [cisco-voip] H323 port list






Thanks.... I was looking for the trunk group command.....
 
Scott

 
On 10/11/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com > wrote: 

I have used trunk groups for PRI but not FXO.  I believe that others on 
the list have mentioned it can be done with FXO as well.  Did you check 
the archives? 

Carlos 


"Scott Voll" <svoll.voip at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net 
10/11/2007 03:12 PM 


To
"cisco voip list" <cisco-voip at puck.nether.net> 
cc

Subject
[cisco-voip] H323 port list









How do you setup a H323 VGW that has 12 FXO into a list so you don't have 
to make a dial peer for each port? 
 
Thanks 
 
Scott 
 
 _______________________________________________ 
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From brian.gibson at stifel.com  Thu Oct 11 15:38:50 2007
From: brian.gibson at stifel.com (Gibson, Brian (Jersey City))
Date: Thu, 11 Oct 2007 14:38:50 -0500
Subject: [cisco-voip] CM/IPT monitoring tools
In-Reply-To: <470E5560.1060404@saskin.net>
References: <5CC6FF682D32164BA5F2E830ABDF146001A5B0D1@SNEXCHANGE02.stifelnet.stifel.local>
	<94a1afde0710110900h5ceeb5a5h106f3ebd97f674e@mail.gmail.com>
	<470E5560.1060404@saskin.net>
Message-ID: <5CC6FF682D32164BA5F2E830ABDF146001A5B2AC@SNEXCHANGE02.stifelnet.stifel.local>

How does their app work in the 5.1 environment?  I assume they no longer install a shim on the server?  

-----Original Message-----
From: Matthew Saskin [mailto:matt at saskin.net] 
Sent: Thursday, October 11, 2007 12:55 PM
To: Ed Leatherman
Cc: Gibson, Brian (Jersey City); cisco voip list
Subject: Re: [cisco-voip] CM/IPT monitoring tools

I've become more and more intrigued by the NetQoS product - truly one of 
the few out there that can give insight into voice quality *without* 
relying on CDR/CMR's.

Personally, we run Prognosis internally and for all of our clients.  The 
most recent versions (8.1 for CM 4x and 8 for CM5+) are great - I've yet 
to run into any issues with the agent installation as it is on the Cisco 
"whitelist" which is generally taken as gospel.

With prognosis, you can get as granular with alerting as you want - for 
example you can alert at anything down to a specific single phone 
unregistering - bad call quality from a single phone, etc.

Also for what it's worth, we chose prognosis after doing an extended 
bake-off between it, cisco ops manager, and NetIQ.

-matt

Ed Leatherman wrote:
> NetIQ has a product out there also for managing cisco voip, might be 
> worth looking at and comparing too. We're planning on evaluating it 
> probably this spring. We're in the same boat that you are with 
> monitoring and management tools.
> 
> I evaluated Prognosis a couple years ago but I wasn't thrilled with the 
> interface and setup, also at the time it was a agent loaded on CCM which 
> I didnt like. I understand with CM5 they have by necessity a separate 
> agent off of call manager now. Pre-sales/support didn't really give me a 
> warm and fuzzy feeling during our eval either, but that was just my 
> experience and like I said it was a few years ago. Now that I have a 
> better handle on what I want, maybe it would look better to me :) Last 
> time they contacted me they were trying to use the VTech tragedy as a 
> possible sales point, which REALLY turned me off big time.
> 
> We're actually looking at NetQoS now from a performance monitoring 
> perspective enterprise wide (data network, applications group, etc), 
> looks like its very good a performance monitoring obviously, but not 
> sure yet if it will alarm on IP phone registrations and the like yet. 
> Still need to look at their VoIP potion of the product myself.
> 
> We kinda got burned by Ciscoworks initially (before it was ops manager), 
> we got it when we first implemented VoIP as part of the order, not 
> knowing any better and not having any environment to test it out in. 
> I've heard that its improved, but like Hesham said in his email  have 
> never heard any rave reviews of it so evaluating it isn't a priority 
> right now. Also Cisco doesn't seem to offer any software upgrade 
> assurance with their maintenance packages for ciscoworks.. our net 
> operations is running into this with their ciscoworks LMS, so i'm less 
> inclined to buy the voice solution if i'm going to have to buy it all 
> over again later to get new features and improvements to the product.
> 
> On 10/11/07, *Gibson, Brian (Jersey City)* <brian.gibson at stifel.com 
> <mailto:brian.gibson at stifel.com>> wrote:
> 
>     Hi guys,
> 
>      
> 
>     Can you tell me what monitoring tools you use to keep an eye on your
>     CM/IPT system?  RTMT is fine for broad brush stuff but I am looking
>     for tools that are a bit more granular.
> 
>      
> 
>     I have tried the Cisco Operations Manager and find it to be somewhat
>     lacking.   As with most CiscoWorks tools it doesn't seem to have
>     much in the way of fine tuning capability. 
> 
>      
> 
>     More specifically I am looking for tools that can give me specific
>     alerts on PRIs going down, MGCP gateways losing connection, specific
>     phones deregistering.  But I don't want tools that will give me 100
>     alerts an hour.
> 
>      
> 
>     I have tried to get use my SNMP poller(SolarWinds) to check these
>     statuses but I haven't had much luck in finding the right MIBs.
> 
>      
> 
>     Any suggestions would be greatly appreciated.
> 
>     ********************************************************************************
> 
>     All electronic messages sent and received by Stifel Nicolaus
> 
>     Associates are subject to review by Stifel Nicolaus. Stifel Nicolaus
> 
>     may retain and reproduce electronic messages for state, federal, or
> 
>     other regulatory agencies as required by applicable law.
> 
>     IMPORTANT: Please do not use e-mail to request or authorize the
> 
>     purchase or sale of any security or commodity, send fund transfer
> 
>     instructions, or otherwise conduct any securities transactions. Any
> 
>     requests, orders, instructions, or time-sensitive messages sent by
> 
>     e-mail cannot be accepted or processed by Stifel Nicolaus. The
> 
>     accuracy of any information sent by Stifel Nicolaus through e-mail
> 
>     cannot be warranted or guaranteed by Stifel Nicolaus or its affiliates.
> 
>     Stifel, Nicolaus & Company, Incorporated
> 
>     Member NYSE & SIPC
> 
>     Headquarters: 501 N. Broadway, St. Louis, MO 63102
> 
>     314-342-2000
> 
>     ********************************************************************************
> 
>      
> 
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> 
> -- 
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From ealeatherman at gmail.com  Thu Oct 11 16:13:12 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 11 Oct 2007 16:13:12 -0400
Subject: [cisco-voip] CCVP Certification
In-Reply-To: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
References: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
Message-ID: <94a1afde0710111313w403ad585radbda57cf39c4e70@mail.gmail.com>

IMO practice tests are definitely the way to go, if you already familiar
with all the material. I had to do a little extra studying on the CCNA for
RIP and OSPF since we don't use those here where i work. I wasn't really
sure which test to take after that, I figured call manager (IPT4.x) was what
I had the most experience with. I tried it at networker's this year as my
freebie test and passed it. Just used practice exams and went back thru
callmanager docs to review the topics I found I was weak on.

Those are the only 2 i've done so far. I also have heard people start with
CVOICE, I think that is what i'm going to try to tackle next. The prep
seminar at networker's this year made it sound pretty basic.

GWGK looks like the worst one, but that is my weakest area I think, so ymmv.
Not sure about the new troubleshooting exam, I heard they changed that one
specifically to put something related to Unity on the CCVP track.

On 10/11/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
>
> Well I am finally going to start my CCVP exams...
>
> Helpful hints, studyguides preffered and any friendly advice appreciated.
>
> Thought I would start with CVOICE and QOS first then IPT (4.x)...thoughts?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From jonvoip at gmail.com  Thu Oct 11 16:18:46 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 11 Oct 2007 15:18:46 -0500
Subject: [cisco-voip] CCVP Certification
In-Reply-To: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
References: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
Message-ID: <5d093f9a0710111318y695a877ahed75767b337964d4@mail.gmail.com>

The standard order is:

CVOICE
QOS
CIPT
TUS (aka CIPTT)
GW/GK

As far as study sources, I think the Cisco Press books are pretty decent...

The Cisco Press QOS Exam Cert book is the best Cisco Press book I have
ever read.

You will probably want to take the CIPT 5.X test, but either one
fulfills the requirement.


Jonathan

On 10/11/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> Well I am finally going to start my CCVP exams...
>
> Helpful hints, studyguides preffered and any friendly advice appreciated.
>
> Thought I would start with CVOICE and QOS first then IPT (4.x)...thoughts?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From ealeatherman at gmail.com  Thu Oct 11 16:42:44 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 11 Oct 2007 16:42:44 -0400
Subject: [cisco-voip] CM/IPT monitoring tools
In-Reply-To: <5CC6FF682D32164BA5F2E830ABDF146001A5B2AC@SNEXCHANGE02.stifelnet.stifel.local>
References: <5CC6FF682D32164BA5F2E830ABDF146001A5B0D1@SNEXCHANGE02.stifelnet.stifel.local>
	<94a1afde0710110900h5ceeb5a5h106f3ebd97f674e@mail.gmail.com>
	<470E5560.1060404@saskin.net>
	<5CC6FF682D32164BA5F2E830ABDF146001A5B2AC@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <94a1afde0710111342j207bcfc4wbe6af8d27864c341@mail.gmail.com>

The NetQoS uses a SPAN port on the callmanager network I believe to
intercept SCCP traffic and call quality metrics as they come in from the IP
Phones. I guess prognosis hits the AXL interfaces now to get all it's info?
Not sure which app you are talking about.

On 10/11/07, Gibson, Brian (Jersey City) <brian.gibson at stifel.com> wrote:
>
> How does their app work in the 5.1 environment?  I assume they no longer
> install a shim on the server?
>
> -----Original Message-----
> From: Matthew Saskin [mailto:matt at saskin.net]
> Sent: Thursday, October 11, 2007 12:55 PM
> To: Ed Leatherman
> Cc: Gibson, Brian (Jersey City); cisco voip list
> Subject: Re: [cisco-voip] CM/IPT monitoring tools
>
> I've become more and more intrigued by the NetQoS product - truly one of
> the few out there that can give insight into voice quality *without*
> relying on CDR/CMR's.
>
> Personally, we run Prognosis internally and for all of our clients.  The
> most recent versions (8.1 for CM 4x and 8 for CM5+) are great - I've yet
> to run into any issues with the agent installation as it is on the Cisco
> "whitelist" which is generally taken as gospel.
>
> With prognosis, you can get as granular with alerting as you want - for
> example you can alert at anything down to a specific single phone
> unregistering - bad call quality from a single phone, etc.
>
> Also for what it's worth, we chose prognosis after doing an extended
> bake-off between it, cisco ops manager, and NetIQ.
>
> -matt
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From tim.medley at moxieent.com  Thu Oct 11 16:44:46 2007
From: tim.medley at moxieent.com (tim.medley at moxieent.com)
Date: Thu, 11 Oct 2007 13:44:46 -0700
Subject: [cisco-voip] CCVP Certification
Message-ID: <20071011134446.cf410a99374b2e152b39acaa4abed1f9.06f2c92766.wbe@email.secureserver.net>

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From aizhar at ccbootcamp.com  Thu Oct 11 16:49:03 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Thu, 11 Oct 2007 13:49:03 -0700
Subject: [cisco-voip] CCVP Certification
References: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663561196@nlimail.nlidomain>

Hi,

The order in which we teach (and test) it is:
QOS
CVOICE
CIPT
GWGK
TUC


QOS has no voice in it (aside from basic understanding of jitter and such). CVOICE has QoS in it. TUC is the last one since troubleshooting contains all the others. 

I'd recommend CIPT 5.x (currently 4.x is allowed as well) since TUC require callmanager 5.x knowledge.

The fastest way would be a two weeks bootcamp, but if not then I'd recommend the cisco press books.

HTH,

-------------------------------------------
thanks,
Avner Izhar
CCIE#15999 (Voice), CCVP, CCSI# 31623
Technical Instructor
CCBOOTCAMP - A Cisco Sponsored Organization (SO)
Email: aizhar at ccbootcamp.com
Toll Free: 877-654-2243
Direct: +1-702-968-5100 = Outside the USA
FAX: +1-702-446-8012
YES! We take Cisco Learning Credits!
Register to win a free iPhone! http://www.ccbootcamp.com/iphone.html



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net on behalf of Kevin Dunn
Sent: Thu 10/11/2007 12:28 PM
To: Cisco Voice
Subject: [cisco-voip] CCVP Certification [bcc][faked-from]
 
Well I am finally going to start my CCVP exams...

Helpful hints, studyguides preffered and any friendly advice appreciated.

Thought I would start with CVOICE and QOS first then IPT (4.x)...thoughts?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From svoll.voip at gmail.com  Thu Oct 11 17:05:51 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 11 Oct 2007 14:05:51 -0700
Subject: [cisco-voip] CCVP Certification
In-Reply-To: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
References: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
Message-ID: <f84a38d30710111405pe2e8553m634cec0f0dbf55@mail.gmail.com>

Are you looking for understanding or a Cert.  if your just looking for the
Cert. I would use a test practice like pass 4 sure.

if your looking for the training A boot camp would be good.

I started with what I was strong on.  CIPT.  I just went through the blue
print and read the white pages on what I was weak on..... ie. IPMA, AAR,
etc.

depending on your level of comfort and weather your looking for training or
cert, you might look at the CCIE bootcamp.

Scott


On 10/11/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
>
> Well I am finally going to start my CCVP exams...
>
> Helpful hints, studyguides preffered and any friendly advice appreciated.
>
> Thought I would start with CVOICE and QOS first then IPT (4.x)...thoughts?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jonvoip at gmail.com  Thu Oct 11 17:14:15 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 11 Oct 2007 16:14:15 -0500
Subject: [cisco-voip] CCVP Certification
In-Reply-To: <f84a38d30710111405pe2e8553m634cec0f0dbf55@mail.gmail.com>
References: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
	<f84a38d30710111405pe2e8553m634cec0f0dbf55@mail.gmail.com>
Message-ID: <5d093f9a0710111414o17812de5j4b1fd59a11b31db4@mail.gmail.com>

To be honest, the CCM tests were a joke if you have ever deployed one.

The Cvoice test is pretty basic too.

The QoS test is basically just CBWFQ...

The GW/GK is the only challenging once cuz most people don't touch GKs...


Jonathan

On 10/11/07, Scott Voll <svoll.voip at gmail.com> wrote:
> Are you looking for understanding or a Cert.  if your just looking for the
> Cert. I would use a test practice like pass 4 sure.
>
> if your looking for the training A boot camp would be good.
>
> I started with what I was strong on.  CIPT.  I just went through the blue
> print and read the white pages on what I was weak on..... ie. IPMA, AAR,
> etc.
>
> depending on your level of comfort and weather your looking for training or
> cert, you might look at the CCIE bootcamp.
>
> Scott
>
>
> On 10/11/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> >
> >
> > Well I am finally going to start my CCVP exams...
> >
> > Helpful hints, studyguides preffered and any friendly advice appreciated.
> >
> > Thought I would start with CVOICE and QOS first then IPT (4.x)...thoughts?
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From matt at saskin.net  Thu Oct 11 17:23:18 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 11 Oct 2007 17:23:18 -0400
Subject: [cisco-voip] CCVP Certification
In-Reply-To: <5d093f9a0710111414o17812de5j4b1fd59a11b31db4@mail.gmail.com>
References: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>	<f84a38d30710111405pe2e8553m634cec0f0dbf55@mail.gmail.com>
	<5d093f9a0710111414o17812de5j4b1fd59a11b31db4@mail.gmail.com>
Message-ID: <470E9446.2000708@saskin.net>

Agreed the CVOICE was definitely a walk in the park.

I'm planning on the rest of the tests in the following order:
GWGK - shouldn't be too hard provided you have experience with both 
gateways and gatekeers (as has been mentioned, a lot of people don't 
touch the latter)
CIPT - also should be easy if you've done a CM deployment or use CM a lot
QOS - my weak spot by far
TUC - My thought is to do this last same as doing the CCNP 
troubleshooting exam last, since it somewhat covers all the other exams 
at the same time.

-matt

Jonathan Charles wrote:
> To be honest, the CCM tests were a joke if you have ever deployed one.
> 
> The Cvoice test is pretty basic too.
> 
> The QoS test is basically just CBWFQ...
> 
> The GW/GK is the only challenging once cuz most people don't touch GKs...
> 
> 
> Jonathan
> 
> On 10/11/07, Scott Voll <svoll.voip at gmail.com> wrote:
>> Are you looking for understanding or a Cert.  if your just looking for the
>> Cert. I would use a test practice like pass 4 sure.
>>
>> if your looking for the training A boot camp would be good.
>>
>> I started with what I was strong on.  CIPT.  I just went through the blue
>> print and read the white pages on what I was weak on..... ie. IPMA, AAR,
>> etc.
>>
>> depending on your level of comfort and weather your looking for training or
>> cert, you might look at the CCIE bootcamp.
>>
>> Scott
>>
>>
>> On 10/11/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
>>>
>>> Well I am finally going to start my CCVP exams...
>>>
>>> Helpful hints, studyguides preffered and any friendly advice appreciated.
>>>
>>> Thought I would start with CVOICE and QOS first then IPT (4.x)...thoughts?
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
A

From Brandon.Ogden at amersports.com  Thu Oct 11 17:30:07 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Thu, 11 Oct 2007 14:30:07 -0700
Subject: [cisco-voip] BAT Job Pending
In-Reply-To: <470E9446.2000708@saskin.net>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00CC22DF@seas7633.am.amersports.int>

Hey All, 
I am trying to run a BAT job on a CCM 5.1 and the job sticks in pending
and will not run.  I am using the appropriate template etc.  Any ideas?

From Brandon.Ogden at amersports.com  Thu Oct 11 17:33:49 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Thu, 11 Oct 2007 14:33:49 -0700
Subject: [cisco-voip] BAT Job Pending
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00CC22DF@seas7633.am.amersports.int>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00CC22E7@seas7633.am.amersports.int>

Nevermind.. found it here:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a00807cc050.shtml#prb3

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden, Brandon
Sent: Thursday, October 11, 2007 3:30 PM
To: Cisco Voice
Subject: [cisco-voip] BAT Job Pending

Hey All, 
I am trying to run a BAT job on a CCM 5.1 and the job sticks in pending
and will not run.  I am using the appropriate template etc.  Any ideas?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) 
please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

From jonvoip at gmail.com  Thu Oct 11 17:50:59 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 11 Oct 2007 16:50:59 -0500
Subject: [cisco-voip] CCVP Certification
In-Reply-To: <470E9446.2000708@saskin.net>
References: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
	<f84a38d30710111405pe2e8553m634cec0f0dbf55@mail.gmail.com>
	<5d093f9a0710111414o17812de5j4b1fd59a11b31db4@mail.gmail.com>
	<470E9446.2000708@saskin.net>
Message-ID: <5d093f9a0710111450t365a6723ha1b524b518773fbb@mail.gmail.com>

Actually, the way it is tested on the GWGK exam, a lot of people
haven't touched gateways, either.

I thought it was going to be an MGCP/H.323 exam, but it wasn't. It is
about the relationship between gateways and gatekeepers (if you know
about Gatekeepers, you will know what I am talking about...)


Jonathan

On 10/11/07, Matthew Saskin <matt at saskin.net> wrote:
> Agreed the CVOICE was definitely a walk in the park.
>
> I'm planning on the rest of the tests in the following order:
> GWGK - shouldn't be too hard provided you have experience with both
> gateways and gatekeers (as has been mentioned, a lot of people don't
> touch the latter)
> CIPT - also should be easy if you've done a CM deployment or use CM a lot
> QOS - my weak spot by far
> TUC - My thought is to do this last same as doing the CCNP
> troubleshooting exam last, since it somewhat covers all the other exams
> at the same time.
>
> -matt
>
> Jonathan Charles wrote:
> > To be honest, the CCM tests were a joke if you have ever deployed one.
> >
> > The Cvoice test is pretty basic too.
> >
> > The QoS test is basically just CBWFQ...
> >
> > The GW/GK is the only challenging once cuz most people don't touch GKs...
> >
> >
> > Jonathan
> >
> > On 10/11/07, Scott Voll <svoll.voip at gmail.com> wrote:
> >> Are you looking for understanding or a Cert.  if your just looking for the
> >> Cert. I would use a test practice like pass 4 sure.
> >>
> >> if your looking for the training A boot camp would be good.
> >>
> >> I started with what I was strong on.  CIPT.  I just went through the blue
> >> print and read the white pages on what I was weak on..... ie. IPMA, AAR,
> >> etc.
> >>
> >> depending on your level of comfort and weather your looking for training or
> >> cert, you might look at the CCIE bootcamp.
> >>
> >> Scott
> >>
> >>
> >> On 10/11/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> >>>
> >>> Well I am finally going to start my CCVP exams...
> >>>
> >>> Helpful hints, studyguides preffered and any friendly advice appreciated.
> >>>
> >>> Thought I would start with CVOICE and QOS first then IPT (4.x)...thoughts?
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip at puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> A
>

From jgrace at adcapnet.com  Thu Oct 11 18:42:00 2007
From: jgrace at adcapnet.com (James Grace)
Date: Thu, 11 Oct 2007 18:42:00 -0400
Subject: [cisco-voip] getting a spanlink voice monitor error on CRS server
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E880@mail3.adcapnetsvc.local>

?
my cpu on the CRS server is a  100%.  Its this high all the time.  after looking around i found the below in the event viewer
 
FATAL FCVMS112 splk_pcap_open_live() failed. errorBuf = Error
opening adapter: Access is denied.
 
this server was configured for server base monitoring and then we decided to user desktop base monitoring. i disable the second nic and desktop monitoring is working fine.  Is there something i missed.  Do i need to remove something in the registry or something.
 
James Grace 
 
Systems Engineer
<http://www.adcapnet.com/>  
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
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This original email was sent to the internet for delivery at 11 Oct 2007 18:40:27 -0400
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From mark.mills at baesystems.com  Thu Oct 11 21:41:57 2007
From: mark.mills at baesystems.com (MILLS, Mark)
Date: Fri, 12 Oct 2007 11:11:57 +0930
Subject: [cisco-voip] VG224 buzzing/humming on incoming calls only
In-Reply-To: <483125.82354.qm@web57205.mail.re3.yahoo.com>
References: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com>
	<483125.82354.qm@web57205.mail.re3.yahoo.com>
Message-ID: <54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>


Thanks to those that replied.

The device is grounded, and tip/ring are correct.

I have actually noticed something else with the problem which seems very
strange to me:

If someone tries to dial the VG224 extension (x600) and the analog
handset is physically connected to the VG224 port, the analog handset
rings, and when its answered, the calling party gets the buzzing/humming
noise, and the analog handset hears nothing.

BUT, if the analog handset is NOT connected to the vg224 when the call
attempted, and you then physically plug it in while the ring attempt is
being made, the analog handset starts ringing and the call can be
answered correctly and both parties can hear each other.

Does anyone else have any suggestions?

Cheers,  
  Mark

 

> -----Original Message-----
> From: Paul Choi [mailto:asobihoudai at yahoo.com] 
> Sent: Thursday, 11 October 2007 11:36 PM
> To: Kevin Dunn; MILLS, Mark
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] VG224 buzzing/humming on incoming calls only
> 
> Are tip and ring reversed?
> 
> Paul
> 
> --- Kevin Dunn <cheesevoice at gmail.com> wrote:
> 
> > I just had this happen with a VG248.  Same exact
> > issue, turned out the
> > analog lines had been victims of some lightning, the
> > Telco replaced teh
> > facilities for the analog lines and the crackling
> > went away... Hope that
> > helps.
> > 
> > On 10/11/07, MILLS, Mark <mark.mills at baesystems.com>
> > wrote:
> > >
> > >  Hi All,
> > >
> > > I was hoping I might be able to get some help with
> > the setup of a VG224
> > > that I am doing.  It is connected to CCM6 cluster.
> >  I can place calls FROM
> > > the analog phones connected to the ports, and it
> > all works properly with two
> > > way audio.
> > >
> > > When making calls TO the analog lines on the
> > VG224, the calling party
> > > simply hears a loud buzzing/humming noise and
> > nothing else.  The analog user
> > > can hear nothing.
> > >
> > > This happens for incoming PSTN calls and calls to
> > other extensions to the
> > > VG224 connected devices.
> > >
> > > I have not setup a VG224 for incoming calls
> > before, so there is obviously
> > > something I have missed.  Below is the relevant
> > config parts on the device.
> > > Any help is appreciated!
> > >
> > >
> > >
> > > stcapp ccm-group 1
> > > stcapp
> > > !
> > > voice-port 2/0
> > >  ren 3
> > >  idle-voltage low
> > >  disc_pi_off
> > >  input gain 10
> > >  output attenuation 10
> > >  playout-delay minimum low
> > >  cptone AU
> > >  timing digit 53
> > >  bearer-cap Speech
> > >  station-id number XXXXXXXXXX
> > >  caller-id enable
> > > !
> > > sccp local FastEthernet0/0
> > > sccp ccm 10.194.3.10 identifier 20 version 4.1
> > > sccp ccm 10.194.4.10 identifier 10 version 4.1
> > > sccp
> > > !
> > > sccp ccm group 1
> > >  associate ccm 10 priority 1
> > >  associate ccm 20 priority 2
> > >  switchback method graceful
> > > !
> > > dial-peer voice XXXXXXXXXX pots
> > >  service stcapp
> > >  port 2/23
> > > !
> > >
> > > Cheers,
> > >    Mark
> > >
> > > "Warning:
> > > The information contained in this email and any
> > attached files is
> > > confidential to BAE Systems Australia. If you are
> > not the intended
> > > recipient, any use, disclosure or copying of this
> > email or any
> > > attachments is expressly prohibited.  If you have
> > received this email
> > > in error, please notify us immediately. VIRUS:
> > Every care has been
> > > taken to ensure this email and its attachments are
> > virus free,
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> > email is not the
> > > sender's responsibility.  It is your
> > responsibility to ensure virus
> > > checks are completed before installing any data
> > sent in this email to
> > > your computer."
> > >
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
>        
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"Warning:
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From david.lin at msn.com  Thu Oct 11 21:47:40 2007
From: david.lin at msn.com (David Lin)
Date: Thu, 11 Oct 2007 21:47:40 -0400
Subject: [cisco-voip] dial-peer circulation iusse
Message-ID: <BAY132-W35B0AB39A94636E51A460AFEA00@phx.gbl>


Hi all,
 
In order to conver the SIP call to H323 and G.711 to G.729/723, it means SIP call-->Port 3/1-->port 3/2-->another H323 gateway. I set cross-connection on two T1 interfaces on one Cisco gateway and the dial-peer are as following, 
dial-peer voice 100 voip incoming called-number 999T session protocol sipv2!dial-peer voice 200 pots destination-pattern 200T port 3/1:D prefix 999!dial-peer voice 300 pots incoming called-number 999T direct-inward-dial!dial-peer voice 400 voip destination-pattern 999T session target ipv4:10.10.30.40
 
Somehow the gateway keeps trying the dial-peer before sending out the H323 Call to the far end gateway, but some time not. It looks like the call will exhaust the channel resource. What's wrong in the configuration? Can anyone advise? Thank you.
 
CallID       CID  ccVdb          Port       DSP/Ch  Called #      Codec    Dial-peers0xA09      1D4F 0x64A5B6D0 3/1:D.0   3006/3 *5195226699 g711alaw 100/2000xA0A      1D50 0x64B7ADF0 3/2:D.0   No dsp  5195226699 None     300/2000xA0B      1D50 0x64A5B6D0 3/1:D.1   No dsp *5195226699 None     200/3000xA0C      1D53 0x64B7ADF0 3/2:D.1   No dsp  5195226699 None     300/2000xA0D      1D53 0x64A5B6D0 3/1:D.2   No dsp *5195226699 None     200/3000xA0E      1D56 0x64B7ADF0 3/2:D.2   No dsp  5195226699 None     300/2000xA0F      1D56 0x64A5B6D0 3/1:D.3   No dsp *5195226699 None     200/3000xA10      1D59 0x64B7ADF0 3/2:D.3   3008/2  5195226699 g729r8   300/400
 
0xA2B      1D7A 0x64A5B6D0 3/1:D.4   3009/5 *4163955400 g711ulaw 100/2000xA2C      1D7B 0x64B7ADF0 3/2:D.4   3008/4  4163955400 g729r8   300/4007 active calls found
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From asobihoudai at yahoo.com  Thu Oct 11 21:54:52 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 11 Oct 2007 18:54:52 -0700 (PDT)
Subject: [cisco-voip] VG224 buzzing/humming on incoming calls only
In-Reply-To: <54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>
Message-ID: <970680.70745.qm@web57207.mail.re3.yahoo.com>

Mark,

This is just a guess but are the lines coming in using
the right signaling? Is it possible that
loop-start/ground-start are mixed up?

Paul

--- "MILLS, Mark" <mark.mills at baesystems.com> wrote:

> 
> Thanks to those that replied.
> 
> The device is grounded, and tip/ring are correct.
> 
> I have actually noticed something else with the
> problem which seems very
> strange to me:
> 
> If someone tries to dial the VG224 extension (x600)
> and the analog
> handset is physically connected to the VG224 port,
> the analog handset
> rings, and when its answered, the calling party gets
> the buzzing/humming
> noise, and the analog handset hears nothing.
> 
> BUT, if the analog handset is NOT connected to the
> vg224 when the call
> attempted, and you then physically plug it in while
> the ring attempt is
> being made, the analog handset starts ringing and
> the call can be
> answered correctly and both parties can hear each
> other.
> 
> Does anyone else have any suggestions?
> 
> Cheers,  
>   Mark
> 
>  
> 
> > -----Original Message-----
> > From: Paul Choi [mailto:asobihoudai at yahoo.com] 
> > Sent: Thursday, 11 October 2007 11:36 PM
> > To: Kevin Dunn; MILLS, Mark
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] VG224 buzzing/humming on
> incoming calls only
> > 
> > Are tip and ring reversed?
> > 
> > Paul
> > 
> > --- Kevin Dunn <cheesevoice at gmail.com> wrote:
> > 
> > > I just had this happen with a VG248.  Same exact
> > > issue, turned out the
> > > analog lines had been victims of some lightning,
> the
> > > Telco replaced teh
> > > facilities for the analog lines and the
> crackling
> > > went away... Hope that
> > > helps.
> > > 
> > > On 10/11/07, MILLS, Mark
> <mark.mills at baesystems.com>
> > > wrote:
> > > >
> > > >  Hi All,
> > > >
> > > > I was hoping I might be able to get some help
> with
> > > the setup of a VG224
> > > > that I am doing.  It is connected to CCM6
> cluster.
> > >  I can place calls FROM
> > > > the analog phones connected to the ports, and
> it
> > > all works properly with two
> > > > way audio.
> > > >
> > > > When making calls TO the analog lines on the
> > > VG224, the calling party
> > > > simply hears a loud buzzing/humming noise and
> > > nothing else.  The analog user
> > > > can hear nothing.
> > > >
> > > > This happens for incoming PSTN calls and calls
> to
> > > other extensions to the
> > > > VG224 connected devices.
> > > >
> > > > I have not setup a VG224 for incoming calls
> > > before, so there is obviously
> > > > something I have missed.  Below is the
> relevant
> > > config parts on the device.
> > > > Any help is appreciated!
> > > >
> > > >
> > > >
> > > > stcapp ccm-group 1
> > > > stcapp
> > > > !
> > > > voice-port 2/0
> > > >  ren 3
> > > >  idle-voltage low
> > > >  disc_pi_off
> > > >  input gain 10
> > > >  output attenuation 10
> > > >  playout-delay minimum low
> > > >  cptone AU
> > > >  timing digit 53
> > > >  bearer-cap Speech
> > > >  station-id number XXXXXXXXXX
> > > >  caller-id enable
> > > > !
> > > > sccp local FastEthernet0/0
> > > > sccp ccm 10.194.3.10 identifier 20 version 4.1
> > > > sccp ccm 10.194.4.10 identifier 10 version 4.1
> > > > sccp
> > > > !
> > > > sccp ccm group 1
> > > >  associate ccm 10 priority 1
> > > >  associate ccm 20 priority 2
> > > >  switchback method graceful
> > > > !
> > > > dial-peer voice XXXXXXXXXX pots
> > > >  service stcapp
> > > >  port 2/23
> > > > !
> > > >
> > > > Cheers,
> > > >    Mark
> > > >
> > > > "Warning:
> > > > The information contained in this email and
> any
> > > attached files is
> > > > confidential to BAE Systems Australia. If you
> are
> > > not the intended
> > > > recipient, any use, disclosure or copying of
> this
> > > email or any
> > > > attachments is expressly prohibited.  If you
> have
> > > received this email
> > > > in error, please notify us immediately. VIRUS:
> > > Every care has been
> > > > taken to ensure this email and its attachments
> are
> > > virus free,
> > > > however, any loss or damage incurred in using
> this
> > > email is not the
> > > > sender's responsibility.  It is your
> > > responsibility to ensure virus
> > > > checks are completed before installing any
> data
> > > sent in this email to
> > > > your computer."
> > > >
> > > >
> > > >
> > > >
> _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > 
> > 
> >        
> >
>
______________________________________________________________
> > ______________________
> > Building a website is a piece of cake. Yahoo!
> Small Business 
> > gives you all the tools to get online.
> > http://smallbusiness.yahoo.com/webhosting 
> > 
> "Warning:
> The information contained in this email and any
> attached files is
> confidential to BAE Systems Australia. If you are
> not the intended
> recipient, any use, disclosure or copying of this
> email or any
> attachments is expressly prohibited.  If you have
> received this email
> in error, please notify us immediately. VIRUS: Every
> care has been
> taken to ensure this email and its attachments are
> virus free,
> however, any loss or damage incurred in using this
> email is not the
> sender's responsibility.  It is your responsibility
> to ensure virus
> checks are completed before installing any data sent
> in this email to
> your computer."
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



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From david.lin at msn.com  Thu Oct 11 21:55:27 2007
From: david.lin at msn.com (David Lin)
Date: Thu, 11 Oct 2007 21:55:27 -0400
Subject: [cisco-voip] dial-peer circulation iusse
In-Reply-To: <BAY132-W35B0AB39A94636E51A460AFEA00@phx.gbl>
References: <BAY132-W35B0AB39A94636E51A460AFEA00@phx.gbl>
Message-ID: <BAY132-W2131868899D266054345AFEA00@phx.gbl>


Sry, correction:
dial-peer voice 200 pots destination-pattern 999T   <<<---Not 200T port 3/1:D prefix 999


From: david.lin at msn.comTo: cisco-voip at puck.nether.netDate: Thu, 11 Oct 2007 21:47:40 -0400Subject: [cisco-voip] dial-peer circulation iusse


Hi all, In order to conver the SIP call to H323 and G.711 to G.729/723, it means SIP call-->Port 3/1-->port 3/2-->another H323 gateway. I set cross-connection on two T1 interfaces on one Cisco gateway and the dial-peer are as following, dial-peer voice 100 voip incoming called-number 999T session protocol sipv2!dial-peer voice 200 pots destination-pattern 200T port 3/1:D prefix 999!dial-peer voice 300 pots incoming called-number 999T direct-inward-dial!dial-peer voice 400 voip destination-pattern 999T session target ipv4:10.10.30.40 Somehow the gateway keeps trying the dial-peer before sending out the H323 Call to the far end gateway, but some time not. It looks like the call will exhaust the channel resource. What's wrong in the configuration? Can anyone advise? Thank you. CallID       CID  ccVdb          Port       DSP/Ch  Called #      Codec    Dial-peers0xA09      1D4F 0x64A5B6D0 3/1:D.0   3006/3 *5195226699 g711alaw 100/2000xA0A      1D50 0x64B7ADF0 3/2:D.0   No dsp  5195226699 None     300/2000xA0B      1D50 0x64A5B6D0 3/1:D.1   No dsp *5195226699 None     200/3000xA0C      1D53 0x64B7ADF0 3/2:D.1   No dsp  5195226699 None     300/2000xA0D      1D53 0x64A5B6D0 3/1:D.2   No dsp *5195226699 None     200/3000xA0E      1D56 0x64B7ADF0 3/2:D.2   No dsp  5195226699 None     300/2000xA0F      1D56 0x64A5B6D0 3/1:D.3   No dsp *5195226699 None     200/3000xA10      1D59 0x64B7ADF0 3/2:D.3   3008/2  5195226699 g729r8   300/400 0xA2B      1D7A 0x64A5B6D0 3/1:D.4   3009/5 *4163955400 g711ulaw 100/2000xA2C      1D7B 0x64B7ADF0 3/2:D.4   3008/4  4163955400 g729r8   300/4007 active calls found

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From asobihoudai at yahoo.com  Thu Oct 11 22:37:41 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 11 Oct 2007 19:37:41 -0700 (PDT)
Subject: [cisco-voip] dial-peer circulation iusse
In-Reply-To: <BAY132-W2131868899D266054345AFEA00@phx.gbl>
Message-ID: <958815.21234.qm@web57203.mail.re3.yahoo.com>

Use preference statements to make sure that your pots
dial-peer is first in line.

Paul
--- David Lin <david.lin at msn.com> wrote:

> 
> Sry, correction:
> dial-peer voice 200 pots destination-pattern 999T  
> <<<---Not 200T port 3/1:D prefix 999
> 
> 
> From: david.lin at msn.comTo:
> cisco-voip at puck.nether.netDate: Thu, 11 Oct 2007
> 21:47:40 -0400Subject: [cisco-voip] dial-peer
> circulation iusse
> 
> 
> Hi all, In order to conver the SIP call to H323 and
> G.711 to G.729/723, it means SIP call-->Port
> 3/1-->port 3/2-->another H323 gateway. I set
> cross-connection on two T1 interfaces on one Cisco
> gateway and the dial-peer are as following,
> dial-peer voice 100 voip incoming called-number 999T
> session protocol sipv2!dial-peer voice 200 pots
> destination-pattern 200T port 3/1:D prefix
> 999!dial-peer voice 300 pots incoming called-number
> 999T direct-inward-dial!dial-peer voice 400 voip
> destination-pattern 999T session target
> ipv4:10.10.30.40 Somehow the gateway keeps trying
> the dial-peer before sending out the H323 Call to
> the far end gateway, but some time not. It looks
> like the call will exhaust the channel resource.
> What's wrong in the configuration? Can anyone
> advise? Thank you. CallID       CID  ccVdb         
> Port       DSP/Ch  Called #      Codec   
> Dial-peers0xA09      1D4F 0x64A5B6D0 3/1:D.0  
> 3006/3 *5195226699 g711alaw 100/2000xA0A      1D50
> 0x64B7ADF0 3/2:D.0   No dsp  5195226699 None    
> 300/2000xA0B      1D50 0x64A5B6D0 3/1:D.1   No dsp
> *5195226699 None     200/3000xA0C      1D53
> 0x64B7ADF0 3/2:D.1   No dsp  5195226699 None    
> 300/2000xA0D      1D53 0x64A5B6D0 3/1:D.2   No dsp
> *5195226699 None     200/3000xA0E      1D56
> 0x64B7ADF0 3/2:D.2   No dsp  5195226699 None    
> 300/2000xA0F      1D56 0x64A5B6D0 3/1:D.3   No dsp
> *5195226699 None     200/3000xA10      1D59
> 0x64B7ADF0 3/2:D.3   3008/2  5195226699 g729r8  
> 300/400 0xA2B      1D7A 0x64A5B6D0 3/1:D.4   3009/5
> *4163955400 g711ulaw 100/2000xA2C      1D7B
> 0x64B7ADF0 3/2:D.4   3008/4  4163955400 g729r8  
> 300/4007 active calls found
> 
> Are you ready for Windows Live Messenger Beta 8.5 ?
> Get the latest for free today! 
>
_________________________________________________________________
> R U Ready for Windows Live Messenger Beta 8.5? Try
> it today!
>
http://entertainment.sympatico.msn.ca/WindowsLiveMessenger>
_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
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From david.lin at msn.com  Thu Oct 11 23:07:38 2007
From: david.lin at msn.com (David Lin)
Date: Thu, 11 Oct 2007 23:07:38 -0400
Subject: [cisco-voip] dial-peer circulation iusse
In-Reply-To: <958815.21234.qm@web57203.mail.re3.yahoo.com>
References: <BAY132-W2131868899D266054345AFEA00@phx.gbl>
	<958815.21234.qm@web57203.mail.re3.yahoo.com>
Message-ID: <BAY132-W297B32AE9681AF31C9320FFEA00@phx.gbl>


I have only one dial-peer to pot, only one T1 for terminiating the calls.So I don't think it's the preference problem.
David
> Date: Thu, 11 Oct 2007 19:37:41 -0700> From: asobihoudai at yahoo.com> Subject: Re: [cisco-voip] dial-peer circulation iusse> To: david.lin at msn.com; cisco-voip at puck.nether.net> > Use preference statements to make sure that your pots> dial-peer is first in line.> > Paul
_________________________________________________________________
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From jason.aarons at us.didata.com  Thu Oct 11 23:11:15 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 11 Oct 2007 23:11:15 -0400
Subject: [cisco-voip] VG224 buzzing/humming on incoming calls only
In-Reply-To: <54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>
References: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com><483125.82354.qm@web57205.mail.re3.yahoo.com>
	<54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0071030AF@USNAEXCH.na.didata.local>

I had the exact same issue, it was odd, a buttset worked but the fax
machine you heard hum, tried several versions of IOS before a IOS
downgrade fixed this issue. 

Summary;
Two VG224s on different floors, you can fax or call out, incoming you
hear a buzz on both fax machines and analog polycoms.  Cannot receive
faxes.  Using a Buttset you don't hear any problems. 

Resolution
Changed VG224 IOS from vg224-i6s-mz.124-15.T1.bin to
vg224-i6s-mz.124-9.T5.bin to vg224-i6s-mz.124-6.T8.bin (worked).

Contact TAC and ask them to read case 606774799, it went to the DSP
development manager, since the downgrade fixed the problem and it was
for executive floor at a bank I could not put the IOS back and help TAC
fix the problem.

Mine was SCCP, I want to say it did the same with H323. We did a pcm
capture and the capture showed the fax machine was making the hum, which
proved false.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of MILLS, Mark
Sent: Thursday, October 11, 2007 9:42 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VG224 buzzing/humming on incoming calls only


Thanks to those that replied.

The device is grounded, and tip/ring are correct.

I have actually noticed something else with the problem which seems very
strange to me:

If someone tries to dial the VG224 extension (x600) and the analog
handset is physically connected to the VG224 port, the analog handset
rings, and when its answered, the calling party gets the buzzing/humming
noise, and the analog handset hears nothing.

BUT, if the analog handset is NOT connected to the vg224 when the call
attempted, and you then physically plug it in while the ring attempt is
being made, the analog handset starts ringing and the call can be
answered correctly and both parties can hear each other.

Does anyone else have any suggestions?

Cheers,  
  Mark

 

> -----Original Message-----
> From: Paul Choi [mailto:asobihoudai at yahoo.com] 
> Sent: Thursday, 11 October 2007 11:36 PM
> To: Kevin Dunn; MILLS, Mark
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] VG224 buzzing/humming on incoming calls only
> 
> Are tip and ring reversed?
> 
> Paul
> 
> --- Kevin Dunn <cheesevoice at gmail.com> wrote:
> 
> > I just had this happen with a VG248.  Same exact
> > issue, turned out the
> > analog lines had been victims of some lightning, the
> > Telco replaced teh
> > facilities for the analog lines and the crackling
> > went away... Hope that
> > helps.
> > 
> > On 10/11/07, MILLS, Mark <mark.mills at baesystems.com>
> > wrote:
> > >
> > >  Hi All,
> > >
> > > I was hoping I might be able to get some help with
> > the setup of a VG224
> > > that I am doing.  It is connected to CCM6 cluster.
> >  I can place calls FROM
> > > the analog phones connected to the ports, and it
> > all works properly with two
> > > way audio.
> > >
> > > When making calls TO the analog lines on the
> > VG224, the calling party
> > > simply hears a loud buzzing/humming noise and
> > nothing else.  The analog user
> > > can hear nothing.
> > >
> > > This happens for incoming PSTN calls and calls to
> > other extensions to the
> > > VG224 connected devices.
> > >
> > > I have not setup a VG224 for incoming calls
> > before, so there is obviously
> > > something I have missed.  Below is the relevant
> > config parts on the device.
> > > Any help is appreciated!
> > >
> > >
> > >
> > > stcapp ccm-group 1
> > > stcapp
> > > !
> > > voice-port 2/0
> > >  ren 3
> > >  idle-voltage low
> > >  disc_pi_off
> > >  input gain 10
> > >  output attenuation 10
> > >  playout-delay minimum low
> > >  cptone AU
> > >  timing digit 53
> > >  bearer-cap Speech
> > >  station-id number XXXXXXXXXX
> > >  caller-id enable
> > > !
> > > sccp local FastEthernet0/0
> > > sccp ccm 10.194.3.10 identifier 20 version 4.1
> > > sccp ccm 10.194.4.10 identifier 10 version 4.1
> > > sccp
> > > !
> > > sccp ccm group 1
> > >  associate ccm 10 priority 1
> > >  associate ccm 20 priority 2
> > >  switchback method graceful
> > > !
> > > dial-peer voice XXXXXXXXXX pots
> > >  service stcapp
> > >  port 2/23
> > > !
> > >
> > > Cheers,
> > >    Mark
> > >
> > > "Warning:
> > > The information contained in this email and any
> > attached files is
> > > confidential to BAE Systems Australia. If you are
> > not the intended
> > > recipient, any use, disclosure or copying of this
> > email or any
> > > attachments is expressly prohibited.  If you have
> > received this email
> > > in error, please notify us immediately. VIRUS:
> > Every care has been
> > > taken to ensure this email and its attachments are
> > virus free,
> > > however, any loss or damage incurred in using this
> > email is not the
> > > sender's responsibility.  It is your
> > responsibility to ensure virus
> > > checks are completed before installing any data
> > sent in this email to
> > > your computer."
> > >
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
>        
> ______________________________________________________________
> ______________________
> Building a website is a piece of cake. Yahoo! Small Business 
> gives you all the tools to get online.
> http://smallbusiness.yahoo.com/webhosting 
> 
"Warning:
The information contained in this email and any attached files is
confidential to BAE Systems Australia. If you are not the intended
recipient, any use, disclosure or copying of this email or any
attachments is expressly prohibited.  If you have received this email
in error, please notify us immediately. VIRUS: Every care has been
taken to ensure this email and its attachments are virus free,
however, any loss or damage incurred in using this email is not the
sender's responsibility.  It is your responsibility to ensure virus
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This e-mail communication and any attachments may contain
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From aman.chugh at gmail.com  Fri Oct 12 01:14:39 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Fri, 12 Oct 2007 10:44:39 +0530
Subject: [cisco-voip] CCVP Certification
In-Reply-To: <5d093f9a0710111450t365a6723ha1b524b518773fbb@mail.gmail.com>
References: <84b74aec0710111228y47e2d1a1s1c9f269dc441324a@mail.gmail.com>
	<f84a38d30710111405pe2e8553m634cec0f0dbf55@mail.gmail.com>
	<5d093f9a0710111414o17812de5j4b1fd59a11b31db4@mail.gmail.com>
	<470E9446.2000708@saskin.net>
	<5d093f9a0710111450t365a6723ha1b524b518773fbb@mail.gmail.com>
Message-ID: <d4c697ad0710112214t6f7cc18fh8a04a7d6b8ad29ba@mail.gmail.com>

I did the GW GK exam couple of weeks back and the entire exam is based on
Cisco Voice GW and GK cisco press book, exam is easy if you go through book
, it does have couple of simulations where you need to configure zones on
the Gk , but the book does cover a lot of stuff, Overall I didnt find the
exam that tuff.

Unicast me if you need more help regarding GW GK.

Aman


On 10/12/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Actually, the way it is tested on the GWGK exam, a lot of people
> haven't touched gateways, either.
>
> I thought it was going to be an MGCP/H.323 exam, but it wasn't. It is
> about the relationship between gateways and gatekeepers (if you know
> about Gatekeepers, you will know what I am talking about...)
>
>
> Jonathan
>
> On 10/11/07, Matthew Saskin <matt at saskin.net> wrote:
> > Agreed the CVOICE was definitely a walk in the park.
> >
> > I'm planning on the rest of the tests in the following order:
> > GWGK - shouldn't be too hard provided you have experience with both
> > gateways and gatekeers (as has been mentioned, a lot of people don't
> > touch the latter)
> > CIPT - also should be easy if you've done a CM deployment or use CM a
> lot
> > QOS - my weak spot by far
> > TUC - My thought is to do this last same as doing the CCNP
> > troubleshooting exam last, since it somewhat covers all the other exams
> > at the same time.
> >
> > -matt
> >
> > Jonathan Charles wrote:
> > > To be honest, the CCM tests were a joke if you have ever deployed one.
> > >
> > > The Cvoice test is pretty basic too.
> > >
> > > The QoS test is basically just CBWFQ...
> > >
> > > The GW/GK is the only challenging once cuz most people don't touch
> GKs...
> > >
> > >
> > > Jonathan
> > >
> > > On 10/11/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > >> Are you looking for understanding or a Cert.  if your just looking
> for the
> > >> Cert. I would use a test practice like pass 4 sure.
> > >>
> > >> if your looking for the training A boot camp would be good.
> > >>
> > >> I started with what I was strong on.  CIPT.  I just went through the
> blue
> > >> print and read the white pages on what I was weak on..... ie. IPMA,
> AAR,
> > >> etc.
> > >>
> > >> depending on your level of comfort and weather your looking for
> training or
> > >> cert, you might look at the CCIE bootcamp.
> > >>
> > >> Scott
> > >>
> > >>
> > >> On 10/11/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> > >>>
> > >>> Well I am finally going to start my CCVP exams...
> > >>>
> > >>> Helpful hints, studyguides preffered and any friendly advice
> appreciated.
> > >>>
> > >>> Thought I would start with CVOICE and QOS first then IPT (4.x
> )...thoughts?
> > >>> _______________________________________________
> > >>> cisco-voip mailing list
> > >>> cisco-voip at puck.nether.net
> > >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>>
> > >>
> > >> _______________________________________________
> > >> cisco-voip mailing list
> > >> cisco-voip at puck.nether.net
> > >> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > A
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From mark.mills at baesystems.com  Fri Oct 12 04:01:38 2007
From: mark.mills at baesystems.com (MILLS, Mark)
Date: Fri, 12 Oct 2007 17:31:38 +0930
Subject: [cisco-voip] VG224 buzzing/humming on incoming calls only
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0071030AF@USNAEXCH.na.didata.local>
References: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com><483125.82354.qm@web57205.mail.re3.yahoo.com>
	<54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>
	<C1FE15183DA37645BC0633BC604E44F0071030AF@USNAEXCH.na.didata.local>
Message-ID: <54DD8B39132EFA4293A5C0565004E5460A05FA34@SBEMAIL1.au.baesystems.com>

Fantastic.. This was the problem!

I have rolled back to the older IOS image, and it is now working
successfully.

Its logged with the TAC and hopefully they fix this bug soon.

Thanks so much for everyones advice and assistance.

Cheers,
   Mark 

> -----Original Message-----
> From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com] 
> Sent: Friday, 12 October 2007 12:41 PM
> To: MILLS, Mark; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] VG224 buzzing/humming on incoming calls only
> 
> I had the exact same issue, it was odd, a buttset worked but the fax
> machine you heard hum, tried several versions of IOS before a IOS
> downgrade fixed this issue. 
> 
> Summary;
> Two VG224s on different floors, you can fax or call out, incoming you
> hear a buzz on both fax machines and analog polycoms.  Cannot receive
> faxes.  Using a Buttset you don't hear any problems. 
> 
> Resolution
> Changed VG224 IOS from vg224-i6s-mz.124-15.T1.bin to
> vg224-i6s-mz.124-9.T5.bin to vg224-i6s-mz.124-6.T8.bin (worked).
> 
> Contact TAC and ask them to read case 606774799, it went to the DSP
> development manager, since the downgrade fixed the problem and it was
> for executive floor at a bank I could not put the IOS back 
> and help TAC
> fix the problem.
> 
> Mine was SCCP, I want to say it did the same with H323. We did a pcm
> capture and the capture showed the fax machine was making the 
> hum, which
> proved false.
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of MILLS, Mark
> Sent: Thursday, October 11, 2007 9:42 PM
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] VG224 buzzing/humming on incoming calls only
> 
> 
> Thanks to those that replied.
> 
> The device is grounded, and tip/ring are correct.
> 
> I have actually noticed something else with the problem which 
> seems very
> strange to me:
> 
> If someone tries to dial the VG224 extension (x600) and the analog
> handset is physically connected to the VG224 port, the analog handset
> rings, and when its answered, the calling party gets the 
> buzzing/humming
> noise, and the analog handset hears nothing.
> 
> BUT, if the analog handset is NOT connected to the vg224 when the call
> attempted, and you then physically plug it in while the ring 
> attempt is
> being made, the analog handset starts ringing and the call can be
> answered correctly and both parties can hear each other.
> 
> Does anyone else have any suggestions?
> 
> Cheers,  
>   Mark
> 
>  
> 
> > -----Original Message-----
> > From: Paul Choi [mailto:asobihoudai at yahoo.com] 
> > Sent: Thursday, 11 October 2007 11:36 PM
> > To: Kevin Dunn; MILLS, Mark
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] VG224 buzzing/humming on incoming 
> calls only
> > 
> > Are tip and ring reversed?
> > 
> > Paul
> > 
> > --- Kevin Dunn <cheesevoice at gmail.com> wrote:
> > 
> > > I just had this happen with a VG248.  Same exact
> > > issue, turned out the
> > > analog lines had been victims of some lightning, the
> > > Telco replaced teh
> > > facilities for the analog lines and the crackling
> > > went away... Hope that
> > > helps.
> > > 
> > > On 10/11/07, MILLS, Mark <mark.mills at baesystems.com>
> > > wrote:
> > > >
> > > >  Hi All,
> > > >
> > > > I was hoping I might be able to get some help with
> > > the setup of a VG224
> > > > that I am doing.  It is connected to CCM6 cluster.
> > >  I can place calls FROM
> > > > the analog phones connected to the ports, and it
> > > all works properly with two
> > > > way audio.
> > > >
> > > > When making calls TO the analog lines on the
> > > VG224, the calling party
> > > > simply hears a loud buzzing/humming noise and
> > > nothing else.  The analog user
> > > > can hear nothing.
> > > >
> > > > This happens for incoming PSTN calls and calls to
> > > other extensions to the
> > > > VG224 connected devices.
> > > >
> > > > I have not setup a VG224 for incoming calls
> > > before, so there is obviously
> > > > something I have missed.  Below is the relevant
> > > config parts on the device.
> > > > Any help is appreciated!
> > > >
> > > >
> > > >
> > > > stcapp ccm-group 1
> > > > stcapp
> > > > !
> > > > voice-port 2/0
> > > >  ren 3
> > > >  idle-voltage low
> > > >  disc_pi_off
> > > >  input gain 10
> > > >  output attenuation 10
> > > >  playout-delay minimum low
> > > >  cptone AU
> > > >  timing digit 53
> > > >  bearer-cap Speech
> > > >  station-id number XXXXXXXXXX
> > > >  caller-id enable
> > > > !
> > > > sccp local FastEthernet0/0
> > > > sccp ccm 10.194.3.10 identifier 20 version 4.1
> > > > sccp ccm 10.194.4.10 identifier 10 version 4.1
> > > > sccp
> > > > !
> > > > sccp ccm group 1
> > > >  associate ccm 10 priority 1
> > > >  associate ccm 20 priority 2
> > > >  switchback method graceful
> > > > !
> > > > dial-peer voice XXXXXXXXXX pots
> > > >  service stcapp
> > > >  port 2/23
> > > > !
> > > >
> > > > Cheers,
> > > >    Mark
> > > >
> > > > "Warning:
> > > > The information contained in this email and any
> > > attached files is
> > > > confidential to BAE Systems Australia. If you are
> > > not the intended
> > > > recipient, any use, disclosure or copying of this
> > > email or any
> > > > attachments is expressly prohibited.  If you have
> > > received this email
> > > > in error, please notify us immediately. VIRUS:
> > > Every care has been
> > > > taken to ensure this email and its attachments are
> > > virus free,
> > > > however, any loss or damage incurred in using this
> > > email is not the
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> > > > checks are completed before installing any data
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> > > >
> > > >
> > > >
> > > > _______________________________________________
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> > > > cisco-voip at puck.nether.net
> > > >
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> > > >
> > > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > 
> > 
> >        
> > ______________________________________________________________
> > ______________________
> > Building a website is a piece of cake. Yahoo! Small Business 
> > gives you all the tools to get online.
> > http://smallbusiness.yahoo.com/webhosting 
> > 
> "Warning:
> The information contained in this email and any attached files is
> confidential to BAE Systems Australia. If you are not the intended
> recipient, any use, disclosure or copying of this email or any
> attachments is expressly prohibited.  If you have received this email
> in error, please notify us immediately. VIRUS: Every care has been
> taken to ensure this email and its attachments are virus free,
> however, any loss or damage incurred in using this email is not the
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> checks are completed before installing any data sent in this email to
> your computer."
> 
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> -----------------------------------------
> Disclaimer:
> 
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
> 
"Warning:
The information contained in this email and any attached files is
confidential to BAE Systems Australia. If you are not the intended
recipient, any use, disclosure or copying of this email or any
attachments is expressly prohibited.  If you have received this email
in error, please notify us immediately. VIRUS: Every care has been
taken to ensure this email and its attachments are virus free,
however, any loss or damage incurred in using this email is not the
sender's responsibility.  It is your responsibility to ensure virus
checks are completed before installing any data sent in this email to
your computer."


From SCASPER at mtb.com  Fri Oct 12 08:36:39 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Fri, 12 Oct 2007 08:36:39 -0400
Subject: [cisco-voip] Software MTP will not register
In-Reply-To: <fb5f789e0710110221m3aa3e45dh5dbd9ccc93feccfa@mail.gmail.com>
References: <001601c80b58$91589b70$0b000a0a@ano>
	<C1FE15183DA37645BC0633BC604E44F0070BC2D6@USNAEXCH.na.didata.local>
	<470D1C38.9874.0039.0@mtb.com>
	<fb5f789e0710110221m3aa3e45dh5dbd9ccc93feccfa@mail.gmail.com>
Message-ID: <470F3216.9874.0039.0@mtb.com>

Hello Patrick,
 
That was it! Thank you for your help.
 
Steve

>>> "Patrick Fischer" <myciscovoip at gmail.com> 10/11/2007 5:21 AM >>>
Hi Steven,

try to use "sccp ccm  ( http://10.95.3.18/ )x.x.x.x identifier x version 4.1" instead. Otherwise the gateway will use the default which is AFAIK CCM 3.1 which did not now anything about software MTP.

Regards,
Patrick

Change your 

2007/10/11, STEVEN CASPER <SCASPER at mtb.com>: 

Can not get an MTP to register with CCM  4.1.3sr5d.  Router IOS is 12.4.11T.  Need the MTP to provide Hold for phones using SIP trunking thru a IPIPGW. I have made sure the name in the router and CCM match. What am I missing? 
 
Getting these errors in debug:
 
*Oct 10 16:26:37.395: sym_xapp_xlate_rpm_device_type: Unsupported appl type 1for ccm verion 2
*Oct 10 16:26:37.395: sccp_send_register_msg: SCCP appl service 1 is xlated to an invalid device type 2
 
Show sccp 
 
Software MTP Oper State: ACTIVE_IN_PROGRESS - Cause Code: NOT_REGIS_WITH_CCM
Active Call Manager: 10.95.3.18, Port Number: 2000 
TCP Link Status: CONNECTED, Profile Identifier: 2
Reported Max Streams: 48, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period:
 
Here is a config snip
 
sccp local Loopback0
sccp ccm 10.95.3.18 identifier 1 
sccp ccm 10.91.4.10 identifier 2 
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 2 register IPTRUNK_MTP
 
!
dspfarm profile 2 mtp
 codec g711ulaw
 maximum sessions software 24
 associate application SCCP
 
Thanks for any advice!
 
 
 
 
 
 


Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026 
************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information. 
There are risks associated with the use of electronic transmission. The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission. 
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This email may contain privileged and/or confidential information that is intended solely for the use of the addressee.  If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission.  If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy.  This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
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From wsisk at cisco.com  Fri Oct 12 09:41:55 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 12 Oct 2007 09:41:55 -0400
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
References: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
Message-ID: <470F79A3.6010900@cisco.com>

CSCsk86705 is now open for this problem.

The resolution for CSCsi65663 was supposed to add another value for 
'Show Call Forwarding' Enterprise Parameter called 'Show Only Forward 
All'.  The value was originally added but then inadvertently removed.  
It should be present and we are working on it.

/Wes

Ted Nugent wrote:
> I upgraded a customer to 5.13 for various bug fixes and now the 
> CFwdALL no longer shows in the CCMUser page. I opened a TAC case and 
> I'm being told it's a WAD! CFwdAll has been in the CCMUser page for as 
> long as I can remember and now it's gone and I'm expected to tell my 
> customers that Cisco decided you no longer need that? I've got many 
> customers that use this religiously and if this is actually a WAD then 
> they're stuck on 5.12 and lower? That just doesn't add up?
>
>  
>
> Wes have you heard anything about this and/or can you shed some light 
> on it? I understand that having all the CFwd options in CCMUser is not 
> the best solution but CFwdAll is really the only reason any of my 
> customers use it?
>  
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From svoll.voip at gmail.com  Fri Oct 12 10:56:24 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 12 Oct 2007 07:56:24 -0700
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <470F79A3.6010900@cisco.com>
References: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
	<470F79A3.6010900@cisco.com>
Message-ID: <f84a38d30710120756w5ec51d97t7d28c7c01aee644e@mail.gmail.com>

What would we do without Wes?

You da Man!

Scott


On 10/12/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> CSCsk86705 is now open for this problem.
>
> The resolution for CSCsi65663 was supposed to add another value for 'Show
> Call Forwarding' Enterprise Parameter called 'Show Only Forward All'.  The
> value was originally added but then inadvertently removed.  It should be
> present and we are working on it.
>
> /Wes
>
> Ted Nugent wrote:
>
> I upgraded a customer to 5.13 for various bug fixes and now the CFwdALL no
> longer shows in the CCMUser page. I opened a TAC case and I'm being told
> it's a WAD! CFwdAll has been in the CCMUser page for as long as I can
> remember and now it's gone and I'm expected to tell my customers that Cisco
> decided you no longer need that? I've got many customers that use this
> religiously and if this is actually a WAD then they're stuck on 5.12 and
> lower? That just doesn't add up?
>
>
> Wes have you heard anything about this and/or can you shed some light on
> it? I understand that having all the CFwd options in CCMUser is not the best
> solution but CFwdAll is really the only reason any of my customers use it?
>
>
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From d4rkf1ber at gmail.com  Fri Oct 12 10:57:10 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Fri, 12 Oct 2007 09:57:10 -0500
Subject: [cisco-voip] Emergency Responder Issue with Inbound Calls?
Message-ID: <a1b07ea60710120757m1c8a2f2cpf544375ddd544f88@mail.gmail.com>

I have my ER up and working partially it would seem.  I have ERL's
defined, and phones placed in them.  I have my ELINS and such working
so that when a call goes out the CPN is changed to one of the DID's
and I do reach the correct PSAP.

So far so good.

However, I tested a return call and I have an issue.

Can't seem to reach the user who called 911.  I can see the call hit
the gateway so I know its not a telco issue regarding my DIDs.

Kinda thinking it might be a issue with significant digits?  Sig
digits on these PRI's are all 4 since they handle our normal DID's as
well for office extensions.

Any thoughts on how to best confirm that sig digits are the issue?
Any insight as to some of the ramifications of adjusting the sig
digits on the gateway and what else would have to be adjusted?

From svoll.voip at gmail.com  Fri Oct 12 11:02:56 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 12 Oct 2007 08:02:56 -0700
Subject: [cisco-voip] Emergency Responder Issue with Inbound Calls?
In-Reply-To: <a1b07ea60710120757m1c8a2f2cpf544375ddd544f88@mail.gmail.com>
References: <a1b07ea60710120757m1c8a2f2cpf544375ddd544f88@mail.gmail.com>
Message-ID: <f84a38d30710120802i4e87467cm1ecb8c66bf215a84@mail.gmail.com>

I'm about a month ahead of you. and had the same issue

you want your translation pattern to grab the 4 digits coming in and
translate to the 913XXXXXXXXXX.  not 913XXXX.  That was my fist issue.  then
make sure in CER that you still have it 913XXXXXXXXX and prefix remove 913.
I got all confused with this also as I only had 4 digits coming from the
Telco but CER is still expecting 10 digits.

Hope that helps.

Scott


On 10/12/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
>
> I have my ER up and working partially it would seem.  I have ERL's
> defined, and phones placed in them.  I have my ELINS and such working
> so that when a call goes out the CPN is changed to one of the DID's
> and I do reach the correct PSAP.
>
> So far so good.
>
> However, I tested a return call and I have an issue.
>
> Can't seem to reach the user who called 911.  I can see the call hit
> the gateway so I know its not a telco issue regarding my DIDs.
>
> Kinda thinking it might be a issue with significant digits?  Sig
> digits on these PRI's are all 4 since they handle our normal DID's as
> well for office extensions.
>
> Any thoughts on how to best confirm that sig digits are the issue?
> Any insight as to some of the ramifications of adjusting the sig
> digits on the gateway and what else would have to be adjusted?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From tednugent73 at gmail.com  Fri Oct 12 11:39:04 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Fri, 12 Oct 2007 11:39:04 -0400
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <470F79A3.6010900@cisco.com>
References: <22e6380b0710100905i79d83d7eo8d3b69df7a968bae@mail.gmail.com>
	<470F79A3.6010900@cisco.com>
Message-ID: <22e6380b0710120839y6b661dccna09281579ea7e1fb@mail.gmail.com>

Thanks Wes!

On 10/12/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> CSCsk86705 is now open for this problem.
>
> The resolution for CSCsi65663 was supposed to add another value for 'Show
> Call Forwarding' Enterprise Parameter called 'Show Only Forward All'.  The
> value was originally added but then inadvertently removed.  It should be
> present and we are working on it.
>
> /Wes
>
> Ted Nugent wrote:
>
> I upgraded a customer to 5.13 for various bug fixes and now the CFwdALL no
> longer shows in the CCMUser page. I opened a TAC case and I'm being told
> it's a WAD! CFwdAll has been in the CCMUser page for as long as I can
> remember and now it's gone and I'm expected to tell my customers that Cisco
> decided you no longer need that? I've got many customers that use this
> religiously and if this is actually a WAD then they're stuck on 5.12 and
> lower? That just doesn't add up?
>
>
> Wes have you heard anything about this and/or can you shed some light on
> it? I understand that having all the CFwd options in CCMUser is not the best
> solution but CFwdAll is really the only reason any of my customers use it?
>
>
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From asobihoudai at yahoo.com  Fri Oct 12 13:19:34 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 12 Oct 2007 10:19:34 -0700 (PDT)
Subject: [cisco-voip] CM5.13 No More CFwdAll IN CCMUser?? WAD?
In-Reply-To: <f84a38d30710120756w5ec51d97t7d28c7c01aee644e@mail.gmail.com>
Message-ID: <462588.49596.qm@web57209.mail.re3.yahoo.com>

Wes,

Thanks very much for your assistance!

Paul

--- Scott Voll <svoll.voip at gmail.com> wrote:

> What would we do without Wes?
> 
> You da Man!
> 
> Scott
> 
> 
> On 10/12/07, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > CSCsk86705 is now open for this problem.
> >
> > The resolution for CSCsi65663 was supposed to add
> another value for 'Show
> > Call Forwarding' Enterprise Parameter called 'Show
> Only Forward All'.  The
> > value was originally added but then inadvertently
> removed.  It should be
> > present and we are working on it.
> >
> > /Wes
> >
> > Ted Nugent wrote:
> >
> > I upgraded a customer to 5.13 for various bug
> fixes and now the CFwdALL no
> > longer shows in the CCMUser page. I opened a TAC
> case and I'm being told
> > it's a WAD! CFwdAll has been in the CCMUser page
> for as long as I can
> > remember and now it's gone and I'm expected to
> tell my customers that Cisco
> > decided you no longer need that
 I've got many
> customers that use this
> > religiously and if this is actually a WAD then
> they're stuck on 5.12 and
> > lower? That just doesn't add up?
> >
> >
> > Wes have you heard anything about this and/or can
> you shed some light on
> > it? I understand that having all the CFwd options
> in CCMUser is not the best
> > solution but CFwdAll is really the only reason any
> of my customers use it

> >
> >
> >
> > ------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing
>
listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
____________________________________________________________________________________
Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. 
http://answers.yahoo.com/dir/?link=list&sid=396545433

From mark.ridgway at qatar.tamu.edu  Fri Oct 12 14:30:45 2007
From: mark.ridgway at qatar.tamu.edu (Ridgway, Mark)
Date: Fri, 12 Oct 2007 21:30:45 +0300
Subject: [cisco-voip] 7931/ Download User Guide from CCMUser page
Message-ID: <0A2EE4E514A44D4B9C8FE1A12837201E0378D762@mail.qatar.tamu.edu>

I have a CallManager 6.0  running the 7906, 7961, and 7931 phones. I
have the CCMUser page configured. I am unable to download the user guide
for the 7931 phone. It works for the 7906 and 7961 phones. The ccmuser
webpage gives the following error when I try to open the Phone Users
Guide for the 7931G. 

HTTP Status 404 -/PhoneGuides/locales/en_us is not available

How do I change the link for the User Guide?

Thanks in advance
Mark
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From wsisk at cisco.com  Fri Oct 12 15:57:52 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 12 Oct 2007 15:57:52 -0400
Subject: [cisco-voip] 7931/ Download User Guide from CCMUser page
In-Reply-To: <0A2EE4E514A44D4B9C8FE1A12837201E0378D762@mail.qatar.tamu.edu>
References: <0A2EE4E514A44D4B9C8FE1A12837201E0378D762@mail.qatar.tamu.edu>
Message-ID: <470FD1C0.9070908@cisco.com>

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From benjamin.kappauf at googlemail.com  Fri Oct 12 16:41:20 2007
From: benjamin.kappauf at googlemail.com (Benjamin Kappauf)
Date: Fri, 12 Oct 2007 22:41:20 +0200
Subject: [cisco-voip] call-manager-fallback wrong phone dn
In-Reply-To: <feidjf$kkr$1@sea.gmane.org>
References: <feidjf$kkr$1@sea.gmane.org>
Message-ID: <feom6s$6lq$1@sea.gmane.org>

Benjamin Kappauf schrieb:
> What is the problem if in case of a fallback all my phones are 
> registered fine to the voice router. But all phones don't get the 
> correct DN. Most of them get a 65xx DN. A few are with ext. mobility 
> activ. And they all have got DNs out of 4440 - 4449
> 
> Btw. no 65xx number exist on the callmanager.
> 
> Thanks alot for any help!
> 
> 
> ----------------------------------
> call-manager-fallback
>   secondary-dialtone 9
>   max-conferences 8 gain -6
>   transfer-system full-consult
>   ip source-address x.x.x.x port 2000
>   max-ephones 40
>   max-dn 80 dual-line
>   system message primary SRST Fallback
>   system message secondary SRST
>   time-format 24
>   date-format dd-mm-yy

Today I downgraded the voice gateway to 
c2800nm-spservicesk9-mz.124-9.T5.bin and I did a reset on all phones.

Afterwards I initialised SRST. But I still get the wrong DNs.

What cause a phone on fallback to get a wrong dn?

Thank you


From lelio at uoguelph.ca  Fri Oct 12 18:47:58 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 12 Oct 2007 18:47:58 -0400
Subject: [cisco-voip] Cisco Unified Workspace Licensing
Message-ID: <BAYC1-PASMTP13227CEEA1C4BBF5D81A90B9A00@CEZ.ICE>

anyone else see this? i think this is pretty new...
  Cisco Unified Workspace Licensing

  http://www.cisco.com/en/US/customer/products/ps9156/index.html
apparently you can get licence packs that give you a whole bunch of stuff like callmanager, voicemail, mobility, etc at a cheaper cost than buying them individually.

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From VLoschiavo at data-corporation.com  Fri Oct 12 18:54:40 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Fri, 12 Oct 2007 18:54:40 -0400
Subject: [cisco-voip] Cisco Unified Workspace Licensing
In-Reply-To: <BAYC1-PASMTP13227CEEA1C4BBF5D81A90B9A00@CEZ.ICE>
References: <BAYC1-PASMTP13227CEEA1C4BBF5D81A90B9A00@CEZ.ICE>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF57136B9CCB@MAIL02.hq.data-corporation.com>

Yeah...it's CUWL  (pronounced "cool").

 

But seriously.....

 

1)       250 user minimum.

2)       Standard (no meetingplace express, others?)

3)       Pro (meetingplace, others?) 

 

 

____________________________________________________________
Vincent Loschiavo
Senior Consultant
Datacorp
18503 Pines Blvd, Suite 213, Pembroke Pines, FL, 33029
"Cisco Advanced Unified Communications Partner"
Cell Phone:        786-282-1164

Office Phone:     954-671-5669
Main Office:      305-594-6933
____________________________________________________________ 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, October 12, 2007 6:48 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco Unified Workspace Licensing

 

anyone else see this? i think this is pretty new...

	Cisco Unified Workspace Licensing

	 

	http://www.cisco.com/en/US/customer/products/ps9156/index.html

apparently you can get licence packs that give you a whole bunch of
stuff like callmanager, voicemail, mobility, etc at a cheaper cost than
buying them individually.

 

 

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From svoll.voip at gmail.com  Fri Oct 12 19:02:13 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 12 Oct 2007 16:02:13 -0700
Subject: [cisco-voip] Cisco Unified Workspace Licensing
In-Reply-To: <BAYC1-PASMTP13227CEEA1C4BBF5D81A90B9A00@CEZ.ICE>
References: <BAYC1-PASMTP13227CEEA1C4BBF5D81A90B9A00@CEZ.ICE>
Message-ID: <f84a38d30710121602n1024b068g56f567e2c9230edf@mail.gmail.com>

ya.... My SE was here a week or so telling me about it.  problem is I
already have my licensing.  Doesn't help me much.

but basically if you buy a phone DLU and Unity VM box you get everything
else for $50.  of course IPCCx not included.

Scott


On 10/12/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  anyone else see this? i think this is pretty new...
>
> Cisco Unified Workspace Licensing
>
> http://www.cisco.com/en/US/customer/products/ps9156/index.html
>
> apparently you can get licence packs that give you a whole bunch of stuff
> like callmanager, voicemail, mobility, etc at a cheaper cost than buying
> them individually.
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From lelio at uoguelph.ca  Fri Oct 12 20:06:00 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 12 Oct 2007 20:06:00 -0400
Subject: [cisco-voip] Cisco Unified Workspace Licensing
References: <BAYC1-PASMTP13227CEEA1C4BBF5D81A90B9A00@CEZ.ICE>
	<f84a38d30710121602n1024b068g56f567e2c9230edf@mail.gmail.com>
Message-ID: <BAYC1-PASMTP13105AF44EB6A3DB01777CB9A10@CEZ.ICE>

You can apparrently upgrade or switch to CUWL. I really like the idea of everything else for $50 bucks. And there was a pretty significantly discounted GOV/EDU line item as well. That might make it even more appealing.

  ----- Original Message ----- 
  From: Scott Voll 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, October 12, 2007 7:02 PM
  Subject: Re: [cisco-voip] Cisco Unified Workspace Licensing


  ya.... My SE was here a week or so telling me about it.  problem is I already have my licensing.  Doesn't help me much.

  but basically if you buy a phone DLU and Unity VM box you get everything else for $50.  of course IPCCx not included.

  Scott

   
  On 10/12/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote: 
    anyone else see this? i think this is pretty new...
      Cisco Unified Workspace Licensing

      http://www.cisco.com/en/US/customer/products/ps9156/index.html 
    apparently you can get licence packs that give you a whole bunch of stuff like callmanager, voicemail, mobility, etc at a cheaper cost than buying them individually.



    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip 


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From lelio at uoguelph.ca  Fri Oct 12 20:07:02 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 12 Oct 2007 20:07:02 -0400
Subject: [cisco-voip] Cisco Unified Workspace Licensing
Message-ID: <BAYC1-PASMTP03D3F71D0787F4F0AE2DC6B9A10@CEZ.ICE>

...and from the sounds of the little video introduction, it comes with upgrades included at no extra charge.

  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: Scott Voll 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, October 12, 2007 8:06 PM
  Subject: Re: [cisco-voip] Cisco Unified Workspace Licensing


  You can apparrently upgrade or switch to CUWL. I really like the idea of everything else for $50 bucks. And there was a pretty significantly discounted GOV/EDU line item as well. That might make it even more appealing.

    ----- Original Message ----- 
    From: Scott Voll 
    To: Lelio Fulgenzi 
    Cc: cisco-voip at puck.nether.net 
    Sent: Friday, October 12, 2007 7:02 PM
    Subject: Re: [cisco-voip] Cisco Unified Workspace Licensing


    ya.... My SE was here a week or so telling me about it.  problem is I already have my licensing.  Doesn't help me much.

    but basically if you buy a phone DLU and Unity VM box you get everything else for $50.  of course IPCCx not included.

    Scott

     
    On 10/12/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote: 
      anyone else see this? i think this is pretty new...
        Cisco Unified Workspace Licensing

        http://www.cisco.com/en/US/customer/products/ps9156/index.html 
      apparently you can get licence packs that give you a whole bunch of stuff like callmanager, voicemail, mobility, etc at a cheaper cost than buying them individually.



      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip 


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From lelio at uoguelph.ca  Fri Oct 12 20:07:48 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 12 Oct 2007 20:07:48 -0400
Subject: [cisco-voip] Cisco Unified Workspace Licensing
References: <BAYC1-PASMTP13227CEEA1C4BBF5D81A90B9A00@CEZ.ICE>
	<CB4BC8C24EF6E149AED9FDB9B9AF57136B9CCB@MAIL02.hq.data-corporation.com>
Message-ID: <BAYC1-PASMTP079D3CDBF522D7694534BBB9A10@CEZ.ICE>

i still like 'CUPID' ;)
  ----- Original Message ----- 
  From: Vince Loschiavo 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Friday, October 12, 2007 6:54 PM
  Subject: RE: [cisco-voip] Cisco Unified Workspace Licensing


  Yeah.it's CUWL  (pronounced "cool").

   

  But seriously...

   

  1)       250 user minimum.

  2)       Standard (no meetingplace express, others?)

  3)       Pro (meetingplace, others?) 

   

   

  ____________________________________________________________
  Vincent Loschiavo
  Senior Consultant
  Datacorp
  18503 Pines Blvd, Suite 213, Pembroke Pines, FL, 33029
  "Cisco Advanced Unified Communications Partner"
  Cell Phone:        786-282-1164

  Office Phone:     954-671-5669
  Main Office:      305-594-6933
  ____________________________________________________________ 


------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Friday, October 12, 2007 6:48 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Cisco Unified Workspace Licensing

   

  anyone else see this? i think this is pretty new...

    Cisco Unified Workspace Licensing

     

    http://www.cisco.com/en/US/customer/products/ps9156/index.html

  apparently you can get licence packs that give you a whole bunch of stuff like callmanager, voicemail, mobility, etc at a cheaper cost than buying them individually.

   

   
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From asobihoudai at yahoo.com  Fri Oct 12 20:21:20 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 12 Oct 2007 17:21:20 -0700 (PDT)
Subject: [cisco-voip] CME as SRST questions
In-Reply-To: <BAYC1-PASMTP079D3CDBF522D7694534BBB9A10@CEZ.ICE>
Message-ID: <682417.38484.qm@web57215.mail.re3.yahoo.com>

Since CCME as SRST has been available since 4.0, I
have read through the documentation and I have several
questions.

If CME is being used as SRST, is it a required
double-whammy where a phone license must be purchased
for use under CCM *AND* CCME? [my guess would be yes]

How in the heck do you configure the DHCP pool when
using this configuration? Under CCME, you point the
option 150 line to be the router itself, under CCM, we
point it to the external TFTP server which may or may
not be one of the CCM boxes.

Lastly, if anyone has tried this out...how well does
it work?

Cheers,
Paul


       
____________________________________________________________________________________
Need a vacation? Get great deals
to amazing places on Yahoo! Travel.
http://travel.yahoo.com/

From cisco1.charles at gmail.com  Fri Oct 12 23:31:07 2007
From: cisco1.charles at gmail.com (charles george)
Date: Fri, 12 Oct 2007 23:31:07 -0400
Subject: [cisco-voip] test
Message-ID: <9552cdf0710122031j3422b249we5c48e09cb87d043@mail.gmail.com>

-- 
Regards,
Charles
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From nalakak at sg.ibm.com  Sat Oct 13 16:05:11 2007
From: nalakak at sg.ibm.com (Nalaka Kariyawasam)
Date: Sun, 14 Oct 2007 04:05:11 +0800
Subject: [cisco-voip] Nalaka Kariyawasam is out of the office.
Message-ID: <OFCEFE31C4.C87DF252-ON48257373.006E5690-48257373.006E5690@sg.ibm.com>


I will be out of the office starting  10/13/2007 and will not return until
10/26/2007.

I will respond to your message when I return.
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From Brandon.Ogden at amersports.com  Sat Oct 13 18:52:15 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Sat, 13 Oct 2007 15:52:15 -0700
Subject: [cisco-voip] IPCC Extension
In-Reply-To: <OFCEFE31C4.C87DF252-ON48257373.006E5690-48257373.006E5690@sg.ibm.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>

So I just had to rebuild my publisher (5.1) and I am trying to setup my
IPCC agents again.. but I am not showing an option in the End User,
Extension associations that will allow me to specify the extension for
the IPCC.  I seem to remember reading that there is a sequel statement
you can run on the Call manager that enables you to setup extensions to
IPCC.  Anyone help?  Thanks.

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From wsisk at cisco.com  Sat Oct 13 23:58:27 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Sat, 13 Oct 2007 23:58:27 -0400
Subject: [cisco-voip] IPCC Extension
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>
References: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>
Message-ID: <471193E3.7000806@cisco.com>

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From mmilanovic at ctiusa.com  Sun Oct 14 00:10:12 2007
From: mmilanovic at ctiusa.com (Mladen Milanovic)
Date: Sun, 14 Oct 2007 00:10:12 -0400
Subject: [cisco-voip] Cisco TSP (Dialing from MS Outlook) large deployment
In-Reply-To: <471193E3.7000806@cisco.com>
References: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>
	<471193E3.7000806@cisco.com>
Message-ID: <4761C60AD1E74A4D8606CAF48582265AF5E2B6@EXCHANGE.ctiusa.com>

Guys,
I have to deploy dialing from MS Outlook Contacts feature for more than 500
users. Is there any "smart" way to instal Cisco TSP and configure it without
revisiting every PC. I am running a CCM 5.1(2)
 
 
Thanks
 
Mladen
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From mmilanovic at ctiusa.com  Sat Oct 13 23:44:12 2007
From: mmilanovic at ctiusa.com (Mladen Milanovic)
Date: Sat, 13 Oct 2007 23:44:12 -0400
Subject: [cisco-voip] FW: Cisco TSP (Dialing from MS Outlook) large
	deployment
Message-ID: <4761C60AD1E74A4D8606CAF48582265AF5E2B4@EXCHANGE.ctiusa.com>

Guys,
I have to deploy dialing from MS Outlook Contacts feature for more than 500
users. Is there any "smart" way to instal Cisco TSP and configure it without
revisiting every PC. I am running a CCM 5.1(2)
 
 
Thanks
 
Mladen
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From jonvoip at gmail.com  Sun Oct 14 14:51:37 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 14 Oct 2007 13:51:37 -0500
Subject: [cisco-voip] IPCC Extension
In-Reply-To: <471193E3.7000806@cisco.com>
References: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>
	<471193E3.7000806@cisco.com>
Message-ID: <5d093f9a0710141151y65daaaa1we8097718814688d5@mail.gmail.com>

How is this actually set? Is it during the install of IPCC 4.5 and it
creates the field on the CCM user page?


Jonathan

On 10/13/07, Wes Sisk <wsisk at cisco.com> wrote:
>
>  Brandon,
>
>  Did you perform a DRS restore or reinstall IPCCX after rebuilding the pub?
> Either of those actions should set the values properly.  That is really the
> best way to get all dependencies corrected.  Otherwise, here is the manual
> process:
>
>  1. Under system->enterprise parameters check for:
>  IPCC Express Installed
>  what is value, true or false?
>
>  2. If false you should find it F in processconfig as well. confirm the
> paramname like '%nstalled%' in processconfig table:
>  admin:run sql select paramname,paramvalue from processconfig where
> paramname like '%nstalled%'
>  paramname        paramvalue
>  ================ ==========
>  AAInstalledFlag  F
>  IAQInstalledFlag F
>
>  3. force it:
>  Either re-run the IPCCX install, it should update this value via AXL during
> install.  Otherwise, force it yourself:
>  run sql update processconfig set paramvalue="T" where paramname like
> '%nstalled%'
>
>  /Wes
>
>
>  Ogden, Brandon wrote:
>
>
>
> So I just had to rebuild my publisher (5.1) and I am trying to setup my IPCC
> agents again.. but I am not showing an option in the End User, Extension
> associations that will allow me to specify the extension for the IPCC.  I
> seem to remember reading that there is a sequel statement you can run on the
> Call manager that enables you to setup extensions to IPCC.  Anyone help?
> Thanks. ________________________________
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From wsisk at cisco.com  Sun Oct 14 15:09:29 2007
From: wsisk at cisco.com (Wesley Sisk)
Date: Sun, 14 Oct 2007 15:09:29 -0400 (EDT)
Subject: [cisco-voip] IPCC Extension
In-Reply-To: <5d093f9a0710141151y65daaaa1we8097718814688d5@mail.gmail.com>
References: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>
	<471193E3.7000806@cisco.com>
	<5d093f9a0710141151y65daaaa1we8097718814688d5@mail.gmail.com>
Message-ID: <Pine.GSO.4.64.0710141508110.2241@clubhouse-6.cisco.com>

feature and fields are always there in cm.  it is only exposed/displayed 
when the parameter is set T in processconfig table.

ipccx install should update param to T.

Or, if Brandon were doing a rebuild, then the DRS restore would have 
restored the value from his previous system.

thus my curiosity of how he got into this state.

/wes

On Sun, 14 Oct 2007, Jonathan Charles wrote:

> How is this actually set? Is it during the install of IPCC 4.5 and it
> creates the field on the CCM user page?
>
>
> Jonathan
>
> On 10/13/07, Wes Sisk <wsisk at cisco.com> wrote:
>>
>>  Brandon,
>>
>>  Did you perform a DRS restore or reinstall IPCCX after rebuilding the pub?
>> Either of those actions should set the values properly.  That is really the
>> best way to get all dependencies corrected.  Otherwise, here is the manual
>> process:
>>
>>  1. Under system->enterprise parameters check for:
>>  IPCC Express Installed
>>  what is value, true or false?
>>
>>  2. If false you should find it F in processconfig as well. confirm the
>> paramname like '%nstalled%' in processconfig table:
>>  admin:run sql select paramname,paramvalue from processconfig where
>> paramname like '%nstalled%'
>>  paramname        paramvalue
>>  ================ ==========
>>  AAInstalledFlag  F
>>  IAQInstalledFlag F
>>
>>  3. force it:
>>  Either re-run the IPCCX install, it should update this value via AXL during
>> install.  Otherwise, force it yourself:
>>  run sql update processconfig set paramvalue="T" where paramname like
>> '%nstalled%'
>>
>>  /Wes
>>
>>
>>  Ogden, Brandon wrote:
>>
>>
>>
>> So I just had to rebuild my publisher (5.1) and I am trying to setup my IPCC
>> agents again.. but I am not showing an option in the End User, Extension
>> associations that will allow me to specify the extension for the IPCC.  I
>> seem to remember reading that there is a sequel statement you can run on the
>> Call manager that enables you to setup extensions to IPCC.  Anyone help?
>> Thanks. ________________________________
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>

From jonvoip at gmail.com  Sun Oct 14 16:05:42 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 14 Oct 2007 15:05:42 -0500
Subject: [cisco-voip] IPCC Extension
In-Reply-To: <Pine.GSO.4.64.0710141508110.2241@clubhouse-6.cisco.com>
References: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>
	<471193E3.7000806@cisco.com>
	<5d093f9a0710141151y65daaaa1we8097718814688d5@mail.gmail.com>
	<Pine.GSO.4.64.0710141508110.2241@clubhouse-6.cisco.com>
Message-ID: <5d093f9a0710141305r5c875179o13e3b77378615a14@mail.gmail.com>

Well, that is the question.

It seems like it should be an exposed Enterprise Parameter (CRS
Installed or something...)... for some reason that sounds familiar...
isn't there such a field?



Jonathan

On 10/14/07, Wesley Sisk <wsisk at cisco.com> wrote:
> feature and fields are always there in cm.  it is only exposed/displayed
> when the parameter is set T in processconfig table.
>
> ipccx install should update param to T.
>
> Or, if Brandon were doing a rebuild, then the DRS restore would have
> restored the value from his previous system.
>
> thus my curiosity of how he got into this state.
>
> /wes
>
> On Sun, 14 Oct 2007, Jonathan Charles wrote:
>
> > How is this actually set? Is it during the install of IPCC 4.5 and it
> > creates the field on the CCM user page?
> >
> >
> > Jonathan
> >
> > On 10/13/07, Wes Sisk <wsisk at cisco.com> wrote:
> >>
> >>  Brandon,
> >>
> >>  Did you perform a DRS restore or reinstall IPCCX after rebuilding the pub?
> >> Either of those actions should set the values properly.  That is really the
> >> best way to get all dependencies corrected.  Otherwise, here is the manual
> >> process:
> >>
> >>  1. Under system->enterprise parameters check for:
> >>  IPCC Express Installed
> >>  what is value, true or false?
> >>
> >>  2. If false you should find it F in processconfig as well. confirm the
> >> paramname like '%nstalled%' in processconfig table:
> >>  admin:run sql select paramname,paramvalue from processconfig where
> >> paramname like '%nstalled%'
> >>  paramname        paramvalue
> >>  ================ ==========
> >>  AAInstalledFlag  F
> >>  IAQInstalledFlag F
> >>
> >>  3. force it:
> >>  Either re-run the IPCCX install, it should update this value via AXL during
> >> install.  Otherwise, force it yourself:
> >>  run sql update processconfig set paramvalue="T" where paramname like
> >> '%nstalled%'
> >>
> >>  /Wes
> >>
> >>
> >>  Ogden, Brandon wrote:
> >>
> >>
> >>
> >> So I just had to rebuild my publisher (5.1) and I am trying to setup my IPCC
> >> agents again.. but I am not showing an option in the End User, Extension
> >> associations that will allow me to specify the extension for the IPCC.  I
> >> seem to remember reading that there is a sequel statement you can run on the
> >> Call manager that enables you to setup extensions to IPCC.  Anyone help?
> >> Thanks. ________________________________
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
>

From jonvoip at gmail.com  Sun Oct 14 18:19:42 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 14 Oct 2007 17:19:42 -0500
Subject: [cisco-voip] IPCC Extension
In-Reply-To: <5d093f9a0710141305r5c875179o13e3b77378615a14@mail.gmail.com>
References: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>
	<471193E3.7000806@cisco.com>
	<5d093f9a0710141151y65daaaa1we8097718814688d5@mail.gmail.com>
	<Pine.GSO.4.64.0710141508110.2241@clubhouse-6.cisco.com>
	<5d093f9a0710141305r5c875179o13e3b77378615a14@mail.gmail.com>
Message-ID: <5d093f9a0710141519i66814925g91aca95e41d2d7e0@mail.gmail.com>

Yeah, as I thought...

IPCC Express Installed:
Required Field    	
This parameter is true when a Cisco IPCC Express system is integrated
with this Cisco CallManager cluster. This is a read only flag. It can
be modified only by CRS when it has been installed. If this flag is
true, then the IPCC Extension can be specified on the User
Configuration page

  	This is a required field.
  	Default:  False

However, it is not modifiable... at least from Ent Params... Though I
understand why it can't be changed (it would nuke out the CRS agent
settings if someone accidentally changed it from true to false).


Jonathan

On 10/14/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> Well, that is the question.
>
> It seems like it should be an exposed Enterprise Parameter (CRS
> Installed or something...)... for some reason that sounds familiar...
> isn't there such a field?
>
>
>
> Jonathan
>
> On 10/14/07, Wesley Sisk <wsisk at cisco.com> wrote:
> > feature and fields are always there in cm.  it is only exposed/displayed
> > when the parameter is set T in processconfig table.
> >
> > ipccx install should update param to T.
> >
> > Or, if Brandon were doing a rebuild, then the DRS restore would have
> > restored the value from his previous system.
> >
> > thus my curiosity of how he got into this state.
> >
> > /wes
> >
> > On Sun, 14 Oct 2007, Jonathan Charles wrote:
> >
> > > How is this actually set? Is it during the install of IPCC 4.5 and it
> > > creates the field on the CCM user page?
> > >
> > >
> > > Jonathan
> > >
> > > On 10/13/07, Wes Sisk <wsisk at cisco.com> wrote:
> > >>
> > >>  Brandon,
> > >>
> > >>  Did you perform a DRS restore or reinstall IPCCX after rebuilding the pub?
> > >> Either of those actions should set the values properly.  That is really the
> > >> best way to get all dependencies corrected.  Otherwise, here is the manual
> > >> process:
> > >>
> > >>  1. Under system->enterprise parameters check for:
> > >>  IPCC Express Installed
> > >>  what is value, true or false?
> > >>
> > >>  2. If false you should find it F in processconfig as well. confirm the
> > >> paramname like '%nstalled%' in processconfig table:
> > >>  admin:run sql select paramname,paramvalue from processconfig where
> > >> paramname like '%nstalled%'
> > >>  paramname        paramvalue
> > >>  ================ ==========
> > >>  AAInstalledFlag  F
> > >>  IAQInstalledFlag F
> > >>
> > >>  3. force it:
> > >>  Either re-run the IPCCX install, it should update this value via AXL during
> > >> install.  Otherwise, force it yourself:
> > >>  run sql update processconfig set paramvalue="T" where paramname like
> > >> '%nstalled%'
> > >>
> > >>  /Wes
> > >>
> > >>
> > >>  Ogden, Brandon wrote:
> > >>
> > >>
> > >>
> > >> So I just had to rebuild my publisher (5.1) and I am trying to setup my IPCC
> > >> agents again.. but I am not showing an option in the End User, Extension
> > >> associations that will allow me to specify the extension for the IPCC.  I
> > >> seem to remember reading that there is a sequel statement you can run on the
> > >> Call manager that enables you to setup extensions to IPCC.  Anyone help?
> > >> Thanks. ________________________________
> > >>
> > >> _______________________________________________
> > >> cisco-voip mailing list
> > >> cisco-voip at puck.nether.net
> > >> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>
> > >> _______________________________________________
> > >> cisco-voip mailing list
> > >> cisco-voip at puck.nether.net
> > >> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>
> > >
> >
>

From jonvoip at gmail.com  Sun Oct 14 18:25:00 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 14 Oct 2007 17:25:00 -0500
Subject: [cisco-voip] call-manager-fallback wrong phone dn
In-Reply-To: <feom6s$6lq$1@sea.gmane.org>
References: <feidjf$kkr$1@sea.gmane.org> <feom6s$6lq$1@sea.gmane.org>
Message-ID: <5d093f9a0710141525l5ac25ba4y3c7fc876a9ad76c4@mail.gmail.com>

Post the entire config... It almost seems like you have a bunch of
ephone-dn's on the router and it is auto-assigning them...



Jonathan

On 10/12/07, Benjamin Kappauf <benjamin.kappauf at googlemail.com> wrote:
> Benjamin Kappauf schrieb:
> > What is the problem if in case of a fallback all my phones are
> > registered fine to the voice router. But all phones don't get the
> > correct DN. Most of them get a 65xx DN. A few are with ext. mobility
> > activ. And they all have got DNs out of 4440 - 4449
> >
> > Btw. no 65xx number exist on the callmanager.
> >
> > Thanks alot for any help!
> >
> >
> > ----------------------------------
> > call-manager-fallback
> >   secondary-dialtone 9
> >   max-conferences 8 gain -6
> >   transfer-system full-consult
> >   ip source-address x.x.x.x port 2000
> >   max-ephones 40
> >   max-dn 80 dual-line
> >   system message primary SRST Fallback
> >   system message secondary SRST
> >   time-format 24
> >   date-format dd-mm-yy
>
> Today I downgraded the voice gateway to
> c2800nm-spservicesk9-mz.124-9.T5.bin and I did a reset on all phones.
>
> Afterwards I initialised SRST. But I still get the wrong DNs.
>
> What cause a phone on fallback to get a wrong dn?
>
> Thank you
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From mark.mills at baesystems.com  Sun Oct 14 19:40:16 2007
From: mark.mills at baesystems.com (MILLS, Mark)
Date: Mon, 15 Oct 2007 09:10:16 +0930
Subject: [cisco-voip] Hookflash issue on VG224
In-Reply-To: <CB4BC8C24EF6E149AED9FDB9B9AF57136B9CCB@MAIL02.hq.data-corporation.com>
References: <BAYC1-PASMTP13227CEEA1C4BBF5D81A90B9A00@CEZ.ICE>
	<CB4BC8C24EF6E149AED9FDB9B9AF57136B9CCB@MAIL02.hq.data-corporation.com>
Message-ID: <54DD8B39132EFA4293A5C0565004E5460A118D47@SBEMAIL1.au.baesystems.com>


Hi All,
 
I seem to have an issue with hookflash not working on an analog phone
connected to a VG224.

I have tried changing the timing hookflash-in settings on the voice
port, but it does not seem to help too much.   I even extended them out
to the far ends of 1550 and 1 to see if it helps, and no go.

I am not exactly sure what I am doing wrong?

The config on the voice-port is below:

voice-port 2/22
 ren 3
 idle-voltage low
 disc_pi_off
 input gain 10
 output attenuation 10
 playout-delay minimum low
 cptone AU
 timing hookflash-in 1550 0
 timing digit 53
 bearer-cap Speech
 station-id number xxxxxx
 caller-id enable
!


Below is the output of debug vpm signal if its any help?

Oct 15 08:21:28.381 CST: [2/22] c2400_report_sig_state: ABCD = 0x4
Oct 15 08:21:28.381 CST: port 22 changed state new state = 4
Oct 15 08:21:28.385 CST: htsp_process_event: [2/22, FXSLS_CONNECT,
E_DSP_SIG_010
0]fxsls_offhook_onhook, HF duration Max=1550, HF duration Min=1
Oct 15 08:21:28.385 CST: htsp_timer - 1550 msec
Oct 15 08:21:28.485 CST: [2/22] c2400_report_sig_state: ABCD = 0xC
Oct 15 08:21:28.485 CST: port 22 changed state new state = 12
Oct 15 08:21:28.489 CST: htsp_process_event: [2/22, FXSLS_CONNECT,
E_DSP_SIG_110
0]fxsls_offhook_offhookhtsp_hook_flash_ind
Oct 15 08:21:28.489 CST: htsp_timer_stop
Oct 15 08:21:29.573 CST: htsp_digit_ready(2/22): digit = 6
Oct 15 08:21:29.773 CST: htsp_digit_ready(2/22): digit = 7
Oct 15 08:21:30.193 CST: htsp_digit_ready(2/22): digit = 6

Any pointers in where to look is appreciated, you guys are very helpful
:)


Thanks,
   Mark
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From ml at iridescent.org  Sun Oct 14 21:07:41 2007
From: ml at iridescent.org (Lawrence E. Bakst)
Date: Sun, 14 Oct 2007 18:07:41 -0700
Subject: [cisco-voip] Unity Express: NM-CUE vs AIM-CUE
Message-ID: <p0624080fc338644c27ff@[192.168.0.7]>

1. For a very small installation, 5 phones in a home, is there any reason not to prefer the AIM-CUE over the NM-CUE for running Unity Express?

I would like to do this on a 2821 and would prefer to use the single NM slot for a NM-HDA I have on hand and use an AIM-CUE For Unity Express. However if the AIM-CUE is going to be headache, I'll flush the NM-HDA and get some VIC2 cards or the EVM stuff and run a NM-CUE in the NM slot.

Jim's post kind of scared me a bit.

At 8:17 AM -0400 7/30/05, Jim McBurnett wrote:
>If there are expectations that there might be more than a dozen messages a day, consider the NM-CUE.
>The AIM-CUE has a flash drive, and is subject to flash wear....
>Also, look into a good Battery backup. I would reccomend a Liebert, with the cable that plugs in the AUX port to shutdown CUE..
>If you don't then CUE can crash on you if the system is not shutdown properly.....
>
>jim

2. With the NM-CUE do you really have to shutdown via aux port (or console?) on the NM-CUE before you turn power off on the router? That is kind of a drag. Is the also true for the AIM-CUE? Does the AIM-CUE also have console ports?

3. How prone to corruption are these cards? I have a UPS but not one that knows how to shutdown the NM-CUE via AUX-port.

4. I also heard that the AIM-CUE is very slow to boot up after a reboot/crash and that the NM-CUE boot up much faster. Is that true? Anyone comparative boot times for crash and no crash scenarios?

5. If the flash on the AIM-CUE does get corrupted what is the procedure for reinstalling? Anything stopping me from cloning the CF card when I get it and having that on hand if the CF card on the AIM-CUE gets corrupted? That would seem to be the "easy button". If that works, it would seem that AIM-CUE might be better than the NM-CUE, if it gets corrupted.

6. Anyone want to share negative AIM-CUE experiences?

I know CME pretty well but I don't have any Unity Express experience and would prefer not to make a mistake here. Any advice appreciated.

Best,

leb


From asobihoudai at yahoo.com  Sun Oct 14 21:44:09 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Sun, 14 Oct 2007 18:44:09 -0700 (PDT)
Subject: [cisco-voip] Unity Express: NM-CUE vs AIM-CUE
In-Reply-To: <p0624080fc338644c27ff@[192.168.0.7]>
Message-ID: <576772.72632.qm@web57213.mail.re3.yahoo.com>


--- "Lawrence E. Bakst" <ml at iridescent.org> wrote:

> 1. For a very small installation, 5 phones in a
> home, is there any reason not to prefer the AIM-CUE
> over the NM-CUE for running Unity Express?

For an installation for 5 people, the AIM should be
fine for your needs. I just don't like it because it's
slow as molasses in winter if you have to reboot it.

> I would like to do this on a 2821 and would prefer
> to use the single NM slot for a NM-HDA I have on
> hand and use an AIM-CUE For Unity Express. However
> if the AIM-CUE is going to be headache, I'll flush
> the NM-HDA and get some VIC2 cards or the EVM stuff
> and run a NM-CUE in the NM slot.
> 
> Jim's post kind of scared me a bit.
> 
> At 8:17 AM -0400 7/30/05, Jim McBurnett wrote:
> >If there are expectations that there might be more
> than a dozen messages a day, consider the NM-CUE.
> >The AIM-CUE has a flash drive, and is subject to
> flash wear....
> >Also, look into a good Battery backup. I would
> reccomend a Liebert, with the cable that plugs in
> the AUX port to shutdown CUE..
> >If you don't then CUE can crash on you if the
> system is not shutdown properly.....
> >
> >jim
> 
> 2. With the NM-CUE do you really have to shutdown
> via aux port (or console?) on the NM-CUE before you
> turn power off on the router? That is kind of a
> drag. Is the also true for the AIM-CUE? Does the
> AIM-CUE also have console ports?

You don't have to have an AUX port for shutdown. You
can shut the module down via the command line.
e.g.,

service-engine service-module 1/0 shutdown

> 3. How prone to corruption are these cards? I have a
> UPS but not one that knows how to shutdown the
> NM-CUE via AUX-port.

I have never had it become corrupt after a year of
regular usage.

> 4. I also heard that the AIM-CUE is very slow to
> boot up after a reboot/crash and that the NM-CUE
> boot up much faster. Is that true? Anyone
> comparative boot times for crash and no crash
> scenarios?

The NM is faster because it uses a regular harddisk.
The AIM uses a compact flash disk. I don't have any
times per se but I'd guess that the NM takes only
5mins or so to reboot while the AIM can take up to 15
mins.

> 5. If the flash on the AIM-CUE does get corrupted
> what is the procedure for reinstalling? Anything
> stopping me from cloning the CF card when I get it
> and having that on hand if the CF card on the
> AIM-CUE gets corrupted? That would seem to be the
> "easy button". If that works, it would seem that
> AIM-CUE might be better than the NM-CUE, if it gets
> corrupted.

I would guess that the answer is nothing is preventing
you from making a clone of the system but then again
I've never tried it so good luck with that if you go
that route.

Paul
> 
> 6. Anyone want to share negative AIM-CUE
> experiences?
> 
> I know CME pretty well but I don't have any Unity
> Express experience and would prefer not to make a
> mistake here. Any advice appreciated.
> 
> Best,
> 
> leb
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________
Need a vacation? Get great deals
to amazing places on Yahoo! Travel.
http://travel.yahoo.com/

From jonvoip at gmail.com  Mon Oct 15 00:54:32 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 14 Oct 2007 23:54:32 -0500
Subject: [cisco-voip] Triple Unity servers...
In-Reply-To: <5d093f9a0709190931h28adab0boec9034c1fdab9660@mail.gmail.com>
References: <5d093f9a0709190931h28adab0boec9034c1fdab9660@mail.gmail.com>
Message-ID: <5d093f9a0710142154w43a9f31cqa0e2b30bdfdae838@mail.gmail.com>

So, I have to do this on Tuesday, and was wondering if anyone had any ideas...

The plan is to install the primary and secondary Unity servers without
TTS and then run the Failoverconfig.exe to mate them together...
Pretty simple (I hope).

The question is about this third Unity server, the dedicated TTS
box... how do I link that to the Unity twins?

Not finding a lot of love on Cisco's website here... (no mention
anywhere of the concept of a dedicated TTS box)...




Jonathan

On 9/19/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> Cust ordered 3 Unity servers, a  primary, a secondary and a dedicated TTS box.
>
> Not sure how to install the TTS box. Do I set it up normally (SQL,
> Exchange tools, etc.) and then install Unity but only select TTS (and
> not select Unity)?
>
> Also, would I NOT select TTS on the two primaries?
>
> Just curious, I didn't sell this monster...
>
>
>
>
>
> Jonathan
>

From aman.chugh at gmail.com  Mon Oct 15 02:33:41 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Mon, 15 Oct 2007 12:03:41 +0530
Subject: [cisco-voip] Triple Unity servers...
In-Reply-To: <5d093f9a0710142154w43a9f31cqa0e2b30bdfdae838@mail.gmail.com>
References: <5d093f9a0709190931h28adab0boec9034c1fdab9660@mail.gmail.com>
	<5d093f9a0710142154w43a9f31cqa0e2b30bdfdae838@mail.gmail.com>
Message-ID: <d4c697ad0710142333g3dbebd24g26934f58becd9547@mail.gmail.com>

Not Sure if this would be a supported TAC deployement.

I am thinking that you can run the unity setup and enable TTS setting only
on the third box but It would be interesting to learn if three unity servers
can exist toegther in the failover setup.

Do let us know what happens next !

Aman


On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> So, I have to do this on Tuesday, and was wondering if anyone had any
> ideas...
>
> The plan is to install the primary and secondary Unity servers without
> TTS and then run the Failoverconfig.exe to mate them together...
> Pretty simple (I hope).
>
> The question is about this third Unity server, the dedicated TTS
> box... how do I link that to the Unity twins?
>
> Not finding a lot of love on Cisco's website here... (no mention
> anywhere of the concept of a dedicated TTS box)...
>
>
>
>
> Jonathan
>
> On 9/19/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > Cust ordered 3 Unity servers, a  primary, a secondary and a dedicated
> TTS box.
> >
> > Not sure how to install the TTS box. Do I set it up normally (SQL,
> > Exchange tools, etc.) and then install Unity but only select TTS (and
> > not select Unity)?
> >
> > Also, would I NOT select TTS on the two primaries?
> >
> > Just curious, I didn't sell this monster...
> >
> >
> >
> >
> >
> > Jonathan
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jonvoip at gmail.com  Mon Oct 15 02:48:01 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 15 Oct 2007 01:48:01 -0500
Subject: [cisco-voip] Triple Unity servers...
In-Reply-To: <d4c697ad0710142333g3dbebd24g26934f58becd9547@mail.gmail.com>
References: <5d093f9a0709190931h28adab0boec9034c1fdab9660@mail.gmail.com>
	<5d093f9a0710142154w43a9f31cqa0e2b30bdfdae838@mail.gmail.com>
	<d4c697ad0710142333g3dbebd24g26934f58becd9547@mail.gmail.com>
Message-ID: <5d093f9a0710142348g557b30e8h2f63af13e1425790@mail.gmail.com>

Well, those are the questions:

A: How is this done, if it can be done at all.

B: Would TAC support afterwards.


Jonathan

On 10/15/07, Aman Chugh <aman.chugh at gmail.com> wrote:
> Not Sure if this would be a supported TAC deployement.
>
> I am thinking that you can run the unity setup and enable TTS setting only
> on the third box but It would be interesting to learn if three unity servers
> can exist toegther in the failover setup.
>
> Do let us know what happens next !
>
> Aman
>
>
> On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > So, I have to do this on Tuesday, and was wondering if anyone had any
> ideas...
> >
> > The plan is to install the primary and secondary Unity servers without
> > TTS and then run the Failoverconfig.exe to mate them together...
> > Pretty simple (I hope).
> >
> > The question is about this third Unity server, the dedicated TTS
> > box... how do I link that to the Unity twins?
> >
> > Not finding a lot of love on Cisco's website here... (no mention
> > anywhere of the concept of a dedicated TTS box)...
> >
> >
> >
> >
> > Jonathan
> >
> > On 9/19/07, Jonathan Charles < jonvoip at gmail.com> wrote:
> > > Cust ordered 3 Unity servers, a  primary, a secondary and a dedicated
> TTS box.
> > >
> > > Not sure how to install the TTS box. Do I set it up normally (SQL,
> > > Exchange tools, etc.) and then install Unity but only select TTS (and
> > > not select Unity)?
> > >
> > > Also, would I NOT select TTS on the two primaries?
> > >
> > > Just curious, I didn't sell this monster...
> > >
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From aka at veco.ru  Mon Oct 15 03:08:45 2007
From: aka at veco.ru (Andrey Koklin)
Date: Mon, 15 Oct 2007 11:08:45 +0400
Subject: [cisco-voip] Problems with Cisco VT Advantage soft
Message-ID: <471311FD.8040501@veco.ru>

We have small installation with about 20 VT Advantage cameras.
And it seems, there is a number of bugs in CUVA soft under Windows XP.

With previous version 2.0.2, there were periodical blue deaths,
influenced by
CDP packet driver. With new 2.0.3, it seems to be cured, but there remain
several other unsolved items.

1. (The most serious). On a number of workstations, CUVA soft refuses
often to see IP phone.
I can't see any system here. Sometimes it finds phone, while often the
soft doesn't see phone,
and it could appear again after several restarts without any feasible
system.
With firewall disabled, I do packets monitoring with wireshark. CUVA
sends CDP packets
every second and gets replies from IP phone. Nevertheless, it can't
establish connection.

2. The soft often can't set/remember camera settings, e.g. anti-flicker
option "50 Hz (European)",
which results in bad video quality.


From ano-1 at worldonline.dk  Mon Oct 15 05:56:37 2007
From: ano-1 at worldonline.dk (ano-1 at worldonline.dk)
Date: Mon, 15 Oct 2007 11:56:37 +0200
Subject: [cisco-voip] Problems with calling another SIP number
Message-ID: <1192442197.471339558af07@webmail.tiscali.dk>

Hey,

Can somebody help me?

When i try to call another SIP number from the same SIP provider I get the 
failur tone. When I call all other number it works fine.

Thanks,

Michael



From benjamin.kappauf at googlemail.com  Mon Oct 15 06:31:12 2007
From: benjamin.kappauf at googlemail.com (Benjamin Kappauf)
Date: Mon, 15 Oct 2007 12:31:12 +0200
Subject: [cisco-voip] call-manager-fallback wrong phone dn
In-Reply-To: <5d093f9a0710141525l5ac25ba4y3c7fc876a9ad76c4@mail.gmail.com>
References: <feidjf$kkr$1@sea.gmane.org> <feom6s$6lq$1@sea.gmane.org>
	<5d093f9a0710141525l5ac25ba4y3c7fc876a9ad76c4@mail.gmail.com>
Message-ID: <fevfis$4cp$1@ger.gmane.org>

hi,

As you can see I am using two Callmanagers (Auto registration is off on 
both of them)

A interesting thing is...

Callmanagers (Site A)
Voice gateway (Site B)

Before I started one admin from site A logged his extensionmob account 
(1700) onto a phone (Phone 1) on site B (changed DP). This phone keeps 
it number on fallback! Then I thought I did something wrong on the other 
phones. So I logged this extensionmob account onto another phone (Phone 
2) on site B. After fallback the old phone (Phone 1) had again the 1700 
and the phone 2 had again its 65xx number.

So it seems as "something" learned once all phone DNs and on fallback 
they are getting up again. It can't be the phones; I did a hard reset on 
them.  After that I did a restart on that voice gateway. So even the 
voice gateway should now have lost every temporally information about 
any phone and phone DN.


Current configuration : 3483 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname x
!

logging buffered 51200 warnings
!
no aaa new-model
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
!
!
ip cef
!
!
no ip domain lookup
ip domain name yourdomain.com
multilink bundle-name authenticated
!
isdn switch-type primary-ni
!
voice-card 0
  dspfarm
  dsp services dspfarm
!
!
voice call send-alert
voice call convert-discpi-to-prog
!
voice service voip
  allow-connections h323 to h323
  supplementary-service h450.12
  h323
!
!
voice class codec 1
  codec preference 1 g711ulaw
  codec preference 2 g711alaw
  codec preference 3 g729r8
!
!
!
voice class h323 1
  h225 timeout tcp establish 3
   call start slow
   call preserve
!
!
!
!
!
!
voice translation-rule 1
  rule 1 /^1284\(...\)$/ /4\1/
!
voice translation-rule 2
  rule 3 /\(.+\)/ /9011\1/ type international international
!
!
voice translation-profile Incoming
  translate calling 2
  translate called 1
!
!
!
!
!
!
archive
  log config
   hidekeys
!
!
controller T1 0/0/0
  framing esf
  linecode b8zs
  pri-group timeslots 1-24
!
!
!
!
!
interface GigabitEthernet0/0
  description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$$ES_LAN$
  ip address 10.40.2.15 255.255.255.0
  duplex auto
  speed auto
  h323-gateway voip bind srcaddr 10.40.2.15
!
interface GigabitEthernet0/1
  no ip address
  shutdown
  duplex auto
  speed auto
!
interface Serial0/0/0:23
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn timer T310 60000
  isdn incoming-voice voice
  isdn sending-complete
  no cdp enable
!
ip route 0.0.0.0 0.0.0.0 10.40.2.1
!
!
ip http server
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
!
!
!
control-plane
!
!
!
voice-port 0/0/0:23
!
!
!
!
!
dial-peer voice 10 pots
  translation-profile incoming Incoming
  incoming called-number .
  direct-inward-dial
  port 0/0/0:23
!
dial-peer voice 40 voip
  description * Primary CM *
  destination-pattern 44..
  progress_ind setup enable 3
  progress_ind connect enable 8
  progress_ind disconnect enable 8
  modem passthrough nse codec g711ulaw
  voice-class codec 1
  voice-class h323 1
  session target ipv4:10.22.33.1
  dtmf-relay h245-alphanumeric
  fax rate disable
  no vad
!
dial-peer voice 50 voip
  description * Secondary CM *
  translation-profile incoming incoming
  preference 1
  destination-pattern 44..
  progress_ind setup enable 3
  progress_ind connect enable 8
  progress_ind disconnect enable 8
  modem passthrough nse codec g711ulaw
  voice-class codec 1
  voice-class h323 1
  session target ipv4:10.22.33.2
  dtmf-relay h245-alphanumeric
  fax rate disable
  no vad
!
dial-peer voice 1 pots
  destination-pattern 9T
  port 0/0/0:23
!
!
!
call-manager-fallback
  secondary-dialtone 9
  max-conferences 8 gain -6
  transfer-system full-consult
  ip source-address 10.40.2.15 port 2000
  system message SRST Fallback
  max-ephones 28
  max-dn 60 dual-line
!
banner login ^CSTOP!^C
!
line con 0
  login local
line aux 0
line vty 0 4
  login local
  transport input telnet
line vty 5 15
  login local
  transport input telnet
!
scheduler allocate 20000 1000
!
end



Jonathan Charles schrieb:
> Post the entire config... It almost seems like you have a bunch of
> ephone-dn's on the router and it is auto-assigning them...
> 
> 
> 
> Jonathan
> 
> On 10/12/07, Benjamin Kappauf <benjamin.kappauf at googlemail.com> wrote:
>> Benjamin Kappauf schrieb:
>>> What is the problem if in case of a fallback all my phones are
>>> registered fine to the voice router. But all phones don't get the
>>> correct DN. Most of them get a 65xx DN. A few are with ext. mobility
>>> activ. And they all have got DNs out of 4440 - 4449
>>>
>>> Btw. no 65xx number exist on the callmanager.
>>>
>>> Thanks alot for any help!
>>>
>>>
>>> ----------------------------------
>>> call-manager-fallback
>>>   secondary-dialtone 9
>>>   max-conferences 8 gain -6
>>>   transfer-system full-consult
>>>   ip source-address x.x.x.x port 2000
>>>   max-ephones 40
>>>   max-dn 80 dual-line
>>>   system message primary SRST Fallback
>>>   system message secondary SRST
>>>   time-format 24
>>>   date-format dd-mm-yy
>> Today I downgraded the voice gateway to
>> c2800nm-spservicesk9-mz.124-9.T5.bin and I did a reset on all phones.
>>
>> Afterwards I initialised SRST. But I still get the wrong DNs.
>>
>> What cause a phone on fallback to get a wrong dn?
>>
>> Thank you
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>


From iandupreez at gmail.com  Mon Oct 15 07:38:19 2007
From: iandupreez at gmail.com (Ian du Preez)
Date: Mon, 15 Oct 2007 13:38:19 +0200
Subject: [cisco-voip] Connection system message store
Message-ID: <b56232420710150438y10446f69m889b01755649c70@mail.gmail.com>

Is there any way of moving a Connection 1.x system message store to a diff
drive or partition?
My drive is running out of space, and I have another with 60G availible.

Thanks
Ian
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From akoul at dataways.gr  Mon Oct 15 07:53:39 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Mon, 15 Oct 2007 14:53:39 +0300
Subject: [cisco-voip] 7970 logo turned to Black and White after using BLFs
Message-ID: <471354C3.4050709@dataways.gr>

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From wsisk at cisco.com  Mon Oct 15 09:38:07 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 15 Oct 2007 09:38:07 -0400
Subject: [cisco-voip] IPCC Extension
In-Reply-To: <5d093f9a0710141519i66814925g91aca95e41d2d7e0@mail.gmail.com>
References: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>	
	<471193E3.7000806@cisco.com>	
	<5d093f9a0710141151y65daaaa1we8097718814688d5@mail.gmail.com>	
	<Pine.GSO.4.64.0710141508110.2241@clubhouse-6.cisco.com>	
	<5d093f9a0710141305r5c875179o13e3b77378615a14@mail.gmail.com>
	<5d093f9a0710141519i66814925g91aca95e41d2d7e0@mail.gmail.com>
Message-ID: <47136D3F.2060001@cisco.com>

IPCCX install updates it by AXL at install time.
/wes

Jonathan Charles wrote:
> Yeah, as I thought...
>
> IPCC Express Installed:
> Required Field    	
> This parameter is true when a Cisco IPCC Express system is integrated
> with this Cisco CallManager cluster. This is a read only flag. It can
> be modified only by CRS when it has been installed. If this flag is
> true, then the IPCC Extension can be specified on the User
> Configuration page
>
>   	This is a required field.
>   	Default:  False
>
> However, it is not modifiable... at least from Ent Params... Though I
> understand why it can't be changed (it would nuke out the CRS agent
> settings if someone accidentally changed it from true to false).
>
>
> Jonathan
>
> On 10/14/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>   
>> Well, that is the question.
>>
>> It seems like it should be an exposed Enterprise Parameter (CRS
>> Installed or something...)... for some reason that sounds familiar...
>> isn't there such a field?
>>
>>
>>
>> Jonathan
>>
>> On 10/14/07, Wesley Sisk <wsisk at cisco.com> wrote:
>>     
>>> feature and fields are always there in cm.  it is only exposed/displayed
>>> when the parameter is set T in processconfig table.
>>>
>>> ipccx install should update param to T.
>>>
>>> Or, if Brandon were doing a rebuild, then the DRS restore would have
>>> restored the value from his previous system.
>>>
>>> thus my curiosity of how he got into this state.
>>>
>>> /wes
>>>
>>> On Sun, 14 Oct 2007, Jonathan Charles wrote:
>>>
>>>       
>>>> How is this actually set? Is it during the install of IPCC 4.5 and it
>>>> creates the field on the CCM user page?
>>>>
>>>>
>>>> Jonathan
>>>>
>>>> On 10/13/07, Wes Sisk <wsisk at cisco.com> wrote:
>>>>         
>>>>>  Brandon,
>>>>>
>>>>>  Did you perform a DRS restore or reinstall IPCCX after rebuilding the pub?
>>>>> Either of those actions should set the values properly.  That is really the
>>>>> best way to get all dependencies corrected.  Otherwise, here is the manual
>>>>> process:
>>>>>
>>>>>  1. Under system->enterprise parameters check for:
>>>>>  IPCC Express Installed
>>>>>  what is value, true or false?
>>>>>
>>>>>  2. If false you should find it F in processconfig as well. confirm the
>>>>> paramname like '%nstalled%' in processconfig table:
>>>>>  admin:run sql select paramname,paramvalue from processconfig where
>>>>> paramname like '%nstalled%'
>>>>>  paramname        paramvalue
>>>>>  ================ ==========
>>>>>  AAInstalledFlag  F
>>>>>  IAQInstalledFlag F
>>>>>
>>>>>  3. force it:
>>>>>  Either re-run the IPCCX install, it should update this value via AXL during
>>>>> install.  Otherwise, force it yourself:
>>>>>  run sql update processconfig set paramvalue="T" where paramname like
>>>>> '%nstalled%'
>>>>>
>>>>>  /Wes
>>>>>
>>>>>
>>>>>  Ogden, Brandon wrote:
>>>>>
>>>>>
>>>>>
>>>>> So I just had to rebuild my publisher (5.1) and I am trying to setup my IPCC
>>>>> agents again.. but I am not showing an option in the End User, Extension
>>>>> associations that will allow me to specify the extension for the IPCC.  I
>>>>> seem to remember reading that there is a sequel statement you can run on the
>>>>> Call manager that enables you to setup extensions to IPCC.  Anyone help?
>>>>> Thanks. ________________________________
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>           

From Brandon.Ogden at amersports.com  Mon Oct 15 09:51:33 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Mon, 15 Oct 2007 06:51:33 -0700
Subject: [cisco-voip] IPCC Extension
In-Reply-To: <Pine.GSO.4.64.0710141508110.2241@clubhouse-6.cisco.com>
References: <D646304AFC55DB4EBB92329BCC800A00D09610@seas7633.am.amersports.int>
	<471193E3.7000806@cisco.com>
	<5d093f9a0710141151y65daaaa1we8097718814688d5@mail.gmail.com>
	<Pine.GSO.4.64.0710141508110.2241@clubhouse-6.cisco.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D0971E@seas7633.am.amersports.int>

Yes.. I did a rebuild...  I am new to this company.. but in the past all
administration of these systems has been handled by the contractor who
installed the system about 6 months ago.  Somewhere between the
contractor and the IT Department, no one had done a backup.  WE had a
UPS Failure last week which corrupted the FileSystem on the
Publisher...and yes.. they are all backed up now. :)    

Brandon Ogden
Amer Sports 
801-624-7529
www.amersports.com
http://www.linkedin.com/in/brandonogden
-----Original Message-----
From: Wesley Sisk [mailto:wsisk at cisco.com] 
Sent: Sunday, October 14, 2007 1:09 PM
To: Jonathan Charles
Cc: Ogden, Brandon; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC Extension

feature and fields are always there in cm.  it is only exposed/displayed

when the parameter is set T in processconfig table.

ipccx install should update param to T.

Or, if Brandon were doing a rebuild, then the DRS restore would have 
restored the value from his previous system.

thus my curiosity of how he got into this state.

/wes

On Sun, 14 Oct 2007, Jonathan Charles wrote:

> How is this actually set? Is it during the install of IPCC 4.5 and it
> creates the field on the CCM user page?
>
>
> Jonathan
>
> On 10/13/07, Wes Sisk <wsisk at cisco.com> wrote:
>>
>>  Brandon,
>>
>>  Did you perform a DRS restore or reinstall IPCCX after rebuilding
the pub?
>> Either of those actions should set the values properly.  That is
really the
>> best way to get all dependencies corrected.  Otherwise, here is the
manual
>> process:
>>
>>  1. Under system->enterprise parameters check for:
>>  IPCC Express Installed
>>  what is value, true or false?
>>
>>  2. If false you should find it F in processconfig as well. confirm
the
>> paramname like '%nstalled%' in processconfig table:
>>  admin:run sql select paramname,paramvalue from processconfig where
>> paramname like '%nstalled%'
>>  paramname        paramvalue
>>  ================ ==========
>>  AAInstalledFlag  F
>>  IAQInstalledFlag F
>>
>>  3. force it:
>>  Either re-run the IPCCX install, it should update this value via AXL
during
>> install.  Otherwise, force it yourself:
>>  run sql update processconfig set paramvalue="T" where paramname like
>> '%nstalled%'
>>
>>  /Wes
>>
>>
>>  Ogden, Brandon wrote:
>>
>>
>>
>> So I just had to rebuild my publisher (5.1) and I am trying to setup
my IPCC
>> agents again.. but I am not showing an option in the End User,
Extension
>> associations that will allow me to specify the extension for the
IPCC.  I
>> seem to remember reading that there is a sequel statement you can run
on the
>> Call manager that enables you to setup extensions to IPCC.  Anyone
help?
>> Thanks. ________________________________
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>

From jonvoip at gmail.com  Mon Oct 15 10:14:29 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 15 Oct 2007 09:14:29 -0500
Subject: [cisco-voip] call-manager-fallback wrong phone dn
In-Reply-To: <fevfis$4cp$1@ger.gmane.org>
References: <feidjf$kkr$1@sea.gmane.org> <feom6s$6lq$1@sea.gmane.org>
	<5d093f9a0710141525l5ac25ba4y3c7fc876a9ad76c4@mail.gmail.com>
	<fevfis$4cp$1@ger.gmane.org>
Message-ID: <5d093f9a0710150714u4842cbbcy9d8ff2d4c4cbe110@mail.gmail.com>

Did you write mem while in fallback at some point?



Jonathan

On 10/15/07, Benjamin Kappauf <benjamin.kappauf at googlemail.com> wrote:
> hi,
>
> As you can see I am using two Callmanagers (Auto registration is off on
> both of them)
>
> A interesting thing is...
>
> Callmanagers (Site A)
> Voice gateway (Site B)
>
> Before I started one admin from site A logged his extensionmob account
> (1700) onto a phone (Phone 1) on site B (changed DP). This phone keeps
> it number on fallback! Then I thought I did something wrong on the other
> phones. So I logged this extensionmob account onto another phone (Phone
> 2) on site B. After fallback the old phone (Phone 1) had again the 1700
> and the phone 2 had again its 65xx number.
>
> So it seems as "something" learned once all phone DNs and on fallback
> they are getting up again. It can't be the phones; I did a hard reset on
> them.  After that I did a restart on that voice gateway. So even the
> voice gateway should now have lost every temporally information about
> any phone and phone DN.
>
>
> Current configuration : 3483 bytes
> !
> version 12.4
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname x
> !
>
> logging buffered 51200 warnings
> !
> no aaa new-model
> network-clock-participate wic 0
> network-clock-select 1 T1 0/0/0
> !
> !
> ip cef
> !
> !
> no ip domain lookup
> ip domain name yourdomain.com
> multilink bundle-name authenticated
> !
> isdn switch-type primary-ni
> !
> voice-card 0
>   dspfarm
>   dsp services dspfarm
> !
> !
> voice call send-alert
> voice call convert-discpi-to-prog
> !
> voice service voip
>   allow-connections h323 to h323
>   supplementary-service h450.12
>   h323
> !
> !
> voice class codec 1
>   codec preference 1 g711ulaw
>   codec preference 2 g711alaw
>   codec preference 3 g729r8
> !
> !
> !
> voice class h323 1
>   h225 timeout tcp establish 3
>    call start slow
>    call preserve
> !
> !
> !
> !
> !
> !
> voice translation-rule 1
>   rule 1 /^1284\(...\)$/ /4\1/
> !
> voice translation-rule 2
>   rule 3 /\(.+\)/ /9011\1/ type international international
> !
> !
> voice translation-profile Incoming
>   translate calling 2
>   translate called 1
> !
> !
> !
> !
> !
> !
> archive
>   log config
>    hidekeys
> !
> !
> controller T1 0/0/0
>   framing esf
>   linecode b8zs
>   pri-group timeslots 1-24
> !
> !
> !
> !
> !
> interface GigabitEthernet0/0
>   description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$$ES_LAN$
>   ip address 10.40.2.15 255.255.255.0
>   duplex auto
>   speed auto
>   h323-gateway voip bind srcaddr 10.40.2.15
> !
> interface GigabitEthernet0/1
>   no ip address
>   shutdown
>   duplex auto
>   speed auto
> !
> interface Serial0/0/0:23
>   no ip address
>   encapsulation hdlc
>   isdn switch-type primary-ni
>   isdn timer T310 60000
>   isdn incoming-voice voice
>   isdn sending-complete
>   no cdp enable
> !
> ip route 0.0.0.0 0.0.0.0 10.40.2.1
> !
> !
> ip http server
> ip http authentication local
> no ip http secure-server
> ip http timeout-policy idle 60 life 86400 requests 10000
> !
> !
> !
> !
> control-plane
> !
> !
> !
> voice-port 0/0/0:23
> !
> !
> !
> !
> !
> dial-peer voice 10 pots
>   translation-profile incoming Incoming
>   incoming called-number .
>   direct-inward-dial
>   port 0/0/0:23
> !
> dial-peer voice 40 voip
>   description * Primary CM *
>   destination-pattern 44..
>   progress_ind setup enable 3
>   progress_ind connect enable 8
>   progress_ind disconnect enable 8
>   modem passthrough nse codec g711ulaw
>   voice-class codec 1
>   voice-class h323 1
>   session target ipv4:10.22.33.1
>   dtmf-relay h245-alphanumeric
>   fax rate disable
>   no vad
> !
> dial-peer voice 50 voip
>   description * Secondary CM *
>   translation-profile incoming incoming
>   preference 1
>   destination-pattern 44..
>   progress_ind setup enable 3
>   progress_ind connect enable 8
>   progress_ind disconnect enable 8
>   modem passthrough nse codec g711ulaw
>   voice-class codec 1
>   voice-class h323 1
>   session target ipv4:10.22.33.2
>   dtmf-relay h245-alphanumeric
>   fax rate disable
>   no vad
> !
> dial-peer voice 1 pots
>   destination-pattern 9T
>   port 0/0/0:23
> !
> !
> !
> call-manager-fallback
>   secondary-dialtone 9
>   max-conferences 8 gain -6
>   transfer-system full-consult
>   ip source-address 10.40.2.15 port 2000
>   system message SRST Fallback
>   max-ephones 28
>   max-dn 60 dual-line
> !
> banner login ^CSTOP!^C
> !
> line con 0
>   login local
> line aux 0
> line vty 0 4
>   login local
>   transport input telnet
> line vty 5 15
>   login local
>   transport input telnet
> !
> scheduler allocate 20000 1000
> !
> end
>
>
>
> Jonathan Charles schrieb:
> > Post the entire config... It almost seems like you have a bunch of
> > ephone-dn's on the router and it is auto-assigning them...
> >
> >
> >
> > Jonathan
> >
> > On 10/12/07, Benjamin Kappauf <benjamin.kappauf at googlemail.com> wrote:
> >> Benjamin Kappauf schrieb:
> >>> What is the problem if in case of a fallback all my phones are
> >>> registered fine to the voice router. But all phones don't get the
> >>> correct DN. Most of them get a 65xx DN. A few are with ext. mobility
> >>> activ. And they all have got DNs out of 4440 - 4449
> >>>
> >>> Btw. no 65xx number exist on the callmanager.
> >>>
> >>> Thanks alot for any help!
> >>>
> >>>
> >>> ----------------------------------
> >>> call-manager-fallback
> >>>   secondary-dialtone 9
> >>>   max-conferences 8 gain -6
> >>>   transfer-system full-consult
> >>>   ip source-address x.x.x.x port 2000
> >>>   max-ephones 40
> >>>   max-dn 80 dual-line
> >>>   system message primary SRST Fallback
> >>>   system message secondary SRST
> >>>   time-format 24
> >>>   date-format dd-mm-yy
> >> Today I downgraded the voice gateway to
> >> c2800nm-spservicesk9-mz.124-9.T5.bin and I did a reset on all phones.
> >>
> >> Afterwards I initialised SRST. But I still get the wrong DNs.
> >>
> >> What cause a phone on fallback to get a wrong dn?
> >>
> >> Thank you
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From rratliff at cisco.com  Mon Oct 15 11:44:51 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 15 Oct 2007 11:44:51 -0400
Subject: [cisco-voip] Problems with Cisco VT Advantage soft
In-Reply-To: <471311FD.8040501@veco.ru>
References: <471311FD.8040501@veco.ru>
Message-ID: <F94FFB17-AC2C-46CF-95C4-D60051F7B617@cisco.com>

For problem 1 from the PC try to telnet from the PC to the phone on  
port 4224.   The VTA should get the IP address of the phone from the  
CDP message and then use a TCP connection on port 4224 for the CAST  
messaging.  If it can't get that connection established then it will  
not "see" the phone correctly.

-Ryan

On Oct 15, 2007, at 3:08 AM, Andrey Koklin wrote:

We have small installation with about 20 VT Advantage cameras.
And it seems, there is a number of bugs in CUVA soft under Windows XP.

With previous version 2.0.2, there were periodical blue deaths,
influenced by
CDP packet driver. With new 2.0.3, it seems to be cured, but there  
remain
several other unsolved items.

1. (The most serious). On a number of workstations, CUVA soft refuses
often to see IP phone.
I can't see any system here. Sometimes it finds phone, while often the
soft doesn't see phone,
and it could appear again after several restarts without any feasible
system.
With firewall disabled, I do packets monitoring with wireshark. CUVA
sends CDP packets
every second and gets replies from IP phone. Nevertheless, it can't
establish connection.

2. The soft often can't set/remember camera settings, e.g. anti-flicker
option "50 Hz (European)",
which results in bad video quality.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From rkulagow at gmail.com  Mon Oct 15 12:16:22 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Mon, 15 Oct 2007 11:16:22 -0500
Subject: [cisco-voip] Problems with Cisco VT Advantage soft
In-Reply-To: <F94FFB17-AC2C-46CF-95C4-D60051F7B617@cisco.com>
References: <471311FD.8040501@veco.ru>
	<F94FFB17-AC2C-46CF-95C4-D60051F7B617@cisco.com>
Message-ID: <47139256.4000403@gmail.com>

Ryan Ratliff wrote:
> For problem 1 from the PC try to telnet from the PC to the phone on  
> port 4224.   The VTA should get the IP address of the phone from the  
> CDP message and then use a TCP connection on port 4224 for the CAST  
> messaging.  If it can't get that connection established then it will  
> not "see" the phone correctly.

Is there a workaround / feature planned where VTA can work where the PC 
isn't hanging off of the phone?  We have separate voice/data 
infrastructures, but it's all 100/FDX to the desktop, GigE uplinks to 
core, so bandwidth shouldn't be an issue.

And IP Communicator on the desktop means lots and lots of USB headsets 
and microphones would need to be purchased, so not going there.

From Brandon.Ogden at amersports.com  Mon Oct 15 12:21:52 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Mon, 15 Oct 2007 09:21:52 -0700
Subject: [cisco-voip] IPCC Whisper
In-Reply-To: <47139256.4000403@gmail.com>
References: <471311FD.8040501@veco.ru><F94FFB17-AC2C-46CF-95C4-D60051F7B617@cisco.com>
	<47139256.4000403@gmail.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D0999F@seas7633.am.amersports.int>

Does anyone know where I can setup a Queue Whisper in IPCC 4.5.  Thanks


Brandon Ogden
Amer Sports 
801-624-7529
www.amersports.com
http://www.linkedin.com/in/brandonogden


From rratliff at cisco.com  Mon Oct 15 12:38:41 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 15 Oct 2007 12:38:41 -0400
Subject: [cisco-voip] Problems with Cisco VT Advantage soft
In-Reply-To: <47139256.4000403@gmail.com>
References: <471311FD.8040501@veco.ru>
	<F94FFB17-AC2C-46CF-95C4-D60051F7B617@cisco.com>
	<47139256.4000403@gmail.com>
Message-ID: <60E98FF1-CE96-4506-BCDC-D764F5C634CE@cisco.com>

I couldn't tell you if any such enhancement to CUVA is planned.   It  
certainly doesn't seem too difficult a problem technically.  That  
sounds like a good question for one of the Cisco speakers at CIPTUG.

-Ryan

On Oct 15, 2007, at 12:16 PM, Robert Kulagowski wrote:

Ryan Ratliff wrote:
> For problem 1 from the PC try to telnet from the PC to the phone on
> port 4224.   The VTA should get the IP address of the phone from the
> CDP message and then use a TCP connection on port 4224 for the CAST
> messaging.  If it can't get that connection established then it will
> not "see" the phone correctly.

Is there a workaround / feature planned where VTA can work where the PC
isn't hanging off of the phone?  We have separate voice/data
infrastructures, but it's all 100/FDX to the desktop, GigE uplinks to
core, so bandwidth shouldn't be an issue.

And IP Communicator on the desktop means lots and lots of USB headsets
and microphones would need to be purchased, so not going there.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From i.anderson at lancaster.ac.uk  Mon Oct 15 12:10:05 2007
From: i.anderson at lancaster.ac.uk (Anderson, Ian)
Date: Mon, 15 Oct 2007 17:10:05 +0100
Subject: [cisco-voip] user access to ccmuser web pages
In-Reply-To: <A58F94A553BA4742821241C8F3E76C9F04494A3F@EX2.ACDM.DS.SAIT.CA>
References: <A58F94A553BA4742821241C8F3E76C9F04494A00@EX2.ACDM.DS.SAIT.CA><46FD447F.8080303@cisco.com><A58F94A553BA4742821241C8F3E76C9F04494A27@EX2.ACDM.DS.SAIT.CA><46FD4FEE.2050809@cisco.com>
	<A58F94A553BA4742821241C8F3E76C9F04494A3F@EX2.ACDM.DS.SAIT.CA>
Message-ID: <84C837A579BB6B41993A00F5256767590375B2F1@exchange-be3.lancs.local>

Hi 

 

We route all requests for ccmuser through a squid reverse proxy running
under linux...  Works fine and achieves what you are after

 

/Ian

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Eric Pedersen
Sent: 28 September 2007 21:07
To: Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] user access to ccmuser web pages

 

Thanks Wes.  Filtering management IP address is standard security
practice on routers and switches, and is easy to implement.  Do you know
if there is a feature request for something similar in callmanager?

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: September 28, 2007 13:03
To: Eric Pedersen
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] user access to ccmuser web pages

Eric,

Good clarification.  Nothing built into the product to allow this, but
sounds like a good use of a proxy server. AONS/firewall would not work
because it's https and encrypted on the wire.

/Wes

Eric Pedersen wrote: 

I wasn't clear enough.  We have a limited range of IP addresses that are
trusted for callmanager administration, and we have larger IP ranges
where our general user population reside.  I would like to filter what
networks can access ccmadmin, os admin, etc. so that the general user
population can't even get to the login screen.  Because ccmadmin and
ccmuser use the same tcp ports, and I haven't found any way to change
this, I cannot simply filter admin access with router ACLs.

 

Simple username and password authentication isn't a particularly secure
way to protect such a key piece of infrastructure ... you're just one
accidental password disclosure or web server bug away from a hacked
callmanager. 

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: September 28, 2007 12:14
To: Eric Pedersen
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] user access to ccmuser web pages

check out the "Standard CCM End Users" group.

Eric Pedersen wrote: 

I'm using callmanager 5.1.  I want to enable general user access to the
callmanager ccmuser web pages.  I have not seen any way to allow this
without also giving access to ccmadmin/osadmin/etc. web pages, which I
don't want to do for obvious security reasons.  Is there a way to do
this?

 

Thanks,

Eric



________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From voicenoob at gmail.com  Mon Oct 15 13:12:01 2007
From: voicenoob at gmail.com (Voice Noob)
Date: Mon, 15 Oct 2007 12:12:01 -0500
Subject: [cisco-voip] CUPS 1.3 with UCM 6.01 no users show up on CUPS server.
Message-ID: <535625f70710151012j3383fe70l8f7b1fa5802690a2@mail.gmail.com>

I am trying to get CUPS to work for the first time. I have the server up and
running but I can't import any users. I have assigned the capabilities to
the users in Call Manager and CUPS sees the phones but does not see any of
my users. I have restarted the SYNC agent several times. Does anyone have
any experience with this product?
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From MillerS at DicksteinShapiro.COM  Mon Oct 15 13:33:10 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 15 Oct 2007 13:33:10 -0400
Subject: [cisco-voip] Quick Question About Orphaned WAV Files
Message-ID: <421CAA1A5300B84A87D200D70998578263DDFE@DCEX2.DSMO.COM>

TAC tells me that the orphaned wav file errors I see when I run DBWalker
can be easily fixed with that program and that there should be no
negative repercussions.  Can I get an "amen" on that before I do it?  He
also said that I should run DBWalker without correction first, then run
with correction. Does that also make sense?  Thank you!
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

--------------------------------------------------------
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named above. This communication may contain material 
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privileges. If you are not the intended recipient or person 
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to the intended recipient, you have received this 
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this e-mail message and any attached files is strictly 
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From MillerS at DicksteinShapiro.COM  Mon Oct 15 13:44:40 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 15 Oct 2007 13:44:40 -0400
Subject: [cisco-voip] Quick Question About Orphaned WAV Files
References: <421CAA1A5300B84A87D200D70998578263DDFE@DCEX2.DSMO.COM>
	<BD89624EE83BD349BD31A08B453DE24401C682F4@uogexb201.cfs.uoguelph.ca>
Message-ID: <421CAA1A5300B84A87D200D70998578263DE00@DCEX2.DSMO.COM>

Thank you!  Does everyone else agree?
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

________________________________

From: Rasim Duric [mailto:rduric at uoguelph.ca] 
Sent: Monday, October 15, 2007 1:43 PM
To: Miller, Steve
Subject: RE: [cisco-voip] Quick Question About Orphaned WAV Files



It shouldn't be a problem. I've been regularly doing these tasks without
consequences.

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, October 15, 2007 1:33 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Quick Question About Orphaned WAV Files

 

TAC tells me that the orphaned wav file errors I see when I run DBWalker
can be easily fixed with that program and that there should be no
negative repercussions.  Can I get an "amen" on that before I do it?  He
also said that I should run DBWalker without correction first, then run
with correction. Does that also make sense?  Thank you!

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
responsible for delivering this confidential communication 
to the intended recipient, you have received this 
communication in error, and any review, use, dissemination, 
forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
network.  If you have received this confidential 
communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
the original message. 
 
To reply to our email administrator directly, send an email 
to postmaster at dicksteinshapiro.com
 
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
 
========================================================================
======

--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
responsible for delivering this confidential communication 
to the intended recipient, you have received this 
communication in error, and any review, use, dissemination, 
forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
network.  If you have received this confidential 
communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
the original message. 

To reply to our email administrator directly, send an email 
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From rkulagow at gmail.com  Mon Oct 15 13:45:50 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Mon, 15 Oct 2007 12:45:50 -0500
Subject: [cisco-voip] Problems with Cisco VT Advantage soft
In-Reply-To: <60E98FF1-CE96-4506-BCDC-D764F5C634CE@cisco.com>
References: <471311FD.8040501@veco.ru>
	<F94FFB17-AC2C-46CF-95C4-D60051F7B617@cisco.com>
	<47139256.4000403@gmail.com>
	<60E98FF1-CE96-4506-BCDC-D764F5C634CE@cisco.com>
Message-ID: <4713A74E.9050101@gmail.com>

Ryan Ratliff wrote:
> I couldn't tell you if any such enhancement to CUVA is planned.   It 
> certainly doesn't seem too difficult a problem technically.  That sounds 
> like a good question for one of the Cisco speakers at CIPTUG.

CIPTUG sounds great except for the divorce which would accompany my 
going to a conference instead of going trick-or-treating with the kids.

From asobihoudai at yahoo.com  Mon Oct 15 13:51:30 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 15 Oct 2007 10:51:30 -0700 (PDT)
Subject: [cisco-voip] Problems with Cisco VT Advantage soft
In-Reply-To: <4713A74E.9050101@gmail.com>
Message-ID: <4218.96416.qm@web57206.mail.re3.yahoo.com>

You can always take the little ones along with you and
tell your better half that CIPTUG is almost like
trick-or-treating except you get coporate schwag
instead.



--- Robert Kulagowski <rkulagow at gmail.com> wrote:

> Ryan Ratliff wrote:
> > I couldn't tell you if any such enhancement to
> CUVA is planned.   It 
> > certainly doesn't seem too difficult a problem
> technically.  That sounds 
> > like a good question for one of the Cisco speakers
> at CIPTUG.
> 
> CIPTUG sounds great except for the divorce which
> would accompany my 
> going to a conference instead of going
> trick-or-treating with the kids.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



      ____________________________________________________________________________________
Shape Yahoo! in your own image.  Join our Network Research Panel today!   http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 



From kenjohnson at letu.edu  Mon Oct 15 14:04:31 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Mon, 15 Oct 2007 13:04:31 -0500
Subject: [cisco-voip] Quick Question About Orphaned WAV Files
In-Reply-To: <421CAA1A5300B84A87D200D70998578263DE00@DCEX2.DSMO.COM>
References: <421CAA1A5300B84A87D200D70998578263DDFE@DCEX2.DSMO.COM><BD89624EE83BD349BD31A08B453DE24401C682F4@uogexb201.cfs.uoguelph.ca>
	<421CAA1A5300B84A87D200D70998578263DE00@DCEX2.DSMO.COM>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EFCD772D@rachael-reborn.letnet.net>

Yes - I've done that option more than once. The only caveat is once I
did it stupidly while we had many subscribers temporarily removed for a
different issue - that's NOT a good time to do it. I had no way to
re-import the wav files manually after without going to backups...what a
pain J

 

Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
 E-mail: kenjohnson at letu.edu
 Helpdesk: (903) 233-3500
 Phone: (903) 233-3520
 Web: http://www.letu.edu/infotech/
<BLOCKED::http://www.letu.edu/infotech/> 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, October 15, 2007 12:45 PM
To: Rasim Duric
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Quick Question About Orphaned WAV Files

 

Thank you!  Does everyone else agree?

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

 

________________________________

From: Rasim Duric [mailto:rduric at uoguelph.ca] 
Sent: Monday, October 15, 2007 1:43 PM
To: Miller, Steve
Subject: RE: [cisco-voip] Quick Question About Orphaned WAV Files

It shouldn't be a problem. I've been regularly doing these tasks without
consequences.

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, October 15, 2007 1:33 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Quick Question About Orphaned WAV Files

 

TAC tells me that the orphaned wav file errors I see when I run DBWalker
can be easily fixed with that program and that there should be no
negative repercussions.  Can I get an "amen" on that before I do it?  He
also said that I should run DBWalker without correction first, then run
with correction. Does that also make sense?  Thank you!

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
responsible for delivering this confidential communication 
to the intended recipient, you have received this 
communication in error, and any review, use, dissemination, 
forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
network.  If you have received this confidential 
communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
the original message. 
 
To reply to our email administrator directly, send an email 
to postmaster at dicksteinshapiro.com
 
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
 
========================================================================
======
--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
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to the intended recipient, you have received this 
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forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
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immediately by reply e-mail message and permanently delete 
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To reply to our email administrator directly, send an email 
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Not spam
<http://antispam.letu.edu/canit/b.php?i=8111365&m=52aed317b8a3&c=n> 
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<http://antispam.letu.edu/canit/b.php?i=8111365&m=52aed317b8a3&c=f> 

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From MillerS at DicksteinShapiro.COM  Mon Oct 15 14:05:47 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 15 Oct 2007 14:05:47 -0400
Subject: [cisco-voip] Quick Question About Orphaned WAV Files
References: <421CAA1A5300B84A87D200D70998578263DDFE@DCEX2.DSMO.COM><BD89624EE83BD349BD31A08B453DE24401C682F4@uogexb201.cfs.uoguelph.ca>
	<421CAA1A5300B84A87D200D70998578263DE00@DCEX2.DSMO.COM>
	<04B6D8CDD22FD34993C47E33E21F82EFCD772D@rachael-reborn.letnet.net>
Message-ID: <421CAA1A5300B84A87D200D70998578263DE06@DCEX2.DSMO.COM>

Cool.  Thanks everyone.
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

________________________________

From: Johnson, Ken [mailto:kenjohnson at letu.edu] 
Sent: Monday, October 15, 2007 2:05 PM
To: Miller, Steve; Rasim Duric
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Quick Question About Orphaned WAV Files



Yes - I've done that option more than once. The only caveat is once I
did it stupidly while we had many subscribers temporarily removed for a
different issue - that's NOT a good time to do it. I had no way to
re-import the wav files manually after without going to backups...what a
pain J

 

Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
 E-mail: kenjohnson at letu.edu
 Helpdesk: (903) 233-3500
 Phone: (903) 233-3520
 Web: http://www.letu.edu/infotech/
<BLOCKED::http://www.letu.edu/infotech/> 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, October 15, 2007 12:45 PM
To: Rasim Duric
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Quick Question About Orphaned WAV Files

 

Thank you!  Does everyone else agree?

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

 

________________________________

From: Rasim Duric [mailto:rduric at uoguelph.ca] 
Sent: Monday, October 15, 2007 1:43 PM
To: Miller, Steve
Subject: RE: [cisco-voip] Quick Question About Orphaned WAV Files

It shouldn't be a problem. I've been regularly doing these tasks without
consequences.

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, October 15, 2007 1:33 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Quick Question About Orphaned WAV Files

 

TAC tells me that the orphaned wav file errors I see when I run DBWalker
can be easily fixed with that program and that there should be no
negative repercussions.  Can I get an "amen" on that before I do it?  He
also said that I should run DBWalker without correction first, then run
with correction. Does that also make sense?  Thank you!

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
responsible for delivering this confidential communication 
to the intended recipient, you have received this 
communication in error, and any review, use, dissemination, 
forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
network.  If you have received this confidential 
communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
the original message. 
 
To reply to our email administrator directly, send an email 
to postmaster at dicksteinshapiro.com
 
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
 
========================================================================
======
--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
responsible for delivering this confidential communication 
to the intended recipient, you have received this 
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this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
network.  If you have received this confidential 
communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
the original message. 
 
To reply to our email administrator directly, send an email 
to postmaster at dicksteinshapiro.com
 
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
 
========================================================================
======
________________________________



Spam <http://antispam.letu.edu/canit/b.php?i=8111365&m=52aed317b8a3&c=s>

Not spam
<http://antispam.letu.edu/canit/b.php?i=8111365&m=52aed317b8a3&c=n> 
Forget previous vote
<http://antispam.letu.edu/canit/b.php?i=8111365&m=52aed317b8a3&c=f> 


--------------------------------------------------------
This e-mail message and any attached files are confidential
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privileges. If you are not the intended recipient or person 
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to the intended recipient, you have received this 
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this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
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communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
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From adman at adman.net  Mon Oct 15 14:17:15 2007
From: adman at adman.net (Adam Blomfield)
Date: Mon, 15 Oct 2007 13:17:15 -0500
Subject: [cisco-voip] 7970 logo turned to Black and White after using
	BLFs
In-Reply-To: <471354C3.4050709@dataways.gr>
References: <471354C3.4050709@dataways.gr>
Message-ID: <c6f32ce40710151117h31cc2c7ar8f135b4e974eea2e@mail.gmail.com>

I doesn't actually turn black and white... what happens is it puts a line
across the screen next to each active button that has a translucent
background the image behind. Once you have used all your lines the entire
phone desktop background has that washed out appearance.

On 10/15/07, Anthony Kouloglou <akoul at dataways.gr> wrote:
>
>  Hi,
> i have been using a 7970 for over a year and i have used a nice colored
> image for background image.
> Yesterday, after, i have used 6 buttons as BLFs, the image turned in Black
> and white!!
> CCM version: *5.0.4.3110-1*
> 7970:*SCCP70.8-3-2S*
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From cisco1.charles at gmail.com  Mon Oct 15 14:19:47 2007
From: cisco1.charles at gmail.com (charles george)
Date: Mon, 15 Oct 2007 14:19:47 -0400
Subject: [cisco-voip] cuc 6.x
Message-ID: <9552cdf0710151119kb67d76fq433640cd887171c6@mail.gmail.com>

how do i clear all dns entries via cli in ccm 6.x

I added a dns server via cli and I get error expected to see 1 mandortory
field and 0 non-mandatory fields but I ahve 2 entries

-- 
Regards,
Charles
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From klong at olotp.org  Mon Oct 15 13:20:22 2007
From: klong at olotp.org (klong at olotp.org)
Date: Mon, 15 Oct 2007 10:20:22 -0700
Subject: [cisco-voip] Veritas and CCM4.3.2(sr2)/appliance OS v4.3.a SR9
Message-ID: <20071015102022.97296067075fa8c76087192ed0e74e4e.ce0ff1b39a.wbe@email.secureserver.net>

Hello,

We will be deploying CCM4.3.2(sr2)/appliance OS v4.3.a SR9.  Our policy is
to install the Veritas agent on our servers for back up/disaster recovery.
Will using Veritas cause problems?  Does anyone have a link to the Cisco
documentation that speaks to any issues or capabilities?

Thanks in advance,

K.A. Long

From jonvoip at gmail.com  Mon Oct 15 15:32:23 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 15 Oct 2007 14:32:23 -0500
Subject: [cisco-voip] Triple Unity servers...
In-Reply-To: <5d093f9a0710142348g557b30e8h2f63af13e1425790@mail.gmail.com>
References: <5d093f9a0709190931h28adab0boec9034c1fdab9660@mail.gmail.com>
	<5d093f9a0710142154w43a9f31cqa0e2b30bdfdae838@mail.gmail.com>
	<d4c697ad0710142333g3dbebd24g26934f58becd9547@mail.gmail.com>
	<5d093f9a0710142348g557b30e8h2f63af13e1425790@mail.gmail.com>
Message-ID: <5d093f9a0710151232h35a08a1fg58b782ab738f6d1f@mail.gmail.com>

OK, I opened a TAC case and here is the response:


"Unfortunately, you cannot install TTS server on a dedicated hardware
box, even on CU ver 5.0. Only Cisco Unity Connection allows you to do
that."


Jonathan

On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> Well, those are the questions:
>
> A: How is this done, if it can be done at all.
>
> B: Would TAC support afterwards.
>
>
> Jonathan
>
> On 10/15/07, Aman Chugh <aman.chugh at gmail.com> wrote:
> > Not Sure if this would be a supported TAC deployement.
> >
> > I am thinking that you can run the unity setup and enable TTS setting only
> > on the third box but It would be interesting to learn if three unity servers
> > can exist toegther in the failover setup.
> >
> > Do let us know what happens next !
> >
> > Aman
> >
> >
> > On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > > So, I have to do this on Tuesday, and was wondering if anyone had any
> > ideas...
> > >
> > > The plan is to install the primary and secondary Unity servers without
> > > TTS and then run the Failoverconfig.exe to mate them together...
> > > Pretty simple (I hope).
> > >
> > > The question is about this third Unity server, the dedicated TTS
> > > box... how do I link that to the Unity twins?
> > >
> > > Not finding a lot of love on Cisco's website here... (no mention
> > > anywhere of the concept of a dedicated TTS box)...
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On 9/19/07, Jonathan Charles < jonvoip at gmail.com> wrote:
> > > > Cust ordered 3 Unity servers, a  primary, a secondary and a dedicated
> > TTS box.
> > > >
> > > > Not sure how to install the TTS box. Do I set it up normally (SQL,
> > > > Exchange tools, etc.) and then install Unity but only select TTS (and
> > > > not select Unity)?
> > > >
> > > > Also, would I NOT select TTS on the two primaries?
> > > >
> > > > Just curious, I didn't sell this monster...
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
>

From svoll.voip at gmail.com  Mon Oct 15 16:24:18 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 15 Oct 2007 13:24:18 -0700
Subject: [cisco-voip] Veritas and CCM4.3.2(sr2)/appliance OS v4.3.a SR9
In-Reply-To: <20071015102022.97296067075fa8c76087192ed0e74e4e.ce0ff1b39a.wbe@email.secureserver.net>
References: <20071015102022.97296067075fa8c76087192ed0e74e4e.ce0ff1b39a.wbe@email.secureserver.net>
Message-ID: <f84a38d30710151324u4ba8b894iaa481136e17b9de7@mail.gmail.com>

use the BARS built into the Call Manager and upload to network storage and
run veritas from there.

I do not believe veritas on Call Manager is supported by TAC.

Scott


On 10/15/07, klong at olotp.org <klong at olotp.org> wrote:
>
> Hello,
>
> We will be deploying CCM4.3.2(sr2)/appliance OS v4.3.a SR9.  Our policy is
> to install the Veritas agent on our servers for back up/disaster recovery.
> Will using Veritas cause problems?  Does anyone have a link to the Cisco
> documentation that speaks to any issues or capabilities?
>
> Thanks in advance,
>
> K.A. Long
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From pat-cv at wcyv.com  Mon Oct 15 16:53:38 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Mon, 15 Oct 2007 16:53:38 -0400
Subject: [cisco-voip] Triple Unity servers...
In-Reply-To: <5d093f9a0710151232h35a08a1fg58b782ab738f6d1f@mail.gmail.com>
References: <5d093f9a0709190931h28adab0boec9034c1fdab9660@mail.gmail.com>	<5d093f9a0710142154w43a9f31cqa0e2b30bdfdae838@mail.gmail.com>	<d4c697ad0710142333g3dbebd24g26934f58becd9547@mail.gmail.com>	<5d093f9a0710142348g557b30e8h2f63af13e1425790@mail.gmail.com>
	<5d093f9a0710151232h35a08a1fg58b782ab738f6d1f@mail.gmail.com>
Message-ID: <4713D352.1000303@wcyv.com>

They're correct, there is no such thing in the Unity world at the 
moment. I believe you (or whoever ordered this) is thinking of the Voice 
Recognition server, new to 5.0. In order to get voice recognition 
features, you have to use another server. Installation procedure here:

http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/installation/guide/umexfo/5xcuigumefoappc.html

-------- Original Message  --------
From: "Jonathan Charles" <jonvoip at gmail.com>
To: "Aman Chugh" <aman.chugh at gmail.com>
Cc: Cisco Voip Mailing list <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Triple Unity servers...
Date: 10/15/2007 3:32 PM

> OK, I opened a TAC case and here is the response:
> 
> 
> "Unfortunately, you cannot install TTS server on a dedicated hardware
> box, even on CU ver 5.0. Only Cisco Unity Connection allows you to do
> that."
> 
> 
> Jonathan
> 
> On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>> Well, those are the questions:
>>
>> A: How is this done, if it can be done at all.
>>
>> B: Would TAC support afterwards.
>>
>>
>> Jonathan
>>
>> On 10/15/07, Aman Chugh <aman.chugh at gmail.com> wrote:
>>> Not Sure if this would be a supported TAC deployement.
>>>
>>> I am thinking that you can run the unity setup and enable TTS setting only
>>> on the third box but It would be interesting to learn if three unity servers
>>> can exist toegther in the failover setup.
>>>
>>> Do let us know what happens next !
>>>
>>> Aman
>>>
>>>
>>> On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>>>> So, I have to do this on Tuesday, and was wondering if anyone had any
>>> ideas...
>>>> The plan is to install the primary and secondary Unity servers without
>>>> TTS and then run the Failoverconfig.exe to mate them together...
>>>> Pretty simple (I hope).
>>>>
>>>> The question is about this third Unity server, the dedicated TTS
>>>> box... how do I link that to the Unity twins?
>>>>
>>>> Not finding a lot of love on Cisco's website here... (no mention
>>>> anywhere of the concept of a dedicated TTS box)...
>>>>
>>>>
>>>>
>>>>
>>>> Jonathan
>>>>
>>>> On 9/19/07, Jonathan Charles < jonvoip at gmail.com> wrote:
>>>>> Cust ordered 3 Unity servers, a  primary, a secondary and a dedicated
>>> TTS box.
>>>>> Not sure how to install the TTS box. Do I set it up normally (SQL,
>>>>> Exchange tools, etc.) and then install Unity but only select TTS (and
>>>>> not select Unity)?
>>>>>
>>>>> Also, would I NOT select TTS on the two primaries?
>>>>>
>>>>> Just curious, I didn't sell this monster...
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Jonathan
>>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From tim.reimers at asheville.k12.nc.us  Mon Oct 15 16:47:58 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon, 15 Oct 2007 16:47:58 -0400
Subject: [cisco-voip] Veritas and CCM4.3.2(sr2)/appliance OS v4.3.a SR9
In-Reply-To: <f84a38d30710151324u4ba8b894iaa481136e17b9de7@mail.gmail.com>
References: <20071015102022.97296067075fa8c76087192ed0e74e4e.ce0ff1b39a.wbe@email.secureserver.net>
	<f84a38d30710151324u4ba8b894iaa481136e17b9de7@mail.gmail.com>
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1AF1F@webmail.asheville.k12.nc.us>

That's what we do

 

Push backups to the Unity server, where we do run an Exchange agent to
grab mailboxes.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Monday, October 15, 2007 4:24 PM
To: klong at olotp.org
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Veritas and CCM4.3.2(sr2)/appliance OS v4.3.a
SR9

 

use the BARS built into the Call Manager and upload to network storage
and run veritas from there.

 

I do not believe veritas on Call Manager is supported by TAC.

 

Scott

 

On 10/15/07, klong at olotp.org <klong at olotp.org> wrote: 

Hello,

We will be deploying CCM4.3.2(sr2)/appliance OS v4.3.a SR9.  Our policy
is
to install the Veritas agent on our servers for back up/disaster
recovery. 
Will using Veritas cause problems?  Does anyone have a link to the Cisco
documentation that speaks to any issues or capabilities?

Thanks in advance,

K.A. Long
_______________________________________________ 
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

 

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From jonvoip at gmail.com  Mon Oct 15 17:39:57 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 15 Oct 2007 16:39:57 -0500
Subject: [cisco-voip] Quick Question About Orphaned WAV Files
In-Reply-To: <04B6D8CDD22FD34993C47E33E21F82EFCD772D@rachael-reborn.letnet.net>
References: <421CAA1A5300B84A87D200D70998578263DDFE@DCEX2.DSMO.COM>
	<BD89624EE83BD349BD31A08B453DE24401C682F4@uogexb201.cfs.uoguelph.ca>
	<421CAA1A5300B84A87D200D70998578263DE00@DCEX2.DSMO.COM>
	<04B6D8CDD22FD34993C47E33E21F82EFCD772D@rachael-reborn.letnet.net>
Message-ID: <5d093f9a0710151439t7d0798dra00543ec602e5e83@mail.gmail.com>

Well, they should be in the stream files directory in the backup...
you could just manually add them... or find some newbie to do it.


Jonathan

On 10/15/07, Johnson, Ken <kenjohnson at letu.edu> wrote:
>
>
>
>
> Yes ? I've done that option more than once. The only caveat is once I did it
> stupidly while we had many subscribers temporarily removed for a different
> issue ? that's NOT a good time to do it. I had no way to re-import the wav
> files manually after without going to backups?what a pain J
>
>
>
> Ken Johnson
>     Mgr. Network Services,
>     Information Technology
>     LeTourneau University
>  _________________________________
>   E-mail: kenjohnson at letu.edu
>   Helpdesk: (903) 233-3500
>   Phone: (903) 233-3520
>   Web: http://www.letu.edu/infotech/
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Miller, Steve
>  Sent: Monday, October 15, 2007 12:45 PM
>  To: Rasim Duric
>  Cc: cisco-voip at puck.nether.net
>  Subject: Re: [cisco-voip] Quick Question About Orphaned WAV Files
>
>
>
>
> Thank you!  Does everyone else agree?
>
>
>
>
> Steve Miller
>  Telecom Engineer
>  Dickstein Shapiro LLP
>  1825 Eye Street NW | Washington, DC 20006
>  Tel (202) 420-3370| Fax (202) 330-5607
>  MillerS at dicksteinshapiro.com
>
>
>
>
>
>  ________________________________
>
>
> From: Rasim Duric [mailto:rduric at uoguelph.ca]
>  Sent: Monday, October 15, 2007 1:43 PM
>  To: Miller, Steve
>  Subject: RE: [cisco-voip] Quick Question About Orphaned WAV Files
>
> It shouldn't be a problem. I've been regularly doing these tasks without
> consequences.
>
>
>
> Rasim Duric
>  Network Analyst (CCS)
>  University of Guelph
>  Guelph, N1G 2W1, ON
>  519-824-4120x53146
>  rduric at uoguelph.ca
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Miller, Steve
>  Sent: Monday, October 15, 2007 1:33 PM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] Quick Question About Orphaned WAV Files
>
>
>
>
> TAC tells me that the orphaned wav file errors I see when I run DBWalker can
> be easily fixed with that program and that there should be no negative
> repercussions.  Can I get an "amen" on that before I do it?  He also said
> that I should run DBWalker without correction first, then run with
> correction. Does that also make sense?  Thank you!
>
>
>
>
> Steve Miller
>  Telecom Engineer
>  Dickstein Shapiro LLP
>  1825 Eye Street NW | Washington, DC 20006
>  Tel (202) 420-3370| Fax (202) 330-5607
>  MillerS at dicksteinshapiro.com
>
>
>   --------------------------------------------------------
> This e-mail message and any attached files are confidential
> and are intended solely for the use of the addressee(s)
> named above. This communication may contain material
> protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
> responsible for delivering this confidential communication
> to the intended recipient, you have received this
> communication in error, and any review, use, dissemination,
> forwarding, printing, copying, or other distribution of
> this e-mail message and any attached files is strictly
> prohibited. Dickstein Shapiro reserves the right to monitor
> any communication that is created, received, or sent on its
> network.  If you have received this confidential
> communication in error, please notify the sender
> immediately by reply e-mail message and permanently delete
> the original message.
>
> To reply to our email administrator directly, send an email
> to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
>
> ==============================================================================
> --------------------------------------------------------
> This e-mail message and any attached files are confidential
> and are intended solely for the use of the addressee(s)
> named above. This communication may contain material
> protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
> responsible for delivering this confidential communication
> to the intended recipient, you have received this
> communication in error, and any review, use, dissemination,
> forwarding, printing, copying, or other distribution of
> this e-mail message and any attached files is strictly
> prohibited. Dickstein Shapiro reserves the right to monitor
> any communication that is created, received, or sent on its
> network.  If you have received this confidential
> communication in error, please notify the sender
> immediately by reply e-mail message and permanently delete
> the original message.
>
> To reply to our email administrator directly, send an email
> to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
>
> ==============================================================================
>
>
>  ________________________________
>
>
>
>  Spam
>  Not spam
>  Forget previous vote
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From cshaffer at gmail.com  Mon Oct 15 18:37:06 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Mon, 15 Oct 2007 18:37:06 -0400
Subject: [cisco-voip] Home user
Message-ID: <000601c80f7b$ec317f40$c4947dc0$@com>

I was wondering want everyone out there is using for the situation where you
have someone on your CCM or CCME that has 1 phone at a home office.
Something tells me an ASA is overkill and I haven't found solid information
that any of the 87x routers support tagging QoS of packets going through the
VPN tunnel. We would obviously like to have QoS in place even though it's
not respected at their ISP just to make sure the VPN/Voice packets are
leaving their routers first as a best effort to get some quality. 

 

Thanks

 

 

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From kenjohnson at letu.edu  Mon Oct 15 18:58:14 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Mon, 15 Oct 2007 17:58:14 -0500
Subject: [cisco-voip] Quick Question About Orphaned WAV Files
In-Reply-To: <5d093f9a0710151439t7d0798dra00543ec602e5e83@mail.gmail.com>
References: <421CAA1A5300B84A87D200D70998578263DDFE@DCEX2.DSMO.COM>
	<BD89624EE83BD349BD31A08B453DE24401C682F4@uogexb201.cfs.uoguelph.ca>
	<421CAA1A5300B84A87D200D70998578263DE00@DCEX2.DSMO.COM>
	<04B6D8CDD22FD34993C47E33E21F82EFCD772D@rachael-reborn.letnet.net>
	<5d093f9a0710151439t7d0798dra00543ec602e5e83@mail.gmail.com>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EFCD7886@rachael-reborn.letnet.net>

Yeah - that's what we did = - grabbed them off the backup - and yes, a
student worker came in handy ;-)

Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
 E-mail: kenjohnson at letu.edu
 Helpdesk: (903) 233-3500
 Phone: (903) 233-3520
 Web: http://www.letu.edu/infotech/

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Monday, October 15, 2007 4:40 PM
To: Johnson, Ken
Cc: Miller, Steve; Rasim Duric; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Quick Question About Orphaned WAV Files

Well, they should be in the stream files directory in the backup...
you could just manually add them... or find some newbie to do it.


Jonathan

On 10/15/07, Johnson, Ken <kenjohnson at letu.edu> wrote:
>
>
>
>
> Yes - I've done that option more than once. The only caveat is once I
did it
> stupidly while we had many subscribers temporarily removed for a
different
> issue - that's NOT a good time to do it. I had no way to re-import the
wav
> files manually after without going to backups...what a pain J
>
>
>
> Ken Johnson
>     Mgr. Network Services,
>     Information Technology
>     LeTourneau University
>  _________________________________
>   E-mail: kenjohnson at letu.edu
>   Helpdesk: (903) 233-3500
>   Phone: (903) 233-3520
>   Web: http://www.letu.edu/infotech/
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Miller, Steve
>  Sent: Monday, October 15, 2007 12:45 PM
>  To: Rasim Duric
>  Cc: cisco-voip at puck.nether.net
>  Subject: Re: [cisco-voip] Quick Question About Orphaned WAV Files
>
>
>
>
> Thank you!  Does everyone else agree?
>
>
>
>
> Steve Miller
>  Telecom Engineer
>  Dickstein Shapiro LLP
>  1825 Eye Street NW | Washington, DC 20006
>  Tel (202) 420-3370| Fax (202) 330-5607
>  MillerS at dicksteinshapiro.com
>
>
>
>
>
>  ________________________________
>
>
> From: Rasim Duric [mailto:rduric at uoguelph.ca]
>  Sent: Monday, October 15, 2007 1:43 PM
>  To: Miller, Steve
>  Subject: RE: [cisco-voip] Quick Question About Orphaned WAV Files
>
> It shouldn't be a problem. I've been regularly doing these tasks
without
> consequences.
>
>
>
> Rasim Duric
>  Network Analyst (CCS)
>  University of Guelph
>  Guelph, N1G 2W1, ON
>  519-824-4120x53146
>  rduric at uoguelph.ca
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Miller, Steve
>  Sent: Monday, October 15, 2007 1:33 PM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] Quick Question About Orphaned WAV Files
>
>
>
>
> TAC tells me that the orphaned wav file errors I see when I run
DBWalker can
> be easily fixed with that program and that there should be no negative
> repercussions.  Can I get an "amen" on that before I do it?  He also
said
> that I should run DBWalker without correction first, then run with
> correction. Does that also make sense?  Thank you!
>
>
>
>
> Steve Miller
>  Telecom Engineer
>  Dickstein Shapiro LLP
>  1825 Eye Street NW | Washington, DC 20006
>  Tel (202) 420-3370| Fax (202) 330-5607
>  MillerS at dicksteinshapiro.com
>
>
>   --------------------------------------------------------
> This e-mail message and any attached files are confidential
> and are intended solely for the use of the addressee(s)
> named above. This communication may contain material
> protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
> responsible for delivering this confidential communication
> to the intended recipient, you have received this
> communication in error, and any review, use, dissemination,
> forwarding, printing, copying, or other distribution of
> this e-mail message and any attached files is strictly
> prohibited. Dickstein Shapiro reserves the right to monitor
> any communication that is created, received, or sent on its
> network.  If you have received this confidential
> communication in error, please notify the sender
> immediately by reply e-mail message and permanently delete
> the original message.
>
> To reply to our email administrator directly, send an email
> to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
>
>
========================================================================
======
> --------------------------------------------------------
> This e-mail message and any attached files are confidential
> and are intended solely for the use of the addressee(s)
> named above. This communication may contain material
> protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
> responsible for delivering this confidential communication
> to the intended recipient, you have received this
> communication in error, and any review, use, dissemination,
> forwarding, printing, copying, or other distribution of
> this e-mail message and any attached files is strictly
> prohibited. Dickstein Shapiro reserves the right to monitor
> any communication that is created, received, or sent on its
> network.  If you have received this confidential
> communication in error, please notify the sender
> immediately by reply e-mail message and permanently delete
> the original message.
>
> To reply to our email administrator directly, send an email
> to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
>
>
========================================================================
======
>
>
>  ________________________________
>
>
>
>  Spam
>  Not spam
>  Forget previous vote
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


-- 
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From SCASPER at mtb.com  Mon Oct 15 18:57:57 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Mon, 15 Oct 2007 18:57:57 -0400
Subject: [cisco-voip]   Locations with H323 and IPIPGW
In-Reply-To: <4704BB73.9874.0039.0@mtb.com>
References: <FA233791F398B848B978464507739E8F0946FC@EUBEBRUSVEX1.eu.didata.local>
	<4704BB73.9874.0039.0@mtb.com>
Message-ID: <4713B835.9874.0039.0@mtb.com>

 I am still working on implementing SIP trunking using an IPIPGW and have discovered during testing  that the Call Manager Locations construct is not working to provide Call Admission Control.
 
topology is:
 
 CCM 4.1.3 ----H323------IPIPGW-------SIP------Carrier
 
 All outbound calls are pointed to the Carrier. I thought that since the phones are communicating with Call Manager and the Call Manager knows that inbound and outbound calls are using the associated IP-IP Gateway that this should work. Does this require a Gatekeeper?
 
As always any advice would be appreciated.
 
Regards,

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee.  If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission.  If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy.  This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
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From jgrace at adcapnet.com  Mon Oct 15 19:06:56 2007
From: jgrace at adcapnet.com (James Grace)
Date: Mon, 15 Oct 2007 19:06:56 -0400
Subject: [cisco-voip] callmanager Device license units matrix
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E895@mail3.adcapnetsvc.local>

?
do anybody know where can i find a callmanager Device license units matrix.  we have all types of ip phone and dont have a clue on how many DL to buy for callmanager 6.0
 
James Grace 
 
Systems Engineer
<http://www.adcapnet.com/>  
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
recipient is prohibited. If you received this message in error, please contact the sender and delete the
material from all computers."

This original email was sent to the internet for delivery at 15 Oct 2007 19:07:17 -0400
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From jgrace at adcapnet.com  Mon Oct 15 19:43:23 2007
From: jgrace at adcapnet.com (James Grace)
Date: Mon, 15 Oct 2007 19:43:23 -0400
Subject: [cisco-voip] terminiating SIP to an ASA
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E898@mail3.adcapnetsvc.local>

?
Has anybody out there worked with terminating SIP to an ASA.  An if so can you please give me the gotchas.  
 
James Grace 
 
Systems Engineer
<http://www.adcapnet.com/>  
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
recipient is prohibited. If you received this message in error, please contact the sender and delete the
material from all computers."

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From patrick.diener at gmail.com  Tue Oct 16 00:27:07 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Tue, 16 Oct 2007 06:27:07 +0200
Subject: [cisco-voip] SIP Integration
In-Reply-To: <001b01c80b68$dab88810$3900490a@acg.arlington.local>
References: <001b01c80b68$dab88810$3900490a@acg.arlington.local>
Message-ID: <c0deb5b80710152127r7f13f6c4gcc725a64c52de57f@mail.gmail.com>

the doc you refer to just states that you can not integrate cisco
unity and cme 3.3 using sip. configuring a trunk to a sip provider
should work fine cause sip trunk functionality is rather a function of
ios call routing than cme...

Regards
Patrick

On 10/10/07, Curt Shaffer <cshaffer at gmail.com> wrote:
> I'm reading some docs on Cisco about what the minimum version of Call
> Manager Express I will need to utilize SIP trunks. From what I am seeing it
> states 3.4, 4.0, and 4.1. It states for SIP integration. Here is the page
> I'm referring to:
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/unity/compatibility/matrix/cus
> iptrunkmtx.html
>
>
>
> My VAR is stating that 3.4 does not support SIP trunk functionality.
> Basically I want to establish some SIP trunks for testing from our Call
> Manger Express to SIP providers. Can anyone point to a definitive answer to
> this question for me?
>
> Thanks
>
> Curt
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From zohaibshabir at gmail.com  Tue Oct 16 03:38:04 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Tue, 16 Oct 2007 12:38:04 +0500
Subject: [cisco-voip] terminiating SIP to an ASA
In-Reply-To: <49AFE548BC7F7E49A1A4A2F28AF1914901E898@mail3.adcapnetsvc.local>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E898@mail3.adcapnetsvc.local>
Message-ID: <1ff12970710160038r1a13d46bm9a771b237db73c38@mail.gmail.com>

james
I have once terminated sip on asa 5510 but it didn't work for me. then i
opened a tac case he checked my config and said that there is no problem
with my config, but still sip packets wasn't translating into  private  ip
address. Actually i have public ip address on my outside and private ip
address on my inside network, and asa was not able to translate sip packets
from public ip into private ip whereas h323 was working fine on asa. I
actually see the sip packets on voice gateway 5400 with public ip address,
whereas 5400 was configured to entertain local ip address, so sip packets
were not matching any dial-peer on the gateway. In the end i told my service
provider to give me h323 calls rather than sip.

On 10/16/07, James Grace <jgrace at adcapnet.com> wrote:
>
> ? Has anybody out there worked with terminating SIP to an ASA.  An if so
> can you please give me the gotchas.
>
>  James Grace
>
> Systems Engineer
>  <http://www.adcapnet.com/>
>
> Disclaimer For ADCAP Network Systems
>
> "The information transmitted is intended only for the person or entity to which it is addressed and may
> contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
> or taking of any action in reliance upon this information by persons or entities other than the intended
> recipient is prohibited. If you received this message in error, please contact the sender and delete the
> material from all computers."
>
> This original email was sent to the internet for delivery at 15 Oct 2007 19:44:17 -0400
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Regards,
Zohaib Shabir
Network Engineer(Voice)
DWP Group, TECH Division
Ph:+92-302-8232689
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From rschuknecht at gmx.de  Tue Oct 16 03:40:42 2007
From: rschuknecht at gmx.de (Robert Schuknecht)
Date: Tue, 16 Oct 2007 09:40:42 +0200
Subject: [cisco-voip] Antw:  callmanager Device license units matrix
In-Reply-To: c81cdb3ce393e6fa47259e084d3b54d3
References: c81cdb3ce393e6fa47259e084d3b54d3
Message-ID: <4714871A.5E3C.0004.0@gmx.de>

James,

please have a look at the attachment. The List is from an CUCM 6.01a. I hope that helps.

/Robert

>>> James Grace<jgrace at adcapnet.com> schrieb am Dienstag, 16. Oktober 2007 um 01:06
in Nachricht c81cdb3ce393e6fa47259e084d3b54d3:
> ?
> do anybody know where can i find a callmanager Device license units matrix.  
> we have all types of ip phone and dont have a clue on how many DL to buy for 
> callmanager 6.0
>  
> James Grace 
>  
> Systems Engineer
> <http://www.adcapnet.com/>  
> Disclaimer For ADCAP Network Systems
> 
> "The information transmitted is intended only for the person or entity to 
> which it is addressed and may
> contain confidential and/or privileged material. Any review, retransmission, 
> dissemination or other use of,
> or taking of any action in reliance upon this information by persons or 
> entities other than the intended
> recipient is prohibited. If you received this message in error, please 
> contact the sender and delete the
> material from all computers."
> 
> This original email was sent to the internet for delivery at 15 Oct 2007 
> 19:07:17 -0400
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From ErasmuE4 at telkom.co.za  Tue Oct 16 03:52:25 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Tue, 16 Oct 2007 09:52:25 +0200
Subject: [cisco-voip] Antw:  callmanager Device license units matrix
In-Reply-To: <4714871A.5E3C.0004.0@gmx.de>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01A92557@CNTRRA20-XCS01.telkom.co.za>


When purchasing license units also investigate the new adjunct
licences Cisco now has to make licences more affordable for people who
uses more than just the plastic phone/ (example phone and communicator
and mobile client)



Erik erasmus

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Schuknecht
Sent: Tuesday, October 16, 2007 9:41 AM
To: cisco-voip at puck.nether.net; jgrace at adcapnet.com
Subject: [cisco-voip] Antw: callmanager Device license units matrix

James,

please have a look at the attachment. The List is from an CUCM 6.01a.
I hope that helps.

/Robert

>>> James Grace<jgrace at adcapnet.com> schrieb am Dienstag, 16. Oktober
2007 um 01:06
in Nachricht c81cdb3ce393e6fa47259e084d3b54d3:
> ?
> do anybody know where can i find a callmanager Device license units
matrix.  
> we have all types of ip phone and dont have a clue on how many DL to
buy for 
> callmanager 6.0
>  
> James Grace 
>  
> Systems Engineer
> <http://www.adcapnet.com/>  
> Disclaimer For ADCAP Network Systems
> 
> "The information transmitted is intended only for the person or
entity to 
> which it is addressed and may
> contain confidential and/or privileged material. Any review,
retransmission, 
> dissemination or other use of,
> or taking of any action in reliance upon this information by persons
or 
> entities other than the intended
> recipient is prohibited. If you received this message in error,
please 
> contact the sender and delete the
> material from all computers."
> 
> This original email was sent to the internet for delivery at 15 Oct
2007 
> 19:07:17 -0400

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail is subject to the Telkom SA electronic communication legal notice, available at :
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

From sajishr at ti.com  Tue Oct 16 03:55:29 2007
From: sajishr at ti.com (R, Sajish)
Date: Tue, 16 Oct 2007 13:25:29 +0530
Subject: [cisco-voip] cisco-voip Digest, Vol 56, Issue 43
In-Reply-To: <mailman.777.1191874215.13771.cisco-voip@puck.nether.net>
Message-ID: <91C99CB54D033E469E6D6702BAE57447E6DDA9@dbde01.ent.ti.com>

Hi,

How to upgrade firmware for 7960 phone??/

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
cisco-voip-request at puck.nether.net
Sent: Tuesday, October 09, 2007 1:40 AM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 56, Issue 43

Send cisco-voip mailing list submissions to
	cisco-voip at puck.nether.net

To subscribe or unsubscribe via the World Wide Web, visit
	https://puck.nether.net/mailman/listinfo/cisco-voip
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of cisco-voip digest..."


Today's Topics:

   1. 5.1 Speed Dial Presence (Jason Aarons (US))
   2. Re: 5.1 Speed Dial Presence (Wes Sisk)
   3. Re: 5.1 Speed Dial Presence (Jason Aarons (US))


----------------------------------------------------------------------

Message: 1
Date: Mon, 8 Oct 2007 15:53:55 -0400
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
Subject: [cisco-voip] 5.1 Speed Dial Presence
To: <cisco-voip at puck.nether.net>
Message-ID:
	
<C1FE15183DA37645BC0633BC604E44F00707EA18 at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="us-ascii"

I think what they want is DSS/BLF, does it require a dedicated presence
server or can CCM  5.1 do this natively?

Cisco Unified CallManager Speed Dial Presence 

Cisco Unified CallManager supports the ability for a speed dial to have
presence capabilities via a busy lamp field (BLF) speed dial. BLF speed
dials work as both a speed dial and a presence indicator. Only the
system administrator can configure a BLF speed dial; a system user is
not allowed to configure a BLF speed dial. 

States are: 

Idle 

Busy 

Unknown 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementa
tion_design_guide_chapter09186a008070bc02.html#wp1070507
<javascript:newWin('http://www.cisco.com/en/US/products/sw/voicesw/ps556
/products_implementation_design_guide_chapter09186a008070bc02.html#wp107
0507')> 

 

From: Jason Aarons (US) 
Sent: Monday, October 08, 2007 2:05 PM
To: cisco-voip at puck.nether.net
Subject: 79x0 vs 79x1 (BLF?)

 

Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to
have "auto dial and appearance", they have been using the 7940 a week,
along with CCMUser and don't think AbbrDial/FastDials/Personal Directory
meets their needs. Their old phone had lots of buttons, etc.

What is the CallManager 5.1 feature called where a button can be both a
Shared Line and Speed Dial?  I'm trying to find the URL if it requires a
79x1 or would it work with a 7960.

What are the caveats? Can they setup/manage via CCMUser? 

I've been stuck on 4.1 a long time and recall seeing something about
this in 5.x, but can't find the details.  I know 4.1 you could have
either a shared line or a speed dial but not both.

 -jason

 




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Message: 2
Date: Mon, 08 Oct 2007 16:01:12 -0400
From: Wes Sisk <wsisk at cisco.com>
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence
To: "Jason Aarons (US)" <jason.aarons at us.didata.com>
Cc: cisco-voip at puck.nether.net
Message-ID: <470A8C88.30703 at cisco.com>
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Message: 3
Date: Mon, 8 Oct 2007 16:08:57 -0400
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence
To: "Wes Sisk" <wsisk at cisco.com>
Cc: cisco-voip at puck.nether.net
Message-ID:
	
<C1FE15183DA37645BC0633BC604E44F00707EA77 at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="us-ascii"

Any benefits if they used 79x1 over 7960 for Speed Dial Presence? They
are about to order more 7960s over 7961s due to costs. I wish there was
some motivating  one-reason issue to get customers to buy 79x1s. I know
the benefits of 79x1, and would only buy this model myself, but costs
are sometimes the winner with customers over logic. Can't we end of life
the 7960s and make the 7961 same price!

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Monday, October 08, 2007 4:01 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Speed Dial Presence

 

native.  no CUPS required. /wes

Jason Aarons (US) wrote: 

I think what they want is DSS/BLF, does it require a dedicated presence
server or can CCM  5.1 do this natively?

Cisco Unified CallManager Speed Dial Presence 

Cisco Unified CallManager supports the ability for a speed dial to have
presence capabilities via a busy lamp field (BLF) speed dial. BLF speed
dials work as both a speed dial and a presence indicator. Only the
system administrator can configure a BLF speed dial; a system user is
not allowed to configure a BLF speed dial. 

States are: 

Idle 

Busy 

Unknown 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementa
tion_design_guide_chapter09186a008070bc02.html#wp1070507
<javascript:newWin('http://www.cisco.com/en/US/products/sw/voicesw/ps556
/products_implementation_design_guide_chapter09186a008070bc02.html#wp107
0507')> 

 

From: Jason Aarons (US) 
Sent: Monday, October 08, 2007 2:05 PM
To: cisco-voip at puck.nether.net
Subject: 79x0 vs 79x1 (BLF?)

 

Customer has all SCCP 79x0s with CCM 5.1 and are asking for a button to
have "auto dial and appearance", they have been using the 7940 a week,
along with CCMUser and don't think AbbrDial/FastDials/Personal Directory
meets their needs. Their old phone had lots of buttons, etc.

What is the CallManager 5.1 feature called where a button can be both a
Shared Line and Speed Dial?  I'm trying to find the URL if it requires a
79x1 or would it work with a 7960.

What are the caveats? Can they setup/manage via CCMUser? 

I've been stuck on 4.1 a long time and recall seeing something about
this in 5.x, but can't find the details.  I know 4.1 you could have
either a shared line or a speed dial but not both.

 -jason

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 

 


________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



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End of cisco-voip Digest, Vol 56, Issue 43
******************************************

From olivers at fiu.edu  Tue Oct 16 03:43:23 2007
From: olivers at fiu.edu (Oliver Setien)
Date: Tue, 16 Oct 2007 03:43:23 -0400
Subject: [cisco-voip] Oliver Setien is out of the office.
Message-ID: <OF76D7EC41.AB8E5B99-ON85257376.002A6C94-85257376.002A6C94@fiu.edu>


I will be out of the office starting  10/16/2007 and will not return until
10/17/2007.

I will respond to your message when I return.


From benjamin.kappauf at googlemail.com  Tue Oct 16 05:18:03 2007
From: benjamin.kappauf at googlemail.com (Benjamin Kappauf)
Date: Tue, 16 Oct 2007 11:18:03 +0200
Subject: [cisco-voip] call-manager-fallback wrong phone dn
In-Reply-To: <5d093f9a0710150714u4842cbbcy9d8ff2d4c4cbe110@mail.gmail.com>
References: <feidjf$kkr$1@sea.gmane.org>
	<feom6s$6lq$1@sea.gmane.org>	<5d093f9a0710141525l5ac25ba4y3c7fc876a9ad76c4@mail.gmail.com>	<fevfis$4cp$1@ger.gmane.org>
	<5d093f9a0710150714u4842cbbcy9d8ff2d4c4cbe110@mail.gmail.com>
Message-ID: <ff1vln$ovs$1@ger.gmane.org>

No, but I believe it is the only thing what could help. Erase the 
configuration, Reboot, and reconfigure it from scratch.

But this voice gateway is in use and i am only able to connect to it by 
remote over LAN.



cheers
ben

Jonathan Charles schrieb:
> Did you write mem while in fallback at some point?
> 
> 
> 
> Jonathan
> 
> On 10/15/07, Benjamin Kappauf <benjamin.kappauf at googlemail.com> wrote:
>> hi,
>>
>> As you can see I am using two Callmanagers (Auto registration is off on
>> both of them)
>>
>> A interesting thing is...
>>
>> Callmanagers (Site A)
>> Voice gateway (Site B)
>>
>> Before I started one admin from site A logged his extensionmob account
>> (1700) onto a phone (Phone 1) on site B (changed DP). This phone keeps
>> it number on fallback! Then I thought I did something wrong on the other
>> phones. So I logged this extensionmob account onto another phone (Phone
>> 2) on site B. After fallback the old phone (Phone 1) had again the 1700
>> and the phone 2 had again its 65xx number.
>>
>> So it seems as "something" learned once all phone DNs and on fallback
>> they are getting up again. It can't be the phones; I did a hard reset on
>> them.  After that I did a restart on that voice gateway. So even the
>> voice gateway should now have lost every temporally information about
>> any phone and phone DN.
>>
>>
>> Current configuration : 3483 bytes
>> !
>> version 12.4
>> service timestamps debug datetime msec
>> service timestamps log datetime msec
>> no service password-encryption
>> !
>> hostname x
>> !
>>
>> logging buffered 51200 warnings
>> !
>> no aaa new-model
>> network-clock-participate wic 0
>> network-clock-select 1 T1 0/0/0
>> !
>> !
>> ip cef
>> !
>> !
>> no ip domain lookup
>> ip domain name yourdomain.com
>> multilink bundle-name authenticated
>> !
>> isdn switch-type primary-ni
>> !
>> voice-card 0
>>   dspfarm
>>   dsp services dspfarm
>> !
>> !
>> voice call send-alert
>> voice call convert-discpi-to-prog
>> !
>> voice service voip
>>   allow-connections h323 to h323
>>   supplementary-service h450.12
>>   h323
>> !
>> !
>> voice class codec 1
>>   codec preference 1 g711ulaw
>>   codec preference 2 g711alaw
>>   codec preference 3 g729r8
>> !
>> !
>> !
>> voice class h323 1
>>   h225 timeout tcp establish 3
>>    call start slow
>>    call preserve
>> !
>> !
>> !
>> !
>> !
>> !
>> voice translation-rule 1
>>   rule 1 /^1284\(...\)$/ /4\1/
>> !
>> voice translation-rule 2
>>   rule 3 /\(.+\)/ /9011\1/ type international international
>> !
>> !
>> voice translation-profile Incoming
>>   translate calling 2
>>   translate called 1
>> !
>> !
>> !
>> !
>> !
>> !
>> archive
>>   log config
>>    hidekeys
>> !
>> !
>> controller T1 0/0/0
>>   framing esf
>>   linecode b8zs
>>   pri-group timeslots 1-24
>> !
>> !
>> !
>> !
>> !
>> interface GigabitEthernet0/0
>>   description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$$ES_LAN$
>>   ip address 10.40.2.15 255.255.255.0
>>   duplex auto
>>   speed auto
>>   h323-gateway voip bind srcaddr 10.40.2.15
>> !
>> interface GigabitEthernet0/1
>>   no ip address
>>   shutdown
>>   duplex auto
>>   speed auto
>> !
>> interface Serial0/0/0:23
>>   no ip address
>>   encapsulation hdlc
>>   isdn switch-type primary-ni
>>   isdn timer T310 60000
>>   isdn incoming-voice voice
>>   isdn sending-complete
>>   no cdp enable
>> !
>> ip route 0.0.0.0 0.0.0.0 10.40.2.1
>> !
>> !
>> ip http server
>> ip http authentication local
>> no ip http secure-server
>> ip http timeout-policy idle 60 life 86400 requests 10000
>> !
>> !
>> !
>> !
>> control-plane
>> !
>> !
>> !
>> voice-port 0/0/0:23
>> !
>> !
>> !
>> !
>> !
>> dial-peer voice 10 pots
>>   translation-profile incoming Incoming
>>   incoming called-number .
>>   direct-inward-dial
>>   port 0/0/0:23
>> !
>> dial-peer voice 40 voip
>>   description * Primary CM *
>>   destination-pattern 44..
>>   progress_ind setup enable 3
>>   progress_ind connect enable 8
>>   progress_ind disconnect enable 8
>>   modem passthrough nse codec g711ulaw
>>   voice-class codec 1
>>   voice-class h323 1
>>   session target ipv4:10.22.33.1
>>   dtmf-relay h245-alphanumeric
>>   fax rate disable
>>   no vad
>> !
>> dial-peer voice 50 voip
>>   description * Secondary CM *
>>   translation-profile incoming incoming
>>   preference 1
>>   destination-pattern 44..
>>   progress_ind setup enable 3
>>   progress_ind connect enable 8
>>   progress_ind disconnect enable 8
>>   modem passthrough nse codec g711ulaw
>>   voice-class codec 1
>>   voice-class h323 1
>>   session target ipv4:10.22.33.2
>>   dtmf-relay h245-alphanumeric
>>   fax rate disable
>>   no vad
>> !
>> dial-peer voice 1 pots
>>   destination-pattern 9T
>>   port 0/0/0:23
>> !
>> !
>> !
>> call-manager-fallback
>>   secondary-dialtone 9
>>   max-conferences 8 gain -6
>>   transfer-system full-consult
>>   ip source-address 10.40.2.15 port 2000
>>   system message SRST Fallback
>>   max-ephones 28
>>   max-dn 60 dual-line
>> !
>> banner login ^CSTOP!^C
>> !
>> line con 0
>>   login local
>> line aux 0
>> line vty 0 4
>>   login local
>>   transport input telnet
>> line vty 5 15
>>   login local
>>   transport input telnet
>> !
>> scheduler allocate 20000 1000
>> !
>> end
>>
>>
>>
>> Jonathan Charles schrieb:
>>> Post the entire config... It almost seems like you have a bunch of
>>> ephone-dn's on the router and it is auto-assigning them...
>>>
>>>
>>>
>>> Jonathan
>>>
>>> On 10/12/07, Benjamin Kappauf <benjamin.kappauf at googlemail.com> wrote:
>>>> Benjamin Kappauf schrieb:
>>>>> What is the problem if in case of a fallback all my phones are
>>>>> registered fine to the voice router. But all phones don't get the
>>>>> correct DN. Most of them get a 65xx DN. A few are with ext. mobility
>>>>> activ. And they all have got DNs out of 4440 - 4449
>>>>>
>>>>> Btw. no 65xx number exist on the callmanager.
>>>>>
>>>>> Thanks alot for any help!
>>>>>
>>>>>
>>>>> ----------------------------------
>>>>> call-manager-fallback
>>>>>   secondary-dialtone 9
>>>>>   max-conferences 8 gain -6
>>>>>   transfer-system full-consult
>>>>>   ip source-address x.x.x.x port 2000
>>>>>   max-ephones 40
>>>>>   max-dn 80 dual-line
>>>>>   system message primary SRST Fallback
>>>>>   system message secondary SRST
>>>>>   time-format 24
>>>>>   date-format dd-mm-yy
>>>> Today I downgraded the voice gateway to
>>>> c2800nm-spservicesk9-mz.124-9.T5.bin and I did a reset on all phones.
>>>>
>>>> Afterwards I initialised SRST. But I still get the wrong DNs.
>>>>
>>>> What cause a phone on fallback to get a wrong dn?
>>>>
>>>> Thank you
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>


From lemon at lemon.za.net  Tue Oct 16 08:01:01 2007
From: lemon at lemon.za.net (Lemon)
Date: Tue, 16 Oct 2007 13:01:01 +0100
Subject: [cisco-voip] RAID-Controller Firmware and RAID BIOS on IBM Servers
	Upgrade or Downgrade Procedure
Message-ID: <94e031d30710160501g294bbe46ibc27146a0af0b078@mail.gmail.com>

Trying to do:

http://www.ciscosystems.or.at/en/US/products/ps6509/products_tech_note09186a008059a81d.shtml

Is there another way of doing this without having a 3-1/2" 1.44 MB
diskette? I mean the server does not even have a drive for this.

Thnx

Charl

From baryonyx5 at gmail.com  Tue Oct 16 08:42:48 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Tue, 16 Oct 2007 07:42:48 -0500
Subject: [cisco-voip] IPCC 4.0 Create XML Document access denied
Message-ID: <a749d0620710160542k357a9178j9bc87cfbf0e85446@mail.gmail.com>

I did an upgrade from IPCC 3.5.4 to 4.0.5 last night.  Many of the existing
scripts call a subflow called schedule.aef.  this script checks an xml file
on the C drive for holidays, open/closed hours.  It worked perfect in
3.5but after the upgrade I am getting an access denied error in the
mivr logs
and in the script editor while debugging.  If I run filemon while debubbing
the script I can see the user CRSAdministrator  tries to open the xml file
but is denied.  My file permissions are correct.  I did notice that
CRSAdministrator had the deny local logon right assigned in the local sec
policy.  I removed that and rebooted the server but am still getting the
same error.  There is currently no CSA running.

Does anyone have any ideas on how this is supposed to work in CRS 4?  It
seems strange that the account used to run the scripts would be denied local
file access but I can't find anything in the documentation about it.

thanks

-- 
kris seraphine
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From craig at staffin.org  Tue Oct 16 08:50:12 2007
From: craig at staffin.org (Craig Staffin)
Date: Tue, 16 Oct 2007 07:50:12 -0500
Subject: [cisco-voip] Mobility 6.0.1a
Message-ID: <9b76d88c0710160550t3c4d83date31c05a96c01614f@mail.gmail.com>

Strange issue.

So I have configured Mobility on our internal system and gotten it working
correctly.

However I now have two clients where when I configure it CM never places the
call out to the cell phone I never even see it on the gateway.

There are no ACL's in play on the mobility profile.

Has anyone had issues getting this to work before?

-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003
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From jsteinberg at gmail.com  Tue Oct 16 09:12:17 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 16 Oct 2007 09:12:17 -0400
Subject: [cisco-voip] Mobility 6.0.1a
In-Reply-To: <9b76d88c0710160550t3c4d83date31c05a96c01614f@mail.gmail.com>
References: <9b76d88c0710160550t3c4d83date31c05a96c01614f@mail.gmail.com>
Message-ID: <f1b65f880710160612x2c8d3512h82f13e6346a9f6e0@mail.gmail.com>

Have you enabled mobility via the checkbox on the end user's config page?
Have you configured the rerouting CSS on the user's RDP with the
proper partitions to route calls to the gateway?
Does the remote destination number have your external access digits
(91, for example) or are you prefixing this info using the app dial
rules?

Justin


On 10/16/07, Craig Staffin <craig at staffin.org> wrote:
> Strange issue.
>
> So I have configured Mobility on our internal system and gotten it working
> correctly.
>
> However I now have two clients where when I configure it CM never places the
> call out to the cell phone I never even see it on the gateway.
>
> There are no ACL's in play on the mobility profile.
>
> Has anyone had issues getting this to work before?
>
> --
> Craig Staffin
> Craig at staffin.org
> (H) 262-437-7313
> (C) 262-613-6003
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From MLinsemier at apcapital.com  Tue Oct 16 09:16:57 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 16 Oct 2007 09:16:57 -0400
Subject: [cisco-voip] Home user
In-Reply-To: <000601c80f7b$ec317f40$c4947dc0$@com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
Message-ID: <AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>

We currently have about 40 production remote home teleworkers that have
been deployed using Cisco 871/877 wireless routers and a 7960 phones.
We are using a Cisco 3845 series router at the head-end so that we can
control QoS tagging on the egress / ingress points of both sides of the
VPN tunnel.  We are using a phase 2 DMVPN solution dual-homed to two
sites to provide secure redundant connectivity.

 

It took me a bit to tweak my router configurations (I started on Cisco
831/837 routers) to get the results that we wanted, but all and all our
users are happy.  There is the occasional jitter and packet loss (it is
the Internet mind you) but g.729 is working quite well coupled with
business cable and DSL services.

 

If you have any other questions, feel free to ask.

 

Matt

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Curt Shaffer
Sent: Monday, October 15, 2007 6:37 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Home user

 

I was wondering want everyone out there is using for the situation where
you have someone on your CCM or CCME that has 1 phone at a home office.
Something tells me an ASA is overkill and I haven't found solid
information that any of the 87x routers support tagging QoS of packets
going through the VPN tunnel. We would obviously like to have QoS in
place even though it's not respected at their ISP just to make sure the
VPN/Voice packets are leaving their routers first as a best effort to
get some quality. 

 

Thanks

 

 


CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may
be protected by one or more legal privileges. It is intended
solely for the use of the addressee identified above. If you
are not the intended recipient, any use, disclosure, copying
or distribution of this communication is UNAUTHORIZED. Neither
this information block, the typed name of the sender, nor
anything else in this message is intended to constitute an
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contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete
this communication from your computer. Thank you.
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From cheesevoice at gmail.com  Tue Oct 16 09:32:45 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Tue, 16 Oct 2007 08:32:45 -0500
Subject: [cisco-voip] from the Newbie
Message-ID: <84b74aec0710160632m3f882d42n1b39a60b33f1cb39@mail.gmail.com>

So I am curious bynature, and just starting my CCVP training.  So some of my
questions will fall in the "WTF" category...

For example.  I am on the CCM (4.2) Publisher (we have a pub and one sub)
and I thought it would be cool to look at my phones config file.

So I went to C: Program Files>Cisco>TFTPPath (where the load files and bin
files are all at) butI saw no SEP(MAC).cnf.xml files.  So i thought maybe I
am not looking in the right spot, and I searched the hard drives for
.cnf.xml  and I found nothing.

If you could be so kind as to point me in the right direction...
Yes I SHOULD be reading my CVOICE books but this is bugging the crap out of
me.

Thanks guys

Kevin
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From craig at staffin.org  Tue Oct 16 09:36:26 2007
From: craig at staffin.org (Craig Staffin)
Date: Tue, 16 Oct 2007 08:36:26 -0500
Subject: [cisco-voip] Mobility 6.0.1a
In-Reply-To: <f1b65f880710160612x2c8d3512h82f13e6346a9f6e0@mail.gmail.com>
References: <9b76d88c0710160550t3c4d83date31c05a96c01614f@mail.gmail.com>
	<f1b65f880710160612x2c8d3512h82f13e6346a9f6e0@mail.gmail.com>
Message-ID: <9b76d88c0710160636w438c43fclf694b6be2b57bd3a@mail.gmail.com>

Justin,

Right as I got your email I figured it out.  Its the rerouting CSS

Wes or Ryan maybe you can answer me what the CSS on the remote destination
profile is for????  Considering that the rerouting CSS is not mandatory this
kind of seems like an over sight.

Craig

On 10/16/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> Have you enabled mobility via the checkbox on the end user's config page?
> Have you configured the rerouting CSS on the user's RDP with the
> proper partitions to route calls to the gateway?
> Does the remote destination number have your external access digits
> (91, for example) or are you prefixing this info using the app dial
> rules?
>
> Justin
>
>
> On 10/16/07, Craig Staffin <craig at staffin.org> wrote:
> > Strange issue.
> >
> > So I have configured Mobility on our internal system and gotten it
> working
> > correctly.
> >
> > However I now have two clients where when I configure it CM never places
> the
> > call out to the cell phone I never even see it on the gateway.
> >
> > There are no ACL's in play on the mobility profile.
> >
> > Has anyone had issues getting this to work before?
> >
> > --
> > Craig Staffin
> > Craig at staffin.org
> > (H) 262-437-7313
> > (C) 262-613-6003
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>



-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003
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From craig at staffin.org  Tue Oct 16 09:38:06 2007
From: craig at staffin.org (Craig Staffin)
Date: Tue, 16 Oct 2007 08:38:06 -0500
Subject: [cisco-voip] from the Newbie
In-Reply-To: <84b74aec0710160632m3f882d42n1b39a60b33f1cb39@mail.gmail.com>
References: <84b74aec0710160632m3f882d42n1b39a60b33f1cb39@mail.gmail.com>
Message-ID: <9b76d88c0710160638g4a132beam6770e6616ddc89d3@mail.gmail.com>

These are stored in memory to make the TFTP server faster.

THere is a service param that states something like "write TFTP files to
Memory" Default is true set this to false

Craig

On 10/16/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
>
> So I am curious bynature, and just starting my CCVP training.  So some of
> my questions will fall in the "WTF" category...
>
> For example.  I am on the CCM (4.2) Publisher (we have a pub and one sub)
> and I thought it would be cool to look at my phones config file.
>
> So I went to C: Program Files>Cisco>TFTPPath (where the load files and bin
> files are all at) butI saw no SEP(MAC).cnf.xml files.  So i thought maybe I
> am not looking in the right spot, and I searched the hard drives for
> .cnf.xml  and I found nothing.
>
> If you could be so kind as to point me in the right direction...
> Yes I SHOULD be reading my CVOICE books but this is bugging the crap out
> of me.
>
> Thanks guys
>
> Kevin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003
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From jgrace at adcapnet.com  Tue Oct 16 09:41:06 2007
From: jgrace at adcapnet.com (James Grace)
Date: Tue, 16 Oct 2007 09:41:06 -0400
Subject: [cisco-voip] CM 6.0 unable to Cfwdall
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E89C@mail3.adcapnetsvc.local>

?
for some reason im not able to hit the  Cfwdall button in CM 6.0.  I can dial any pattern i want to, but when i press cfwdall its not allowing any digits.  It give me "your call can not be completed.  Never ran into this on previous ver.  
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
recipient is prohibited. If you received this message in error, please contact the sender and delete the
material from all computers."

This original email was sent to the internet for delivery at 16 Oct 2007 09:39:53 -0400
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From pwalenta at wi.rr.com  Tue Oct 16 09:41:38 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 16 Oct 2007 08:41:38 -0500
Subject: [cisco-voip] from the Newbie
In-Reply-To: <84b74aec0710160632m3f882d42n1b39a60b33f1cb39@mail.gmail.com>
References: <84b74aec0710160632m3f882d42n1b39a60b33f1cb39@mail.gmail.com>
Message-ID: <00b001c80ffa$470faf30$0232a8c0@walentawk1>

By default CCM caches all config files in memory.  There's an option (I
think it's under services for tftp) that forces CCM to write out the files.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kevin Dunn
Sent: Tuesday, October 16, 2007 8:33 AM
To: Cisco Voice
Subject: [cisco-voip] from the Newbie


So I am curious bynature, and just starting my CCVP training.  So some of my
questions will fall in the "WTF" category...
 
For example.  I am on the CCM (4.2) Publisher (we have a pub and one sub)
and I thought it would be cool to look at my phones config file.
 
So I went to C: Program Files>Cisco>TFTPPath (where the load files and bin
files are all at) butI saw no SEP(MAC).cnf.xml files.  So i thought maybe I
am not looking in the right spot, and I searched the hard drives for
.cnf.xml  and I found nothing.  
 
If you could be so kind as to point me in the right direction...
Yes I SHOULD be reading my CVOICE books but this is bugging the crap out of
me.
 
Thanks guys
 
Kevin
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From jsteinberg at gmail.com  Tue Oct 16 09:44:42 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 16 Oct 2007 09:44:42 -0400
Subject: [cisco-voip] Mobility 6.0.1a
In-Reply-To: <9b76d88c0710160636w438c43fclf694b6be2b57bd3a@mail.gmail.com>
References: <9b76d88c0710160550t3c4d83date31c05a96c01614f@mail.gmail.com>
	<f1b65f880710160612x2c8d3512h82f13e6346a9f6e0@mail.gmail.com>
	<9b76d88c0710160636w438c43fclf694b6be2b57bd3a@mail.gmail.com>
Message-ID: <f1b65f880710160644j6ca09db7rc82c9d4b6d62f38f@mail.gmail.com>

RDP settings (as I understand it, and I could be wrong):

CSS - used for routing 'inbound' calls placed from PSTN via mobile voice access
Rerouting CSS - used for routing 'outbound' calls via mobile connnect

This is tricky, b/c the RDP CSS is used for routing inbound calls via
mobile voice access - not the physical gateway CSS.  IT is kinda cool
b/c if you call in via your cell through MVA and you call someone
on-net, it is supposed to look to them like you are calling from
on-net/your deskphone.

I haven't yet testing MVA with CM6 though.

Justin

On 10/16/07, Craig Staffin <craig at staffin.org> wrote:
> Justin,
>
> Right as I got your email I figured it out.  Its the rerouting CSS
>
> Wes or Ryan maybe you can answer me what the CSS on the remote destination
> profile is for????  Considering that the rerouting CSS is not mandatory this
> kind of seems like an over sight.
>
> Craig
>
>
> On 10/16/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
> > Have you enabled mobility via the checkbox on the end user's config page?
> > Have you configured the rerouting CSS on the user's RDP with the
> > proper partitions to route calls to the gateway?
> > Does the remote destination number have your external access digits
> > (91, for example) or are you prefixing this info using the app dial
> > rules?
> >
> > Justin
> >
> >
> > On 10/16/07, Craig Staffin <craig at staffin.org> wrote:
> > > Strange issue.
> > >
> > > So I have configured Mobility on our internal system and gotten it
> working
> > > correctly.
> > >
> > > However I now have two clients where when I configure it CM never places
> the
> > > call out to the cell phone I never even see it on the gateway.
> > >
> > > There are no ACL's in play on the mobility profile.
> > >
> > > Has anyone had issues getting this to work before?
> > >
> > > --
> > > Craig Staffin
> > > Craig at staffin.org
> > > (H) 262-437-7313
> > > (C) 262-613-6003
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
>
>
>
> --
>
> Craig Staffin
>  Craig at staffin.org
> (H) 262-437-7313
> (C) 262-613-6003

From Matt.Slaga at us.didata.com  Tue Oct 16 09:58:35 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 16 Oct 2007 09:58:35 -0400
Subject: [cisco-voip] CM 6.0 unable to Cfwdall
In-Reply-To: <49AFE548BC7F7E49A1A4A2F28AF1914901E89C@mail3.adcapnetsvc.local>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E89C@mail3.adcapnetsvc.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0071B68B8@USNAEXCH.na.didata.local>

Check your CFA CSS.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, October 16, 2007 9:41 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CM 6.0 unable to Cfwdall

 

?

for some reason im not able to hit the  Cfwdall button in CM 6.0.  I can
dial any pattern i want to, but when i press cfwdall its not allowing
any digits.  It give me "your call can not be completed.  Never ran into
this on previous ver.  

 

James Grace 

 

Systems Engineer

Adcap Network Systems, Inc.

6865 Shiloh Road East, Suite 250

Alpharetta, GA 30005

 

office: 678-456-6986

direct: 678-456-6761

jgrace at adcapnet.com

www.adcapnet.com <http://www.adcapnet.com/> 

Disclaimer For ADCAP Network Systems
 
"The information transmitted is intended only for the person or entity
to which it is addressed and may
contain confidential and/or privileged material. Any review,
retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or
entities other than the intended
recipient is prohibited. If you received this message in error, please
contact the sender and delete the
material from all computers."
 
This original email was sent to the internet for delivery at 16 Oct 2007
09:39:53 -0400



-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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From ealeatherman at gmail.com  Tue Oct 16 10:20:20 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 16 Oct 2007 10:20:20 -0400
Subject: [cisco-voip] from the Newbie
In-Reply-To: <84b74aec0710160632m3f882d42n1b39a60b33f1cb39@mail.gmail.com>
References: <84b74aec0710160632m3f882d42n1b39a60b33f1cb39@mail.gmail.com>
Message-ID: <94a1afde0710160720s5b8c40b3ke7f706878c584caf@mail.gmail.com>

If you just want to look at a config, i think you could use a TFTP client
issue a reguest for one as if you were a phone, rather than changing any
params in CCM.

example in windows:
C:\>tftp 10.0.0.2 GET SEP000F3487EA29.cnf.xml


On 10/16/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
>
> So I am curious bynature, and just starting my CCVP training.  So some of
> my questions will fall in the "WTF" category...
>
> For example.  I am on the CCM (4.2) Publisher (we have a pub and one sub)
> and I thought it would be cool to look at my phones config file.
>
> So I went to C: Program Files>Cisco>TFTPPath (where the load files and bin
> files are all at) butI saw no SEP(MAC).cnf.xml files.  So i thought maybe I
> am not looking in the right spot, and I searched the hard drives for
> .cnf.xml  and I found nothing.
>
> If you could be so kind as to point me in the right direction...
> Yes I SHOULD be reading my CVOICE books but this is bugging the crap out
> of me.
>
> Thanks guys
>
> Kevin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From kenjohnson at letu.edu  Tue Oct 16 10:56:05 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Tue, 16 Oct 2007 09:56:05 -0500
Subject: [cisco-voip] Home user
In-Reply-To: <000601c80f7b$ec317f40$c4947dc0$@com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EFCD795B@rachael-reborn.letnet.net>

Couldn't say but we've used 3002 VPN hardware clients for years and more
recently 871 routers without any problems whatsoever. Mostly 7970's on
the other end.

 

Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
 E-mail: kenjohnson at letu.edu
 Helpdesk: (903) 233-3500
 Phone: (903) 233-3520
 Web: http://www.letu.edu/infotech/
<BLOCKED::http://www.letu.edu/infotech/> 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Curt Shaffer
Sent: Monday, October 15, 2007 5:37 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Home user

 

I was wondering want everyone out there is using for the situation where
you have someone on your CCM or CCME that has 1 phone at a home office.
Something tells me an ASA is overkill and I haven't found solid
information that any of the 87x routers support tagging QoS of packets
going through the VPN tunnel. We would obviously like to have QoS in
place even though it's not respected at their ISP just to make sure the
VPN/Voice packets are leaving their routers first as a best effort to
get some quality. 

 

Thanks

 

 

________________________________


Spam <http://antispam.letu.edu/canit/b.php?i=8121476&m=f46c512813c5&c=s>

Not spam
<http://antispam.letu.edu/canit/b.php?i=8121476&m=f46c512813c5&c=n> 
Forget previous vote
<http://antispam.letu.edu/canit/b.php?i=8121476&m=f46c512813c5&c=f> 

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From Brandon.Ogden at amersports.com  Tue Oct 16 11:04:52 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Tue, 16 Oct 2007 08:04:52 -0700
Subject: [cisco-voip] Subscriber Rebuild
In-Reply-To: <04B6D8CDD22FD34993C47E33E21F82EFCD795B@rachael-reborn.letnet.net>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<04B6D8CDD22FD34993C47E33E21F82EFCD795B@rachael-reborn.letnet.net>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D0A180@seas7633.am.amersports.int>

I just rebuilt my publisher this weekend.. Everything has been running
well, so I am getting ready to rebuild the subscriber (which has been
offline with the old config as a reference and fall back) in the next
day or two.  Does it come on line pretty seamlessly?  Is there anything
I need to know... our Telecom contractor told me it was pretty smooth
and simply install, but he has been know to lie.. LOL  Thanks all!  

 

BTW.. its 5.1

 

Brandon Ogden

Amer Sports 

801-624-7529

www.amersports.com

http://www.linkedin.com/in/brandonogden

________________________________

 

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From svoll.voip at gmail.com  Tue Oct 16 11:55:39 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 16 Oct 2007 08:55:39 -0700
Subject: [cisco-voip] Home user
In-Reply-To: <000601c80f7b$ec317f40$c4947dc0$@com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
Message-ID: <f84a38d30710160855p8e84cfg570aead9a09b0769@mail.gmail.com>

We have a combanation of 877, ASA5505, and more recently and will continue
forward, Cisco's Phone Proxy.  No VPN neccessary!!  But the nice part about
the ASA is that it has two PoE ports.  but I'm not much for troubleshooting
VPN Lan2Lan connections so the Phone proxy is nice.  I guess it's all
personal preference.  Remimber to put a sticker on the phones not to dial
911 or have CER that can take care of it.

Scott


On 10/15/07, Curt Shaffer <cshaffer at gmail.com> wrote:
>
>  I was wondering want everyone out there is using for the situation where
> you have someone on your CCM or CCME that has 1 phone at a home office.
> Something tells me an ASA is overkill and I haven't found solid information
> that any of the 87x routers support tagging QoS of packets going through the
> VPN tunnel. We would obviously like to have QoS in place even though it's
> not respected at their ISP just to make sure the VPN/Voice packets are
> leaving their routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jsteinberg at gmail.com  Tue Oct 16 12:17:02 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 16 Oct 2007 12:17:02 -0400
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
Message-ID: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com>

Has anyone been successful with this?

I am trying to get this to work but can't figure out what the 'Trust
Certificate Subject CN' field is for on the presence gateway page when
integrating to outlook.

thanks.
justin

From jonvoip at gmail.com  Tue Oct 16 13:13:12 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 12:13:12 -0500
Subject: [cisco-voip] Triple Unity servers...
In-Reply-To: <4713D352.1000303@wcyv.com>
References: <5d093f9a0709190931h28adab0boec9034c1fdab9660@mail.gmail.com>
	<5d093f9a0710142154w43a9f31cqa0e2b30bdfdae838@mail.gmail.com>
	<d4c697ad0710142333g3dbebd24g26934f58becd9547@mail.gmail.com>
	<5d093f9a0710142348g557b30e8h2f63af13e1425790@mail.gmail.com>
	<5d093f9a0710151232h35a08a1fg58b782ab738f6d1f@mail.gmail.com>
	<4713D352.1000303@wcyv.com>
Message-ID: <5d093f9a0710161013n6afdbcaesff75a0459e51dded@mail.gmail.com>

The weird question is, why would you want/need TTS in Unity
Connections... if it is voicemail only?


Jonathan

On 10/15/07, Pat Hayes <pat-cv at wcyv.com> wrote:
> They're correct, there is no such thing in the Unity world at the
> moment. I believe you (or whoever ordered this) is thinking of the Voice
> Recognition server, new to 5.0. In order to get voice recognition
> features, you have to use another server. Installation procedure here:
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/installation/guide/umexfo/5xcuigumefoappc.html
>
> -------- Original Message  --------
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "Aman Chugh" <aman.chugh at gmail.com>
> Cc: Cisco Voip Mailing list <cisco-voip at puck.nether.net>
> Subject: [cisco-voip] Triple Unity servers...
> Date: 10/15/2007 3:32 PM
>
> > OK, I opened a TAC case and here is the response:
> >
> >
> > "Unfortunately, you cannot install TTS server on a dedicated hardware
> > box, even on CU ver 5.0. Only Cisco Unity Connection allows you to do
> > that."
> >
> >
> > Jonathan
> >
> > On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >> Well, those are the questions:
> >>
> >> A: How is this done, if it can be done at all.
> >>
> >> B: Would TAC support afterwards.
> >>
> >>
> >> Jonathan
> >>
> >> On 10/15/07, Aman Chugh <aman.chugh at gmail.com> wrote:
> >>> Not Sure if this would be a supported TAC deployement.
> >>>
> >>> I am thinking that you can run the unity setup and enable TTS setting only
> >>> on the third box but It would be interesting to learn if three unity servers
> >>> can exist toegther in the failover setup.
> >>>
> >>> Do let us know what happens next !
> >>>
> >>> Aman
> >>>
> >>>
> >>> On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >>>> So, I have to do this on Tuesday, and was wondering if anyone had any
> >>> ideas...
> >>>> The plan is to install the primary and secondary Unity servers without
> >>>> TTS and then run the Failoverconfig.exe to mate them together...
> >>>> Pretty simple (I hope).
> >>>>
> >>>> The question is about this third Unity server, the dedicated TTS
> >>>> box... how do I link that to the Unity twins?
> >>>>
> >>>> Not finding a lot of love on Cisco's website here... (no mention
> >>>> anywhere of the concept of a dedicated TTS box)...
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> Jonathan
> >>>>
> >>>> On 9/19/07, Jonathan Charles < jonvoip at gmail.com> wrote:
> >>>>> Cust ordered 3 Unity servers, a  primary, a secondary and a dedicated
> >>> TTS box.
> >>>>> Not sure how to install the TTS box. Do I set it up normally (SQL,
> >>>>> Exchange tools, etc.) and then install Unity but only select TTS (and
> >>>>> not select Unity)?
> >>>>>
> >>>>> Also, would I NOT select TTS on the two primaries?
> >>>>>
> >>>>> Just curious, I didn't sell this monster...
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> Jonathan
> >>>>>
> >>>> _______________________________________________
> >>>> cisco-voip mailing list
> >>>> cisco-voip at puck.nether.net
> >>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>>
> >>>
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Tue Oct 16 13:17:15 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 12:17:15 -0500
Subject: [cisco-voip] CM 6.0 unable to Cfwdall
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0071B68B8@USNAEXCH.na.didata.local>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E89C@mail3.adcapnetsvc.local>
	<C1FE15183DA37645BC0633BC604E44F0071B68B8@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0710161017w23b1a3c8r6460fdb138fd27e5@mail.gmail.com>

Well, yes, it is most likely a CSS issue for CFA...

However, before changing the CFA CSS to LD or whatever, make sure you
want to let people do that, because there is a great potential for
toll fraud.

For example, employee has CFA CSS set to everywhere.

He CFA's his phone to Maddrass, goes home and calls his desk phone to
talk to his parents in India... he paid for a local call, you paid for
a call to India.

I would recommend that you create a CFA CSS to include just Voicemail
and local calling.

Maybe include your local area codes (so people can CFA to their
cells), but don't go crazy until you are sure you want to.


Jonathan

On 10/16/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
>
>
>
>
> Check your CFA CSS.
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> James Grace
>  Sent: Tuesday, October 16, 2007 9:41 AM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] CM 6.0 unable to Cfwdall
>
>
>
>
> ?
>
>
>
> for some reason im not able to hit the  Cfwdall button in CM 6.0.  I can
> dial any pattern i want to, but when i press cfwdall its not allowing any
> digits.  It give me "your call can not be completed.  Never ran into this on
> previous ver.
>
>
>
>
>
>
>
> James Grace
>
>
>
>
>
> Systems Engineer
>
>
> Adcap Network Systems, Inc.
>
>
> 6865 Shiloh Road East, Suite 250
>
>
> Alpharetta, GA 30005
>
>
>
>
>
> office: 678-456-6986
>
>
> direct: 678-456-6761
>
>
> jgrace at adcapnet.com
>
>
> www.adcapnet.com Disclaimer For ADCAP Network Systems
>
> "The information transmitted is intended only for the person or entity to
> which it is addressed and may
> contain confidential and/or privileged material. Any review, retransmission,
> dissemination or other use of,
> or taking of any action in reliance upon this information by persons or
> entities other than the intended
> recipient is prohibited. If you received this message in error, please
> contact the sender and delete the
> material from all computers."
>
> This original email was sent to the internet for delivery at 16 Oct 2007
> 09:39:53 -0400
>
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From Matt.Slaga at us.didata.com  Tue Oct 16 13:46:54 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 16 Oct 2007 13:46:54 -0400
Subject: [cisco-voip] CM 6.0 unable to Cfwdall
In-Reply-To: <5d093f9a0710161017w23b1a3c8r6460fdb138fd27e5@mail.gmail.com>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E89C@mail3.adcapnetsvc.local>
	<C1FE15183DA37645BC0633BC604E44F0071B68B8@USNAEXCH.na.didata.local>
	<5d093f9a0710161017w23b1a3c8r6460fdb138fd27e5@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0071FF3DB@USNAEXCH.na.didata.local>

That is why departmental bill-back is such a great thing.  Managers that
see this type of expense will squash it quicker than any IT guy could!

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Tuesday, October 16, 2007 1:17 PM
To: Matt Slaga (US)
Cc: James Grace; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 6.0 unable to Cfwdall

Well, yes, it is most likely a CSS issue for CFA...

However, before changing the CFA CSS to LD or whatever, make sure you
want to let people do that, because there is a great potential for
toll fraud.

For example, employee has CFA CSS set to everywhere.

He CFA's his phone to Maddrass, goes home and calls his desk phone to
talk to his parents in India... he paid for a local call, you paid for
a call to India.

I would recommend that you create a CFA CSS to include just Voicemail
and local calling.

Maybe include your local area codes (so people can CFA to their
cells), but don't go crazy until you are sure you want to.


Jonathan

On 10/16/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
>
>
>
>
> Check your CFA CSS.
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> James Grace
>  Sent: Tuesday, October 16, 2007 9:41 AM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] CM 6.0 unable to Cfwdall
>
>
>
>
> ?
>
>
>
> for some reason im not able to hit the  Cfwdall button in CM 6.0.  I
can
> dial any pattern i want to, but when i press cfwdall its not allowing
any
> digits.  It give me "your call can not be completed.  Never ran into
this on
> previous ver.
>
>
>
>
>
>
>
> James Grace
>
>
>
>
>
> Systems Engineer
>
>
> Adcap Network Systems, Inc.
>
>
> 6865 Shiloh Road East, Suite 250
>
>
> Alpharetta, GA 30005
>
>
>
>
>
> office: 678-456-6986
>
>
> direct: 678-456-6761
>
>
> jgrace at adcapnet.com
>
>
> www.adcapnet.com Disclaimer For ADCAP Network Systems
>
> "The information transmitted is intended only for the person or entity
to
> which it is addressed and may
> contain confidential and/or privileged material. Any review,
retransmission,
> dissemination or other use of,
> or taking of any action in reliance upon this information by persons
or
> entities other than the intended
> recipient is prohibited. If you received this message in error, please
> contact the sender and delete the
> material from all computers."
>
> This original email was sent to the internet for delivery at 16 Oct
2007
> 09:39:53 -0400
>
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
designated
> addressee(s) named above only. If you are not the intended addressee,
you
> are hereby notified that you have received this communication in error
and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this
communication in
> error, please notify us immediately by replying to this message and
deleting
> it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From cheesevoice at gmail.com  Tue Oct 16 14:04:31 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Tue, 16 Oct 2007 13:04:31 -0500
Subject: [cisco-voip] from the Newbie
In-Reply-To: <94a1afde0710160720s5b8c40b3ke7f706878c584caf@mail.gmail.com>
References: <84b74aec0710160632m3f882d42n1b39a60b33f1cb39@mail.gmail.com>
	<94a1afde0710160720s5b8c40b3ke7f706878c584caf@mail.gmail.com>
Message-ID: <84b74aec0710161104p2d011b82m12ffc5253671e70d@mail.gmail.com>

perfect!

thanks so much guys all of those parameters were as you said, however I
followed Ed's advice and just tftp's the file to my harddrive and it worked
great.

You guys are awesome and thanks so much

Kevin


On 10/16/07, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> If you just want to look at a config, i think you could use a TFTP client
> issue a reguest for one as if you were a phone, rather than changing any
> params in CCM.
>
> example in windows:
> C:\>tftp 10.0.0.2 GET SEP000F3487EA29.cnf.xml
>
>
> On 10/16/07, Kevin Dunn <cheesevoice at gmail.com> wrote:
> >
> >  So I am curious bynature, and just starting my CCVP training.  So some
> > of my questions will fall in the "WTF" category...
> >
> > For example.  I am on the CCM (4.2) Publisher (we have a pub and one
> > sub) and I thought it would be cool to look at my phones config file.
> >
> > So I went to C: Program Files>Cisco>TFTPPath (where the load files and
> > bin files are all at) butI saw no SEP(MAC).cnf.xml files.  So i thought
> > maybe I am not looking in the right spot, and I searched the hard drives for
> > .cnf.xml  and I found nothing.
> >
> > If you could be so kind as to point me in the right direction...
> > Yes I SHOULD be reading my CVOICE books but this is bugging the crap out
> > of me.
> >
> > Thanks guys
> >
> > Kevin
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
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From cheesevoice at gmail.com  Tue Oct 16 14:12:15 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Tue, 16 Oct 2007 13:12:15 -0500
Subject: [cisco-voip] Subscriber Rebuild
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D0A180@seas7633.am.amersports.int>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<04B6D8CDD22FD34993C47E33E21F82EFCD795B@rachael-reborn.letnet.net>
	<D646304AFC55DB4EBB92329BCC800A00D0A180@seas7633.am.amersports.int>
Message-ID: <84b74aec0710161112id446d7fyedcc6628ad53cdc6@mail.gmail.com>

I am going to preface most of my replies with "Newbie" so you guys know my
experience level is low... that being said

you already have a publisher in your cluster.  You are going to add a
subscriber to the cluster, that alone isn't much ofan issue, but the phones
will need to register with the subscriber as the publisher houses the actual
up to date database, and the subscriber handles the "day-to-day"
operations... so the phones will need to re-register, but the configuration
process shouldn't be lengthy.

(someone else should confirm my answers)


On 10/16/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>  I just rebuilt my publisher this weekend.. Everything has been running
> well, so I am getting ready to rebuild the subscriber (which has been
> offline with the old config as a reference and fall back) in the next day or
> two.  Does it come on line pretty seamlessly?  Is there anything I need to
> know? our Telecom contractor told me it was pretty smooth and simply
> install, but he has been know to lie.. LOL  Thanks all!
>
>
>
> BTW.. its 5.1
>
>
>
> Brandon Ogden
>
> Amer Sports
>
> 801-624-7529
>
> www.amersports.com
>
> http://www.linkedin.com/in/brandonogden
>  ------------------------------
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Brandon.Ogden at amersports.com  Tue Oct 16 14:16:43 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Tue, 16 Oct 2007 11:16:43 -0700
Subject: [cisco-voip] Subscriber Rebuild
In-Reply-To: <84b74aec0710161112id446d7fyedcc6628ad53cdc6@mail.gmail.com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<04B6D8CDD22FD34993C47E33E21F82EFCD795B@rachael-reborn.letnet.net>
	<D646304AFC55DB4EBB92329BCC800A00D0A180@seas7633.am.amersports.int>
	<84b74aec0710161112id446d7fyedcc6628ad53cdc6@mail.gmail.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D59758@seas7633.am.amersports.int>

Kevin/All, 

That is my understanding too.. and am I right in understanding that the
database transfer between the publisher and subscriber takes place with
little intervention during the process of bringing the subscriber
online?  Thanks!

 

Brandon Ogden

Amer Sports 

801-624-7529

www.amersports.com

http://www.linkedin.com/in/brandonogden

________________________________

From: Kevin Dunn [mailto:cheesevoice at gmail.com] 
Sent: Tuesday, October 16, 2007 12:12 PM
To: Ogden, Brandon
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Subscriber Rebuild

 

I am going to preface most of my replies with "Newbie" so you guys know
my experience level is low... that being said

 

you already have a publisher in your cluster.  You are going to add a
subscriber to the cluster, that alone isn't much ofan issue, but the
phones will need to register with the subscriber as the publisher houses
the actual up to date database, and the subscriber handles the
"day-to-day" operations... so the phones will need to re-register, but
the configuration process shouldn't be lengthy.  

 

(someone else should confirm my answers)

 

On 10/16/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote: 

I just rebuilt my publisher this weekend.. Everything has been running
well, so I am getting ready to rebuild the subscriber (which has been
offline with the old config as a reference and fall back) in the next
day or two.  Does it come on line pretty seamlessly?  Is there anything
I need to know... our Telecom contractor told me it was pretty smooth
and simply install, but he has been know to lie.. LOL  Thanks all!  

 

BTW.. its 5.1

 

Brandon Ogden

Amer Sports 

801-624-7529

www.amersports.com <http://www.amersports.com/> 

http://www.linkedin.com/in/brandonogden 

________________________________

 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

 

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From jburton at NETechCorp.com  Tue Oct 16 14:06:54 2007
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue, 16 Oct 2007 14:06:54 -0400
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
In-Reply-To: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com>
References: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com>
Message-ID: <FDC137145729AD438CDCA0F9FC6F7236013FE273@mail.netechcorp.com>

I ran into a bug with it.  Are you using an extended validation SSL cert
(the one that changes IE7's address bar to green)?

_____________________________ 
Jason Burton 
Systems Engineer
CCIE Voice #17319
NETech Corporation
317.660.3010 - Desk
317.581.1919 - Fax
jburton at netechcorp.com



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Tuesday, October 16, 2007 12:17 PM
To: cisco voip list
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring

Has anyone been successful with this?

I am trying to get this to work but can't figure out what the 'Trust
Certificate Subject CN' field is for on the presence gateway page when
integrating to outlook.

thanks.
justin
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jonvoip at gmail.com  Tue Oct 16 14:51:57 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 13:51:57 -0500
Subject: [cisco-voip] CM 6.0 unable to Cfwdall
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0071FF3DB@USNAEXCH.na.didata.local>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E89C@mail3.adcapnetsvc.local>
	<C1FE15183DA37645BC0633BC604E44F0071B68B8@USNAEXCH.na.didata.local>
	<5d093f9a0710161017w23b1a3c8r6460fdb138fd27e5@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0071FF3DB@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0710161151v2ff2a66dmf57bef2955517902@mail.gmail.com>

Better to block it in the first place... I would much rather have
people whine they couldn't make a call, then have some CIO yelling at
me cuz I allowed the call...

Then again, we make this as part of the design meetings... with
appropriate 'the sky is falling' warnings...

Jonathan

On 10/16/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> That is why departmental bill-back is such a great thing.  Managers that
> see this type of expense will squash it quicker than any IT guy could!
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Tuesday, October 16, 2007 1:17 PM
> To: Matt Slaga (US)
> Cc: James Grace; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CM 6.0 unable to Cfwdall
>
> Well, yes, it is most likely a CSS issue for CFA...
>
> However, before changing the CFA CSS to LD or whatever, make sure you
> want to let people do that, because there is a great potential for
> toll fraud.
>
> For example, employee has CFA CSS set to everywhere.
>
> He CFA's his phone to Maddrass, goes home and calls his desk phone to
> talk to his parents in India... he paid for a local call, you paid for
> a call to India.
>
> I would recommend that you create a CFA CSS to include just Voicemail
> and local calling.
>
> Maybe include your local area codes (so people can CFA to their
> cells), but don't go crazy until you are sure you want to.
>
>
> Jonathan
>
> On 10/16/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> >
> >
> >
> >
> > Check your CFA CSS.
> >
> >
> >
> >
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> > James Grace
> >  Sent: Tuesday, October 16, 2007 9:41 AM
> >  To: cisco-voip at puck.nether.net
> >  Subject: [cisco-voip] CM 6.0 unable to Cfwdall
> >
> >
> >
> >
> > ?
> >
> >
> >
> > for some reason im not able to hit the  Cfwdall button in CM 6.0.  I
> can
> > dial any pattern i want to, but when i press cfwdall its not allowing
> any
> > digits.  It give me "your call can not be completed.  Never ran into
> this on
> > previous ver.
> >
> >
> >
> >
> >
> >
> >
> > James Grace
> >
> >
> >
> >
> >
> > Systems Engineer
> >
> >
> > Adcap Network Systems, Inc.
> >
> >
> > 6865 Shiloh Road East, Suite 250
> >
> >
> > Alpharetta, GA 30005
> >
> >
> >
> >
> >
> > office: 678-456-6986
> >
> >
> > direct: 678-456-6761
> >
> >
> > jgrace at adcapnet.com
> >
> >
> > www.adcapnet.com Disclaimer For ADCAP Network Systems
> >
> > "The information transmitted is intended only for the person or entity
> to
> > which it is addressed and may
> > contain confidential and/or privileged material. Any review,
> retransmission,
> > dissemination or other use of,
> > or taking of any action in reliance upon this information by persons
> or
> > entities other than the intended
> > recipient is prohibited. If you received this message in error, please
> > contact the sender and delete the
> > material from all computers."
> >
> > This original email was sent to the internet for delivery at 16 Oct
> 2007
> > 09:39:53 -0400
> >
> >
> >
> > ________________________________
> >
> >
> >
> >
> >  Disclaimer: This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the
> designated
> > addressee(s) named above only. If you are not the intended addressee,
> you
> > are hereby notified that you have received this communication in error
> and
> > that any use or reproduction of this email or its contents is strictly
> > prohibited and may be unlawful. If you have received this
> communication in
> > error, please notify us immediately by replying to this message and
> deleting
> > it from your computer. Thank you.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>

From martind at cae.com  Tue Oct 16 14:55:29 2007
From: martind at cae.com (Daniel Martin)
Date: Tue, 16 Oct 2007 14:55:29 -0400
Subject: [cisco-voip] Upgrade from V5.1.1
Message-ID: <09881513E25DD549B26BF7F38ECAF8DA498694@caemex01.caecorp.cae.com>

Hi,

 

I am hitting a bug on V5.1.1 (core dumps) and TAC told me I need to
upgrade to V5.1.3. I asked him if I should go to V6.0 and he said yes. I
already have the V6.0 licenses files and CD's.

 

I would like to know if the people on the list recommend I go to V6.0
directly or I should stick with V5.1 release for a little while longer
and what are the reasons for your recommendations.

 

Is there any major issues with V6.0(1a)?

 

Much appreciated.

 

Daniel.

 

Daniel Martin

Groupe Reseau / Networks Group

CAE Inc.

St-Laurent, Quebec Canada

H4T 1G6

Tel:  1-514-341-2000 x4878

Fax: 1-514-340-5583

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From rkulagow at gmail.com  Tue Oct 16 15:03:27 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Tue, 16 Oct 2007 14:03:27 -0500
Subject: [cisco-voip] ISDN overlap receiving digits:  feh!
Message-ID: <47150AFF.8040209@gmail.com>

Thanks to our telco friends in Europe, I'm learning more than I ever 
wanted to know about variable-length dial plans, overlap receiving and 
dribbling in the called number one ISDN message at a time rather than 
all at once.

The Cisco doc on how to handle this 
(http://www.cisco.com/warp/public/788/signalling/isdn_overlap_prob.pdf) 
gives a solution, but it appears to have opened us up to toll-fraud.

Is anyone doing something like a TCL script to capture the inbound 
digits from the telco, build the dialed-number string, evaluate it for 
correctness and then pass it along?  It seems like TCL (or my dream 
PERS: BASIC.  If billg could do it on an Altair, then...) gives me the 
string manipulation routines that are a little more flexible then the 15 
rules available in a voice translation stuff.

None of the sample TCL scripts on the Cisco developer website seem 
oriented to this, and in looking over the developer docs I'm not sure 
that I can have the TCL script do what I want in the first place.

ObConfig:
H.323 dialpeers, 12.4.11T

From svoll.voip at gmail.com  Tue Oct 16 15:05:13 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 16 Oct 2007 12:05:13 -0700
Subject: [cisco-voip] Upgrade from V5.1.1
In-Reply-To: <09881513E25DD549B26BF7F38ECAF8DA498694@caemex01.caecorp.cae.com>
References: <09881513E25DD549B26BF7F38ECAF8DA498694@caemex01.caecorp.cae.com>
Message-ID: <f84a38d30710161205x204b379fuce03e9737cc828db@mail.gmail.com>

As a rule of thumb I (now) always wait for x.1.  my upgrade to IPCCx
4.0.1was 22 hours long with 3 Tac engineers and 6 developers.

if your only running CM then things are not as big of an issue.

but I look at it from the aspect that I have to upgrade IPCCx, Berbee
InformaCast, School Messenger (auto Dialer), XMedius Fax server, etc as part
of my upgrade.  So number one you need to have all your other software
applications support 6.0 and have the time to upgrade them and deal with any
bugs along the way.

With that said....... I think my next upgrade will no longer be from 4.1 to
5.1 but instead to 6.1.  (final desion still not made)

Scott

On 10/16/07, Daniel Martin <martind at cae.com> wrote:
>
>  Hi,
>
>
>
> I am hitting a bug on V5.1.1 (core dumps) and TAC told me I need to
> upgrade to V5.1.3. I asked him if I should go to V6.0 and he said yes. I
> already have the V6.0 licenses files and CD's.
>
>
>
> I would like to know if the people on the list recommend I go to V6.0directly or I should stick with
> V5.1 release for a little while longer and what are the reasons for your
> recommendations.
>
>
>
> Is there any major issues with V6.0(1a)?
>
>
>
> Much appreciated.
>
>
>
> Daniel.
>
>
>
> Daniel Martin
>
> Groupe Reseau / Networks Group
>
> CAE Inc.
>
> St-Laurent, Quebec Canada
>
> H4T 1G6
>
> Tel:  1-514-341-2000 x4878
>
> Fax: 1-514-340-5583
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Matt.Slaga at us.didata.com  Tue Oct 16 15:10:36 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 16 Oct 2007 15:10:36 -0400
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
In-Reply-To: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com>
References: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0071FF5CE@USNAEXCH.na.didata.local>

Not supported until CUPS 6.0(2). 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Tuesday, October 16, 2007 12:17 PM
To: cisco voip list
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring

Has anyone been successful with this?

I am trying to get this to work but can't figure out what the 'Trust
Certificate Subject CN' field is for on the presence gateway page when
integrating to outlook.

thanks.
justin
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From cisco.voip at verizon.net  Tue Oct 16 15:13:02 2007
From: cisco.voip at verizon.net (cisco.voip)
Date: Tue, 16 Oct 2007 15:13:02 -0400
Subject: [cisco-voip] Cisco CDP - 7970
References: <mailman.2034.1192558343.13771.cisco-voip@puck.nether.net>
Message-ID: <000b01c81028$93e761d0$661b1b0a@cs.myharris.net>

Is the Cisco CDP protocol required for the 7970 utilizing the POE switches?
I thought it was how the phone determined the power setting from the switch?

Thanks



From jonvoip at gmail.com  Tue Oct 16 17:06:15 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 16:06:15 -0500
Subject: [cisco-voip] Extended Services for CCM 4.3????
Message-ID: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>

It does not appear to have shipped with it, nor is it on CCO... I am
trying to setup TAPS... any ideas?



Jonathan

From jonvoip at gmail.com  Tue Oct 16 17:30:35 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 16:30:35 -0500
Subject: [cisco-voip] Extended Services for CCM 4.3????
In-Reply-To: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
References: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
Message-ID: <5d093f9a0710161430v7182284je8fb47fd7b7b9211@mail.gmail.com>

OK, more questions...

I know this is going to sound shocking, but I have never used TAPS
before (yes, I have been doing this since 2004, and I have never set
up TAPS... sue me).

I have two CCM 4.3(1) boxes and two IPCC 4.1(1) servers in an HA cluster.

Can I use the IPCC cluster for TAPS and just load the aar file onto
it? It seems too easy, but it sounds like a plan...

Or, MUST I install IPCC on my CCM Publisher and then run TAPS there?

Any ideas....



Jonathan

On 10/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> It does not appear to have shipped with it, nor is it on CCO... I am
> trying to setup TAPS... any ideas?
>
>
>
> Jonathan
>

From matt at saskin.net  Tue Oct 16 17:59:50 2007
From: matt at saskin.net (Matthew Saskin)
Date: Tue, 16 Oct 2007 17:59:50 -0400
Subject: [cisco-voip] Extended Services for CCM 4.3????
In-Reply-To: <5d093f9a0710161430v7182284je8fb47fd7b7b9211@mail.gmail.com>
References: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
	<5d093f9a0710161430v7182284je8fb47fd7b7b9211@mail.gmail.com>
Message-ID: <47153456.3030003@saskin.net>

I believe you can do the former without issue.  Should be a matter of 
running the TAPS installer on the IPCC boxes themselves and pointing 
towards the CM.

-matt

Jonathan Charles wrote:
> OK, more questions...
> 
> I know this is going to sound shocking, but I have never used TAPS
> before (yes, I have been doing this since 2004, and I have never set
> up TAPS... sue me).
> 
> I have two CCM 4.3(1) boxes and two IPCC 4.1(1) servers in an HA cluster.
> 
> Can I use the IPCC cluster for TAPS and just load the aar file onto
> it? It seems too easy, but it sounds like a plan...
> 
> Or, MUST I install IPCC on my CCM Publisher and then run TAPS there?
> 
> Any ideas....
> 
> 
> 
> Jonathan
> 
> On 10/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>> It does not appear to have shipped with it, nor is it on CCO... I am
>> trying to setup TAPS... any ideas?
>>
>>
>>
>> Jonathan
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From jsteinberg at gmail.com  Tue Oct 16 18:05:56 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 16 Oct 2007 18:05:56 -0400
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
In-Reply-To: <FDC137145729AD438CDCA0F9FC6F7236013FE273@mail.netechcorp.com>
References: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com>
	<FDC137145729AD438CDCA0F9FC6F7236013FE273@mail.netechcorp.com>
Message-ID: <f1b65f880710161505m5989eed5leec418d9ecd76b96@mail.gmail.com>

Jason, Are you speaking about the SSL cert that I use for url such as:

https://exchangeserver/Exchange/Jsteinberg/Calendar/


Matt, where did you hear it wasn't working in CUPS 6.0.1 ?  It is
mentioned in the release notes for 6.0.1.  Is there a bug filed
against it?

http://www.cisco.com/en/US/docs/voice_ip_comm/cups/6_0_1/rel_notes/cup601rn.html#wp478329


On 10/16/07, Jason Burton <jburton at netechcorp.com> wrote:
> I ran into a bug with it.  Are you using an extended validation SSL cert
> (the one that changes IE7's address bar to green)?
>
> _____________________________
> Jason Burton
> Systems Engineer
> CCIE Voice #17319
> NETech Corporation
> 317.660.3010 - Desk
> 317.581.1919 - Fax
> jburton at netechcorp.com
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> Steinberg
> Sent: Tuesday, October 16, 2007 12:17 PM
> To: cisco voip list
> Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
>
> Has anyone been successful with this?
>
> I am trying to get this to work but can't figure out what the 'Trust
> Certificate Subject CN' field is for on the presence gateway page when
> integrating to outlook.
>
> thanks.
> justin
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jsteinberg at gmail.com  Tue Oct 16 18:08:11 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 16 Oct 2007 18:08:11 -0400
Subject: [cisco-voip] Triple Unity servers...
In-Reply-To: <5d093f9a0710161013n6afdbcaesff75a0459e51dded@mail.gmail.com>
References: <5d093f9a0709190931h28adab0boec9034c1fdab9660@mail.gmail.com>
	<5d093f9a0710142154w43a9f31cqa0e2b30bdfdae838@mail.gmail.com>
	<d4c697ad0710142333g3dbebd24g26934f58becd9547@mail.gmail.com>
	<5d093f9a0710142348g557b30e8h2f63af13e1425790@mail.gmail.com>
	<5d093f9a0710151232h35a08a1fg58b782ab738f6d1f@mail.gmail.com>
	<4713D352.1000303@wcyv.com>
	<5d093f9a0710161013n6afdbcaesff75a0459e51dded@mail.gmail.com>
Message-ID: <f1b65f880710161508pdf4739di49c06b9c081b8d39@mail.gmail.com>

Uconn supports TTS for exchange:

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/1x/installation/guide/ig_120.html


On 10/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> The weird question is, why would you want/need TTS in Unity
> Connections... if it is voicemail only?
>
>
> Jonathan
>
> On 10/15/07, Pat Hayes <pat-cv at wcyv.com> wrote:
> > They're correct, there is no such thing in the Unity world at the
> > moment. I believe you (or whoever ordered this) is thinking of the Voice
> > Recognition server, new to 5.0. In order to get voice recognition
> > features, you have to use another server. Installation procedure here:
> >
> > http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/installation/guide/umexfo/5xcuigumefoappc.html
> >
> > -------- Original Message  --------
> > From: "Jonathan Charles" <jonvoip at gmail.com>
> > To: "Aman Chugh" <aman.chugh at gmail.com>
> > Cc: Cisco Voip Mailing list <cisco-voip at puck.nether.net>
> > Subject: [cisco-voip] Triple Unity servers...
> > Date: 10/15/2007 3:32 PM
> >
> > > OK, I opened a TAC case and here is the response:
> > >
> > >
> > > "Unfortunately, you cannot install TTS server on a dedicated hardware
> > > box, even on CU ver 5.0. Only Cisco Unity Connection allows you to do
> > > that."
> > >
> > >
> > > Jonathan
> > >
> > > On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >> Well, those are the questions:
> > >>
> > >> A: How is this done, if it can be done at all.
> > >>
> > >> B: Would TAC support afterwards.
> > >>
> > >>
> > >> Jonathan
> > >>
> > >> On 10/15/07, Aman Chugh <aman.chugh at gmail.com> wrote:
> > >>> Not Sure if this would be a supported TAC deployement.
> > >>>
> > >>> I am thinking that you can run the unity setup and enable TTS setting only
> > >>> on the third box but It would be interesting to learn if three unity servers
> > >>> can exist toegther in the failover setup.
> > >>>
> > >>> Do let us know what happens next !
> > >>>
> > >>> Aman
> > >>>
> > >>>
> > >>> On 10/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >>>> So, I have to do this on Tuesday, and was wondering if anyone had any
> > >>> ideas...
> > >>>> The plan is to install the primary and secondary Unity servers without
> > >>>> TTS and then run the Failoverconfig.exe to mate them together...
> > >>>> Pretty simple (I hope).
> > >>>>
> > >>>> The question is about this third Unity server, the dedicated TTS
> > >>>> box... how do I link that to the Unity twins?
> > >>>>
> > >>>> Not finding a lot of love on Cisco's website here... (no mention
> > >>>> anywhere of the concept of a dedicated TTS box)...
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>> Jonathan
> > >>>>
> > >>>> On 9/19/07, Jonathan Charles < jonvoip at gmail.com> wrote:
> > >>>>> Cust ordered 3 Unity servers, a  primary, a secondary and a dedicated
> > >>> TTS box.
> > >>>>> Not sure how to install the TTS box. Do I set it up normally (SQL,
> > >>>>> Exchange tools, etc.) and then install Unity but only select TTS (and
> > >>>>> not select Unity)?
> > >>>>>
> > >>>>> Also, would I NOT select TTS on the two primaries?
> > >>>>>
> > >>>>> Just curious, I didn't sell this monster...
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>> Jonathan
> > >>>>>
> > >>>> _______________________________________________
> > >>>> cisco-voip mailing list
> > >>>> cisco-voip at puck.nether.net
> > >>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>>>
> > >>>
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jsteinberg at gmail.com  Tue Oct 16 18:11:00 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 16 Oct 2007 18:11:00 -0400
Subject: [cisco-voip] Upgrade from V5.1.1
In-Reply-To: <09881513E25DD549B26BF7F38ECAF8DA498694@caemex01.caecorp.cae.com>
References: <09881513E25DD549B26BF7F38ECAF8DA498694@caemex01.caecorp.cae.com>
Message-ID: <f1b65f880710161511x1d6de4a0uf14a9b7ade9960d9@mail.gmail.com>

Did TAC give you specific reason's for pushing you to 6.0?  Did you
ask whether you should upgrade to 5.1.3 or 6?  Or did TAC proactively
recommend 6.0?

I'm in the process of making a similar decision...

Justin

On 10/16/07, Daniel Martin <martind at cae.com> wrote:
>
>
>
>
> Hi,
>
>
>
> I am hitting a bug on V5.1.1 (core dumps) and TAC told me I need to upgrade
> to V5.1.3. I asked him if I should go to V6.0 and he said yes. I already
> have the V6.0 licenses files and CD's.
>
>
>
> I would like to know if the people on the list recommend I go to V6.0
> directly or I should stick with V5.1 release for a little while longer and
> what are the reasons for your recommendations.
>
>
>
> Is there any major issues with V6.0(1a)?
>
>
>
> Much appreciated.
>
>
>
> Daniel.
>
>
>
> Daniel Martin
>
> Groupe Reseau / Networks Group
>
> CAE Inc.
>
> St-Laurent, Quebec Canada
>
> H4T 1G6
>
> Tel:  1-514-341-2000 x4878
>
> Fax: 1-514-340-5583
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jsteinberg at gmail.com  Tue Oct 16 18:13:21 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 16 Oct 2007 18:13:21 -0400
Subject: [cisco-voip] Extended Services for CCM 4.3????
In-Reply-To: <5d093f9a0710161430v7182284je8fb47fd7b7b9211@mail.gmail.com>
References: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
	<5d093f9a0710161430v7182284je8fb47fd7b7b9211@mail.gmail.com>
Message-ID: <f1b65f880710161513i4965fe53h8b2009d0d0142391@mail.gmail.com>

haven't done 4.3

but with 4.1, you install TAPS from the ccmadmin plugin page on the
CCM publisher, and then run the same install (taps from ccmadmin
plugins) on the IPCCX master box.

it's pretty easy.  taps install guide is straightforward.


On 10/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> OK, more questions...
>
> I know this is going to sound shocking, but I have never used TAPS
> before (yes, I have been doing this since 2004, and I have never set
> up TAPS... sue me).
>
> I have two CCM 4.3(1) boxes and two IPCC 4.1(1) servers in an HA cluster.
>
> Can I use the IPCC cluster for TAPS and just load the aar file onto
> it? It seems too easy, but it sounds like a plan...
>
> Or, MUST I install IPCC on my CCM Publisher and then run TAPS there?
>
> Any ideas....
>
>
>
> Jonathan
>
> On 10/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > It does not appear to have shipped with it, nor is it on CCO... I am
> > trying to setup TAPS... any ideas?
> >
> >
> >
> > Jonathan
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jason.aarons at us.didata.com  Tue Oct 16 18:12:58 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 16 Oct 2007 18:12:58 -0400
Subject: [cisco-voip] UCCX 4-5 and UCM 5.1 Speed Dials with Presence
Message-ID: <C1FE15183DA37645BC0633BC604E44F0071FF919@USNAEXCH.na.didata.local>

Currently shared lines are not supported, however, can a manager have a
7914 side car setup with buttons that are Speed Dials with Presence
showing their Agents ICD controlled line?

 

In short, the Supervisor wants to see if an ICD Controlled line is on
the phone.

 

UCCX 4.5(2)SR2 and UCM 5.1.2

 




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From Nunnally at Harding.edu  Tue Oct 16 18:17:54 2007
From: Nunnally at Harding.edu (John Nunnally)
Date: Tue, 16 Oct 2007 17:17:54 -0500
Subject: [cisco-voip] Call handler question - Replacing Nortel express
 messaging?
In-Reply-To: <2B3745BC9D31514381A6FCC5E72D9373026AA0A8@SMOMAIL.jhacorp.com>
References: <040201c805e1$f1d0e9c0$c906010a@harding.edu>
	<2B3745BC9D31514381A6FCC5E72D9373026AA0A8@SMOMAIL.jhacorp.com>
Message-ID: <09a001c81042$65d430b0$c906010a@harding.edu>

Chris, thanks for this reply.  I'm sorry it took me so long to look into it,
but this solution does not completely address my problem.  This allows me to
dial a number with some kind of prefix and be routed to a greeting.  This is
impossible when I am dialing a DID number from outside my call manager.
 
I need a way to call a number in Unity. Once I am talking to Unity, enter an
extension (or mailbox) number and be transferred directly to the greeting
rather than ringing the phone.  I have not found a way to do that.  This is
a feature in Nortel Meridian Mail called "Express Messaging" which is very
useful in a lot of circumstances.  We are movng from Nortel to Unity and I
have not come up with a replacement for this feature.  The natives are
restless.
 
John N.

  _____  

From: Chris D'amore [mailto:CDAmore at jackhenry.com] 
Sent: Wednesday, October 03, 2007 12:59 PM
To: John Nunnally; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Call handler question



Follow this link.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a00800dea82.shtml

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Nunnally
Sent: Wednesday, October 03, 2007 12:22 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call handler question

 

Is there a way in Unity to allow a caller to enter an extension number and
leave a message in a mailbox without ringing the phone?  Nortel calls this
feature "express messaging".

 

In other words, I would like to use "Caller System Transfer" and go directly
to the subscribers greeting instead of transferring the call.

 

Is there a way to do that?

 

JWN

 

NOTICE: This electronic mail message and any files transmitted with it are
intended

exclusively for the individual or entity to which it is addressed. The
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Any unauthorized review, use, printing, saving, copying, disclosure or
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From wsisk at cisco.com  Tue Oct 16 18:26:41 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 16 Oct 2007 18:26:41 -0400
Subject: [cisco-voip] Subscriber Rebuild
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D59758@seas7633.am.amersports.int>
References: <000601c80f7b$ec317f40$c4947dc0$@com>	<04B6D8CDD22FD34993C47E33E21F82EFCD795B@rachael-reborn.letnet.net>	<D646304AFC55DB4EBB92329BCC800A00D0A180@seas7633.am.amersports.int>	<84b74aec0710161112id446d7fyedcc6628ad53cdc6@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D59758@seas7633.am.amersports.int>
Message-ID: <47153AA1.6060306@cisco.com>

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From lists at jerkys.org  Tue Oct 16 18:32:26 2007
From: lists at jerkys.org (Jerky)
Date: Tue, 16 Oct 2007 17:32:26 -0500
Subject: [cisco-voip] Home user
In-Reply-To: <AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
Message-ID: <513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>

This has been kicked around for a while since we moved to CallManager  
but not much thought has been given to it. I'm trying to understand  
how your hardware is setup. How would it look, similar to one of these?

87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <-- 
Ethernet--> LAN

or

87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
<---> ASA or PIX Firewall <--Ethernet--> LAN

Is the 3800 used for all your firewalling needs in lieu of something  
like an ASA or PIX? Sonicwall's are currently in place and haven't  
worked very well for the remote users it was tested with. The  
Sonicwalls we have don't have anything similar to what the 871's seem  
to have in regards to vlans and packet tagging. We would probably  
kick the Sonicwalls out if something else would work better.

jeff

On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:

> We currently have about 40 production remote home teleworkers that  
> have been deployed using Cisco 871/877 wireless routers and a 7960  
> phones.  We are using a Cisco 3845 series router at the head-end so  
> that we can control QoS tagging on the egress / ingress points of  
> both sides of the VPN tunnel.  We are using a phase 2 DMVPN  
> solution dual-homed to two sites to provide secure redundant  
> connectivity.
>
>
>
> It took me a bit to tweak my router configurations (I started on  
> Cisco 831/837 routers) to get the results that we wanted, but all  
> and all our users are happy.  There is the occasional jitter and  
> packet loss (it is the Internet mind you) but g.729 is working  
> quite well coupled with business cable and DSL services.
>
>
>
> If you have any other questions, feel free to ask.
>
>
>
> Matt
>
>
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net] On Behalf Of Curt Shaffer
> Sent: Monday, October 15, 2007 6:37 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Home user
>
>
>
> I was wondering want everyone out there is using for the situation  
> where you have someone on your CCM or CCME that has 1 phone at a  
> home office. Something tells me an ASA is overkill and I haven?t  
> found solid information that any of the 87x routers support tagging  
> QoS of packets going through the VPN tunnel. We would obviously  
> like to have QoS in place even though it?s not respected at their  
> ISP just to make sure the VPN/Voice packets are leaving their  
> routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be  
> protected by one or more legal privileges. It is intended solely  
> for the use of the addressee identified above. If you are not the  
> intended recipient, any use, disclosure, copying or distribution of  
> this communication is UNAUTHORIZED. Neither this information block,  
> the typed name of the sender, nor anything else in this message is  
> intended to constitute an electronic signature unless a specific  
> statement to the contrary is included in this message. If you have  
> received this communication in error, please immediately contact me  
> and delete this communication from your computer. Thank you.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From wsisk at cisco.com  Tue Oct 16 18:32:50 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 16 Oct 2007 18:32:50 -0400
Subject: [cisco-voip] Cisco CDP - 7970
In-Reply-To: <000b01c81028$93e761d0$661b1b0a@cs.myharris.net>
References: <mailman.2034.1192558343.13771.cisco-voip@puck.nether.net>
	<000b01c81028$93e761d0$661b1b0a@cs.myharris.net>
Message-ID: <47153C12.5020200@cisco.com>

1. switch provides max power initially.  max power depends if 
switch/phone is using cisco power or 802.3af (~15.4w)
2. once phone boots it advertises CDP to switch the max power the phone 
will consume (~14.9w)

cdp is not 'required' for inline power. it's only an optimization so the 
client can 'request' less than max power.

/wes

cisco.voip wrote:
> Is the Cisco CDP protocol required for the 7970 utilizing the POE switches?
> I thought it was how the phone determined the power setting from the switch?
>
> Thanks
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Tue Oct 16 18:34:00 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 16 Oct 2007 18:34:00 -0400
Subject: [cisco-voip] Extended Services for CCM 4.3????
In-Reply-To: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
References: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
Message-ID: <47153C58.6040400@cisco.com>

as i recently learned the hard way:

"Bundles that ship with UCM on w2k3 (e.g. 4.3) ship with CCX 4.1 which 
is Win2003 ? they can run co-res."

/Wes

Jonathan Charles wrote:
> It does not appear to have shipped with it, nor is it on CCO... I am
> trying to setup TAPS... any ideas?
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Tue Oct 16 18:42:04 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 16 Oct 2007 18:42:04 -0400
Subject: [cisco-voip] UCCX 4-5 and UCM 5.1 Speed Dials with Presence
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0071FF919@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0071FF919@USNAEXCH.na.didata.local>
Message-ID: <47153E3C.1000000@cisco.com>

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From jonvoip at gmail.com  Tue Oct 16 18:56:52 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 17:56:52 -0500
Subject: [cisco-voip] Extended Services for CCM 4.3????
In-Reply-To: <47153C58.6040400@cisco.com>
References: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
	<47153C58.6040400@cisco.com>
Message-ID: <5d093f9a0710161556l47aea5aam2033ec1a2185d5ab@mail.gmail.com>

But I don't want to install Coresident... TAPS will be used during
install and then not be used again... why have CRS sucking away
resources for nothing...


Jonathan

On 10/16/07, Wes Sisk <wsisk at cisco.com> wrote:
> as i recently learned the hard way:
>
> "Bundles that ship with UCM on w2k3 (e.g. 4.3) ship with CCX 4.1 which
> is Win2003 ? they can run co-res."
>
> /Wes
>
> Jonathan Charles wrote:
> > It does not appear to have shipped with it, nor is it on CCO... I am
> > trying to setup TAPS... any ideas?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From jonvoip at gmail.com  Tue Oct 16 18:57:57 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 17:57:57 -0500
Subject: [cisco-voip] Extended Services for CCM 4.3????
In-Reply-To: <f1b65f880710161513i4965fe53h8b2009d0d0142391@mail.gmail.com>
References: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
	<5d093f9a0710161430v7182284je8fb47fd7b7b9211@mail.gmail.com>
	<f1b65f880710161513i4965fe53h8b2009d0d0142391@mail.gmail.com>
Message-ID: <5d093f9a0710161557i3fc7d5e2xd513056e83a3bdb8@mail.gmail.com>

Someone suggested using the AAR file to create everything does that work?

Also, are the apps and prompts built automagically? Or do I have to
hunt em down...

Also Also, where is this install guide you speak of that says how to
do it on an IPCC cluster?


Jonathan

On 10/16/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
> haven't done 4.3
>
> but with 4.1, you install TAPS from the ccmadmin plugin page on the
> CCM publisher, and then run the same install (taps from ccmadmin
> plugins) on the IPCCX master box.
>
> it's pretty easy.  taps install guide is straightforward.
>
>
> On 10/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > OK, more questions...
> >
> > I know this is going to sound shocking, but I have never used TAPS
> > before (yes, I have been doing this since 2004, and I have never set
> > up TAPS... sue me).
> >
> > I have two CCM 4.3(1) boxes and two IPCC 4.1(1) servers in an HA cluster.
> >
> > Can I use the IPCC cluster for TAPS and just load the aar file onto
> > it? It seems too easy, but it sounds like a plan...
> >
> > Or, MUST I install IPCC on my CCM Publisher and then run TAPS there?
> >
> > Any ideas....
> >
> >
> >
> > Jonathan
> >
> > On 10/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > It does not appear to have shipped with it, nor is it on CCO... I am
> > > trying to setup TAPS... any ideas?
> > >
> > >
> > >
> > > Jonathan
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From wsisk at cisco.com  Tue Oct 16 18:58:53 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 16 Oct 2007 18:58:53 -0400
Subject: [cisco-voip] Extended Services for CCM 4.3????
In-Reply-To: <5d093f9a0710161556l47aea5aam2033ec1a2185d5ab@mail.gmail.com>
References: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>	
	<47153C58.6040400@cisco.com>
	<5d093f9a0710161556l47aea5aam2033ec1a2185d5ab@mail.gmail.com>
Message-ID: <4715422D.7070906@cisco.com>

<supposition>
install, use, uninstall
</supposition>

otherwise get into server license, os license, license license license

/wes

Jonathan Charles wrote:
> But I don't want to install Coresident... TAPS will be used during
> install and then not be used again... why have CRS sucking away
> resources for nothing...
>
>
> Jonathan
>
> On 10/16/07, Wes Sisk <wsisk at cisco.com> wrote:
>   
>> as i recently learned the hard way:
>>
>> "Bundles that ship with UCM on w2k3 (e.g. 4.3) ship with CCX 4.1 which
>> is Win2003 ? they can run co-res."
>>
>> /Wes
>>
>> Jonathan Charles wrote:
>>     
>>> It does not appear to have shipped with it, nor is it on CCO... I am
>>> trying to setup TAPS... any ideas?
>>>
>>>
>>>
>>> Jonathan
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>       

From cshaffer at gmail.com  Tue Oct 16 19:40:52 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Tue, 16 Oct 2007 19:40:52 -0400
Subject: [cisco-voip] Home user
In-Reply-To: <513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
Message-ID: <00ba01c8104d$fe583920$fb08ab60$@com>

Actually due to a difference between who this customer uses for Server
network vs. Phone network; they have been using Sonicwall for normal VPN
connectivity, which I personally do not like based on personal experience.
But the setup will be as follows:

 

87x router or ASA ?-DSL or Cable-? INTERNET ?---PRI/Internet
connection--?2811 router?-Phone LAN

 

From: Jerky [mailto:lists at jerkys.org] 
Sent: Tuesday, October 16, 2007 6:32 PM
To: Linsemier, Matthew
Cc: Curt Shaffer; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Home user

 

This has been kicked around for a while since we moved to CallManager but
not much thought has been given to it. I'm trying to understand how your
hardware is setup. How would it look, similar to one of these?

 

87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845
<--Ethernet--> LAN

 

or

 

87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <---> ASA
or PIX Firewall <--Ethernet--> LAN

 

Is the 3800 used for all your firewalling needs in lieu of something like an
ASA or PIX? Sonicwall's are currently in place and haven't worked very well
for the remote users it was tested with. The Sonicwalls we have don't have
anything similar to what the 871's seem to have in regards to vlans and
packet tagging. We would probably kick the Sonicwalls out if something else
would work better.

 

jeff

 

On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:





We currently have about 40 production remote home teleworkers that have been
deployed using Cisco 871/877 wireless routers and a 7960 phones.  We are
using a Cisco 3845 series router at the head-end so that we can control QoS
tagging on the egress / ingress points of both sides of the VPN tunnel.  We
are using a phase 2 DMVPN solution dual-homed to two sites to provide secure
redundant connectivity.

 

It took me a bit to tweak my router configurations (I started on Cisco
831/837 routers) to get the results that we wanted, but all and all our
users are happy.  There is the occasional jitter and packet loss (it is the
Internet mind you) but g.729 is working quite well coupled with business
cable and DSL services.

 

If you have any other questions, feel free to ask.

 

Matt

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Curt Shaffer
Sent: Monday, October 15, 2007 6:37 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Home user

 

I was wondering want everyone out there is using for the situation where you
have someone on your CCM or CCME that has 1 phone at a home office.
Something tells me an ASA is overkill and I haven?t found solid information
that any of the 87x routers support tagging QoS of packets going through the
VPN tunnel. We would obviously like to have QoS in place even though it?s
not respected at their ISP just to make sure the VPN/Voice packets are
leaving their routers first as a best effort to get some quality. 

 

Thanks

 

 

  _____  

CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may be protected
by one or more legal privileges. It is intended solely for the use of the
addressee identified above. If you are not the intended recipient, any use,
disclosure, copying or distribution of this communication is UNAUTHORIZED.
Neither this information block, the typed name of the sender, nor anything
else in this message is intended to constitute an electronic signature
unless a specific statement to the contrary is included in this message. If
you have received this communication in error, please immediately contact me
and delete this communication from your computer. Thank you. 

  _____  

_______________________________________________

cisco-voip mailing list

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https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From mthompson729 at gmail.com  Tue Oct 16 20:17:34 2007
From: mthompson729 at gmail.com (Michael Thompson)
Date: Tue, 16 Oct 2007 20:17:34 -0400
Subject: [cisco-voip] Extended Services for CCM 4.3????
In-Reply-To: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
References: <5d093f9a0710161406m2a359e50l63ea1b77d99a51fe@mail.gmail.com>
Message-ID: <363f21ee0710161717r4d8985c4h4abe7beb83b07fd@mail.gmail.com>

Jonathan,
   from a technical perspective, the only pieces that are really used are:
Scripts on CRS (IPCC), CTI Ports / CTI RP on CallManager, BAT Plug in (the
works locally to CCM), and that's the bulk of it (no BAT pun intended).
from that position, I have not done what you're talking about but see
absolutely no technical reason it wouldn't work.  All you're doing is
processing the TAPS script / TAPS service on another box.  The only grey
area I can think of is the TAPS service that gets installed on Publisher
when you install it.  That would have to run on the IPCC box, but may not be
an issue.



On 10/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> It does not appear to have shipped with it, nor is it on CCO... I am
> trying to setup TAPS... any ideas?
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From iayap at cisco.com  Tue Oct 16 20:18:55 2007
From: iayap at cisco.com (Ian Yap (iayap))
Date: Tue, 16 Oct 2007 17:18:55 -0700
Subject: [cisco-voip] Question on automating the Phone Services subscription
	process
Message-ID: <2FEA927D4859134F939C41B656EBB21C04E81641@xmb-sjc-21e.amer.cisco.com>

Hi there everyone
 
I was wondering if there is a way to automate this process. I am using a
few phone services developed under CUAE inside the latest CUCM( i.e. I
added the services under Device --> Device Settings --> Phone Services
). I also have auto-registration enabled, therefore, when I register a
Cisco IP phone with my CUCM, it will be assigned an extension
immediately. However, I would need to go to every phone manually (Device
--> Phone) each time to subscribe it to those phone services I mentioned
earlier.
 
I couldn't find a way to define some form of a template that every newly
registered phone could use to automatically subscribe to the services I
wanted. Is there a workaround solution?
 
Thanks
Ian
 
"You can make more friends in two months by becoming really interested
in other people than you can in two years by trying to get other people
interested in you."  --- Dale Carnegie
 
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From mthompson729 at gmail.com  Tue Oct 16 20:20:54 2007
From: mthompson729 at gmail.com (Michael Thompson)
Date: Tue, 16 Oct 2007 20:20:54 -0400
Subject: [cisco-voip] Question on automating the Phone Services
	subscription process
In-Reply-To: <2FEA927D4859134F939C41B656EBB21C04E81641@xmb-sjc-21e.amer.cisco.com>
References: <2FEA927D4859134F939C41B656EBB21C04E81641@xmb-sjc-21e.amer.cisco.com>
Message-ID: <363f21ee0710161720i6240823t2c610350a801d201@mail.gmail.com>

I'm going to base my comment off of assuming that you're talking about CCM
4.X and not 6 (which is the latest).  if BAT is installed, i'm pretty sure
you can select all the phones after they are auto-registered and assign a
service to them at that point.

On 10/16/07, Ian Yap (iayap) <iayap at cisco.com> wrote:
>
>  Hi there everyone
>
> I was wondering if there is a way to automate this process. I am using a
> few phone services developed under CUAE inside the latest CUCM( i.e. I
> added the services under Device --> Device Settings --> Phone Services ). I
> also have auto-registration enabled, therefore, when I register a Cisco IP
> phone with my CUCM, it will be assigned an extension immediately. However, I
> would need to go to every phone manually (Device --> Phone) each time to
> subscribe it to those phone services I mentioned earlier.
>
> I couldn't find a way to define some form of a template that every newly
> registered phone could use to automatically subscribe to the services I
> wanted. Is there a workaround solution?
>
> Thanks
> Ian
>
> "You can make more friends in two months by becoming really interested in
> other people than you can in two years by trying to get other people
> interested in you."  --- Dale Carnegie
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From mthompson729 at gmail.com  Tue Oct 16 20:23:14 2007
From: mthompson729 at gmail.com (Michael Thompson)
Date: Tue, 16 Oct 2007 20:23:14 -0400
Subject: [cisco-voip] Cisco CDP - 7970
In-Reply-To: <000b01c81028$93e761d0$661b1b0a@cs.myharris.net>
References: <mailman.2034.1192558343.13771.cisco-voip@puck.nether.net>
	<000b01c81028$93e761d0$661b1b0a@cs.myharris.net>
Message-ID: <363f21ee0710161723s144434f4k863649f54a1d89cf@mail.gmail.com>

don't underestimate the use of CDP for enterprise (of even a few sites in a
smaller company) and the roll it plays in packet marking for QoS purposes.

On 10/16/07, cisco.voip <cisco.voip at verizon.net> wrote:
>
> Is the Cisco CDP protocol required for the 7970 utilizing the POE
> switches?
> I thought it was how the phone determined the power setting from the
> switch?
>
> Thanks
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From pat-cv at wcyv.com  Tue Oct 16 21:12:17 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Tue, 16 Oct 2007 21:12:17 -0400
Subject: [cisco-voip] Call handler question - Replacing Nortel express
 messaging?
In-Reply-To: <09a001c81042$65d430b0$c906010a@harding.edu>
References: <040201c805e1$f1d0e9c0$c906010a@harding.edu>	<2B3745BC9D31514381A6FCC5E72D9373026AA0A8@SMOMAIL.jhacorp.com>
	<09a001c81042$65d430b0$c906010a@harding.edu>
Message-ID: <47156171.3070405@wcyv.com>

Are you using the transfer to subscriber functionality elsewhere? If 
not, the easiest thing to do is to set all of the subscriber's transfer 
options to 'send to greeting'. Then you could just use any call handler 
with the option to enable callers to enter an extension.

Another option, with similar behavior, that does take a little extra 
user training is #2. From any call handler that is enabled to allow you 
to enter a subscriber's extension, you can add #2 to the end of the dial 
string to bypass the user's transfer options and go straight to their 
greeting. So, to get to 1000's mailbox, you dial 1000#2.

Other that that, Unity really doesn't have a direct equivalent of 
Express Messaging.

-------- Original Message  --------
From: John Nunnally <Nunnally at harding.edu>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call handler question - Replacing Nortel express 
messaging?
Date: 10/16/2007 6:17 PM

> Chris, thanks for this reply.  I'm sorry it took me so long to look into it,
> but this solution does not completely address my problem.  This allows me to
> dial a number with some kind of prefix and be routed to a greeting.  This is
> impossible when I am dialing a DID number from outside my call manager.
>  
> I need a way to call a number in Unity. Once I am talking to Unity, enter an
> extension (or mailbox) number and be transferred directly to the greeting
> rather than ringing the phone.  I have not found a way to do that.  This is
> a feature in Nortel Meridian Mail called "Express Messaging" which is very
> useful in a lot of circumstances.  We are movng from Nortel to Unity and I
> have not come up with a replacement for this feature.  The natives are
> restless.
>  
> John N.
> 
>   _____  
> 
> From: Chris D'amore [mailto:CDAmore at jackhenry.com] 
> Sent: Wednesday, October 03, 2007 12:59 PM
> To: John Nunnally; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Call handler question
> 
> 
> 
> Follow this link.
> 
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
> a00800dea82.shtml
> 
>  
> 
>  
> 
>  
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Nunnally
> Sent: Wednesday, October 03, 2007 12:22 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Call handler question
> 
>  
> 
> Is there a way in Unity to allow a caller to enter an extension number and
> leave a message in a mailbox without ringing the phone?  Nortel calls this
> feature "express messaging".
> 
>  
> 
> In other words, I would like to use "Caller System Transfer" and go directly
> to the subscribers greeting instead of transferring the call.
> 
>  
> 
> Is there a way to do that?
> 
>  
> 
> JWN
> 
>  
> 
> NOTICE: This electronic mail message and any files transmitted with it are
> intended
> 
> exclusively for the individual or entity to which it is addressed. The
> message, 
> 
> together with any attachment, may contain confidential and/or privileged
> information.
> 
> Any unauthorized review, use, printing, saving, copying, disclosure or
> distribution 
> 
> is strictly prohibited. If you have received this message in error, please 
> 
> immediately advise the sender by reply email and delete all copies.
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From aman.chugh at gmail.com  Wed Oct 17 00:23:48 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Wed, 17 Oct 2007 09:53:48 +0530
Subject: [cisco-voip] CM 6.0 unable to Cfwdall
In-Reply-To: <5d093f9a0710161151v2ff2a66dmf57bef2955517902@mail.gmail.com>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E89C@mail3.adcapnetsvc.local>
	<C1FE15183DA37645BC0633BC604E44F0071B68B8@USNAEXCH.na.didata.local>
	<5d093f9a0710161017w23b1a3c8r6460fdb138fd27e5@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0071FF3DB@USNAEXCH.na.didata.local>
	<5d093f9a0710161151v2ff2a66dmf57bef2955517902@mail.gmail.com>
Message-ID: <d4c697ad0710162123l4ded24e9mab649d18e67930c6@mail.gmail.com>

Did you have some one calling Maddrrass , the city is now named Chennai :)

Aman


On 10/17/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Better to block it in the first place... I would much rather have
> people whine they couldn't make a call, then have some CIO yelling at
> me cuz I allowed the call...
>
> Then again, we make this as part of the design meetings... with
> appropriate 'the sky is falling' warnings...
>
> Jonathan
>
> On 10/16/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> > That is why departmental bill-back is such a great thing.  Managers that
> > see this type of expense will squash it quicker than any IT guy could!
> >
> > -----Original Message-----
> > From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > Sent: Tuesday, October 16, 2007 1:17 PM
> > To: Matt Slaga (US)
> > Cc: James Grace; cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] CM 6.0 unable to Cfwdall
> >
> > Well, yes, it is most likely a CSS issue for CFA...
> >
> > However, before changing the CFA CSS to LD or whatever, make sure you
> > want to let people do that, because there is a great potential for
> > toll fraud.
> >
> > For example, employee has CFA CSS set to everywhere.
> >
> > He CFA's his phone to Maddrass, goes home and calls his desk phone to
> > talk to his parents in India... he paid for a local call, you paid for
> > a call to India.
> >
> > I would recommend that you create a CFA CSS to include just Voicemail
> > and local calling.
> >
> > Maybe include your local area codes (so people can CFA to their
> > cells), but don't go crazy until you are sure you want to.
> >
> >
> > Jonathan
> >
> > On 10/16/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> > >
> > >
> > >
> > >
> > > Check your CFA CSS.
> > >
> > >
> > >
> > >
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> > > James Grace
> > >  Sent: Tuesday, October 16, 2007 9:41 AM
> > >  To: cisco-voip at puck.nether.net
> > >  Subject: [cisco-voip] CM 6.0 unable to Cfwdall
> > >
> > >
> > >
> > >
> > > ?
> > >
> > >
> > >
> > > for some reason im not able to hit the  Cfwdall button in CM 6.0.  I
> > can
> > > dial any pattern i want to, but when i press cfwdall its not allowing
> > any
> > > digits.  It give me "your call can not be completed.  Never ran into
> > this on
> > > previous ver.
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > James Grace
> > >
> > >
> > >
> > >
> > >
> > > Systems Engineer
> > >
> > >
> > > Adcap Network Systems, Inc.
> > >
> > >
> > > 6865 Shiloh Road East, Suite 250
> > >
> > >
> > > Alpharetta, GA 30005
> > >
> > >
> > >
> > >
> > >
> > > office: 678-456-6986
> > >
> > >
> > > direct: 678-456-6761
> > >
> > >
> > > jgrace at adcapnet.com
> > >
> > >
> > > www.adcapnet.com Disclaimer For ADCAP Network Systems
> > >
> > > "The information transmitted is intended only for the person or entity
> > to
> > > which it is addressed and may
> > > contain confidential and/or privileged material. Any review,
> > retransmission,
> > > dissemination or other use of,
> > > or taking of any action in reliance upon this information by persons
> > or
> > > entities other than the intended
> > > recipient is prohibited. If you received this message in error, please
> > > contact the sender and delete the
> > > material from all computers."
> > >
> > > This original email was sent to the internet for delivery at 16 Oct
> > 2007
> > > 09:39:53 -0400
> > >
> > >
> > >
> > > ________________________________
> > >
> > >
> > >
> > >
> > >  Disclaimer: This e-mail communication and any attachments may contain
> > > confidential and privileged information and is for use by the
> > designated
> > > addressee(s) named above only. If you are not the intended addressee,
> > you
> > > are hereby notified that you have received this communication in error
> > and
> > > that any use or reproduction of this email or its contents is strictly
> > > prohibited and may be unlawful. If you have received this
> > communication in
> > > error, please notify us immediately by replying to this message and
> > deleting
> > > it from your computer. Thank you.
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the
> > designated addressee(s) named above only.  If you are not the
> > intended addressee, you are hereby notified that you have received
> > this communication in error and that any use or reproduction of
> > this email or its contents is strictly prohibited and may be
> > unlawful.  If you have received this communication in error, please
> > notify us immediately by replying to this message and deleting it
> > from your computer. Thank you.
> >
> _______________________________________________
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>
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From jonvoip at gmail.com  Wed Oct 17 00:35:27 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 23:35:27 -0500
Subject: [cisco-voip] CM 6.0 unable to Cfwdall
In-Reply-To: <d4c697ad0710162123l4ded24e9mab649d18e67930c6@mail.gmail.com>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E89C@mail3.adcapnetsvc.local>
	<C1FE15183DA37645BC0633BC604E44F0071B68B8@USNAEXCH.na.didata.local>
	<5d093f9a0710161017w23b1a3c8r6460fdb138fd27e5@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0071FF3DB@USNAEXCH.na.didata.local>
	<5d093f9a0710161151v2ff2a66dmf57bef2955517902@mail.gmail.com>
	<d4c697ad0710162123l4ded24e9mab649d18e67930c6@mail.gmail.com>
Message-ID: <5d093f9a0710162135i6c8a5645l501d0e257e6f91d5@mail.gmail.com>

Actually it was some guy calling Kerala and talking for hours in Malayalam



Jonathan

On 10/16/07, Aman Chugh <aman.chugh at gmail.com> wrote:
> Did you have some one calling Maddrrass , the city is now named Chennai :)
>
> Aman
>
>
>
> On 10/17/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > Better to block it in the first place... I would much rather have
> > people whine they couldn't make a call, then have some CIO yelling at
> > me cuz I allowed the call...
> >
> > Then again, we make this as part of the design meetings... with
> > appropriate 'the sky is falling' warnings...
> >
> > Jonathan
> >
> > On 10/16/07, Matt Slaga (US) < Matt.Slaga at us.didata.com> wrote:
> > > That is why departmental bill-back is such a great thing.  Managers that
> > > see this type of expense will squash it quicker than any IT guy could!
> > >
> > > -----Original Message-----
> > > From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > > Sent: Tuesday, October 16, 2007 1:17 PM
> > > To: Matt Slaga (US)
> > > Cc: James Grace; cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] CM 6.0 unable to Cfwdall
> > >
> > > Well, yes, it is most likely a CSS issue for CFA...
> > >
> > > However, before changing the CFA CSS to LD or whatever, make sure you
> > > want to let people do that, because there is a great potential for
> > > toll fraud.
> > >
> > > For example, employee has CFA CSS set to everywhere.
> > >
> > > He CFA's his phone to Maddrass, goes home and calls his desk phone to
> > > talk to his parents in India... he paid for a local call, you paid for
> > > a call to India.
> > >
> > > I would recommend that you create a CFA CSS to include just Voicemail
> > > and local calling.
> > >
> > > Maybe include your local area codes (so people can CFA to their
> > > cells), but don't go crazy until you are sure you want to.
> > >
> > >
> > > Jonathan
> > >
> > > On 10/16/07, Matt Slaga (US) < Matt.Slaga at us.didata.com> wrote:
> > > >
> > > >
> > > >
> > > >
> > > > Check your CFA CSS.
> > > >
> > > >
> > > >
> > > >
> > > > From: cisco-voip-bounces at puck.nether.net
> > > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf
> Of
> > > > James Grace
> > > >  Sent: Tuesday, October 16, 2007 9:41 AM
> > > >  To: cisco-voip at puck.nether.net
> > > >  Subject: [cisco-voip] CM 6.0 unable to Cfwdall
> > > >
> > > >
> > > >
> > > >
> > > > ?
> > > >
> > > >
> > > >
> > > > for some reason im not able to hit the  Cfwdall button in CM 6.0.  I
> > > can
> > > > dial any pattern i want to, but when i press cfwdall its not allowing
> > > any
> > > > digits.  It give me "your call can not be completed.  Never ran into
> > > this on
> > > > previous ver.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > James Grace
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Systems Engineer
> > > >
> > > >
> > > > Adcap Network Systems, Inc.
> > > >
> > > >
> > > > 6865 Shiloh Road East, Suite 250
> > > >
> > > >
> > > > Alpharetta, GA 30005
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > office: 678-456-6986
> > > >
> > > >
> > > > direct: 678-456-6761
> > > >
> > > >
> > > > jgrace at adcapnet.com
> > > >
> > > >
> > > > www.adcapnet.com Disclaimer For ADCAP Network Systems
> > > >
> > > > "The information transmitted is intended only for the person or entity
> > > to
> > > > which it is addressed and may
> > > > contain confidential and/or privileged material. Any review,
> > > retransmission,
> > > > dissemination or other use of,
> > > > or taking of any action in reliance upon this information by persons
> > > or
> > > > entities other than the intended
> > > > recipient is prohibited. If you received this message in error, please
> > > > contact the sender and delete the
> > > > material from all computers."
> > > >
> > > > This original email was sent to the internet for delivery at 16 Oct
> > > 2007
> > > > 09:39:53 -0400
> > > >
> > > >
> > > >
> > > > ________________________________
> > > >
> > > >
> > > >
> > > >
> > > >  Disclaimer: This e-mail communication and any attachments may contain
> > > > confidential and privileged information and is for use by the
> > > designated
> > > > addressee(s) named above only. If you are not the intended addressee,
> > > you
> > > > are hereby notified that you have received this communication in error
> > > and
> > > > that any use or reproduction of this email or its contents is strictly
> > > > prohibited and may be unlawful. If you have received this
> > > communication in
> > > > error, please notify us immediately by replying to this message and
> > > deleting
> > > > it from your computer. Thank you.
> > > > _______________________________________________
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> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > > -----------------------------------------
> > > Disclaimer:
> > >
> > > This e-mail communication and any attachments may contain
> > > confidential and privileged information and is for use by the
> > > designated addressee(s) named above only.  If you are not the
> > > intended addressee, you are hereby notified that you have received
> > > this communication in error and that any use or reproduction of
> > > this email or its contents is strictly prohibited and may be
> > > unlawful.  If you have received this communication in error, please
> > > notify us immediately by replying to this message and deleting it
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> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From jonvoip at gmail.com  Wed Oct 17 00:45:57 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 16 Oct 2007 23:45:57 -0500
Subject: [cisco-voip] Upgrade from V5.1.1
In-Reply-To: <09881513E25DD549B26BF7F38ECAF8DA498694@caemex01.caecorp.cae.com>
References: <09881513E25DD549B26BF7F38ECAF8DA498694@caemex01.caecorp.cae.com>
Message-ID: <5d093f9a0710162145t1b6abbectfca0c25f40be559d@mail.gmail.com>

You do realize that CCM 6 is on New Product Hold until the end of the
year, right (and will be until 6.1 comes out in January)?

Sorry, not NPH... Cisco is calling it Controlled Introduction now...
cuz NPH had too many negative connotations (e.g. Cisco using you to
field test their stuff).


Jonathan

On 10/16/07, Daniel Martin <martind at cae.com> wrote:
>
>
>
>
> Hi,
>
>
>
> I am hitting a bug on V5.1.1 (core dumps) and TAC told me I need to upgrade
> to V5.1.3. I asked him if I should go to V6.0 and he said yes. I already
> have the V6.0 licenses files and CD's.
>
>
>
> I would like to know if the people on the list recommend I go to V6.0
> directly or I should stick with V5.1 release for a little while longer and
> what are the reasons for your recommendations.
>
>
>
> Is there any major issues with V6.0(1a)?
>
>
>
> Much appreciated.
>
>
>
> Daniel.
>
>
>
> Daniel Martin
>
> Groupe Reseau / Networks Group
>
> CAE Inc.
>
> St-Laurent, Quebec Canada
>
> H4T 1G6
>
> Tel:  1-514-341-2000 x4878
>
> Fax: 1-514-340-5583
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From asobihoudai at yahoo.com  Wed Oct 17 01:15:23 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 16 Oct 2007 22:15:23 -0700 (PDT)
Subject: [cisco-voip] Upgrade from V5.1.1
In-Reply-To: <5d093f9a0710162145t1b6abbectfca0c25f40be559d@mail.gmail.com>
Message-ID: <289295.54822.qm@web57202.mail.re3.yahoo.com>

Neil Patrick Harris?

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> You do realize that CCM 6 is on New Product Hold
> until the end of the
> year, right (and will be until 6.1 comes out in
> January)?
> 
> Sorry, not NPH... Cisco is calling it Controlled
> Introduction now...
> cuz NPH had too many negative connotations (e.g.
> Cisco using you to
> field test their stuff).
> 
> 
> Jonathan
> 
> On 10/16/07, Daniel Martin <martind at cae.com> wrote:
> >
> >
> >
> >
> > Hi,
> >
> >
> >
> > I am hitting a bug on V5.1.1 (core dumps) and TAC
> told me I need to upgrade
> > to V5.1.3. I asked him if I should go to V6.0 and
> he said yes. I already
> > have the V6.0 licenses files and CD's.
> >
> >
> >
> > I would like to know if the people on the list
> recommend I go to V6.0
> > directly or I should stick with V5.1 release for a
> little while longer and
> > what are the reasons for your recommendations.
> >
> >
> >
> > Is there any major issues with V6.0(1a)?
> >
> >
> >
> > Much appreciated.
> >
> >
> >
> > Daniel.
> >
> >
> >
> > Daniel Martin
> >
> > Groupe Reseau / Networks Group
> >
> > CAE Inc.
> >
> > St-Laurent, Quebec Canada
> >
> > H4T 1G6
> >
> > Tel:  1-514-341-2000 x4878
> >
> > Fax: 1-514-340-5583
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



      ____________________________________________________________________________________
Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more!
http://tv.yahoo.com/collections/3658 

From lists at jerkys.org  Wed Oct 17 03:02:25 2007
From: lists at jerkys.org (Jerky)
Date: Wed, 17 Oct 2007 02:02:25 -0500
Subject: [cisco-voip] Home user
In-Reply-To: <00ba01c8104d$fe583920$fb08ab60$@com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
	<00ba01c8104d$fe583920$fb08ab60$@com>
Message-ID: <A8D8E089-82BF-4389-B533-3C53D4F88C1B@jerkys.org>

I haven't been particularly fond of the Sonicwalls either but they  
pre-date me and have been inherited.

currently it looks like this:
Sonicwall (10 user type)?-DSL or Cable-? INTERNET ?---T1 Internet-- 
?26xx router ?--Ethernet---?Sonicwall (VX/PRO type)?--- LAN
                                                      |______?--- DMZ  
network

I'll have to take a closer look at a 2800 router for this. I have one  
available that I typically use in the voice lab. Should the 2800 be  
able to handle things like setting up a DMZ and one-to-one NAT  
mappings (I'm not sure if that is the same term used in the Cisco  
world for it) to internal hosts. From the initial reading I've done  
about it seems there's a fine line between the firewall appliances  
like the Sonicwall and ASA or PIX and 2800/3800 routers since it  
seems the ISRs have hardware level encryption built in.

Thanks,
jeff


On Oct 16, 2007, at 6:40 PM, Curt Shaffer wrote:

> Actually due to a difference between who this customer uses for  
> Server network vs. Phone network; they have been using Sonicwall  
> for normal VPN connectivity, which I personally do not like based  
> on personal experience. But the setup will be as follows:
>
>
>
> 87x router or ASA ?-DSL or Cable-? INTERNET ?---PRI/Internet  
> connection--?2811 router?-Phone LAN
>
>
>
> From: Jerky [mailto:lists at jerkys.org]
> Sent: Tuesday, October 16, 2007 6:32 PM
> To: Linsemier, Matthew
> Cc: Curt Shaffer; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Home user
>
>
>
> This has been kicked around for a while since we moved to  
> CallManager but not much thought has been given to it. I'm trying  
> to understand how your hardware is setup. How would it look,  
> similar to one of these?
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
> <--Ethernet--> LAN
>
>
>
> or
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
> <---> ASA or PIX Firewall <--Ethernet--> LAN
>
>
>
> Is the 3800 used for all your firewalling needs in lieu of  
> something like an ASA or PIX? Sonicwall's are currently in place  
> and haven't worked very well for the remote users it was tested  
> with. The Sonicwalls we have don't have anything similar to what  
> the 871's seem to have in regards to vlans and packet tagging. We  
> would probably kick the Sonicwalls out if something else would work  
> better.
>
>
>
> jeff
>
>
>
> On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
>
>
>
>
> We currently have about 40 production remote home teleworkers that  
> have been deployed using Cisco 871/877 wireless routers and a 7960  
> phones.  We are using a Cisco 3845 series router at the head-end so  
> that we can control QoS tagging on the egress / ingress points of  
> both sides of the VPN tunnel.  We are using a phase 2 DMVPN  
> solution dual-homed to two sites to provide secure redundant  
> connectivity.
>
>
>
> It took me a bit to tweak my router configurations (I started on  
> Cisco 831/837 routers) to get the results that we wanted, but all  
> and all our users are happy.  There is the occasional jitter and  
> packet loss (it is the Internet mind you) but g.729 is working  
> quite well coupled with business cable and DSL services.
>
>
>
> If you have any other questions, feel free to ask.
>
>
>
> Matt
>
>
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net] On Behalf Of Curt Shaffer
> Sent: Monday, October 15, 2007 6:37 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Home user
>
>
>
> I was wondering want everyone out there is using for the situation  
> where you have someone on your CCM or CCME that has 1 phone at a  
> home office. Something tells me an ASA is overkill and I haven?t  
> found solid information that any of the 87x routers support tagging  
> QoS of packets going through the VPN tunnel. We would obviously  
> like to have QoS in place even though it?s not respected at their  
> ISP just to make sure the VPN/Voice packets are leaving their  
> routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be  
> protected by one or more legal privileges. It is intended solely  
> for the use of the addressee identified above. If you are not the  
> intended recipient, any use, disclosure, copying or distribution of  
> this communication is UNAUTHORIZED. Neither this information block,  
> the typed name of the sender, nor anything else in this message is  
> intended to constitute an electronic signature unless a specific  
> statement to the contrary is included in this message. If you have  
> received this communication in error, please immediately contact me  
> and delete this communication from your computer. Thank you.
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>

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From martind at cae.com  Wed Oct 17 07:36:21 2007
From: martind at cae.com (Daniel Martin)
Date: Wed, 17 Oct 2007 07:36:21 -0400
Subject: [cisco-voip] Upgrade from V5.1.1
In-Reply-To: <f1b65f880710161511x1d6de4a0uf14a9b7ade9960d9@mail.gmail.com>
References: <09881513E25DD549B26BF7F38ECAF8DA498694@caemex01.caecorp.cae.com>
	<f1b65f880710161511x1d6de4a0uf14a9b7ade9960d9@mail.gmail.com>
Message-ID: <09881513E25DD549B26BF7F38ECAF8DA4986FA@caemex01.caecorp.cae.com>

Justin,

He initially specified to go to 5.1.3. I then asked if going to v6.0.1a
was fine and he said yes.


-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Tuesday, October 16, 2007 6:11 PM
To: Daniel Martin
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Upgrade from V5.1.1

Did TAC give you specific reason's for pushing you to 6.0?  Did you
ask whether you should upgrade to 5.1.3 or 6?  Or did TAC proactively
recommend 6.0?

I'm in the process of making a similar decision...

Justin

On 10/16/07, Daniel Martin <martind at cae.com> wrote:
>
>
>
>
> Hi,
>
>
>
> I am hitting a bug on V5.1.1 (core dumps) and TAC told me I need to
upgrade
> to V5.1.3. I asked him if I should go to V6.0 and he said yes. I
already
> have the V6.0 licenses files and CD's.
>
>
>
> I would like to know if the people on the list recommend I go to V6.0
> directly or I should stick with V5.1 release for a little while longer
and
> what are the reasons for your recommendations.
>
>
>
> Is there any major issues with V6.0(1a)?
>
>
>
> Much appreciated.
>
>
>
> Daniel.
>
>
>
> Daniel Martin
>
> Groupe Reseau / Networks Group
>
> CAE Inc.
>
> St-Laurent, Quebec Canada
>
> H4T 1G6
>
> Tel:  1-514-341-2000 x4878
>
> Fax: 1-514-340-5583
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From Matt.Slaga at us.didata.com  Wed Oct 17 09:26:19 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 17 Oct 2007 09:26:19 -0400
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
In-Reply-To: <f1b65f880710161505m5989eed5leec418d9ecd76b96@mail.gmail.com>
References: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com><FDC137145729AD438CDCA0F9FC6F7236013FE273@mail.netechcorp.com>
	<f1b65f880710161505m5989eed5leec418d9ecd76b96@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0071FFC39@USNAEXCH.na.didata.local>

I checked with the BU when I had trouble getting it to work.  The
response I received from their group is that is it not supported until
6.0(2).  It is not documented anywhere.  Have any of you been able to
get this part to work?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Tuesday, October 16, 2007 6:06 PM
To: Jason Burton
Cc: cisco voip list
Subject: Re: [cisco-voip] CUPS6 integration to Exchange Calendaring

Jason, Are you speaking about the SSL cert that I use for url such as:

https://exchangeserver/Exchange/Jsteinberg/Calendar/


Matt, where did you hear it wasn't working in CUPS 6.0.1 ?  It is
mentioned in the release notes for 6.0.1.  Is there a bug filed
against it?

http://www.cisco.com/en/US/docs/voice_ip_comm/cups/6_0_1/rel_notes/cup60
1rn.html#wp478329


On 10/16/07, Jason Burton <jburton at netechcorp.com> wrote:
> I ran into a bug with it.  Are you using an extended validation SSL
cert
> (the one that changes IE7's address bar to green)?
>
> _____________________________
> Jason Burton
> Systems Engineer
> CCIE Voice #17319
> NETech Corporation
> 317.660.3010 - Desk
> 317.581.1919 - Fax
> jburton at netechcorp.com
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> Steinberg
> Sent: Tuesday, October 16, 2007 12:17 PM
> To: cisco voip list
> Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
>
> Has anyone been successful with this?
>
> I am trying to get this to work but can't figure out what the 'Trust
> Certificate Subject CN' field is for on the presence gateway page when
> integrating to outlook.
>
> thanks.
> justin
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From mbennett at als-xtn.com  Wed Oct 17 10:23:30 2007
From: mbennett at als-xtn.com (Micah Bennett)
Date: Wed, 17 Oct 2007 09:23:30 -0500
Subject: [cisco-voip] IPCC Scripting with Sub-Flows
Message-ID: <160C116A79522E4B9404937D2E28811D092E05@exchange02.alsnet.als-xtn.com>


Good morning all.

We have IPCC Premium.  I am starting to explore the use of subflows in
some of our script applications.  I am going to use the subflow to check
for things like a date / time comparison for playing our Holiday "We're
Closed" message.  

I have heard from various sources to be careful when using subflows in
the scripting applications because they can cause the call to be counted
twice in the reporting functions.  

Is this true?  Is there a way to avoid this?  

We have a very report driven call center and it is imperative that our
call counts be accurate or we may incur penalties from the clients we
support.  Any feedback on subflow experience in the group is
appreciated.  

Thanks.
Micah Bennett
Telecommunications Admin
Automated License Systems
3055 Lebanon Rd, Ste 2301
Nashville, TN  37214
Extension: 2357
Office: 615-263-4066
Mobile: 615-207-0318
Email: micah.bennett at als-xtn.com
Website: www.als-xtn.com
 









The information contained in this email and any attachments is confidential and may be subject to copyright or other intellectual property protection. If you are not the intended recipient, you are not authorized to use or disclose this information, and we request that you notify us by reply mail or telephone and delete the original message from your mail system.
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From jason.aarons at us.didata.com  Wed Oct 17 10:30:48 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 17 Oct 2007 10:30:48 -0400
Subject: [cisco-voip] CCM trace files gathered via RTMT show timestamps 1
	hour off
Message-ID: <C1FE15183DA37645BC0633BC604E44F0071FFDE6@USNAEXCH.na.didata.local>

The log file creation date is correct on my PC (it's likely created by
RTMT?), but when I open the file the logged data shows 1 hour behind.

 

I ssh in to the CallManager (5.1.2) and is there a show command or
someplace to see what time the server thinks it is?

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
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From svoll.voip at gmail.com  Wed Oct 17 10:40:12 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 17 Oct 2007 07:40:12 -0700
Subject: [cisco-voip] Call handler question - Replacing Nortel express
	messaging?
In-Reply-To: <47156171.3070405@wcyv.com>
References: <040201c805e1$f1d0e9c0$c906010a@harding.edu>
	<2B3745BC9D31514381A6FCC5E72D9373026AA0A8@SMOMAIL.jhacorp.com>
	<09a001c81042$65d430b0$c906010a@harding.edu>
	<47156171.3070405@wcyv.com>
Message-ID: <f84a38d30710170740k7ce5b095h5b53ef61ecc22670@mail.gmail.com>

you can also setup the *extension feature to send calls directly to vm from
a phone but not from the call handler.

Scott


On 10/16/07, Pat Hayes <pat-cv at wcyv.com> wrote:
>
> Are you using the transfer to subscriber functionality elsewhere? If
> not, the easiest thing to do is to set all of the subscriber's transfer
> options to 'send to greeting'. Then you could just use any call handler
> with the option to enable callers to enter an extension.
>
> Another option, with similar behavior, that does take a little extra
> user training is #2. From any call handler that is enabled to allow you
> to enter a subscriber's extension, you can add #2 to the end of the dial
> string to bypass the user's transfer options and go straight to their
> greeting. So, to get to 1000's mailbox, you dial 1000#2.
>
> Other that that, Unity really doesn't have a direct equivalent of
> Express Messaging.
>
> -------- Original Message  --------
> From: John Nunnally <Nunnally at harding.edu>
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Call handler question - Replacing Nortel express
> messaging?
> Date: 10/16/2007 6:17 PM
>
> > Chris, thanks for this reply.  I'm sorry it took me so long to look into
> it,
> > but this solution does not completely address my problem.  This allows
> me to
> > dial a number with some kind of prefix and be routed to a
> greeting.  This is
> > impossible when I am dialing a DID number from outside my call manager.
> >
> > I need a way to call a number in Unity. Once I am talking to Unity,
> enter an
> > extension (or mailbox) number and be transferred directly to the
> greeting
> > rather than ringing the phone.  I have not found a way to do that.  This
> is
> > a feature in Nortel Meridian Mail called "Express Messaging" which is
> very
> > useful in a lot of circumstances.  We are movng from Nortel to Unity and
> I
> > have not come up with a replacement for this feature.  The natives are
> > restless.
> >
> > John N.
> >
> >   _____
> >
> > From: Chris D'amore [mailto:CDAmore at jackhenry.com]
> > Sent: Wednesday, October 03, 2007 12:59 PM
> > To: John Nunnally; cisco-voip at puck.nether.net
> > Subject: RE: [cisco-voip] Call handler question
> >
> >
> >
> > Follow this link.
> >
> >
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
> > a00800dea82.shtml
> >
> >
> >
> >
> >
> >
> >
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Nunnally
> > Sent: Wednesday, October 03, 2007 12:22 PM
> > To: cisco-voip at puck.nether.net
> > Subject: [cisco-voip] Call handler question
> >
> >
> >
> > Is there a way in Unity to allow a caller to enter an extension number
> and
> > leave a message in a mailbox without ringing the phone?  Nortel calls
> this
> > feature "express messaging".
> >
> >
> >
> > In other words, I would like to use "Caller System Transfer" and go
> directly
> > to the subscribers greeting instead of transferring the call.
> >
> >
> >
> > Is there a way to do that?
> >
> >
> >
> > JWN
> >
> >
> >
> > NOTICE: This electronic mail message and any files transmitted with it
> are
> > intended
> >
> > exclusively for the individual or entity to which it is addressed. The
> > message,
> >
> > together with any attachment, may contain confidential and/or privileged
> > information.
> >
> > Any unauthorized review, use, printing, saving, copying, disclosure or
> > distribution
> >
> > is strictly prohibited. If you have received this message in error,
> please
> >
> > immediately advise the sender by reply email and delete all copies.
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From svoll.voip at gmail.com  Wed Oct 17 10:46:46 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 17 Oct 2007 07:46:46 -0700
Subject: [cisco-voip] Home user
In-Reply-To: <A8D8E089-82BF-4389-B533-3C53D4F88C1B@jerkys.org>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
	<00ba01c8104d$fe583920$fb08ab60$@com>
	<A8D8E089-82BF-4389-B533-3C53D4F88C1B@jerkys.org>
Message-ID: <f84a38d30710170746g2e0c973asb8f486ca3d938790@mail.gmail.com>

if your trying to get around the Firewall and lets say your Call manager is
on the DMZ or LAN I think the Phone Proxy is a great option.  No VPN. Well
priced (believe it or not...... ) and easy to use.

I got the Phone proxy with the default 25 phone license pack and had it up
and running in less then 4 hours.  reading the doc's to get it setup.  No
VPN overhead or troubleshooting.

Just my two cents.

Scott


On 10/17/07, Jerky <lists at jerkys.org> wrote:
>
> I haven't been particularly fond of the Sonicwalls either but they
> pre-date me and have been inherited.
>  currently it looks like this:
> Sonicwall (10 user type)?-DSL or Cable-? INTERNET ?---T1 Internet--?26xx
> router ?--Ethernet---?Sonicwall (VX/PRO type)?--- LAN
>                                                      |______?--- DMZ
> network
>
>
> I'll have to take a closer look at a 2800 router for this. I have one
> available that I typically use in the voice lab. Should the 2800 be able to
> handle things like setting up a DMZ and one-to-one NAT mappings (I'm not
> sure if that is the same term used in the Cisco world for it) to internal
> hosts. From the initial reading I've done about it seems there's a fine line
> between the firewall appliances like the Sonicwall and ASA or PIX and
> 2800/3800 routers since it seems the ISRs have hardware level encryption
> built in.
>
>
> Thanks,
> jeff
>
>
>
>  On Oct 16, 2007, at 6:40 PM, Curt Shaffer wrote:
>
>  Actually due to a difference between who this customer uses for Server
> network vs. Phone network; they have been using Sonicwall for normal VPN
> connectivity, which I personally do not like based on personal experience.
> But the setup will be as follows:
>
>
>
> 87x router or ASA ?-DSL or Cable-? INTERNET ?---PRI/Internet connection--?2811
> router?-Phone LAN
>
>
>
> *From:* Jerky [mailto:lists at jerkys.org <lists at jerkys.org>]
> *Sent:* Tuesday, October 16, 2007 6:32 PM
> *To:* Linsemier, Matthew
> *Cc:* Curt Shaffer; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Home user
>
>
>
> This has been kicked around for a while since we moved to CallManager but
> not much thought has been given to it. I'm trying to understand how your
> hardware is setup. How would it look, similar to one of these?
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845
> <--Ethernet--> LAN
>
>
>
> or
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <--->
> ASA or PIX Firewall <--Ethernet--> LAN
>
>
>
> Is the 3800 used for all your firewalling needs in lieu of something like
> an ASA or PIX? Sonicwall's are currently in place and haven't worked very
> well for the remote users it was tested with. The Sonicwalls we have don't
> have anything similar to what the 871's seem to have in regards to vlans and
> packet tagging. We would probably kick the Sonicwalls out if something else
> would work better.
>
>
>
> jeff
>
>
>
> On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
>
>
>
>  We currently have about 40 production remote home teleworkers that have
> been deployed using Cisco 871/877 wireless routers and a 7960 phones.  We
> are using a Cisco 3845 series router at the head-end so that we can control
> QoS tagging on the egress / ingress points of both sides of the VPN tunnel.
> We are using a phase 2 DMVPN solution dual-homed to two sites to provide
> secure redundant connectivity.
>
>
>
> It took me a bit to tweak my router configurations (I started on Cisco
> 831/837 routers) to get the results that we wanted, but all and all our
> users are happy.  There is the occasional jitter and packet loss (it is the
> Internet mind you) but g.729 is working quite well coupled with business
> cable and DSL services.
>
>
>
> If you have any other questions, feel free to ask.
>
>
>
> Matt
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [
> mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether.net>]
> *On Behalf Of *Curt Shaffer
> *Sent:* Monday, October 15, 2007 6:37 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Home user
>
>
>
> I was wondering want everyone out there is using for the situation where
> you have someone on your CCM or CCME that has 1 phone at a home office.
> Something tells me an ASA is overkill and I haven't found solid information
> that any of the 87x routers support tagging QoS of packets going through the
> VPN tunnel. We would obviously like to have QoS in place even though it's
> not respected at their ISP just to make sure the VPN/Voice packets are
> leaving their routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>  ------------------------------
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be
> protected by one or more legal privileges. It is intended solely for the use
> of the addressee identified above. If you are not the intended recipient,
> any use, disclosure, copying or distribution of this communication is
> UNAUTHORIZED. Neither this information block, the typed name of the sender,
> nor anything else in this message is intended to constitute an electronic
> signature unless a specific statement to the contrary is included in this
> message. If you have received this communication in error, please
> immediately contact me and delete this communication from your computer.
> Thank you.
>  ------------------------------
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Nunnally at Harding.edu  Wed Oct 17 10:59:49 2007
From: Nunnally at Harding.edu (John Nunnally)
Date: Wed, 17 Oct 2007 09:59:49 -0500
Subject: [cisco-voip] Call handler question - Replacing Nortel express
 messaging?
In-Reply-To: <f84a38d30710170740k7ce5b095h5b53ef61ecc22670@mail.gmail.com>
References: <040201c805e1$f1d0e9c0$c906010a@harding.edu>
	<2B3745BC9D31514381A6FCC5E72D9373026AA0A8@SMOMAIL.jhacorp.com>
	<09a001c81042$65d430b0$c906010a@harding.edu>
	<47156171.3070405@wcyv.com>
	<f84a38d30710170740k7ce5b095h5b53ef61ecc22670@mail.gmail.com>
Message-ID: <0a3b01c810ce$5d8f29f0$c906010a@harding.edu>

I'm embarrassed to say I was not aware of the built-in "#2" feature.  Other
than being a training issue, that feature meets our needs.
 
Using the asterisk-extension configuration documented by Cisco can be
confusing when asterisk is used elsewhere to escape to a password prompt.
Using a different prefix works, but this whole concept does not work if you
need to reach a greeting directly by dialing a DID number from outside.
Since "#2" is implemented in Unity, it doesn't matter how you get routed
there.
 
Thanks to everyone for the help.
 
John N.  

  _____  

From: Scott Voll [mailto:svoll.voip at gmail.com] 
Sent: Wednesday, October 17, 2007 9:40 AM
To: Pat Hayes
Cc: John Nunnally; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call handler question - Replacing Nortel express
messaging?


you can also setup the *extension feature to send calls directly to vm from
a phone but not from the call handler.
 
Scott

 
On 10/16/07, Pat Hayes <pat-cv at wcyv.com> wrote: 

Are you using the transfer to subscriber functionality elsewhere? If
not, the easiest thing to do is to set all of the subscriber's transfer 
options to 'send to greeting'. Then you could just use any call handler
with the option to enable callers to enter an extension.

Another option, with similar behavior, that does take a little extra
user training is #2. From any call handler that is enabled to allow you
to enter a subscriber's extension, you can add #2 to the end of the dial
string to bypass the user's transfer options and go straight to their 
greeting. So, to get to 1000's mailbox, you dial 1000#2.

Other that that, Unity really doesn't have a direct equivalent of
Express Messaging.

-------- Original Message  --------
From: John Nunnally < Nunnally at harding.edu>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call handler question - Replacing Nortel express 
messaging?
Date: 10/16/2007 6:17 PM

> Chris, thanks for this reply.  I'm sorry it took me so long to look into
it,
> but this solution does not completely address my problem.  This allows me
to
> dial a number with some kind of prefix and be routed to a greeting.  This
is
> impossible when I am dialing a DID number from outside my call manager.
>
> I need a way to call a number in Unity. Once I am talking to Unity, enter
an 
> extension (or mailbox) number and be transferred directly to the greeting
> rather than ringing the phone.  I have not found a way to do that.  This
is
> a feature in Nortel Meridian Mail called "Express Messaging" which is very

> useful in a lot of circumstances.  We are movng from Nortel to Unity and I
> have not come up with a replacement for this feature.  The natives are
> restless.
>
> John N.
>
>   _____ 
>
> From: Chris D'amore [mailto:CDAmore at jackhenry.com]
> Sent: Wednesday, October 03, 2007 12:59 PM
> To: John Nunnally; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Call handler question
>
>
>
> Follow this link.
>
>
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
> a00800dea82.shtml
>
>
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Nunnally
> Sent: Wednesday, October 03, 2007 12:22 PM 
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Call handler question
>
>
>
> Is there a way in Unity to allow a caller to enter an extension number and

> leave a message in a mailbox without ringing the phone?  Nortel calls this
> feature "express messaging".
>
>
>
> In other words, I would like to use "Caller System Transfer" and go
directly 
> to the subscribers greeting instead of transferring the call.
>
>
>
> Is there a way to do that?
>
>
>
> JWN
>
>
>
> NOTICE: This electronic mail message and any files transmitted with it are

> intended
>
> exclusively for the individual or entity to which it is addressed. The
> message,
>
> together with any attachment, may contain confidential and/or privileged
> information. 
>
> Any unauthorized review, use, printing, saving, copying, disclosure or
> distribution
>
> is strictly prohibited. If you have received this message in error, please
>
> immediately advise the sender by reply email and delete all copies. 
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list 
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



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From MLinsemier at apcapital.com  Wed Oct 17 11:19:07 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 17 Oct 2007 11:19:07 -0400
Subject: [cisco-voip] Home user
In-Reply-To: <513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
Message-ID: <AB0A35F2486F81468971AA68558BE719073BF05C@smieln01winm01.micoanet.com>

In our environment we utilize PIX firewalls (still have to upgrade to
ASA's) to handle our firewall needs and then use the 3800 series router
just to terminate the DMVPN home users.  They are deployed in parallel
and sit behind a perimeter screening router (another 3800 series
router).  We shied away from using the PIX for the simple fact that
while it would preserve QoS markings, we couldn't do any remarking or
shaping in the device.  Maybe this has changed in the ASA, but I don't
think you have the control like you do in IOS (such as qos pre-classify,
shaping, policing,  etc.).  Depending on how many tunnels you plan on
using, you could use a router much smaller than a 3800 series to
terminate the end nodes.

 

On the home user end we have the Cisco 871/877 routers configured to
support wired and wireless connections using three VLANS.  We have a
VLAN configured for corporate connectivity, one VLAN configured as a
voice VLAN, and then a VLAN configured for untrusted traffic.  One
Ethernet port on the router provides connectivity to the corporate and
voice VLANS, while the remaining three are configured as untrusted.
Similarly with Wireless, we extend PEAP authentication from the
headquarters and authenticate users to the corporate VLAN, and use a
WPA-PSK to secure the untrusted connections.  This way the users plug in
their phone, then their laptop/docking station to port 0, and any other
home devices can be connected to port 1-3 or use the wireless WPA-PSK
network and be logically segregated (using ACL's) from any data on the
corporate network.  This way we can also control QoS and mark down all
traffic that enters the router from the untrusted network.  So when said
employees son or daughter starts downing a 2 gig torrent from a home PC,
they don't kill the voice or impact the corporate workflow.  Eventually
we will be implementing 802.1x on the corporate port for additional
security, but have had mixed results of getting it to work with Windows
XP.


Hope this helps.

 

Matt

 

 

From: Jerky [mailto:lists at jerkys.org] 
Sent: Tuesday, October 16, 2007 6:32 PM
To: Linsemier, Matthew
Cc: Curt Shaffer; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Home user

 

This has been kicked around for a while since we moved to CallManager
but not much thought has been given to it. I'm trying to understand how
your hardware is setup. How would it look, similar to one of these?

 

87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845
<--Ethernet--> LAN

 

or

 

87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <--->
ASA or PIX Firewall <--Ethernet--> LAN

 

Is the 3800 used for all your firewalling needs in lieu of something
like an ASA or PIX? Sonicwall's are currently in place and haven't
worked very well for the remote users it was tested with. The Sonicwalls
we have don't have anything similar to what the 871's seem to have in
regards to vlans and packet tagging. We would probably kick the
Sonicwalls out if something else would work better.

 

jeff

 

On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:





We currently have about 40 production remote home teleworkers that have
been deployed using Cisco 871/877 wireless routers and a 7960 phones.
We are using a Cisco 3845 series router at the head-end so that we can
control QoS tagging on the egress / ingress points of both sides of the
VPN tunnel.  We are using a phase 2 DMVPN solution dual-homed to two
sites to provide secure redundant connectivity.

 

It took me a bit to tweak my router configurations (I started on Cisco
831/837 routers) to get the results that we wanted, but all and all our
users are happy.  There is the occasional jitter and packet loss (it is
the Internet mind you) but g.729 is working quite well coupled with
business cable and DSL services.

 

If you have any other questions, feel free to ask.

 

Matt

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Curt Shaffer
Sent: Monday, October 15, 2007 6:37 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Home user

 

I was wondering want everyone out there is using for the situation where
you have someone on your CCM or CCME that has 1 phone at a home office.
Something tells me an ASA is overkill and I haven't found solid
information that any of the 87x routers support tagging QoS of packets
going through the VPN tunnel. We would obviously like to have QoS in
place even though it's not respected at their ISP just to make sure the
VPN/Voice packets are leaving their routers first as a best effort to
get some quality. 

 

Thanks

 

 

________________________________

CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may be
protected by one or more legal privileges. It is intended solely for the
use of the addressee identified above. If you are not the intended
recipient, any use, disclosure, copying or distribution of this
communication is UNAUTHORIZED. Neither this information block, the typed
name of the sender, nor anything else in this message is intended to
constitute an electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete this
communication from your computer. Thank you. 

________________________________

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

 


CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may
be protected by one or more legal privileges. It is intended
solely for the use of the addressee identified above. If you
are not the intended recipient, any use, disclosure, copying
or distribution of this communication is UNAUTHORIZED. Neither
this information block, the typed name of the sender, nor
anything else in this message is intended to constitute an
electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete
this communication from your computer. Thank you.
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From lelio at uoguelph.ca  Wed Oct 17 11:51:20 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 17 Oct 2007 11:51:20 -0400
Subject: [cisco-voip] bulk edit vg224??
Message-ID: <020f01c810d5$8f97ae20$56196883@cfs.uoguelph.ca>

I can't remember - can I bulk load VG224?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR
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From wsisk at cisco.com  Wed Oct 17 13:26:30 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 17 Oct 2007 13:26:30 -0400
Subject: [cisco-voip] bulk edit vg224??
In-Reply-To: <020f01c810d5$8f97ae20$56196883@cfs.uoguelph.ca>
References: <020f01c810d5$8f97ae20$56196883@cfs.uoguelph.ca>
Message-ID: <471645C6.6030305@cisco.com>

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From lists at jerkys.org  Wed Oct 17 13:43:44 2007
From: lists at jerkys.org (Jerky)
Date: Wed, 17 Oct 2007 12:43:44 -0500
Subject: [cisco-voip] Home user
In-Reply-To: <f84a38d30710170746g2e0c973asb8f486ca3d938790@mail.gmail.com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
	<00ba01c8104d$fe583920$fb08ab60$@com>
	<A8D8E089-82BF-4389-B533-3C53D4F88C1B@jerkys.org>
	<f84a38d30710170746g2e0c973asb8f486ca3d938790@mail.gmail.com>
Message-ID: <814895AC-04B0-4383-A7D7-88A90A3767C9@jerkys.org>

I saw your mention of that earlier. It sounds interesting and fairly  
simple. I haven't gotten to spend much time reading up on it yet so  
correct me if I'm wrong but it sounds like this is geared towards  
phones only. Something  that's starting to get getting kicked around  
also is IPCCX which I think has been rebadged as UPCCX to keep with  
the whole Unified naming scheme. It's another thing I've just started  
to look into but from what I get there is a desktop client that goes  
with it, which I'm not sure what the communication protocols it uses  
yet. I think the goal is to have a few home based service people that  
aren't out on calls use their down time to help man the phones via an  
IP phone at home.


jeff

On Oct 17, 2007, at 9:46 AM, Scott Voll wrote:

> if your trying to get around the Firewall and lets say your Call  
> manager is on the DMZ or LAN I think the Phone Proxy is a great  
> option.  No VPN. Well priced (believe it or not...... ) and easy to  
> use.
>
> I got the Phone proxy with the default 25 phone license pack and  
> had it up and running in less then 4 hours.  reading the doc's to  
> get it setup.  No VPN overhead or troubleshooting.
>
> Just my two cents.
>
> Scott
>
>
> On 10/17/07, Jerky <lists at jerkys.org> wrote:
> I haven't been particularly fond of the Sonicwalls either but they  
> pre-date me and have been inherited.
>
> currently it looks like this:
> Sonicwall (10 user type) ?-DSL or Cable- ? INTERNET ?---T1  
> Internet-- ?26xx router ?--Ethernet--- ?Sonicwall (VX/PRO type)?  
> --- LAN
>                                                      |______? ---  
> DMZ network
>
>
> I'll have to take a closer look at a 2800 router for this. I have  
> one available that I typically use in the voice lab. Should the  
> 2800 be able to handle things like setting up a DMZ and one-to-one  
> NAT mappings (I'm not sure if that is the same term used in the  
> Cisco world for it) to internal hosts. From the initial reading  
> I've done about it seems there's a fine line between the firewall  
> appliances like the Sonicwall and ASA or PIX and 2800/3800 routers  
> since it seems the ISRs have hardware level encryption built in.
>
>
> Thanks,
> jeff
>
>
>
> On Oct 16, 2007, at 6:40 PM, Curt Shaffer wrote:
>
>> Actually due to a difference between who this customer uses for  
>> Server network vs. Phone network; they have been using Sonicwall  
>> for normal VPN connectivity, which I personally do not like based  
>> on personal experience. But the setup will be as follows:
>>
>>
>> 87x router or ASA ?-DSL or Cable- ? INTERNET ?---PRI/Internet  
>> connection-- ?2811 router ? -Phone LAN
>>
>>
>> From: Jerky [mailto:lists at jerkys.org]
>> Sent: Tuesday, October 16, 2007 6:32 PM
>> To: Linsemier, Matthew
>> Cc: Curt Shaffer; cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] Home user
>>
>>
>> This has been kicked around for a while since we moved to  
>> CallManager but not much thought has been given to it. I'm trying  
>> to understand how your hardware is setup. How would it look,  
>> similar to one of these?
>>
>>
>> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
>> <--Ethernet--> LAN
>>
>>
>> or
>>
>>
>> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
>> <---> ASA or PIX Firewall <--Ethernet--> LAN
>>
>>
>> Is the 3800 used for all your firewalling needs in lieu of  
>> something like an ASA or PIX? Sonicwall's are currently in place  
>> and haven't worked very well for the remote users it was tested  
>> with. The Sonicwalls we have don't have anything similar to what  
>> the 871's seem to have in regards to vlans and packet tagging. We  
>> would probably kick the Sonicwalls out if something else would  
>> work better.
>>
>>
>> jeff
>>
>>
>> On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
>>
>>
>>
>>
>> We currently have about 40 production remote home teleworkers that  
>> have been deployed using Cisco 871/877 wireless routers and a 7960  
>> phones.  We are using a Cisco 3845 series router at the head-end  
>> so that we can control QoS tagging on the egress / ingress points  
>> of both sides of the VPN tunnel.  We are using a phase 2 DMVPN  
>> solution dual-homed to two sites to provide secure redundant  
>> connectivity.
>>
>>
>> It took me a bit to tweak my router configurations (I started on  
>> Cisco 831/837 routers) to get the results that we wanted, but all  
>> and all our users are happy.  There is the occasional jitter and  
>> packet loss (it is the Internet mind you) but g.729 is working  
>> quite well coupled with business cable and DSL services.
>>
>>
>> If you have any other questions, feel free to ask.
>>
>>
>> Matt
>>
>>
>> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
>> bounces at puck.nether.net ] On Behalf Of Curt Shaffer
>> Sent: Monday, October 15, 2007 6:37 PM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] Home user
>>
>>
>> I was wondering want everyone out there is using for the situation  
>> where you have someone on your CCM or CCME that has 1 phone at a  
>> home office. Something tells me an ASA is overkill and I haven't  
>> found solid information that any of the 87x routers support  
>> tagging QoS of packets going through the VPN tunnel. We would  
>> obviously like to have QoS in place even though it's not respected  
>> at their ISP just to make sure the VPN/Voice packets are leaving  
>> their routers first as a best effort to get some quality.
>>
>>
>> Thanks
>>
>>
>>
>> CONFIDENTIALITY STATEMENT
>> This communication and any attachments are CONFIDENTIAL and may be  
>> protected by one or more legal privileges. It is intended solely  
>> for the use of the addressee identified above. If you are not the  
>> intended recipient, any use, disclosure, copying or distribution  
>> of this communication is UNAUTHORIZED. Neither this information  
>> block, the typed name of the sender, nor anything else in this  
>> message is intended to constitute an electronic signature unless a  
>> specific statement to the contrary is included in this message. If  
>> you have received this communication in error, please immediately  
>> contact me and delete this communication from your computer. Thank  
>> you.
>>
>> _______________________________________________
>>
>> cisco-voip mailing list
>>
>> cisco-voip at puck.nether.net
>>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

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From svoll.voip at gmail.com  Wed Oct 17 13:50:08 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 17 Oct 2007 10:50:08 -0700
Subject: [cisco-voip] Home user
In-Reply-To: <814895AC-04B0-4383-A7D7-88A90A3767C9@jerkys.org>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
	<00ba01c8104d$fe583920$fb08ab60$@com>
	<A8D8E089-82BF-4389-B533-3C53D4F88C1B@jerkys.org>
	<f84a38d30710170746g2e0c973asb8f486ca3d938790@mail.gmail.com>
	<814895AC-04B0-4383-A7D7-88A90A3767C9@jerkys.org>
Message-ID: <f84a38d30710171050u5d1de854jf1292f1943339a95@mail.gmail.com>

Not sure if it's TAC supported but I just tested it the other day because we
want the helpdesk to answer calls if we are snowed out and it does work.
our agents log into the client via Citrix (not sure if that's tac supported
either).  All and all it does work.  CM 4.1.3 / IPCCx 4.0.1 / Unity 4.0.5

Not that we don't also give the option to VPN the computer in, but the Voice
doesn't have the extra overhead of IPSEC.

Scott


On 10/17/07, Jerky <lists at jerkys.org> wrote:
>
> I saw your mention of that earlier. It sounds interesting and fairly
> simple. I haven't gotten to spend much time reading up on it yet so correct
> me if I'm wrong but it sounds like this is geared towards phones only.
> Something  that's starting to get getting kicked around also is IPCCX which
> I think has been rebadged as UPCCX to keep with the whole Unified naming
> scheme. It's another thing I've just started to look into but from what I
> get there is a desktop client that goes with it, which I'm not sure what
> the communication protocols it uses yet. I think the goal is to have a few
> home based service people that aren't out on calls use their down time to
> help man the phones via an IP phone at home.
>
>
> jeff
>
>  On Oct 17, 2007, at 9:46 AM, Scott Voll wrote:
>
>  if your trying to get around the Firewall and lets say your Call manager
> is on the DMZ or LAN I think the Phone Proxy is a great option.  No VPN.
> Well priced (believe it or not...... ) and easy to use.
>
> I got the Phone proxy with the default 25 phone license pack and had it up
> and running in less then 4 hours.  reading the doc's to get it setup.  No
> VPN overhead or troubleshooting.
>
> Just my two cents.
>
> Scott
>
>
> On 10/17/07, Jerky <lists at jerkys.org> wrote:
> >
> > I haven't been particularly fond of the Sonicwalls either but they
> > pre-date me and have been inherited.
> >  currently it looks like this:
> > Sonicwall (10 user type) ?-DSL or Cable- ? INTERNET ?---T1 Internet-- ?26xx
> > router ?--Ethernet--- ?Sonicwall (VX/PRO type)? --- LAN
> >                                                      |______? --- DMZ
> > network
> >
> >
> > I'll have to take a closer look at a 2800 router for this. I have one
> > available that I typically use in the voice lab. Should the 2800 be able to
> > handle things like setting up a DMZ and one-to-one NAT mappings (I'm not
> > sure if that is the same term used in the Cisco world for it) to internal
> > hosts. From the initial reading I've done about it seems there's a fine line
> > between the firewall appliances like the Sonicwall and ASA or PIX and
> > 2800/3800 routers since it seems the ISRs have hardware level encryption
> > built in.
> >
> >
> > Thanks,
> > jeff
> >
> >
> >
> >  On Oct 16, 2007, at 6:40 PM, Curt Shaffer wrote:
> >
> >  Actually due to a difference between who this customer uses for Server
> > network vs. Phone network; they have been using Sonicwall for normal VPN
> > connectivity, which I personally do not like based on personal experience.
> > But the setup will be as follows:
> >
> >
> >
> > 87x router or ASA ?-DSL or Cable- ? INTERNET ?---PRI/Internet
> > connection-- ?2811 router ? -Phone LAN
> >
> >
> >
> > *From:* Jerky [mailto:lists at jerkys.org <lists at jerkys.org>]
> > *Sent:* Tuesday, October 16, 2007 6:32 PM
> > *To:* Linsemier, Matthew
> > *Cc: *Curt Shaffer; cisco-voip at puck.nether.net
> > *Subject:* Re: [cisco-voip] Home user
> >
> >
> >
> > This has been kicked around for a while since we moved to CallManager
> > but not much thought has been given to it. I'm trying to understand how your
> > hardware is setup. How would it look, similar to one of these?
> >
> >
> >
> > 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845
> > <--Ethernet--> LAN
> >
> >
> >
> > or
> >
> >
> >
> > 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <--->
> > ASA or PIX Firewall <--Ethernet--> LAN
> >
> >
> >
> > Is the 3800 used for all your firewalling needs in lieu of something
> > like an ASA or PIX? Sonicwall's are currently in place and haven't worked
> > very well for the remote users it was tested with. The Sonicwalls we have
> > don't have anything similar to what the 871's seem to have in regards to
> > vlans and packet tagging. We would probably kick the Sonicwalls out if
> > something else would work better.
> >
> >
> >
> > jeff
> >
> >
> >
> > On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
> >
> >
> >
> >  We currently have about 40 production remote home teleworkers that have
> > been deployed using Cisco 871/877 wireless routers and a 7960 phones.  We
> > are using a Cisco 3845 series router at the head-end so that we can control
> > QoS tagging on the egress / ingress points of both sides of the VPN tunnel.
> > We are using a phase 2 DMVPN solution dual-homed to two sites to provide
> > secure redundant connectivity.
> >
> >
> >
> > It took me a bit to tweak my router configurations (I started on Cisco
> > 831/837 routers) to get the results that we wanted, but all and all our
> > users are happy.  There is the occasional jitter and packet loss (it is the
> > Internet mind you) but g.729 is working quite well coupled with business
> > cable and DSL services.
> >
> >
> >
> > If you have any other questions, feel free to ask.
> >
> >
> >
> > Matt
> >
> >
> >
> > *From: *cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net
> > <cisco-voip-bounces at puck.nether.net>] *On Behalf Of *Curt Shaffer
> > *Sent:* Monday, October 15, 2007 6:37 PM
> > *To:* cisco-voip at puck.nether.net
> > *Subject:* [cisco-voip] Home user
> >
> >
> >
> > I was wondering want everyone out there is using for the situation where
> > you have someone on your CCM or CCME that has 1 phone at a home office.
> > Something tells me an ASA is overkill and I haven't found solid information
> > that any of the 87x routers support tagging QoS of packets going through the
> > VPN tunnel. We would obviously like to have QoS in place even though it's
> > not respected at their ISP just to make sure the VPN/Voice packets are
> > leaving their routers first as a best effort to get some quality.
> >
> >
> >
> > Thanks
> >
> >
> >
> >
> >  ------------------------------
> >
> > CONFIDENTIALITY STATEMENT
> > This communication and any attachments are CONFIDENTIAL and may be
> > protected by one or more legal privileges. It is intended solely for the use
> > of the addressee identified above. If you are not the intended recipient,
> > any use, disclosure, copying or distribution of this communication is
> > UNAUTHORIZED. Neither this information block, the typed name of the sender,
> > nor anything else in this message is intended to constitute an electronic
> > signature unless a specific statement to the contrary is included in this
> > message. If you have received this communication in error, please
> > immediately contact me and delete this communication from your computer.
> > Thank you.
> >  ------------------------------
> >
> > _______________________________________________
> >
> > cisco-voip mailing list
> >
> > cisco-voip at puck.nether.net
> >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
>
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From lists at jerkys.org  Wed Oct 17 13:52:52 2007
From: lists at jerkys.org (Jerky)
Date: Wed, 17 Oct 2007 12:52:52 -0500
Subject: [cisco-voip] Home user
In-Reply-To: <AB0A35F2486F81468971AA68558BE719073BF05C@smieln01winm01.micoanet.com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
	<AB0A35F2486F81468971AA68558BE719073BF05C@smieln01winm01.micoanet.com>
Message-ID: <6A059BC4-F184-4563-B117-1D28BCDAFFA9@jerkys.org>

so it would be more like this:

Cisco 871
      |
DSL CABLE
      |
Internet
      |
T1 Connection (Serial0/0/0)
                     |
    _____ 3800 _____
   |                                  |
ethernet 0/0          ethernet 0/1
|                                    |
PIX/ASA                    3800 (Cisco 871 VPN's terminate here)
|                                    |
LAN(computers)       LAN (Voice)


Homefully my crude diagram makes sense. Do your home users have  
access to any data on the computer network side. Or is the 87x VPNs  
solely for getting to the voice network If users access things on the  
"computer" side would you have a separate tunnel setup just for that?

Thanks for so much helping enlighten me. It's been very helpful.

jeff



On Oct 17, 2007, at 10:19 AM, Linsemier, Matthew wrote:

> In our environment we utilize PIX firewalls (still have to upgrade  
> to ASA?s) to handle our firewall needs and then use the 3800 series  
> router just to terminate the DMVPN home users.  They are deployed  
> in parallel and sit behind a perimeter screening router (another  
> 3800 series router).  We shied away from using the PIX for the  
> simple fact that while it would preserve QoS markings, we couldn?t  
> do any remarking or shaping in the device.  Maybe this has changed  
> in the ASA, but I don?t think you have the control like you do in  
> IOS (such as qos pre-classify, shaping, policing,  etc.).   
> Depending on how many tunnels you plan on using, you could use a  
> router much smaller than a 3800 series to terminate the end nodes.
>
>
>
> On the home user end we have the Cisco 871/877 routers configured  
> to support wired and wireless connections using three VLANS.  We  
> have a VLAN configured for corporate connectivity, one VLAN  
> configured as a voice VLAN, and then a VLAN configured for  
> untrusted traffic.  One Ethernet port on the router provides  
> connectivity to the corporate and voice VLANS, while the remaining  
> three are configured as untrusted.  Similarly with Wireless, we  
> extend PEAP authentication from the headquarters and authenticate  
> users to the corporate VLAN, and use a WPA-PSK to secure the  
> untrusted connections.  This way the users plug in their phone,  
> then their laptop/docking station to port 0, and any other home  
> devices can be connected to port 1-3 or use the wireless WPA-PSK  
> network and be logically segregated (using ACL?s) from any data on  
> the corporate network.  This way we can also control QoS and mark  
> down all traffic that enters the router from the untrusted  
> network.  So when said employees son or daughter starts downing a 2  
> gig torrent from a home PC, they don?t kill the voice or impact the  
> corporate workflow.  Eventually we will be implementing 802.1x on  
> the corporate port for additional security, but have had mixed  
> results of getting it to work with Windows XP.
>
>
> Hope this helps.
>
>
>
> Matt
>
>
>
>
>
> From: Jerky [mailto:lists at jerkys.org]
> Sent: Tuesday, October 16, 2007 6:32 PM
> To: Linsemier, Matthew
> Cc: Curt Shaffer; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Home user
>
>
>
> This has been kicked around for a while since we moved to  
> CallManager but not much thought has been given to it. I'm trying  
> to understand how your hardware is setup. How would it look,  
> similar to one of these?
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
> <--Ethernet--> LAN
>
>
>
> or
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
> <---> ASA or PIX Firewall <--Ethernet--> LAN
>
>
>
> Is the 3800 used for all your firewalling needs in lieu of  
> something like an ASA or PIX? Sonicwall's are currently in place  
> and haven't worked very well for the remote users it was tested  
> with. The Sonicwalls we have don't have anything similar to what  
> the 871's seem to have in regards to vlans and packet tagging. We  
> would probably kick the Sonicwalls out if something else would work  
> better.
>
>
>
> jeff
>
>
>
> On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
>
>
>
>
> We currently have about 40 production remote home teleworkers that  
> have been deployed using Cisco 871/877 wireless routers and a 7960  
> phones.  We are using a Cisco 3845 series router at the head-end so  
> that we can control QoS tagging on the egress / ingress points of  
> both sides of the VPN tunnel.  We are using a phase 2 DMVPN  
> solution dual-homed to two sites to provide secure redundant  
> connectivity.
>
>
>
> It took me a bit to tweak my router configurations (I started on  
> Cisco 831/837 routers) to get the results that we wanted, but all  
> and all our users are happy.  There is the occasional jitter and  
> packet loss (it is the Internet mind you) but g.729 is working  
> quite well coupled with business cable and DSL services.
>
>
>
> If you have any other questions, feel free to ask.
>
>
>
> Matt
>
>
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net] On Behalf Of Curt Shaffer
> Sent: Monday, October 15, 2007 6:37 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Home user
>
>
>
> I was wondering want everyone out there is using for the situation  
> where you have someone on your CCM or CCME that has 1 phone at a  
> home office. Something tells me an ASA is overkill and I haven?t  
> found solid information that any of the 87x routers support tagging  
> QoS of packets going through the VPN tunnel. We would obviously  
> like to have QoS in place even though it?s not respected at their  
> ISP just to make sure the VPN/Voice packets are leaving their  
> routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be  
> protected by one or more legal privileges. It is intended solely  
> for the use of the addressee identified above. If you are not the  
> intended recipient, any use, disclosure, copying or distribution of  
> this communication is UNAUTHORIZED. Neither this information block,  
> the typed name of the sender, nor anything else in this message is  
> intended to constitute an electronic signature unless a specific  
> statement to the contrary is included in this message. If you have  
> received this communication in error, please immediately contact me  
> and delete this communication from your computer. Thank you.
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be  
> protected by one or more legal privileges. It is intended solely  
> for the use of the addressee identified above. If you are not the  
> intended recipient, any use, disclosure, copying or distribution of  
> this communication is UNAUTHORIZED. Neither this information block,  
> the typed name of the sender, nor anything else in this message is  
> intended to constitute an electronic signature unless a specific  
> statement to the contrary is included in this message. If you have  
> received this communication in error, please immediately contact me  
> and delete this communication from your computer. Thank you.
>
>
>

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From ano-1 at worldonline.dk  Wed Oct 17 13:59:17 2007
From: ano-1 at worldonline.dk (Katrine og Michael)
Date: Wed, 17 Oct 2007 19:59:17 +0200
Subject: [cisco-voip] Script for AutoAttendant UnityExpress
Message-ID: <005801c810e7$6fb76980$0200000a@ano>

Hello

Can somebody help me?

I really need a script for unity express. The script will be used when a phone i busy. When the phone is busy the calling person will hear a voice saying: press 1 to callcontact, press 2 for try agin and press 3 for end call. When 1 is pressed the calling person will be transferred to callcontact. When 2 is pressed the calling person will put in cue whith music and the it will try to contact the called person again. When 3 is pressed the script will terminate the call.

Can sombody please help me? I?ve attatched my script, but it is not working.

your sincerely

Michael from Denmark :-)

-- 
Jeg beskyttes af den gratis SPAMfighter til privatbrugere.
Den har indtil videre sparet mig for at f? 696 spam-mails.
Betalende brugere f?r ikke denne besked i deres e-mails.
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From svoll.voip at gmail.com  Wed Oct 17 14:05:12 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 17 Oct 2007 11:05:12 -0700
Subject: [cisco-voip] Home user
In-Reply-To: <6A059BC4-F184-4563-B117-1D28BCDAFFA9@jerkys.org>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
	<AB0A35F2486F81468971AA68558BE719073BF05C@smieln01winm01.micoanet.com>
	<6A059BC4-F184-4563-B117-1D28BCDAFFA9@jerkys.org>
Message-ID: <f84a38d30710171105n6977175by2725263a1bc4afc@mail.gmail.com>

no Cisco CPE required.

ip phone
  |
internet connection
  |
  --------------------------- phone proxy
  |                               |
  |                                |
FW / router                  |
  |                                |
internal network ------ voice network

basically you save the money of a cisco CPE by getting the phone proxy.  let
the end users VPN in with the client for data purposes or use Citrix to get
around VPN all together.

the phone proxy has a north / South interface so the only thing going
through it is the authenticated voice traffic.

hope that's understandable.

scott




On 10/17/07, Jerky <lists at jerkys.org> wrote:
>
> so it would be more like this:
>
> Cisco 871
>      |
> DSL CABLE
>      |
> Internet
>      |
> T1 Connection (Serial0/0/0)
>                     |
>    _____ 3800 _____
>   |                                  |
> ethernet 0/0          ethernet 0/1
> |                                    |
> PIX/ASA                    3800 (Cisco 871 VPN's terminate here)
> |                                    |
> LAN(computers)       LAN (Voice)
>
>
>
>
> Homefully my crude diagram makes sense. Do your home users have access to
> any data on the computer network side. Or is the 87x VPNs solely for getting
> to the voice network If users access things on the "computer" side would you
> have a separate tunnel setup just for that?
>
>
> Thanks for so much helping enlighten me. It's been very helpful.
>
>
> jeff
>
>
>
>
>
>
>  On Oct 17, 2007, at 10:19 AM, Linsemier, Matthew wrote:
>
>  In our environment we utilize PIX firewalls (still have to upgrade to
> ASA's) to handle our firewall needs and then use the 3800 series router just
> to terminate the DMVPN home users.  They are deployed in parallel and sit
> behind a perimeter screening router (another 3800 series router).  We shied
> away from using the PIX for the simple fact that while it would preserve QoS
> markings, we couldn't do any remarking or shaping in the device.  Maybe this
> has changed in the ASA, but I don't think you have the control like you do
> in IOS (such as qos pre-classify, shaping, policing,  etc.).  Depending on
> how many tunnels you plan on using, you could use a router much smaller than
> a 3800 series to terminate the end nodes.
>
>
>
> On the home user end we have the Cisco 871/877 routers configured to
> support wired and wireless connections using three VLANS.  We have a VLAN
> configured for corporate connectivity, one VLAN configured as a voice VLAN,
> and then a VLAN configured for untrusted traffic.  One Ethernet port on the
> router provides connectivity to the corporate and voice VLANS, while the
> remaining three are configured as untrusted.  Similarly with Wireless, we
> extend PEAP authentication from the headquarters and authenticate users to
> the corporate VLAN, and use a WPA-PSK to secure the untrusted connections.
> This way the users plug in their phone, then their laptop/docking station to
> port 0, and any other home devices can be connected to port 1-3 or use the
> wireless WPA-PSK network and be logically segregated (using ACL's) from any
> data on the corporate network.  This way we can also control QoS and mark
> down all traffic that enters the router from the untrusted network.  So when
> said employees son or daughter starts downing a 2 gig torrent from a home
> PC, they don't kill the voice or impact the corporate workflow.  Eventually
> we will be implementing 802.1x on the corporate port for additional
> security, but have had mixed results of getting it to work with Windows XP.
>
>
> Hope this helps.
>
>
>
> Matt
>
>
>
>
>
> *From:* Jerky [mailto:lists at jerkys.org <lists at jerkys.org>]
> *Sent:* Tuesday, October 16, 2007 6:32 PM
> *To:* Linsemier, Matthew
> *Cc:* Curt Shaffer; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Home user
>
>
>
> This has been kicked around for a while since we moved to CallManager but
> not much thought has been given to it. I'm trying to understand how your
> hardware is setup. How would it look, similar to one of these?
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845
> <--Ethernet--> LAN
>
>
>
> or
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <--->
> ASA or PIX Firewall <--Ethernet--> LAN
>
>
>
> Is the 3800 used for all your firewalling needs in lieu of something like
> an ASA or PIX? Sonicwall's are currently in place and haven't worked very
> well for the remote users it was tested with. The Sonicwalls we have don't
> have anything similar to what the 871's seem to have in regards to vlans and
> packet tagging. We would probably kick the Sonicwalls out if something else
> would work better.
>
>
>
> jeff
>
>
>
> On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
>
>
>
>  We currently have about 40 production remote home teleworkers that have
> been deployed using Cisco 871/877 wireless routers and a 7960 phones.  We
> are using a Cisco 3845 series router at the head-end so that we can control
> QoS tagging on the egress / ingress points of both sides of the VPN tunnel.
> We are using a phase 2 DMVPN solution dual-homed to two sites to provide
> secure redundant connectivity.
>
>
>
> It took me a bit to tweak my router configurations (I started on Cisco
> 831/837 routers) to get the results that we wanted, but all and all our
> users are happy.  There is the occasional jitter and packet loss (it is the
> Internet mind you) but g.729 is working quite well coupled with business
> cable and DSL services.
>
>
>
> If you have any other questions, feel free to ask.
>
>
>
> Matt
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [
> mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether.net>]
> *On Behalf Of *Curt Shaffer
> *Sent:* Monday, October 15, 2007 6:37 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Home user
>
>
>
> I was wondering want everyone out there is using for the situation where
> you have someone on your CCM or CCME that has 1 phone at a home office.
> Something tells me an ASA is overkill and I haven't found solid information
> that any of the 87x routers support tagging QoS of packets going through the
> VPN tunnel. We would obviously like to have QoS in place even though it's
> not respected at their ISP just to make sure the VPN/Voice packets are
> leaving their routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>  ------------------------------
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be
> protected by one or more legal privileges. It is intended solely for the use
> of the addressee identified above. If you are not the intended recipient,
> any use, disclosure, copying or distribution of this communication is
> UNAUTHORIZED. Neither this information block, the typed name of the sender,
> nor anything else in this message is intended to constitute an electronic
> signature unless a specific statement to the contrary is included in this
> message. If you have received this communication in error, please
> immediately contact me and delete this communication from your computer.
> Thank you.
>  ------------------------------
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> ------------------------------
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be
> protected by one or more legal privileges. It is intended solely for the use
> of the addressee identified above. If you are not the intended recipient,
> any use, disclosure, copying or distribution of this communication is
> UNAUTHORIZED. Neither this information block, the typed name of the sender,
> nor anything else in this message is intended to constitute an electronic
> signature unless a specific statement to the contrary is included in this
> message. If you have received this communication in error, please
> immediately contact me and delete this communication from your computer.
> Thank you.
>
> ------------------------------
>
>
>
>
>
> _______________________________________________
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From lists at jerkys.org  Wed Oct 17 14:10:47 2007
From: lists at jerkys.org (Jerky)
Date: Wed, 17 Oct 2007 13:10:47 -0500
Subject: [cisco-voip] CCM trace files gathered via RTMT show timestamps
	1 hour off
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0071FFDE6@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0071FFDE6@USNAEXCH.na.didata.local>
Message-ID: <A2E4FEE7-B98E-4E26-B3B3-A541D20FC400@jerkys.org>

show status should do it

jeff
On Oct 17, 2007, at 9:30 AM, Jason Aarons (US) wrote:

> The log file creation date is correct on my PC (it?s likely created  
> by RTMT?), but when I open the file the logged data shows 1 hour  
> behind.
>
>
>
> I ssh in to the CallManager (5.1.2) and is there a show command or  
> someplace to see what time the server thinks it is?
>
>
>
>
>
> Disclaimer: This e-mail communication and any attachments may  
> contain confidential and privileged information and is for use by  
> the designated addressee(s) named above only. If you are not the  
> intended addressee, you are hereby notified that you have received  
> this communication in error and that any use or reproduction of  
> this email or its contents is strictly prohibited and may be  
> unlawful. If you have received this communication in error, please  
> notify us immediately by replying to this message and deleting it  
> from your computer. Thank you.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jonvoip at gmail.com  Wed Oct 17 14:30:57 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 17 Oct 2007 13:30:57 -0500
Subject: [cisco-voip] Maximum number of voicemail ports...???
Message-ID: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>

So, I am installing a pair of Unity servers...  (primary and secondary)

CCM version is 4.3(1)sr1

Unity is 5.0(1)

TSP is 8.2(1)

Anyway, here's the problem... customer has a license for 144 voicemail
ports... However, CCM will only let me add 96 per Unity server (if I
try to add any, it doesn't let me), the max appears to be 96...

If that is the case, why can you buy a 144 port license???



Jonathan

From lelio at uoguelph.ca  Wed Oct 17 14:31:17 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 17 Oct 2007 14:31:17 -0400
Subject: [cisco-voip] Maximum number of voicemail ports...???
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
Message-ID: <033601c810eb$e8099440$56196883@cfs.uoguelph.ca>

Maybe you have to buy ports for each server? 144 licence is 72 per server, 
which used to be the max ports per server.

Since you have to buy a vm licence for the failover box, I don't see why 
they wouldn't make you buy a port.

----- Original Message ----- 
From: "Jonathan Charles" <jonvoip at gmail.com>
To: "Cisco Voip Mailing list" <cisco-voip at puck.nether.net>
Sent: Wednesday, October 17, 2007 2:30 PM
Subject: [cisco-voip] Maximum number of voicemail ports...???


> So, I am installing a pair of Unity servers...  (primary and secondary)
>
> CCM version is 4.3(1)sr1
>
> Unity is 5.0(1)
>
> TSP is 8.2(1)
>
> Anyway, here's the problem... customer has a license for 144 voicemail
> ports... However, CCM will only let me add 96 per Unity server (if I
> try to add any, it doesn't let me), the max appears to be 96...
>
> If that is the case, why can you buy a 144 port license???
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From jonvoip at gmail.com  Wed Oct 17 14:41:00 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 17 Oct 2007 13:41:00 -0500
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <033601c810eb$e8099440$56196883@cfs.uoguelph.ca>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
	<033601c810eb$e8099440$56196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0710171141t63ffa7b1wace9ca30136b0292@mail.gmail.com>

Maybe I am reading this wrong... Since it is a failover, does that
mean there are 72 per?

So, on CCM, I configure 72 per integration... ok, makes sense...

But why do all 144 ports show up on the primary?

Granted the failoverconfig thingy is running right now, maybe it splits em then


Jonathan

On 10/17/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> Maybe you have to buy ports for each server? 144 licence is 72 per server,
> which used to be the max ports per server.
>
> Since you have to buy a vm licence for the failover box, I don't see why
> they wouldn't make you buy a port.
>
> ----- Original Message -----
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "Cisco Voip Mailing list" <cisco-voip at puck.nether.net>
> Sent: Wednesday, October 17, 2007 2:30 PM
> Subject: [cisco-voip] Maximum number of voicemail ports...???
>
>
> > So, I am installing a pair of Unity servers...  (primary and secondary)
> >
> > CCM version is 4.3(1)sr1
> >
> > Unity is 5.0(1)
> >
> > TSP is 8.2(1)
> >
> > Anyway, here's the problem... customer has a license for 144 voicemail
> > ports... However, CCM will only let me add 96 per Unity server (if I
> > try to add any, it doesn't let me), the max appears to be 96...
> >
> > If that is the case, why can you buy a 144 port license???
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From wsisk at cisco.com  Wed Oct 17 15:00:49 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 17 Oct 2007 15:00:49 -0400
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
Message-ID: <47165BE1.4040502@cisco.com>

really only allows, or wizard on allows? CSCsi00569

/Wes

Jonathan Charles wrote:
> So, I am installing a pair of Unity servers...  (primary and secondary)
>
> CCM version is 4.3(1)sr1
>
> Unity is 5.0(1)
>
> TSP is 8.2(1)
>
> Anyway, here's the problem... customer has a license for 144 voicemail
> ports... However, CCM will only let me add 96 per Unity server (if I
> try to add any, it doesn't let me), the max appears to be 96...
>
> If that is the case, why can you buy a 144 port license???
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From jonvoip at gmail.com  Wed Oct 17 15:15:56 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 17 Oct 2007 14:15:56 -0500
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <47165BE1.4040502@cisco.com>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
	<47165BE1.4040502@cisco.com>
Message-ID: <5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>

Really only allows.

When I try to add a single port manually it gives me an error saying
the valid range is 1-96

I am pretty sure I am reading this wrong and it is 72 ports per server


Jonathan

On 10/17/07, Wes Sisk <wsisk at cisco.com> wrote:
> really only allows, or wizard on allows? CSCsi00569
>
> /Wes
>
> Jonathan Charles wrote:
> > So, I am installing a pair of Unity servers...  (primary and secondary)
> >
> > CCM version is 4.3(1)sr1
> >
> > Unity is 5.0(1)
> >
> > TSP is 8.2(1)
> >
> > Anyway, here's the problem... customer has a license for 144 voicemail
> > ports... However, CCM will only let me add 96 per Unity server (if I
> > try to add any, it doesn't let me), the max appears to be 96...
> >
> > If that is the case, why can you buy a 144 port license???
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From CarlosOrtiz at bayviewfinancial.com  Wed Oct 17 16:31:23 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 17 Oct 2007 16:31:23 -0400
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>
Message-ID: <OFE65AEA1A.3CEF9E77-ON85257377.006FB930-85257377.0070BCF7@bftg.com>

I would check the parts list on what was ordered.  I am not sure if 
failover licensing and pricing works the same way in 5.x as in 4.x but  I 
once set up a failover server on 4.x that was purchased months after the 
primary server was purchased.  The Cisco partner ended up selling us the 
same top level part number which contained the max number of ports (72) 
AGAIN.  When you would look at the number of ports with license viewer it 
would show 144 when we only had 72 ports. I do not believe the ports are 
licensed per server but by usage. 

In the company that I currently work at we have 64 ports that are usable 
by both Unity servers and the license viewer shows only 64 ports. If this 
holds true in Unity 5 then you may be able to save your customer some 
money for an unnecessary license that they may have received.  If I 
remember correctly this was about a $10,000 license so you may want to 
verify.

Carlos




"Jonathan Charles" <jonvoip at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
10/17/2007 03:15 PM

To
"Wes Sisk" <wsisk at cisco.com>
cc
Cisco Voip Mailing list <cisco-voip at puck.nether.net>
Subject
Re: [cisco-voip] Maximum number of voicemail ports...???






Really only allows.

When I try to add a single port manually it gives me an error saying
the valid range is 1-96

I am pretty sure I am reading this wrong and it is 72 ports per server


Jonathan

On 10/17/07, Wes Sisk <wsisk at cisco.com> wrote:
> really only allows, or wizard on allows? CSCsi00569
>
> /Wes
>
> Jonathan Charles wrote:
> > So, I am installing a pair of Unity servers...  (primary and 
secondary)
> >
> > CCM version is 4.3(1)sr1
> >
> > Unity is 5.0(1)
> >
> > TSP is 8.2(1)
> >
> > Anyway, here's the problem... customer has a license for 144 voicemail
> > ports... However, CCM will only let me add 96 per Unity server (if I
> > try to add any, it doesn't let me), the max appears to be 96...
> >
> > If that is the case, why can you buy a 144 port license???
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From ealeatherman at gmail.com  Wed Oct 17 18:13:41 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 17 Oct 2007 18:13:41 -0400
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
	<47165BE1.4040502@cisco.com>
	<5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>
Message-ID: <94a1afde0710171513l26c179cbt79e59276867fcb67@mail.gmail.com>

could you have unity ports registered to different CCM clusters? this way
you could have a valid Unity license for 114 ports but only have a max of 96
ports on a single CCM.

On 10/17/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Really only allows.
>
> When I try to add a single port manually it gives me an error saying
> the valid range is 1-96
>
> I am pretty sure I am reading this wrong and it is 72 ports per server
>
>
> Jonathan
>
> On 10/17/07, Wes Sisk <wsisk at cisco.com> wrote:
> > really only allows, or wizard on allows? CSCsi00569
> >
> > /Wes
> >
> > Jonathan Charles wrote:
> > > So, I am installing a pair of Unity servers...  (primary and
> secondary)
> > >
> > > CCM version is 4.3(1)sr1
> > >
> > > Unity is 5.0(1)
> > >
> > > TSP is 8.2(1)
> > >
> > > Anyway, here's the problem... customer has a license for 144 voicemail
> > > ports... However, CCM will only let me add 96 per Unity server (if I
> > > try to add any, it doesn't let me), the max appears to be 96...
> > >
> > > If that is the case, why can you buy a 144 port license???
> > >
> > >
> > >
> > > Jonathan
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From pat-cv at wcyv.com  Wed Oct 17 18:50:09 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Wed, 17 Oct 2007 18:50:09 -0400
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>	<47165BE1.4040502@cisco.com>
	<5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>
Message-ID: <471691A1.8060409@wcyv.com>

Unity 5.0(1) does indeed support 144 ports per server:

http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/release/notes/501curelnotes.html#wp409389

I just tried with ccm 5.1(2) in the lab and you're definitely right, 
when you try to create port 97, it throws an error. Unfortunately, it 
looks like the bug Wes cited isn't fixed in anything yet, so, for the 
time being, the only workaround would be to create two separate 
integrations in Unity to the same CCM cluster.

-------- Original Message  --------
From: "Jonathan Charles" <jonvoip at gmail.com>
To: "Wes Sisk" <wsisk at cisco.com>
Cc: Cisco Voip Mailing list <cisco-voip at puck.nether.net>
Subject: Re:[cisco-voip] Maximum number of voicemail ports...???
Date: 10/17/2007 3:15 PM

> Really only allows.
> 
> When I try to add a single port manually it gives me an error saying
> the valid range is 1-96
> 
> I am pretty sure I am reading this wrong and it is 72 ports per server
> 
> 
> Jonathan
> 
> On 10/17/07, Wes Sisk <wsisk at cisco.com> wrote:
>> really only allows, or wizard on allows? CSCsi00569
>>
>> /Wes
>>
>> Jonathan Charles wrote:
>>> So, I am installing a pair of Unity servers...  (primary and secondary)
>>>
>>> CCM version is 4.3(1)sr1
>>>
>>> Unity is 5.0(1)
>>>
>>> TSP is 8.2(1)
>>>
>>> Anyway, here's the problem... customer has a license for 144 voicemail
>>> ports... However, CCM will only let me add 96 per Unity server (if I
>>> try to add any, it doesn't let me), the max appears to be 96...
>>>
>>> If that is the case, why can you buy a 144 port license???
>>>
>>>
>>>
>>> Jonathan
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From jason.aarons at us.didata.com  Wed Oct 17 19:27:12 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 17 Oct 2007 19:27:12 -0400
Subject: [cisco-voip] CCM trace files gathered via RTMT show timestamps
	1 hour off
In-Reply-To: <A2E4FEE7-B98E-4E26-B3B3-A541D20FC400@jerkys.org>
References: <C1FE15183DA37645BC0633BC604E44F0071FFDE6@USNAEXCH.na.didata.local>
	<A2E4FEE7-B98E-4E26-B3B3-A541D20FC400@jerkys.org>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00724B234@USNAEXCH.na.didata.local>

Him this is right, I wonder why the entries in the ccm trace are 1 hour
behind.


   Welcome to the Platform Command Line Interface (version 1.1)

admin:show status

Host Name    : lxccm01
Date         : Wed Oct 17, 2007 19:26:06
Time Zone    : EDT
Locale       : en_US.UTF-8
Product Ver  : 5.1.2.3000-2
Platform Ver : 2.0.0.1-1

-----Original Message-----
From: Jerky [mailto:lists at jerkys.org] 
Sent: Wednesday, October 17, 2007 2:11 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CCM trace files gathered via RTMT show
timestamps 1 hour off

show status should do it

jeff
On Oct 17, 2007, at 9:30 AM, Jason Aarons (US) wrote:

> The log file creation date is correct on my PC (it's likely created  
> by RTMT?), but when I open the file the logged data shows 1 hour  
> behind.
>
>
>
> I ssh in to the CallManager (5.1.2) and is there a show command or  
> someplace to see what time the server thinks it is?
>
>
>
>
>
> Disclaimer: This e-mail communication and any attachments may  
> contain confidential and privileged information and is for use by  
> the designated addressee(s) named above only. If you are not the  
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> this communication in error and that any use or reproduction of  
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> from your computer. Thank you.
>
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip


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From jonvoip at gmail.com  Wed Oct 17 20:20:59 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 17 Oct 2007 19:20:59 -0500
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <471691A1.8060409@wcyv.com>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
	<47165BE1.4040502@cisco.com>
	<5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>
	<471691A1.8060409@wcyv.com>
Message-ID: <5d093f9a0710171720n26bfb455p51ce18522b7c4cd9@mail.gmail.com>

Yes, but CCM does not.

Unity 5.0 liked the 144 ports, but CCM 4.3 did not.




Jonathan

On 10/17/07, Pat Hayes <pat-cv at wcyv.com> wrote:
> Unity 5.0(1) does indeed support 144 ports per server:
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/release/notes/501curelnotes.html#wp409389
>
> I just tried with ccm 5.1(2) in the lab and you're definitely right,
> when you try to create port 97, it throws an error. Unfortunately, it
> looks like the bug Wes cited isn't fixed in anything yet, so, for the
> time being, the only workaround would be to create two separate
> integrations in Unity to the same CCM cluster.
>
> -------- Original Message  --------
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "Wes Sisk" <wsisk at cisco.com>
> Cc: Cisco Voip Mailing list <cisco-voip at puck.nether.net>
> Subject: Re:[cisco-voip] Maximum number of voicemail ports...???
> Date: 10/17/2007 3:15 PM
>
> > Really only allows.
> >
> > When I try to add a single port manually it gives me an error saying
> > the valid range is 1-96
> >
> > I am pretty sure I am reading this wrong and it is 72 ports per server
> >
> >
> > Jonathan
> >
> > On 10/17/07, Wes Sisk <wsisk at cisco.com> wrote:
> >> really only allows, or wizard on allows? CSCsi00569
> >>
> >> /Wes
> >>
> >> Jonathan Charles wrote:
> >>> So, I am installing a pair of Unity servers...  (primary and secondary)
> >>>
> >>> CCM version is 4.3(1)sr1
> >>>
> >>> Unity is 5.0(1)
> >>>
> >>> TSP is 8.2(1)
> >>>
> >>> Anyway, here's the problem... customer has a license for 144 voicemail
> >>> ports... However, CCM will only let me add 96 per Unity server (if I
> >>> try to add any, it doesn't let me), the max appears to be 96...
> >>>
> >>> If that is the case, why can you buy a 144 port license???
> >>>
> >>>
> >>>
> >>> Jonathan
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip at puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>

From asobihoudai at yahoo.com  Wed Oct 17 20:24:27 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 17 Oct 2007 17:24:27 -0700 (PDT)
Subject: [cisco-voip] CCME GUI adding phones "no phone to add"
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00724B234@USNAEXCH.na.didata.local>
Message-ID: <823476.82688.qm@web57215.mail.re3.yahoo.com>

FYI, for anyone that is encountering an error dialog
of "no phones to add" from the CCME GUI, you have to
go into the box and under telephony-service mode, turn
auto-reg-ephone back on.

hope that helps someone!
Paul


      ____________________________________________________________________________________
Shape Yahoo! in your own image.  Join our Network Research Panel today!   http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 



From mthompson729 at gmail.com  Wed Oct 17 20:44:13 2007
From: mthompson729 at gmail.com (Michael Thompson)
Date: Wed, 17 Oct 2007 20:44:13 -0400
Subject: [cisco-voip] Home user
In-Reply-To: <6A059BC4-F184-4563-B117-1D28BCDAFFA9@jerkys.org>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
	<AB0A35F2486F81468971AA68558BE719073BF05C@smieln01winm01.micoanet.com>
	<6A059BC4-F184-4563-B117-1D28BCDAFFA9@jerkys.org>
Message-ID: <363f21ee0710171744m729b24abjb126f569c4b97bf9@mail.gmail.com>

the big reason I'm more of a fan of taking the 871 route is that you have
flexibility of creating a DMZ (using F0/3 I think) for the other home users
on the network (kids at home playing games, etc) and it creates a seperate
segment for SoHo users.

what's sexy about it is that you can be working from home, on your company
laptop for security reasons of course, on a site to site VPN
essentially.  you can control what traffic from that VLANs / Subnets is
allowed across so that only specific application traffic is allowed (Citrix,
EMail, IP Voice, IPCC CAD, etc).  The best part is that since you assign LLQ
to the tunnel interface, you have the full capabilities of the MQC at your
disposal no matter how QoS nutty you get with your DSCP markings (priority
markings, policing, variable drop precedence, etc)...it's all there.  AND
the capper, and ABSOLUTELY critical when talking about using this from home,
you have absolute control over how the bandwidth is used to the internet.
i.e junior downloading music and playing World of Warcraft isn't crippling
your IPCC / Voice traffic.

essentially you create a VPN on the 871 that terminates on the outside
interface of the router.  Then you have GRE terminate on a loopback inside
the router.  You then apply policing on the GRE tunnel equivalent to the
approximate upstream bandwidth of your broadband.  the exceed parameter
calls the MQC / LLQ configuration (it's not invoked unless you start to
approach bandwidth contention).  the 871 is a beefy little box.  we have
customers running this exact config with up to around 5 PCs in the house
(personal and work).  all of the personal computers are on the DMZ and only
the work devices go on the inside interfaces.

all of that being said, you're still at the mercy of the idiot out at the
distribution box pulling your DSL wires because he doesn't get dial tone on
them and he needs copper for the neighbors new phone...but you can't totally
defend against the morons of the world.

On 10/17/07, Jerky <lists at jerkys.org> wrote:
>
> so it would be more like this:
>
> Cisco 871
>      |
> DSL CABLE
>      |
> Internet
>      |
> T1 Connection (Serial0/0/0)
>                     |
>    _____ 3800 _____
>   |                                  |
> ethernet 0/0          ethernet 0/1
> |                                    |
> PIX/ASA                    3800 (Cisco 871 VPN's terminate here)
> |                                    |
> LAN(computers)       LAN (Voice)
>
>
>
>
> Homefully my crude diagram makes sense. Do your home users have access to
> any data on the computer network side. Or is the 87x VPNs solely for getting
> to the voice network If users access things on the "computer" side would you
> have a separate tunnel setup just for that?
>
>
> Thanks for so much helping enlighten me. It's been very helpful.
>
>
> jeff
>
>
>
>
>
>
>  On Oct 17, 2007, at 10:19 AM, Linsemier, Matthew wrote:
>
>  In our environment we utilize PIX firewalls (still have to upgrade to
> ASA's) to handle our firewall needs and then use the 3800 series router just
> to terminate the DMVPN home users.  They are deployed in parallel and sit
> behind a perimeter screening router (another 3800 series router).  We shied
> away from using the PIX for the simple fact that while it would preserve QoS
> markings, we couldn't do any remarking or shaping in the device.  Maybe this
> has changed in the ASA, but I don't think you have the control like you do
> in IOS (such as qos pre-classify, shaping, policing,  etc.).  Depending on
> how many tunnels you plan on using, you could use a router much smaller than
> a 3800 series to terminate the end nodes.
>
>
>
> On the home user end we have the Cisco 871/877 routers configured to
> support wired and wireless connections using three VLANS.  We have a VLAN
> configured for corporate connectivity, one VLAN configured as a voice VLAN,
> and then a VLAN configured for untrusted traffic.  One Ethernet port on the
> router provides connectivity to the corporate and voice VLANS, while the
> remaining three are configured as untrusted.  Similarly with Wireless, we
> extend PEAP authentication from the headquarters and authenticate users to
> the corporate VLAN, and use a WPA-PSK to secure the untrusted connections.
> This way the users plug in their phone, then their laptop/docking station to
> port 0, and any other home devices can be connected to port 1-3 or use the
> wireless WPA-PSK network and be logically segregated (using ACL's) from any
> data on the corporate network.  This way we can also control QoS and mark
> down all traffic that enters the router from the untrusted network.  So when
> said employees son or daughter starts downing a 2 gig torrent from a home
> PC, they don't kill the voice or impact the corporate workflow.  Eventually
> we will be implementing 802.1x on the corporate port for additional
> security, but have had mixed results of getting it to work with Windows XP.
>
>
> Hope this helps.
>
>
>
> Matt
>
>
>
>
>
> *From:* Jerky [mailto:lists at jerkys.org <lists at jerkys.org>]
> *Sent:* Tuesday, October 16, 2007 6:32 PM
> *To:* Linsemier, Matthew
> *Cc:* Curt Shaffer; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Home user
>
>
>
> This has been kicked around for a while since we moved to CallManager but
> not much thought has been given to it. I'm trying to understand how your
> hardware is setup. How would it look, similar to one of these?
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845
> <--Ethernet--> LAN
>
>
>
> or
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <--->
> ASA or PIX Firewall <--Ethernet--> LAN
>
>
>
> Is the 3800 used for all your firewalling needs in lieu of something like
> an ASA or PIX? Sonicwall's are currently in place and haven't worked very
> well for the remote users it was tested with. The Sonicwalls we have don't
> have anything similar to what the 871's seem to have in regards to vlans and
> packet tagging. We would probably kick the Sonicwalls out if something else
> would work better.
>
>
>
> jeff
>
>
>
> On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
>
>
>
>  We currently have about 40 production remote home teleworkers that have
> been deployed using Cisco 871/877 wireless routers and a 7960 phones.  We
> are using a Cisco 3845 series router at the head-end so that we can control
> QoS tagging on the egress / ingress points of both sides of the VPN tunnel.
> We are using a phase 2 DMVPN solution dual-homed to two sites to provide
> secure redundant connectivity.
>
>
>
> It took me a bit to tweak my router configurations (I started on Cisco
> 831/837 routers) to get the results that we wanted, but all and all our
> users are happy.  There is the occasional jitter and packet loss (it is the
> Internet mind you) but g.729 is working quite well coupled with business
> cable and DSL services.
>
>
>
> If you have any other questions, feel free to ask.
>
>
>
> Matt
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [
> mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether.net>]
> *On Behalf Of *Curt Shaffer
> *Sent:* Monday, October 15, 2007 6:37 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Home user
>
>
>
> I was wondering want everyone out there is using for the situation where
> you have someone on your CCM or CCME that has 1 phone at a home office.
> Something tells me an ASA is overkill and I haven't found solid information
> that any of the 87x routers support tagging QoS of packets going through the
> VPN tunnel. We would obviously like to have QoS in place even though it's
> not respected at their ISP just to make sure the VPN/Voice packets are
> leaving their routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>  ------------------------------
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be
> protected by one or more legal privileges. It is intended solely for the use
> of the addressee identified above. If you are not the intended recipient,
> any use, disclosure, copying or distribution of this communication is
> UNAUTHORIZED. Neither this information block, the typed name of the sender,
> nor anything else in this message is intended to constitute an electronic
> signature unless a specific statement to the contrary is included in this
> message. If you have received this communication in error, please
> immediately contact me and delete this communication from your computer.
> Thank you.
>  ------------------------------
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> ------------------------------
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be
> protected by one or more legal privileges. It is intended solely for the use
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> any use, disclosure, copying or distribution of this communication is
> UNAUTHORIZED. Neither this information block, the typed name of the sender,
> nor anything else in this message is intended to constitute an electronic
> signature unless a specific statement to the contrary is included in this
> message. If you have received this communication in error, please
> immediately contact me and delete this communication from your computer.
> Thank you.
>
> ------------------------------
>
>
>
>
>
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jefflin.choi at gmail.com  Wed Oct 17 23:26:45 2007
From: jefflin.choi at gmail.com (Jefflin Choi)
Date: Thu, 18 Oct 2007 11:26:45 +0800
Subject: [cisco-voip] offnet to offnet transfer behavior inquiry
Message-ID: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>

Hi all,

When doing offnet to offnet transfer, will the two offnet calls retain using
the voice port of the gateway?
As far as i know it will but i need clarification/correction.

Will the callmanager keep track of the call transfer on its CDR? Since the
on-net ip phone gets out of the picture, will the CDR be able to have logs
between this call?

We have a situation were we use offnet to offnet transfer and need to have a
clear understanding of how it will affect the billing using CDR logs.


Thanks,
Jeff
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From jefflin.choi at gmail.com  Thu Oct 18 00:46:30 2007
From: jefflin.choi at gmail.com (Jefflin Choi)
Date: Thu, 18 Oct 2007 12:46:30 +0800
Subject: [cisco-voip] offnet to offnet transfer behavior inquiry
In-Reply-To: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
Message-ID: <5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>

I am referring to H323 bdw.

On 10/18/07, Jefflin Choi <jefflin.choi at gmail.com> wrote:
>
> Hi all,
>
> When doing offnet to offnet transfer, will the two offnet calls retain
> using the voice port of the gateway?
> As far as i know it will but i need clarification/correction.
>
> Will the callmanager keep track of the call transfer on its CDR? Since the
> on-net ip phone gets out of the picture, will the CDR be able to have logs
> between this call?
>
> We have a situation were we use offnet to offnet transfer and need to have
> a clear understanding of how it will affect the billing using CDR logs.
>
>
> Thanks,
> Jeff
>
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From Brandon.Ogden at amersports.com  Thu Oct 18 09:52:04 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Thu, 18 Oct 2007 06:52:04 -0700
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>

I rebuilt my subscriber box last night, it seemed to go well.  I was
wondering how long it should take for the phones to register with the
Subscriber rather than the pub.  As of this morning, they are still
registered with the pub.  Thanks! 

 

Brandon 

 

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From svoll.voip at gmail.com  Thu Oct 18 10:53:51 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 18 Oct 2007 07:53:51 -0700
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
Message-ID: <f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>

did you change the order of your servers so it registers with the sub?

Scott


On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>  I rebuilt my subscriber box last night, it seemed to go well.  I was
> wondering how long it should take for the phones to register with the
> Subscriber rather than the pub.  As of this morning, they are still
> registered with the pub.  Thanks!
>
>
>
> Brandon
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From wsisk at cisco.com  Thu Oct 18 11:04:03 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 18 Oct 2007 11:04:03 -0400
Subject: [cisco-voip] offnet to offnet transfer behavior inquiry
In-Reply-To: <5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
Message-ID: <471775E3.9010809@cisco.com>

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From Brandon.Ogden at amersports.com  Thu Oct 18 11:59:47 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Thu, 18 Oct 2007 08:59:47 -0700
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
	<f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>

I just went in and looked at that and my subscriber is not showing up as
an available server in the Call Manager Group.  Any ideas?  It appears
to have replicated correctly, and I was able to use it to modify a
Directory Number.  

 

Brandon 

________________________________

From: Scott Voll [mailto:svoll.voip at gmail.com] 
Sent: Thursday, October 18, 2007 8:54 AM
To: Ogden, Brandon
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Registration to Subscriber

 

did you change the order of your servers so it registers with the sub?

 

Scott

 

On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote: 

I rebuilt my subscriber box last night, it seemed to go well.  I was
wondering how long it should take for the phones to register with the
Subscriber rather than the pub.  As of this morning, they are still
registered with the pub.  Thanks! 

 

Brandon 

 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

 

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From jonvoip at gmail.com  Thu Oct 18 12:14:07 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 11:14:07 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
	<f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
Message-ID: <5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>

Did you reboot the cluster after the rebuild?


Jonathan

On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>
>
>
> I just went in and looked at that and my subscriber is not showing up as an
> available server in the Call Manager Group.  Any ideas?  It appears to have
> replicated correctly, and I was able to use it to modify a Directory Number.
>
>
>
>
>
> Brandon
>
>  ________________________________
>
>
> From: Scott Voll [mailto:svoll.voip at gmail.com]
>  Sent: Thursday, October 18, 2007 8:54 AM
>  To: Ogden, Brandon
>  Cc: cisco-voip at puck.nether.net
>  Subject: Re: [cisco-voip] Registration to Subscriber
>
>
>
>
>
> did you change the order of your servers so it registers with the sub?
>
>
>
>
>
> Scott
>
>
>
>
> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>
>
> I rebuilt my subscriber box last night, it seemed to go well.  I was
> wondering how long it should take for the phones to register with the
> Subscriber rather than the pub.  As of this morning, they are still
> registered with the pub.  Thanks!
>
>
>
>
>
> Brandon
>
>
>
>
>  _______________________________________________
>  cisco-voip mailing list
>  cisco-voip at puck.nether.net
>  https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From lelio at uoguelph.ca  Thu Oct 18 12:15:51 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 18 Oct 2007 12:15:51 -0400
Subject: [cisco-voip] Registration to Subscriber
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com><5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int><f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
Message-ID: <014801c811a2$26d7a520$56196883@cfs.uoguelph.ca>

When you rebuilt your subscriber, did you first delete it or deactivate it? Doing this removes all reference to the subscriber, including callmanager groups and even deletes call park groups. It also resets all enterprise and system parameters to default. It's not a fun process, that's for sure.
  ----- Original Message ----- 
  From: Ogden, Brandon 
  To: Scott Voll 
  Cc: cisco-voip at puck.nether.net 
  Sent: Thursday, October 18, 2007 11:59 AM
  Subject: Re: [cisco-voip] Registration to Subscriber


  I just went in and looked at that and my subscriber is not showing up as an available server in the Call Manager Group.  Any ideas?  It appears to have replicated correctly, and I was able to use it to modify a Directory Number.  

   

  Brandon 


------------------------------------------------------------------------------

  From: Scott Voll [mailto:svoll.voip at gmail.com] 
  Sent: Thursday, October 18, 2007 8:54 AM
  To: Ogden, Brandon
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Registration to Subscriber

   

  did you change the order of your servers so it registers with the sub?

   

  Scott

   

  On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote: 

  I rebuilt my subscriber box last night, it seemed to go well.  I was wondering how long it should take for the phones to register with the Subscriber rather than the pub.  As of this morning, they are still registered with the pub.  Thanks! 

   

  Brandon 

   


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip 

   



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From jonvoip at gmail.com  Thu Oct 18 12:40:05 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 11:40:05 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
	<f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
	<5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
Message-ID: <5d093f9a0710180940g62054a21g7dcc3421d1e53100@mail.gmail.com>

OK, every time I have done a subscriber rebuild, I have had to reboot
the pub to get them to sync up happily...


Jonathan

On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> I did not do a reboot of the publisher, just the sub.
>
> Brandon Ogden
> Amer Sports
> 801-624-7529
> www.amersports.com
> http://www.linkedin.com/in/brandonogden
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Thursday, October 18, 2007 10:14 AM
> To: Ogden, Brandon
> Cc: Scott Voll; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Registration to Subscriber
>
> Did you reboot the cluster after the rebuild?
>
>
> Jonathan
>
> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >
> >
> >
> >
> > I just went in and looked at that and my subscriber is not showing up
> as an
> > available server in the Call Manager Group.  Any ideas?  It appears to
> have
> > replicated correctly, and I was able to use it to modify a Directory
> Number.
> >
> >
> >
> >
> >
> > Brandon
> >
> >  ________________________________
> >
> >
> > From: Scott Voll [mailto:svoll.voip at gmail.com]
> >  Sent: Thursday, October 18, 2007 8:54 AM
> >  To: Ogden, Brandon
> >  Cc: cisco-voip at puck.nether.net
> >  Subject: Re: [cisco-voip] Registration to Subscriber
> >
> >
> >
> >
> >
> > did you change the order of your servers so it registers with the sub?
> >
> >
> >
> >
> >
> > Scott
> >
> >
> >
> >
> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >
> >
> >
> > I rebuilt my subscriber box last night, it seemed to go well.  I was
> > wondering how long it should take for the phones to register with the
> > Subscriber rather than the pub.  As of this morning, they are still
> > registered with the pub.  Thanks!
> >
> >
> >
> >
> >
> > Brandon
> >
> >
> >
> >
> >  _______________________________________________
> >  cisco-voip mailing list
> >  cisco-voip at puck.nether.net
> >  https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error)
> please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden.
>

From Erik.Stillman at Shearman.com  Thu Oct 18 13:12:35 2007
From: Erik.Stillman at Shearman.com (Erik Stillman)
Date: Thu, 18 Oct 2007 13:12:35 -0400
Subject: [cisco-voip] Different ring types for internal/external calls
In-Reply-To: <NYEXM4Tj8Svn3snFjVQ00017c10@NYEXMB2.shearman.com>
Message-ID: <643A3A543DB5224499BC3D73473AEC27869D93@NYEXMB2.shearman.com>

Hey guys, I've got a hot one here. Request has come in to differentiate
calls by ring type.

Is there anything I can do in CCM4.2 to achieve this?

Any help would be great. Thanks!

-Erik


***********************************************************************

This communication is not intended to be used, and cannot be used,
by the recipient or any other person for the purpose of avoiding United
States federal tax penalties that may be imposed on the recipient or
such other person. In addition, if any United States federal tax
advice contained in this communication is used or referred to in
promoting, marketing or recommending any corporation, partnership
or other entity, investment plan, concept, structure or arrangement
(which should be assumed to be the case by a recipient or other
person who is not our client with respect to the subject matter of the
communication), then (i) such tax advice should be construed as written
to support the promotion or marketing of the transactions or matters
addressed by the advice and (ii) the recipient or other person should
seek advice based on the recipient or other person's particular
circumstances from an independent tax advisor. For further information,
please go to http://www.shearman.com/disclaimer/tax_disclosure.html

***********************************************************************

This transmittal and/or attachments may be a confidential
attorney-client communication or may otherwise be privileged or
confidential. If you are not the intended recipient, you are hereby
notified that you have received this transmittal in error; any review,
dissemination, distribution or copying of this transmittal is strictly
prohibited. If you have received this transmittal and/or attachments in
error, please notify us immediately by reply or by telephone (call us
collect at +1 212-848-8400) and immediately delete this message and
all its attachments.

Shearman & Sterling LLP is a limited liability partnership organized in
the United States under the laws of the State of Delaware, which laws
limit the personal liability of partners.

From Brandon.Ogden at amersports.com  Thu Oct 18 12:18:11 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Thu, 18 Oct 2007 09:18:11 -0700
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
	<f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
	<5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>

I did not do a reboot of the publisher, just the sub.

Brandon Ogden
Amer Sports 
801-624-7529
www.amersports.com
http://www.linkedin.com/in/brandonogden

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Thursday, October 18, 2007 10:14 AM
To: Ogden, Brandon
Cc: Scott Voll; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Registration to Subscriber

Did you reboot the cluster after the rebuild?


Jonathan

On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>
>
>
> I just went in and looked at that and my subscriber is not showing up
as an
> available server in the Call Manager Group.  Any ideas?  It appears to
have
> replicated correctly, and I was able to use it to modify a Directory
Number.
>
>
>
>
>
> Brandon
>
>  ________________________________
>
>
> From: Scott Voll [mailto:svoll.voip at gmail.com]
>  Sent: Thursday, October 18, 2007 8:54 AM
>  To: Ogden, Brandon
>  Cc: cisco-voip at puck.nether.net
>  Subject: Re: [cisco-voip] Registration to Subscriber
>
>
>
>
>
> did you change the order of your servers so it registers with the sub?
>
>
>
>
>
> Scott
>
>
>
>
> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>
>
> I rebuilt my subscriber box last night, it seemed to go well.  I was
> wondering how long it should take for the phones to register with the
> Subscriber rather than the pub.  As of this morning, they are still
> registered with the pub.  Thanks!
>
>
>
>
>
> Brandon
>
>
>
>
>  _______________________________________________
>  cisco-voip mailing list
>  cisco-voip at puck.nether.net
>  https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) 
please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

From ErasmuE4 at telkom.co.za  Thu Oct 18 13:22:01 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Thu, 18 Oct 2007 19:22:01 +0200
Subject: [cisco-voip] call manager express blocking outbound caller ID
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D692@CNTRRA20-XCS01.telkom.co.za>


HI
 
Does anyone know how to enable selective caller id blocking on call
manager express ver 4-2
 
I tried adding the caller-id blocking code under telehony service and
also added the line to enable caller ID blocking on the DN but this
does not work and it looks like blocking is on all the time.
 
 
what I want to achieve is..   enable blocking on the system somehow.
Sample user then makes a call to the pstn -- but before making the
call the user types a special code like one would do on some cell
phones. Then the ID displays as private number on the destination
phone. Then -- as an example the person wants to make the next call to
another number he can dial another special code to disble caller id
blocking and the ID will show on the target phone.
 
This basically means the user wants to switch blocking on or off
dynamically from the phone when making calls
 
 
 
 
erik erasmus  


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail is subject to the Telkom SA electronic communication legal notice, available at :
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From csvoip at googlemail.com  Thu Oct 18 13:28:24 2007
From: csvoip at googlemail.com (Nick)
Date: Thu, 18 Oct 2007 18:28:24 +0100
Subject: [cisco-voip] CCM 5 or 6 or vmware player
Message-ID: <475fda330710181028p4ae6e6c2saaa84c5bf2b0de30@mail.gmail.com>

I know that ccm 5 has been well documented about installing it on vmware esc
and gsx server but hasd anyone installed it on vmware player on a pc?
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From Brandon.Ogden at amersports.com  Thu Oct 18 13:34:31 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Thu, 18 Oct 2007 10:34:31 -0700
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com><5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int><f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int><5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>

LOL.. I got it resolved.. Didn't have to reboot.. and before you say
so.. yes I am an idiot.. LOL  The Call Manager Service had not started
for some reason.. I restarted the service and it popped right up, all is
well now.  Thanks all! 
 
Brandon 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden, Brandon
Sent: Thursday, October 18, 2007 10:18 AM
To: Jonathan Charles
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Registration to Subscriber

I did not do a reboot of the publisher, just the sub.

Brandon Ogden
Amer Sports 
801-624-7529
www.amersports.com
http://www.linkedin.com/in/brandonogden

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Thursday, October 18, 2007 10:14 AM
To: Ogden, Brandon
Cc: Scott Voll; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Registration to Subscriber

Did you reboot the cluster after the rebuild?


Jonathan

On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>
>
>
> I just went in and looked at that and my subscriber is not showing up
as an
> available server in the Call Manager Group.  Any ideas?  It appears to
have
> replicated correctly, and I was able to use it to modify a Directory
Number.
>
>
>
>
>
> Brandon
>
>  ________________________________
>
>
> From: Scott Voll [mailto:svoll.voip at gmail.com]
>  Sent: Thursday, October 18, 2007 8:54 AM
>  To: Ogden, Brandon
>  Cc: cisco-voip at puck.nether.net
>  Subject: Re: [cisco-voip] Registration to Subscriber
>
>
>
>
>
> did you change the order of your servers so it registers with the sub?
>
>
>
>
>
> Scott
>
>
>
>
> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>
>
> I rebuilt my subscriber box last night, it seemed to go well.  I was
> wondering how long it should take for the phones to register with the
> Subscriber rather than the pub.  As of this morning, they are still
> registered with the pub.  Thanks!
>
>
>
>
>
> Brandon
>
>
>
>
>  _______________________________________________
>  cisco-voip mailing list
>  cisco-voip at puck.nether.net
>  https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) 
please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From lelio at uoguelph.ca  Thu Oct 18 13:34:33 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 18 Oct 2007 13:34:33 -0400
Subject: [cisco-voip] Different ring types for internal/external calls
References: <643A3A543DB5224499BC3D73473AEC27869D93@NYEXMB2.shearman.com>
Message-ID: <016d01c811ad$25914170$56196883@cfs.uoguelph.ca>

As long as you are using a CTI based autoattendant like IPCC Express or 
Unity Express than this should be doable. There is an enterprise or system 
parameter that differentiates external calls with a double ring.

If you are using Unity, then you are out of luck because it uses SCCP and 
transfers calls as if a user is transferring them.

----- Original Message ----- 
From: "Erik Stillman" <Erik.Stillman at Shearman.com>
To: <cisco-voip at puck.nether.net>
Sent: Thursday, October 18, 2007 1:12 PM
Subject: [cisco-voip] Different ring types for internal/external calls


> Hey guys, I've got a hot one here. Request has come in to differentiate
> calls by ring type.
>
> Is there anything I can do in CCM4.2 to achieve this?
>
> Any help would be great. Thanks!
>
> -Erik
>
>
> ***********************************************************************
>
> This communication is not intended to be used, and cannot be used,
> by the recipient or any other person for the purpose of avoiding United
> States federal tax penalties that may be imposed on the recipient or
> such other person. In addition, if any United States federal tax
> advice contained in this communication is used or referred to in
> promoting, marketing or recommending any corporation, partnership
> or other entity, investment plan, concept, structure or arrangement
> (which should be assumed to be the case by a recipient or other
> person who is not our client with respect to the subject matter of the
> communication), then (i) such tax advice should be construed as written
> to support the promotion or marketing of the transactions or matters
> addressed by the advice and (ii) the recipient or other person should
> seek advice based on the recipient or other person's particular
> circumstances from an independent tax advisor. For further information,
> please go to http://www.shearman.com/disclaimer/tax_disclosure.html
>
> ***********************************************************************
>
> This transmittal and/or attachments may be a confidential
> attorney-client communication or may otherwise be privileged or
> confidential. If you are not the intended recipient, you are hereby
> notified that you have received this transmittal in error; any review,
> dissemination, distribution or copying of this transmittal is strictly
> prohibited. If you have received this transmittal and/or attachments in
> error, please notify us immediately by reply or by telephone (call us
> collect at +1 212-848-8400) and immediately delete this message and
> all its attachments.
>
> Shearman & Sterling LLP is a limited liability partnership organized in
> the United States under the laws of the State of Delaware, which laws
> limit the personal liability of partners.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From ErasmuE4 at telkom.co.za  Thu Oct 18 14:03:31 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Thu, 18 Oct 2007 20:03:31 +0200
Subject: [cisco-voip] call manager express blocking outbound caller ID
References: <472E195E692EE74BAE4E7B2880A78C1F84D692@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D697@CNTRRA20-XCS01.telkom.co.za>

I think I messed myself around and should get it working now after
reading the docs again
 
I will remove the caller-id blocking on the dn and just leave the code
under telehony srvice
 
Today I tested using the code plus blocking on the dn  but to an
internal phone -- will try on a pri link tomorrow
 
any advice will be appreciated anyway. 
 
erik

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Erik Erasmus (E)
Sent: Thu 2007-10-18 19:22
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] call manager express blocking outbound caller ID


HI
 
Does anyone know how to enable selective caller id blocking on call
manager express ver 4-2
 
I tried adding the caller-id blocking code under telehony service and
also added the line to enable caller ID blocking on the DN but this
does not work and it looks like blocking is on all the time.
 
 
what I want to achieve is..   enable blocking on the system somehow.
Sample user then makes a call to the pstn -- but before making the
call the user types a special code like one would do on some cell
phones. Then the ID displays as private number on the destination
phone. Then -- as an example the person wants to make the next call to
another number he can dial another special code to disble caller id
blocking and the ID will show on the target phone.
 
This basically means the user wants to switch blocking on or off
dynamically from the phone when making calls
 
 
 
 
erik erasmus  
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail is subject to the Telkom SA electronic communication legal
notice, available at :
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From jonvoip at gmail.com  Thu Oct 18 14:14:32 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 13:14:32 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
	<f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
	<5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
	<D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>
Message-ID: <5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>

If you have a large number of route patterns the CCM service can time
out when starting

There is a service parameter to increase the timeout, but I forgot
what it was...


Jonathan

On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> LOL.. I got it resolved.. Didn't have to reboot.. and before you say
> so.. yes I am an idiot.. LOL  The Call Manager Service had not started
> for some reason.. I restarted the service and it popped right up, all is
> well now.  Thanks all!
>
> Brandon
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden, Brandon
> Sent: Thursday, October 18, 2007 10:18 AM
> To: Jonathan Charles
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Registration to Subscriber
>
> I did not do a reboot of the publisher, just the sub.
>
> Brandon Ogden
> Amer Sports
> 801-624-7529
> www.amersports.com
> http://www.linkedin.com/in/brandonogden
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Thursday, October 18, 2007 10:14 AM
> To: Ogden, Brandon
> Cc: Scott Voll; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Registration to Subscriber
>
> Did you reboot the cluster after the rebuild?
>
>
> Jonathan
>
> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >
> >
> >
> >
> > I just went in and looked at that and my subscriber is not showing up
> as an
> > available server in the Call Manager Group.  Any ideas?  It appears to
> have
> > replicated correctly, and I was able to use it to modify a Directory
> Number.
> >
> >
> >
> >
> >
> > Brandon
> >
> >  ________________________________
> >
> >
> > From: Scott Voll [mailto:svoll.voip at gmail.com]
> >  Sent: Thursday, October 18, 2007 8:54 AM
> >  To: Ogden, Brandon
> >  Cc: cisco-voip at puck.nether.net
> >  Subject: Re: [cisco-voip] Registration to Subscriber
> >
> >
> >
> >
> >
> > did you change the order of your servers so it registers with the sub?
> >
> >
> >
> >
> >
> > Scott
> >
> >
> >
> >
> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >
> >
> >
> > I rebuilt my subscriber box last night, it seemed to go well.  I was
> > wondering how long it should take for the phones to register with the
> > Subscriber rather than the pub.  As of this morning, they are still
> > registered with the pub.  Thanks!
> >
> >
> >
> >
> >
> > Brandon
> >
> >
> >
> >
> >  _______________________________________________
> >  cisco-voip mailing list
> >  cisco-voip at puck.nether.net
> >  https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> This e-mail may contain confidential and/or privileged information. If
> you are not the intended recipient (or have received this e-mail in
> error)
> please notify the sender immediately and destroy this e-mail. Any
> unauthorized copying, disclosure or distribution of the material in this
> e-mail is strictly forbidden.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From lelio at uoguelph.ca  Thu Oct 18 14:17:39 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 18 Oct 2007 14:17:39 -0400
Subject: [cisco-voip] Registration to Subscriber
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com><5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int><f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int><5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int><D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>
	<5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>
Message-ID: <021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca>

what's "a large number of route patterns" ?

100? 1,000? 10,000?

----- Original Message ----- 
From: "Jonathan Charles" <jonvoip at gmail.com>
To: "Ogden, Brandon" <Brandon.Ogden at amersports.com>
Cc: <cisco-voip at puck.nether.net>
Sent: Thursday, October 18, 2007 2:14 PM
Subject: Re: [cisco-voip] Registration to Subscriber


> If you have a large number of route patterns the CCM service can time
> out when starting
> 
> There is a service parameter to increase the timeout, but I forgot
> what it was...
> 
> 
> Jonathan
> 
> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>> LOL.. I got it resolved.. Didn't have to reboot.. and before you say
>> so.. yes I am an idiot.. LOL  The Call Manager Service had not started
>> for some reason.. I restarted the service and it popped right up, all is
>> well now.  Thanks all!
>>
>> Brandon
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden, Brandon
>> Sent: Thursday, October 18, 2007 10:18 AM
>> To: Jonathan Charles
>> Cc: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] Registration to Subscriber
>>
>> I did not do a reboot of the publisher, just the sub.
>>
>> Brandon Ogden
>> Amer Sports
>> 801-624-7529
>> www.amersports.com
>> http://www.linkedin.com/in/brandonogden
>>
>> -----Original Message-----
>> From: Jonathan Charles [mailto:jonvoip at gmail.com]
>> Sent: Thursday, October 18, 2007 10:14 AM
>> To: Ogden, Brandon
>> Cc: Scott Voll; cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] Registration to Subscriber
>>
>> Did you reboot the cluster after the rebuild?
>>
>>
>> Jonathan
>>
>> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>> >
>> >
>> >
>> >
>> > I just went in and looked at that and my subscriber is not showing up
>> as an
>> > available server in the Call Manager Group.  Any ideas?  It appears to
>> have
>> > replicated correctly, and I was able to use it to modify a Directory
>> Number.
>> >
>> >
>> >
>> >
>> >
>> > Brandon
>> >
>> >  ________________________________
>> >
>> >
>> > From: Scott Voll [mailto:svoll.voip at gmail.com]
>> >  Sent: Thursday, October 18, 2007 8:54 AM
>> >  To: Ogden, Brandon
>> >  Cc: cisco-voip at puck.nether.net
>> >  Subject: Re: [cisco-voip] Registration to Subscriber
>> >
>> >
>> >
>> >
>> >
>> > did you change the order of your servers so it registers with the sub?
>> >
>> >
>> >
>> >
>> >
>> > Scott
>> >
>> >
>> >
>> >
>> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>> >
>> >
>> >
>> > I rebuilt my subscriber box last night, it seemed to go well.  I was
>> > wondering how long it should take for the phones to register with the
>> > Subscriber rather than the pub.  As of this morning, they are still
>> > registered with the pub.  Thanks!
>> >
>> >
>> >
>> >
>> >
>> > Brandon
>> >
>> >
>> >
>> >
>> >  _______________________________________________
>> >  cisco-voip mailing list
>> >  cisco-voip at puck.nether.net
>> >  https://puck.nether.net/mailman/listinfo/cisco-voip
>> >
>> >
>> > _______________________________________________
>> > cisco-voip mailing list
>> > cisco-voip at puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >
>>
>> This e-mail may contain confidential and/or privileged information. If
>> you are not the intended recipient (or have received this e-mail in
>> error)
>> please notify the sender immediately and destroy this e-mail. Any
>> unauthorized copying, disclosure or distribution of the material in this
>> e-mail is strictly forbidden.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Thu Oct 18 14:22:44 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 13:22:44 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
	<f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
	<5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
	<D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>
	<5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>
	<021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0710181122l6d58b2b7h84967895def7494f@mail.gmail.com>

It depends

I have seen the service take forever on 100 patterns and start
immediately on 1000 patterns...


Jonathan

On 10/18/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> what's "a large number of route patterns" ?
>
> 100? 1,000? 10,000?
>
> ----- Original Message -----
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "Ogden, Brandon" <Brandon.Ogden at amersports.com>
> Cc: <cisco-voip at puck.nether.net>
> Sent: Thursday, October 18, 2007 2:14 PM
> Subject: Re: [cisco-voip] Registration to Subscriber
>
>
> > If you have a large number of route patterns the CCM service can time
> > out when starting
> >
> > There is a service parameter to increase the timeout, but I forgot
> > what it was...
> >
> >
> > Jonathan
> >
> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >> LOL.. I got it resolved.. Didn't have to reboot.. and before you say
> >> so.. yes I am an idiot.. LOL  The Call Manager Service had not started
> >> for some reason.. I restarted the service and it popped right up, all is
> >> well now.  Thanks all!
> >>
> >> Brandon
> >>
> >> -----Original Message-----
> >> From: cisco-voip-bounces at puck.nether.net
> >> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden, Brandon
> >> Sent: Thursday, October 18, 2007 10:18 AM
> >> To: Jonathan Charles
> >> Cc: cisco-voip at puck.nether.net
> >> Subject: Re: [cisco-voip] Registration to Subscriber
> >>
> >> I did not do a reboot of the publisher, just the sub.
> >>
> >> Brandon Ogden
> >> Amer Sports
> >> 801-624-7529
> >> www.amersports.com
> >> http://www.linkedin.com/in/brandonogden
> >>
> >> -----Original Message-----
> >> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> >> Sent: Thursday, October 18, 2007 10:14 AM
> >> To: Ogden, Brandon
> >> Cc: Scott Voll; cisco-voip at puck.nether.net
> >> Subject: Re: [cisco-voip] Registration to Subscriber
> >>
> >> Did you reboot the cluster after the rebuild?
> >>
> >>
> >> Jonathan
> >>
> >> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >> >
> >> >
> >> >
> >> >
> >> > I just went in and looked at that and my subscriber is not showing up
> >> as an
> >> > available server in the Call Manager Group.  Any ideas?  It appears to
> >> have
> >> > replicated correctly, and I was able to use it to modify a Directory
> >> Number.
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > Brandon
> >> >
> >> >  ________________________________
> >> >
> >> >
> >> > From: Scott Voll [mailto:svoll.voip at gmail.com]
> >> >  Sent: Thursday, October 18, 2007 8:54 AM
> >> >  To: Ogden, Brandon
> >> >  Cc: cisco-voip at puck.nether.net
> >> >  Subject: Re: [cisco-voip] Registration to Subscriber
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > did you change the order of your servers so it registers with the sub?
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > Scott
> >> >
> >> >
> >> >
> >> >
> >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >> >
> >> >
> >> >
> >> > I rebuilt my subscriber box last night, it seemed to go well.  I was
> >> > wondering how long it should take for the phones to register with the
> >> > Subscriber rather than the pub.  As of this morning, they are still
> >> > registered with the pub.  Thanks!
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > Brandon
> >> >
> >> >
> >> >
> >> >
> >> >  _______________________________________________
> >> >  cisco-voip mailing list
> >> >  cisco-voip at puck.nether.net
> >> >  https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >
> >> >
> >> > _______________________________________________
> >> > cisco-voip mailing list
> >> > cisco-voip at puck.nether.net
> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >
> >>
> >> This e-mail may contain confidential and/or privileged information. If
> >> you are not the intended recipient (or have received this e-mail in
> >> error)
> >> please notify the sender immediately and destroy this e-mail. Any
> >> unauthorized copying, disclosure or distribution of the material in this
> >> e-mail is strictly forbidden.
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From Erik.Stillman at Shearman.com  Thu Oct 18 14:25:27 2007
From: Erik.Stillman at Shearman.com (Erik Stillman)
Date: Thu, 18 Oct 2007 14:25:27 -0400
Subject: [cisco-voip] Different ring types for internal/external calls
In-Reply-To: <NYEXM4pmw60lFrDOo9I00017d6c@NYEXMB2.shearman.com>
Message-ID: <643A3A543DB5224499BC3D73473AEC27869DD5@NYEXMB2.shearman.com>

Thanks Lelio, but this request is for the phones to ring differently
when called from the PSTN.

-Erik 

-----Original Message-----
From: "Lelio Fulgenzi" <lelio at uoguelph.ca>@SHSDOMAIN 
Sent: Thursday, October 18, 2007 1:35 PM
To: Erik Stillman; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Different ring types for internal/external
calls

As long as you are using a CTI based autoattendant like IPCC Express or
Unity Express than this should be doable. There is an enterprise or
system parameter that differentiates external calls with a double ring.

If you are using Unity, then you are out of luck because it uses SCCP
and transfers calls as if a user is transferring them.

----- Original Message -----
From: "Erik Stillman" <Erik.Stillman at Shearman.com>
To: <cisco-voip at puck.nether.net>
Sent: Thursday, October 18, 2007 1:12 PM
Subject: [cisco-voip] Different ring types for internal/external calls


> Hey guys, I've got a hot one here. Request has come in to
differentiate
> calls by ring type.
>
> Is there anything I can do in CCM4.2 to achieve this?
>
> Any help would be great. Thanks!
>
> -Erik
>
>
>
***********************************************************************
>
> This communication is not intended to be used, and cannot be used,
> by the recipient or any other person for the purpose of avoiding
United
> States federal tax penalties that may be imposed on the recipient or
> such other person. In addition, if any United States federal tax
> advice contained in this communication is used or referred to in
> promoting, marketing or recommending any corporation, partnership
> or other entity, investment plan, concept, structure or arrangement
> (which should be assumed to be the case by a recipient or other
> person who is not our client with respect to the subject matter of the
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> please go to http://www.shearman.com/disclaimer/tax_disclosure.html
>
>
***********************************************************************
>
> This transmittal and/or attachments may be a confidential
> attorney-client communication or may otherwise be privileged or
> confidential. If you are not the intended recipient, you are hereby
> notified that you have received this transmittal in error; any review,
> dissemination, distribution or copying of this transmittal is strictly
> prohibited. If you have received this transmittal and/or attachments
in
> error, please notify us immediately by reply or by telephone (call us
> collect at +1 212-848-8400) and immediately delete this message and
> all its attachments.
>
> Shearman & Sterling LLP is a limited liability partnership organized
in
> the United States under the laws of the State of Delaware, which laws
> limit the personal liability of partners.
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



From ealeatherman at gmail.com  Thu Oct 18 14:32:30 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 18 Oct 2007 14:32:30 -0400
Subject: [cisco-voip] IPCC Express 4 Reason Codes
Message-ID: <94a1afde0710181132v187a98d8j7b1a8a4194f51c8f@mail.gmail.com>

Hello,

Trying to figure out what the reason code "32758 - Work state timer expired"
is supposed to mean. I have a 15 second work period on one of our CSQ's, and
90% of the time it works great, but there are a few cases where it is
putting the agent into Not Ready state when the timer expires, instead of
Ready. I do not believe this is something the agent is doing and normally
the agent notices and puts themselves back into Ready.. but there are
occasions where they miss it.

I think I may be hitting CSCsk06089, which is exactly the symptoms i have
but the MIVR logs don't match what is in the bug report. I don't want to
open up a TAC case on this until I understand what a legitimate case for
that reason code would be... in case it IS something we're doing to cause
it.

Thanks!

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From tednugent73 at gmail.com  Thu Oct 18 14:39:55 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 18 Oct 2007 14:39:55 -0400
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
Message-ID: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>

I'm not able to find any info on the support of CM 6.x Business Edition and
UCCX? Does anyone know if this is a supported combo with the way that BE is
being marketed? I'm sure it will work however I need to make sure that its
supported or not?
TIA
Ted
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From tednugent73 at gmail.com  Thu Oct 18 14:39:55 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 18 Oct 2007 14:39:55 -0400
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
Message-ID: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>

I'm not able to find any info on the support of CM 6.x Business Edition and
UCCX? Does anyone know if this is a supported combo with the way that BE is
being marketed? I'm sure it will work however I need to make sure that its
supported or not?
TIA
Ted
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From milay at cisco.com  Thu Oct 18 14:42:03 2007
From: milay at cisco.com (Mike Lay (milay))
Date: Thu, 18 Oct 2007 11:42:03 -0700
Subject: [cisco-voip] IPCC Express 4 Reason Codes
In-Reply-To: <94a1afde0710181132v187a98d8j7b1a8a4194f51c8f@mail.gmail.com>
References: <94a1afde0710181132v187a98d8j7b1a8a4194f51c8f@mail.gmail.com>
Message-ID: <413FABF5DCE22849AF86F442E472B78B02C6F386@xmb-sjc-219.amer.cisco.com>

Ed,
 
Here is some information I have found. I am not an IPCCX expert, just
trying to help out.
 
When an agent get off a IPCC routed call, they are automatically put
into work state, which tells them that they have configured seconds to
wrap-up
with what they are doing and after that they will be put into ready
state. After configured seconds, which is when the work state timer
expires, the
system tries to put the agent into ready state but cannot if they are
using the ICD line to make an out bound call and therefore you are
seeing the reason code of 32758 in the report which is what the system
is suppose to do.

You can either increase the wrap up timer or create another line for
these agents so that they can use that line to make outbound calls
instead of using the ICD line.

You can change the wrap up time setting from IPCC AppAdmin page.
(Subsystems->RMCM Subsystem --> CSQ-> Wrap up time)


Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, October 18, 2007 1:33 PM
To: Cisco Voip Mailing list
Subject: [cisco-voip] IPCC Express 4 Reason Codes


Hello,

Trying to figure out what the reason code "32758 - Work state timer
expired" is supposed to mean. I have a 15 second work period on one of
our CSQ's, and 90% of the time it works great, but there are a few cases
where it is putting the agent into Not Ready state when the timer
expires, instead of Ready. I do not believe this is something the agent
is doing and normally the agent notices and puts themselves back into
Ready.. but there are occasions where they miss it. 

I think I may be hitting CSCsk06089, which is exactly the symptoms i
have but the MIVR logs don't match what is in the bug report. I don't
want to open up a TAC case on this until I understand what a legitimate
case for that reason code would be... in case it IS something we're
doing to cause it. 

Thanks!

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 
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From ealeatherman at gmail.com  Thu Oct 18 14:44:20 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 18 Oct 2007 14:44:20 -0400
Subject: [cisco-voip] IPCC Express 4 Reason Codes
In-Reply-To: <413FABF5DCE22849AF86F442E472B78B02C6F386@xmb-sjc-219.amer.cisco.com>
References: <94a1afde0710181132v187a98d8j7b1a8a4194f51c8f@mail.gmail.com>
	<413FABF5DCE22849AF86F442E472B78B02C6F386@xmb-sjc-219.amer.cisco.com>
Message-ID: <94a1afde0710181144r54c77414k9b38616fd18f41c5@mail.gmail.com>

That's exactly what I was looking for, thank you very much Mike. Is that in
a document somewhere on CCO that I can read for future reference? Perhaps I
was looking in the wrong place.

On 10/18/07, Mike Lay (milay) <milay at cisco.com> wrote:
>
>  Ed,
>
> Here is some information I have found. I am not an IPCCX expert, just
> trying to help out.
>
> When an agent get off a IPCC routed call, they are automatically put
> into work state, which tells them that they have configured seconds to
> wrap-up
> with what they are doing and after that they will be put into ready
> state. After configured seconds, which is when the work state timer
> expires, the
> system tries to put the agent into ready state but cannot if they are
> using the ICD line to make an out bound call and therefore you are
> seeing the reason code of 32758 in the report which is what the system
> is suppose to do.
>
> You can either increase the wrap up timer or create another line for
> these agents so that they can use that line to make outbound calls
> instead of using the ICD line.
>
> You can change the wrap up time setting from IPCC AppAdmin page.
> (Subsystems->RMCM Subsystem --> CSQ-> Wrap up time)
>
> Mike
>
>  ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ed Leatherman
> *Sent:* Thursday, October 18, 2007 1:33 PM
> *To:* Cisco Voip Mailing list
> *Subject:* [cisco-voip] IPCC Express 4 Reason Codes
>
> Hello,
>
> Trying to figure out what the reason code "32758 - Work state timer
> expired" is supposed to mean. I have a 15 second work period on one of our
> CSQ's, and 90% of the time it works great, but there are a few cases where
> it is putting the agent into Not Ready state when the timer expires, instead
> of Ready. I do not believe this is something the agent is doing and normally
> the agent notices and puts themselves back into Ready.. but there are
> occasions where they miss it.
>
> I think I may be hitting CSCsk06089, which is exactly the symptoms i have
> but the MIVR logs don't match what is in the bug report. I don't want to
> open up a TAC case on this until I understand what a legitimate case for
> that reason code would be... in case it IS something we're doing to cause
> it.
>
> Thanks!
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Thu Oct 18 15:19:13 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 18 Oct 2007 15:19:13 -0400
Subject: [cisco-voip] Registration to Subscriber
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int>
	<f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
	<5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
	<D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>
	<5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>
	<021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca>
	<5d093f9a0710181122l6d58b2b7h84967895def7494f@mail.gmail.com>
Message-ID: <02ef01c811bb$c5a21730$56196883@cfs.uoguelph.ca>

gotta love the consistency... ;)

----- Original Message ----- 
From: "Jonathan Charles" <jonvoip at gmail.com>
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Cc: "Ogden, Brandon" <Brandon.Ogden at amersports.com>; 
<cisco-voip at puck.nether.net>
Sent: Thursday, October 18, 2007 2:22 PM
Subject: Re: [cisco-voip] Registration to Subscriber


> It depends
>
> I have seen the service take forever on 100 patterns and start
> immediately on 1000 patterns...
>
>
> Jonathan
>
> On 10/18/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>> what's "a large number of route patterns" ?
>>
>> 100? 1,000? 10,000?
>>
>> ----- Original Message -----
>> From: "Jonathan Charles" <jonvoip at gmail.com>
>> To: "Ogden, Brandon" <Brandon.Ogden at amersports.com>
>> Cc: <cisco-voip at puck.nether.net>
>> Sent: Thursday, October 18, 2007 2:14 PM
>> Subject: Re: [cisco-voip] Registration to Subscriber
>>
>>
>> > If you have a large number of route patterns the CCM service can time
>> > out when starting
>> >
>> > There is a service parameter to increase the timeout, but I forgot
>> > what it was...
>> >
>> >
>> > Jonathan
>> >
>> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>> >> LOL.. I got it resolved.. Didn't have to reboot.. and before you say
>> >> so.. yes I am an idiot.. LOL  The Call Manager Service had not started
>> >> for some reason.. I restarted the service and it popped right up, all 
>> >> is
>> >> well now.  Thanks all!
>> >>
>> >> Brandon
>> >>
>> >> -----Original Message-----
>> >> From: cisco-voip-bounces at puck.nether.net
>> >> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden, 
>> >> Brandon
>> >> Sent: Thursday, October 18, 2007 10:18 AM
>> >> To: Jonathan Charles
>> >> Cc: cisco-voip at puck.nether.net
>> >> Subject: Re: [cisco-voip] Registration to Subscriber
>> >>
>> >> I did not do a reboot of the publisher, just the sub.
>> >>
>> >> Brandon Ogden
>> >> Amer Sports
>> >> 801-624-7529
>> >> www.amersports.com
>> >> http://www.linkedin.com/in/brandonogden
>> >>
>> >> -----Original Message-----
>> >> From: Jonathan Charles [mailto:jonvoip at gmail.com]
>> >> Sent: Thursday, October 18, 2007 10:14 AM
>> >> To: Ogden, Brandon
>> >> Cc: Scott Voll; cisco-voip at puck.nether.net
>> >> Subject: Re: [cisco-voip] Registration to Subscriber
>> >>
>> >> Did you reboot the cluster after the rebuild?
>> >>
>> >>
>> >> Jonathan
>> >>
>> >> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > I just went in and looked at that and my subscriber is not showing 
>> >> > up
>> >> as an
>> >> > available server in the Call Manager Group.  Any ideas?  It appears 
>> >> > to
>> >> have
>> >> > replicated correctly, and I was able to use it to modify a Directory
>> >> Number.
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > Brandon
>> >> >
>> >> >  ________________________________
>> >> >
>> >> >
>> >> > From: Scott Voll [mailto:svoll.voip at gmail.com]
>> >> >  Sent: Thursday, October 18, 2007 8:54 AM
>> >> >  To: Ogden, Brandon
>> >> >  Cc: cisco-voip at puck.nether.net
>> >> >  Subject: Re: [cisco-voip] Registration to Subscriber
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > did you change the order of your servers so it registers with the 
>> >> > sub?
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > Scott
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>> >> >
>> >> >
>> >> >
>> >> > I rebuilt my subscriber box last night, it seemed to go well.  I was
>> >> > wondering how long it should take for the phones to register with 
>> >> > the
>> >> > Subscriber rather than the pub.  As of this morning, they are still
>> >> > registered with the pub.  Thanks!
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > Brandon
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >  _______________________________________________
>> >> >  cisco-voip mailing list
>> >> >  cisco-voip at puck.nether.net
>> >> >  https://puck.nether.net/mailman/listinfo/cisco-voip
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > cisco-voip mailing list
>> >> > cisco-voip at puck.nether.net
>> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >> >
>> >>
>> >> This e-mail may contain confidential and/or privileged information. If
>> >> you are not the intended recipient (or have received this e-mail in
>> >> error)
>> >> please notify the sender immediately and destroy this e-mail. Any
>> >> unauthorized copying, disclosure or distribution of the material in 
>> >> this
>> >> e-mail is strictly forbidden.
>> >> _______________________________________________
>> >> cisco-voip mailing list
>> >> cisco-voip at puck.nether.net
>> >> https://puck.nether.net/mailman/listinfo/cisco-voip
>> >>
>> > _______________________________________________
>> > cisco-voip mailing list
>> > cisco-voip at puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >
>>
> 


From jonvoip at gmail.com  Thu Oct 18 15:30:43 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 14:30:43 -0500
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <5d093f9a0710171720n26bfb455p51ce18522b7c4cd9@mail.gmail.com>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
	<47165BE1.4040502@cisco.com>
	<5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>
	<471691A1.8060409@wcyv.com>
	<5d093f9a0710171720n26bfb455p51ce18522b7c4cd9@mail.gmail.com>
Message-ID: <5d093f9a0710181230l53044c45gf093ff2a5f7c50fc@mail.gmail.com>

OK, I opened a TAC case, they said it would be 144 ports per server.

However, CallManager 4.3, 5.0, 5.1 and 6.0 only support a maximum of 96 ports...

(I have confirmed this on all of the above versions of CCM).



Jonathan

On 10/17/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> Yes, but CCM does not.
>
> Unity 5.0 liked the 144 ports, but CCM 4.3 did not.
>
>
>
>
> Jonathan
>
> On 10/17/07, Pat Hayes <pat-cv at wcyv.com> wrote:
> > Unity 5.0(1) does indeed support 144 ports per server:
> >
> > http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/release/notes/501curelnotes.html#wp409389
> >
> > I just tried with ccm 5.1(2) in the lab and you're definitely right,
> > when you try to create port 97, it throws an error. Unfortunately, it
> > looks like the bug Wes cited isn't fixed in anything yet, so, for the
> > time being, the only workaround would be to create two separate
> > integrations in Unity to the same CCM cluster.
> >
> > -------- Original Message  --------
> > From: "Jonathan Charles" <jonvoip at gmail.com>
> > To: "Wes Sisk" <wsisk at cisco.com>
> > Cc: Cisco Voip Mailing list <cisco-voip at puck.nether.net>
> > Subject: Re:[cisco-voip] Maximum number of voicemail ports...???
> > Date: 10/17/2007 3:15 PM
> >
> > > Really only allows.
> > >
> > > When I try to add a single port manually it gives me an error saying
> > > the valid range is 1-96
> > >
> > > I am pretty sure I am reading this wrong and it is 72 ports per server
> > >
> > >
> > > Jonathan
> > >
> > > On 10/17/07, Wes Sisk <wsisk at cisco.com> wrote:
> > >> really only allows, or wizard on allows? CSCsi00569
> > >>
> > >> /Wes
> > >>
> > >> Jonathan Charles wrote:
> > >>> So, I am installing a pair of Unity servers...  (primary and secondary)
> > >>>
> > >>> CCM version is 4.3(1)sr1
> > >>>
> > >>> Unity is 5.0(1)
> > >>>
> > >>> TSP is 8.2(1)
> > >>>
> > >>> Anyway, here's the problem... customer has a license for 144 voicemail
> > >>> ports... However, CCM will only let me add 96 per Unity server (if I
> > >>> try to add any, it doesn't let me), the max appears to be 96...
> > >>>
> > >>> If that is the case, why can you buy a 144 port license???
> > >>>
> > >>>
> > >>>
> > >>> Jonathan
> > >>> _______________________________________________
> > >>> cisco-voip mailing list
> > >>> cisco-voip at puck.nether.net
> > >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>>
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From lelio at uoguelph.ca  Thu Oct 18 14:34:50 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 18 Oct 2007 14:34:50 -0400
Subject: [cisco-voip] Different ring types for internal/external calls
References: <OF7CAE0A96.7759EDFB-ON85257378.006539B3@shearman.com>
Message-ID: <023d01c811b5$929266c0$56196883@cfs.uoguelph.ca>

Sure - it will work with that too. If your PSTN calls are not directed to an 
auto-attendant and are DIDs, that's even better.

...funny thing is, I can't find that service parameter now!

----- Original Message ----- 
From: "Erik Stillman" <erik.stillman at shearman.com>
To: <lelio at uoguelph.ca>; <cisco-voip at puck.nether.net>
Sent: Thursday, October 18, 2007 2:25 PM
Subject: RE: [cisco-voip] Different ring types for internal/external calls


> Thanks Lelio, but this request is for the phones to ring differently
> when called from the PSTN.
>
> -Erik
>
> -----Original Message-----
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>@SHSDOMAIN
> Sent: Thursday, October 18, 2007 1:35 PM
> To: Erik Stillman; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Different ring types for internal/external
> calls
>
> As long as you are using a CTI based autoattendant like IPCC Express or
> Unity Express than this should be doable. There is an enterprise or
> system parameter that differentiates external calls with a double ring.
>
> If you are using Unity, then you are out of luck because it uses SCCP
> and transfers calls as if a user is transferring them.
>
> ----- Original Message -----
> From: "Erik Stillman" <Erik.Stillman at Shearman.com>
> To: <cisco-voip at puck.nether.net>
> Sent: Thursday, October 18, 2007 1:12 PM
> Subject: [cisco-voip] Different ring types for internal/external calls
>
>
>> Hey guys, I've got a hot one here. Request has come in to
> differentiate
>> calls by ring type.
>>
>> Is there anything I can do in CCM4.2 to achieve this?
>>
>> Any help would be great. Thanks!
>>
>> -Erik
>>
>>
>>
> ***********************************************************************
>>
>> This communication is not intended to be used, and cannot be used,
>> by the recipient or any other person for the purpose of avoiding
> United
>> States federal tax penalties that may be imposed on the recipient or
>> such other person. In addition, if any United States federal tax
>> advice contained in this communication is used or referred to in
>> promoting, marketing or recommending any corporation, partnership
>> or other entity, investment plan, concept, structure or arrangement
>> (which should be assumed to be the case by a recipient or other
>> person who is not our client with respect to the subject matter of the
>> communication), then (i) such tax advice should be construed as
> written
>> to support the promotion or marketing of the transactions or matters
>> addressed by the advice and (ii) the recipient or other person should
>> seek advice based on the recipient or other person's particular
>> circumstances from an independent tax advisor. For further
> information,
>> please go to http://www.shearman.com/disclaimer/tax_disclosure.html
>>
>>
> ***********************************************************************
>>
>> This transmittal and/or attachments may be a confidential
>> attorney-client communication or may otherwise be privileged or
>> confidential. If you are not the intended recipient, you are hereby
>> notified that you have received this transmittal in error; any review,
>> dissemination, distribution or copying of this transmittal is strictly
>> prohibited. If you have received this transmittal and/or attachments
> in
>> error, please notify us immediately by reply or by telephone (call us
>> collect at +1 212-848-8400) and immediately delete this message and
>> all its attachments.
>>
>> Shearman & Sterling LLP is a limited liability partnership organized
> in
>> the United States under the laws of the State of Delaware, which laws
>> limit the personal liability of partners.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
>
> 


From d4rkf1ber at gmail.com  Thu Oct 18 15:59:44 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Thu, 18 Oct 2007 14:59:44 -0500
Subject: [cisco-voip] OT: Creating your own site to site PRI's?
Message-ID: <a1b07ea60710181259m63a3a4adq96ffe705ff7da7f6@mail.gmail.com>

I need to come up with a way to create my own PRI's between two
locations.  These will have to be stand alone, PRI's not related or
controlled by CallManager.  Its to tie a remote office PBX shelf unit
into the main office PBX.

See the call center still runs on an older PBX, we utilize CallManager
for everything else.  The remote office will be relocating soon and
well we have alot of fiber between the new location and the main
office, so ideally I would prefer to come up with our own PRI solution
rather then have to pay for and rely on BS/AT&T for PRIs.

Any thoughts or suggestions?

From wsisk at cisco.com  Thu Oct 18 16:05:45 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 18 Oct 2007 16:05:45 -0400
Subject: [cisco-voip] Different ring types for internal/external calls
In-Reply-To: <643A3A543DB5224499BC3D73473AEC27869DD5@NYEXMB2.shearman.com>
References: <643A3A543DB5224499BC3D73473AEC27869DD5@NYEXMB2.shearman.com>
Message-ID: <4717BC99.5070300@cisco.com>

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From lelio at uoguelph.ca  Thu Oct 18 16:06:17 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 18 Oct 2007 16:06:17 -0400
Subject: [cisco-voip] OT: Creating your own site to site PRI's?
References: <a1b07ea60710181259m63a3a4adq96ffe705ff7da7f6@mail.gmail.com>
Message-ID: <032801c811c2$57d4ca70$56196883@cfs.uoguelph.ca>

T1 extenders. We use them and have been happy with them. If it's your fibre, 
that's what I would do.

Something like this should do:

http://www.rad.com/Article/0,6583,33171-E1_T1_Optical_Line_Multiplexers,00.html


you can send up to four T1s over a fibre pair with this.


----- Original Message ----- 
From: "Dark Fiber" <d4rkf1ber at gmail.com>
To: <cisco-voip at puck.nether.net>
Sent: Thursday, October 18, 2007 3:59 PM
Subject: [cisco-voip] OT: Creating your own site to site PRI's?


>I need to come up with a way to create my own PRI's between two
> locations.  These will have to be stand alone, PRI's not related or
> controlled by CallManager.  Its to tie a remote office PBX shelf unit
> into the main office PBX.
>
> See the call center still runs on an older PBX, we utilize CallManager
> for everything else.  The remote office will be relocating soon and
> well we have alot of fiber between the new location and the main
> office, so ideally I would prefer to come up with our own PRI solution
> rather then have to pay for and rely on BS/AT&T for PRIs.
>
> Any thoughts or suggestions?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From wsisk at cisco.com  Thu Oct 18 16:08:53 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 18 Oct 2007 16:08:53 -0400
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <5d093f9a0710181230l53044c45gf093ff2a5f7c50fc@mail.gmail.com>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>	
	<47165BE1.4040502@cisco.com>	
	<5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>	
	<471691A1.8060409@wcyv.com>	
	<5d093f9a0710171720n26bfb455p51ce18522b7c4cd9@mail.gmail.com>
	<5d093f9a0710181230l53044c45gf093ff2a5f7c50fc@mail.gmail.com>
Message-ID: <4717BD55.7040406@cisco.com>

This is fixed by CSCsk20791

Integrated-releases:
004.003(001.117) 004.003(001.127) 005.001(002.9901.105) 
005.001(002.9901.107) 005.001(002.9901.108) 005.001(003.9901.009) 
005.001(003.9901.012) 006.001(000.9901.402) 006.001(000.9901.404) 
006.001(000.9901.421) 04.1(03)SR06

It's already fixed in 4.1.3sr6
With Jonathan's case we should get this fix into 5.1.3es, 5.1.2es, 
6.0.1es, 4.2.3es and 4.3.1es.

/Wes

Jonathan Charles wrote:
> OK, I opened a TAC case, they said it would be 144 ports per server.
>
> However, CallManager 4.3, 5.0, 5.1 and 6.0 only support a maximum of 96 ports...
>
> (I have confirmed this on all of the above versions of CCM).
>
>
>
> Jonathan
>
> On 10/17/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>   
>> Yes, but CCM does not.
>>
>> Unity 5.0 liked the 144 ports, but CCM 4.3 did not.
>>
>>
>>
>>
>> Jonathan
>>
>> On 10/17/07, Pat Hayes <pat-cv at wcyv.com> wrote:
>>     
>>> Unity 5.0(1) does indeed support 144 ports per server:
>>>
>>> http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/release/notes/501curelnotes.html#wp409389
>>>
>>> I just tried with ccm 5.1(2) in the lab and you're definitely right,
>>> when you try to create port 97, it throws an error. Unfortunately, it
>>> looks like the bug Wes cited isn't fixed in anything yet, so, for the
>>> time being, the only workaround would be to create two separate
>>> integrations in Unity to the same CCM cluster.
>>>
>>> -------- Original Message  --------
>>> From: "Jonathan Charles" <jonvoip at gmail.com>
>>> To: "Wes Sisk" <wsisk at cisco.com>
>>> Cc: Cisco Voip Mailing list <cisco-voip at puck.nether.net>
>>> Subject: Re:[cisco-voip] Maximum number of voicemail ports...???
>>> Date: 10/17/2007 3:15 PM
>>>
>>>       
>>>> Really only allows.
>>>>
>>>> When I try to add a single port manually it gives me an error saying
>>>> the valid range is 1-96
>>>>
>>>> I am pretty sure I am reading this wrong and it is 72 ports per server
>>>>
>>>>
>>>> Jonathan
>>>>
>>>> On 10/17/07, Wes Sisk <wsisk at cisco.com> wrote:
>>>>         
>>>>> really only allows, or wizard on allows? CSCsi00569
>>>>>
>>>>> /Wes
>>>>>
>>>>> Jonathan Charles wrote:
>>>>>           
>>>>>> So, I am installing a pair of Unity servers...  (primary and secondary)
>>>>>>
>>>>>> CCM version is 4.3(1)sr1
>>>>>>
>>>>>> Unity is 5.0(1)
>>>>>>
>>>>>> TSP is 8.2(1)
>>>>>>
>>>>>> Anyway, here's the problem... customer has a license for 144 voicemail
>>>>>> ports... However, CCM will only let me add 96 per Unity server (if I
>>>>>> try to add any, it doesn't let me), the max appears to be 96...
>>>>>>
>>>>>> If that is the case, why can you buy a 144 port license???
>>>>>>
>>>>>>
>>>>>>
>>>>>> Jonathan
>>>>>> _______________________________________________
>>>>>> cisco-voip mailing list
>>>>>> cisco-voip at puck.nether.net
>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>
>>>>>>             
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>         

From matt at saskin.net  Thu Oct 18 16:09:53 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 18 Oct 2007 16:09:53 -0400
Subject: [cisco-voip] OT: Creating your own site to site PRI's?
In-Reply-To: <032801c811c2$57d4ca70$56196883@cfs.uoguelph.ca>
References: <a1b07ea60710181259m63a3a4adq96ffe705ff7da7f6@mail.gmail.com>
	<032801c811c2$57d4ca70$56196883@cfs.uoguelph.ca>
Message-ID: <4717BD91.9040702@saskin.net>

Failing (or as an alternative to) that, you can do it with just H.323 
gateways pointing to each other via dial-peers.

-matt

Lelio Fulgenzi wrote:
> T1 extenders. We use them and have been happy with them. If it's your fibre, 
> that's what I would do.
> 
> Something like this should do:
> 
> http://www.rad.com/Article/0,6583,33171-E1_T1_Optical_Line_Multiplexers,00.html
> 
> 
> you can send up to four T1s over a fibre pair with this.
> 
> 
> ----- Original Message ----- 
> From: "Dark Fiber" <d4rkf1ber at gmail.com>
> To: <cisco-voip at puck.nether.net>
> Sent: Thursday, October 18, 2007 3:59 PM
> Subject: [cisco-voip] OT: Creating your own site to site PRI's?
> 
> 
>> I need to come up with a way to create my own PRI's between two
>> locations.  These will have to be stand alone, PRI's not related or
>> controlled by CallManager.  Its to tie a remote office PBX shelf unit
>> into the main office PBX.
>>
>> See the call center still runs on an older PBX, we utilize CallManager
>> for everything else.  The remote office will be relocating soon and
>> well we have alot of fiber between the new location and the main
>> office, so ideally I would prefer to come up with our own PRI solution
>> rather then have to pay for and rely on BS/AT&T for PRIs.
>>
>> Any thoughts or suggestions?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip




From jonvoip at gmail.com  Thu Oct 18 16:43:00 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 15:43:00 -0500
Subject: [cisco-voip] Maximum number of voicemail ports...???
In-Reply-To: <4717BD55.7040406@cisco.com>
References: <5d093f9a0710171130q5246b85fm452d29d11629b524@mail.gmail.com>
	<47165BE1.4040502@cisco.com>
	<5d093f9a0710171215h7e4a6908oce93a978f22f2fe3@mail.gmail.com>
	<471691A1.8060409@wcyv.com>
	<5d093f9a0710171720n26bfb455p51ce18522b7c4cd9@mail.gmail.com>
	<5d093f9a0710181230l53044c45gf093ff2a5f7c50fc@mail.gmail.com>
	<4717BD55.7040406@cisco.com>
Message-ID: <5d093f9a0710181343l2ca0354bu3ae3f1192807600b@mail.gmail.com>

Yeah, they sent me ES21 which fixes SA WEB...




Jonathan

On 10/18/07, Wes Sisk <wsisk at cisco.com> wrote:
> This is fixed by CSCsk20791
>
> Integrated-releases:
> 004.003(001.117) 004.003(001.127) 005.001(002.9901.105)
> 005.001(002.9901.107) 005.001(002.9901.108) 005.001(003.9901.009)
> 005.001(003.9901.012) 006.001(000.9901.402) 006.001(000.9901.404)
> 006.001(000.9901.421) 04.1(03)SR06
>
> It's already fixed in 4.1.3sr6
> With Jonathan's case we should get this fix into 5.1.3es, 5.1.2es,
> 6.0.1es, 4.2.3es and 4.3.1es.
>
> /Wes
>
> Jonathan Charles wrote:
> > OK, I opened a TAC case, they said it would be 144 ports per server.
> >
> > However, CallManager 4.3, 5.0, 5.1 and 6.0 only support a maximum of 96 ports...
> >
> > (I have confirmed this on all of the above versions of CCM).
> >
> >
> >
> > Jonathan
> >
> > On 10/17/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> >> Yes, but CCM does not.
> >>
> >> Unity 5.0 liked the 144 ports, but CCM 4.3 did not.
> >>
> >>
> >>
> >>
> >> Jonathan
> >>
> >> On 10/17/07, Pat Hayes <pat-cv at wcyv.com> wrote:
> >>
> >>> Unity 5.0(1) does indeed support 144 ports per server:
> >>>
> >>> http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/release/notes/501curelnotes.html#wp409389
> >>>
> >>> I just tried with ccm 5.1(2) in the lab and you're definitely right,
> >>> when you try to create port 97, it throws an error. Unfortunately, it
> >>> looks like the bug Wes cited isn't fixed in anything yet, so, for the
> >>> time being, the only workaround would be to create two separate
> >>> integrations in Unity to the same CCM cluster.
> >>>
> >>> -------- Original Message  --------
> >>> From: "Jonathan Charles" <jonvoip at gmail.com>
> >>> To: "Wes Sisk" <wsisk at cisco.com>
> >>> Cc: Cisco Voip Mailing list <cisco-voip at puck.nether.net>
> >>> Subject: Re:[cisco-voip] Maximum number of voicemail ports...???
> >>> Date: 10/17/2007 3:15 PM
> >>>
> >>>
> >>>> Really only allows.
> >>>>
> >>>> When I try to add a single port manually it gives me an error saying
> >>>> the valid range is 1-96
> >>>>
> >>>> I am pretty sure I am reading this wrong and it is 72 ports per server
> >>>>
> >>>>
> >>>> Jonathan
> >>>>
> >>>> On 10/17/07, Wes Sisk <wsisk at cisco.com> wrote:
> >>>>
> >>>>> really only allows, or wizard on allows? CSCsi00569
> >>>>>
> >>>>> /Wes
> >>>>>
> >>>>> Jonathan Charles wrote:
> >>>>>
> >>>>>> So, I am installing a pair of Unity servers...  (primary and secondary)
> >>>>>>
> >>>>>> CCM version is 4.3(1)sr1
> >>>>>>
> >>>>>> Unity is 5.0(1)
> >>>>>>
> >>>>>> TSP is 8.2(1)
> >>>>>>
> >>>>>> Anyway, here's the problem... customer has a license for 144 voicemail
> >>>>>> ports... However, CCM will only let me add 96 per Unity server (if I
> >>>>>> try to add any, it doesn't let me), the max appears to be 96...
> >>>>>>
> >>>>>> If that is the case, why can you buy a 144 port license???
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Jonathan
> >>>>>> _______________________________________________
> >>>>>> cisco-voip mailing list
> >>>>>> cisco-voip at puck.nether.net
> >>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>>>>
> >>>>>>
> >>>> _______________________________________________
> >>>> cisco-voip mailing list
> >>>> cisco-voip at puck.nether.net
> >>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>>
>

From Erik.Stillman at Shearman.com  Thu Oct 18 16:54:21 2007
From: Erik.Stillman at Shearman.com (Erik Stillman)
Date: Thu, 18 Oct 2007 16:54:21 -0400
Subject: [cisco-voip] Different ring types for internal/external calls
In-Reply-To: <NYEXM4domP0l0HQFDMz00017f35@NYEXMB2.shearman.com>
Message-ID: <643A3A543DB5224499BC3D73473AEC27869E63@NYEXMB2.shearman.com>

Genius Wes!
 
I did a hard shut no/shut of MGCP and it works!

________________________________

From: Wes Sisk <wsisk at cisco.com>@SHSDOMAIN 
Sent: Thursday, October 18, 2007 4:06 PM
To: Erik Stillman
Cc: Lelio Fulgenzi; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Different ring types for internal/external
calls


In your PSTN gateway configuration make sure the 'Call Classification'
is set to OffNet or use system default.
Make sure advanced CM service parameter "Call Classification" = OffNet.

/Wes

Erik Stillman wrote: 

	Thanks Lelio, but this request is for the phones to ring
differently
	when called from the PSTN.
	
	-Erik 
	
	-----Original Message-----
	From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
<mailto:lelio at uoguelph.ca> @SHSDOMAIN 
	Sent: Thursday, October 18, 2007 1:35 PM
	To: Erik Stillman; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Different ring types for
internal/external
	calls
	
	As long as you are using a CTI based autoattendant like IPCC
Express or
	Unity Express than this should be doable. There is an enterprise
or
	system parameter that differentiates external calls with a
double ring.
	
	If you are using Unity, then you are out of luck because it uses
SCCP
	and transfers calls as if a user is transferring them.
	
	----- Original Message -----
	From: "Erik Stillman" <Erik.Stillman at Shearman.com>
<mailto:Erik.Stillman at Shearman.com> 
	To: <cisco-voip at puck.nether.net>
<mailto:cisco-voip at puck.nether.net> 
	Sent: Thursday, October 18, 2007 1:12 PM
	Subject: [cisco-voip] Different ring types for internal/external
calls
	
	
	  

		Hey guys, I've got a hot one here. Request has come in
to
		    

	differentiate
	  

		calls by ring type.
		
		Is there anything I can do in CCM4.2 to achieve this?
		
		Any help would be great. Thanks!
		
		-Erik
		
		
		
		    

	
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From wsisk at cisco.com  Thu Oct 18 17:12:08 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 18 Oct 2007 17:12:08 -0400
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
In-Reply-To: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>
References: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>
Message-ID: <4717CC28.70806@cisco.com>

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From jonvoip at gmail.com  Thu Oct 18 17:31:10 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 16:31:10 -0500
Subject: [cisco-voip] Unity 5.0 fix for SA WEB page not coming up in Windows
	2003
Message-ID: <5d093f9a0710181431m79362397m637e5dc7884062b3@mail.gmail.com>

Call TAC and request Unity 5.0(1) ES 21

That supposedly fixes the issue

Unity 4.2(1) ES116 fixes it for 4.2




Jonathan

From tednugent73 at gmail.com  Thu Oct 18 17:37:23 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 18 Oct 2007 17:37:23 -0400
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
In-Reply-To: <4717CC28.70806@cisco.com>
References: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>
	<4717CC28.70806@cisco.com>
Message-ID: <22e6380b0710181437y51f08477w9b4180d34ab7c888@mail.gmail.com>

Thanks for looking into that Wes however my question was more is integrating
IPCCx with CMBE supported on 2 separate platforms, CM on one server and
IPCCE on another. I know CUCM-BE does not include IPCCx. This may be a
stupid question however one would have guessed that BE would support ICT but
apparently it doesn't even though there's nothing preventing you from
setting them up... I'm just trying to prevent my sales guys from selling BE
and IPCCx and then finding out that Cisco has no intentions of supporting
that setup. Thanks!
Ted




On 10/18/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> from the PM: "IPCC Express is not included in the CUCMBE."
>
> /Wes
>
> Ted Nugent wrote:
>
>  I'm not able to find any info on the support of CM 6.x Business Edition
> and UCCX? Does anyone know if this is a supported combo with the way that BE
> is being marketed? I'm sure it will work however I need to make sure that
> its supported or not?
> TIA
> Ted
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From wsisk at cisco.com  Thu Oct 18 17:41:32 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 18 Oct 2007 17:41:32 -0400
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
In-Reply-To: <22e6380b0710181437y51f08477w9b4180d34ab7c888@mail.gmail.com>
References: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>	
	<4717CC28.70806@cisco.com>
	<22e6380b0710181437y51f08477w9b4180d34ab7c888@mail.gmail.com>
Message-ID: <4717D30C.5060705@cisco.com>

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From Andrew.Riley at getronics.com  Thu Oct 18 18:43:35 2007
From: Andrew.Riley at getronics.com (Riley, Andrew)
Date: Fri, 19 Oct 2007 08:43:35 +1000
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
References: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com><4717CC28.70806@cisco.com>
	<22e6380b0710181437y51f08477w9b4180d34ab7c888@mail.gmail.com>
Message-ID: <A49371C6FF101B46BD3E3A8C89985F023DCAA9@excausy200.australia.unity>

Great question.. I know a few sales guys that will try to do this. 

I would assume that you could considering the primary dependancy on CCM would just be CTI and DC Directory.. 

I am also interested to know the answer to this.

Andrew Riley


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net on behalf of Ted Nugent
Sent: Fri 10/19/2007 7:37 AM
To: Wes Sisk
Cc: cisco voip list
Subject: Re: [cisco-voip] CUCM-BE 6.x support for UCCX?
 
Thanks for looking into that Wes however my question was more is integrating
IPCCx with CMBE supported on 2 separate platforms, CM on one server and
IPCCE on another. I know CUCM-BE does not include IPCCx. This may be a
stupid question however one would have guessed that BE would support ICT but
apparently it doesn't even though there's nothing preventing you from
setting them up... I'm just trying to prevent my sales guys from selling BE
and IPCCx and then finding out that Cisco has no intentions of supporting
that setup. Thanks!
Ted




On 10/18/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> from the PM: "IPCC Express is not included in the CUCMBE."
>
> /Wes
>
> Ted Nugent wrote:
>
>  I'm not able to find any info on the support of CM 6.x Business Edition
> and UCCX? Does anyone know if this is a supported combo with the way that BE
> is being marketed? I'm sure it will work however I need to make sure that
> its supported or not?
> TIA
> Ted
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
>
>


From jonvoip at gmail.com  Thu Oct 18 19:11:30 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 18:11:30 -0500
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
In-Reply-To: <A49371C6FF101B46BD3E3A8C89985F023DCAA9@excausy200.australia.unity>
References: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>
	<4717CC28.70806@cisco.com>
	<22e6380b0710181437y51f08477w9b4180d34ab7c888@mail.gmail.com>
	<A49371C6FF101B46BD3E3A8C89985F023DCAA9@excausy200.australia.unity>
Message-ID: <5d093f9a0710181611w269513eai4ec60572d5eae1f9@mail.gmail.com>

CUCMBE = Cucumber

Spread it around, I would like to see this official by the end of October.



Jonathan

On 10/18/07, Riley, Andrew <Andrew.Riley at getronics.com> wrote:
> Great question.. I know a few sales guys that will try to do this.
>
> I would assume that you could considering the primary dependancy on CCM would just be CTI and DC Directory..
>
> I am also interested to know the answer to this.
>
> Andrew Riley
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net on behalf of Ted Nugent
> Sent: Fri 10/19/2007 7:37 AM
> To: Wes Sisk
> Cc: cisco voip list
> Subject: Re: [cisco-voip] CUCM-BE 6.x support for UCCX?
>
> Thanks for looking into that Wes however my question was more is integrating
> IPCCx with CMBE supported on 2 separate platforms, CM on one server and
> IPCCE on another. I know CUCM-BE does not include IPCCx. This may be a
> stupid question however one would have guessed that BE would support ICT but
> apparently it doesn't even though there's nothing preventing you from
> setting them up... I'm just trying to prevent my sales guys from selling BE
> and IPCCx and then finding out that Cisco has no intentions of supporting
> that setup. Thanks!
> Ted
>
>
>
>
> On 10/18/07, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > from the PM: "IPCC Express is not included in the CUCMBE."
> >
> > /Wes
> >
> > Ted Nugent wrote:
> >
> >  I'm not able to find any info on the support of CM 6.x Business Edition
> > and UCCX? Does anyone know if this is a supported combo with the way that BE
> > is being marketed? I'm sure it will work however I need to make sure that
> > its supported or not?
> > TIA
> > Ted
> >
> > ------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Thu Oct 18 19:12:32 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 18:12:32 -0500
Subject: [cisco-voip] CCME GUI adding phones "no phone to add"
In-Reply-To: <823476.82688.qm@web57215.mail.re3.yahoo.com>
References: <C1FE15183DA37645BC0633BC604E44F00724B234@USNAEXCH.na.didata.local>
	<823476.82688.qm@web57215.mail.re3.yahoo.com>
Message-ID: <5d093f9a0710181612i6f003565q4128e7fdc24865a1@mail.gmail.com>

People are actually out there configuring CCME via the GUI????

Wow.


Jonathan

On 10/17/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> FYI, for anyone that is encountering an error dialog
> of "no phones to add" from the CCME GUI, you have to
> go into the box and under telephony-service mode, turn
> auto-reg-ephone back on.
>
> hope that helps someone!
> Paul
>
>
>       ____________________________________________________________________________________
> Shape Yahoo! in your own image.  Join our Network Research Panel today!   http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From kenjohnson at letu.edu  Thu Oct 18 19:31:16 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Thu, 18 Oct 2007 18:31:16 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <02ef01c811bb$c5a21730$56196883@cfs.uoguelph.ca>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com><5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int><f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int><5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int><D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int><5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com><021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca><5d093f9a0710181122l6d58b2b7h84967895def7494f@mail.gmail.com>
	<02ef01c811bb$c5a21730$56196883@cfs.uoguelph.ca>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EF88DAA8@rachael-reborn.letnet.net>

Actually we've had this problem sporadically with our subscriber since
some 4.2(X) patch with the CCM service not consistently starting on
bootup (even with the service recovery processes set). It always starts
okay manually once the monitor emails remind us of the issue - and it
never affects the publisher, nor does it affect any other service or
write anything useful in the error log.

Since it never fails while running, only fails to come up with a reboot
or other CCM restart which almost never happens - even though TAC could
certainly fix it for us, we ignored it and focused on more pressing
cases figuring it would be a non-issue with the wipe-and-reload CM6.0
process....

Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
 E-mail: kenjohnson at letu.edu
 Helpdesk: (903) 233-3500
 Phone: (903) 233-3520
 Web: http://www.letu.edu/infotech/
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Thursday, October 18, 2007 2:19 PM
To: Jonathan Charles
Cc: Ogden, Brandon; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Registration to Subscriber

gotta love the consistency... ;)

----- Original Message -----
From: "Jonathan Charles" <jonvoip at gmail.com>
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Cc: "Ogden, Brandon" <Brandon.Ogden at amersports.com>;
<cisco-voip at puck.nether.net>
Sent: Thursday, October 18, 2007 2:22 PM
Subject: Re: [cisco-voip] Registration to Subscriber


> It depends
>
> I have seen the service take forever on 100 patterns and start 
> immediately on 1000 patterns...
>
>
> Jonathan
>
> On 10/18/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>> what's "a large number of route patterns" ?
>>
>> 100? 1,000? 10,000?
>>
>> ----- Original Message -----
>> From: "Jonathan Charles" <jonvoip at gmail.com>
>> To: "Ogden, Brandon" <Brandon.Ogden at amersports.com>
>> Cc: <cisco-voip at puck.nether.net>
>> Sent: Thursday, October 18, 2007 2:14 PM
>> Subject: Re: [cisco-voip] Registration to Subscriber
>>
>>
>> > If you have a large number of route patterns the CCM service can 
>> > time out when starting
>> >
>> > There is a service parameter to increase the timeout, but I forgot 
>> > what it was...
>> >
>> >
>> > Jonathan
>> >
>> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>> >> LOL.. I got it resolved.. Didn't have to reboot.. and before you 
>> >> say so.. yes I am an idiot.. LOL  The Call Manager Service had not

>> >> started for some reason.. I restarted the service and it popped 
>> >> right up, all is well now.  Thanks all!
>> >>
>> >> Brandon
>> >>
>> >> -----Original Message-----
>> >> From: cisco-voip-bounces at puck.nether.net
>> >> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden, 
>> >> Brandon
>> >> Sent: Thursday, October 18, 2007 10:18 AM
>> >> To: Jonathan Charles
>> >> Cc: cisco-voip at puck.nether.net
>> >> Subject: Re: [cisco-voip] Registration to Subscriber
>> >>
>> >> I did not do a reboot of the publisher, just the sub.
>> >>
>> >> Brandon Ogden
>> >> Amer Sports
>> >> 801-624-7529
>> >> www.amersports.com
>> >> http://www.linkedin.com/in/brandonogden
>> >>
>> >> -----Original Message-----
>> >> From: Jonathan Charles [mailto:jonvoip at gmail.com]
>> >> Sent: Thursday, October 18, 2007 10:14 AM
>> >> To: Ogden, Brandon
>> >> Cc: Scott Voll; cisco-voip at puck.nether.net
>> >> Subject: Re: [cisco-voip] Registration to Subscriber
>> >>
>> >> Did you reboot the cluster after the rebuild?
>> >>
>> >>
>> >> Jonathan
>> >>
>> >> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > I just went in and looked at that and my subscriber is not 
>> >> > showing up
>> >> as an
>> >> > available server in the Call Manager Group.  Any ideas?  It 
>> >> > appears to
>> >> have
>> >> > replicated correctly, and I was able to use it to modify a 
>> >> > Directory
>> >> Number.
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > Brandon
>> >> >
>> >> >  ________________________________
>> >> >
>> >> >
>> >> > From: Scott Voll [mailto:svoll.voip at gmail.com]
>> >> >  Sent: Thursday, October 18, 2007 8:54 AM
>> >> >  To: Ogden, Brandon
>> >> >  Cc: cisco-voip at puck.nether.net
>> >> >  Subject: Re: [cisco-voip] Registration to Subscriber
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > did you change the order of your servers so it registers with 
>> >> > the sub?
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > Scott
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com>
wrote:
>> >> >
>> >> >
>> >> >
>> >> > I rebuilt my subscriber box last night, it seemed to go well.  I

>> >> > was wondering how long it should take for the phones to register

>> >> > with the Subscriber rather than the pub.  As of this morning, 
>> >> > they are still registered with the pub.  Thanks!
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > Brandon
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >  _______________________________________________
>> >> >  cisco-voip mailing list
>> >> >  cisco-voip at puck.nether.net
>> >> >  https://puck.nether.net/mailman/listinfo/cisco-voip
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > cisco-voip mailing list
>> >> > cisco-voip at puck.nether.net
>> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >> >
>> >>
>> >> This e-mail may contain confidential and/or privileged 
>> >> information. If you are not the intended recipient (or have 
>> >> received this e-mail in
>> >> error)
>> >> please notify the sender immediately and destroy this e-mail. Any 
>> >> unauthorized copying, disclosure or distribution of the material 
>> >> in this e-mail is strictly forbidden.
>> >> _______________________________________________
>> >> cisco-voip mailing list
>> >> cisco-voip at puck.nether.net
>> >> https://puck.nether.net/mailman/listinfo/cisco-voip
>> >>
>> > _______________________________________________
>> > cisco-voip mailing list
>> > cisco-voip at puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >
>>
> 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From tednugent73 at gmail.com  Thu Oct 18 19:39:26 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 18 Oct 2007 19:39:26 -0400
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
In-Reply-To: <4717D30C.5060705@cisco.com>
References: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>
	<4717CC28.70806@cisco.com>
	<22e6380b0710181437y51f08477w9b4180d34ab7c888@mail.gmail.com>
	<4717D30C.5060705@cisco.com>
Message-ID: <22e6380b0710181639m18976651y66c8afc3b8b1323e@mail.gmail.com>

Ok sorry... that was actually a clarifcation to you and the Product
Manager... He response led me to beleive that he thought i was asking if
IPCC was included IN BE?

On 10/18/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> Ted,
>
> Good clarification.  It is a bit fuzzy but IPCCX was clear because of the
> extra device weight on the server.
>
> /Wes
>
> Ted Nugent wrote:
>
> Thanks for looking into that Wes however my question was more is
> integrating IPCCx with CMBE supported on 2 separate platforms, CM on one
> server and IPCCE on another. I know CUCM-BE does not include IPCCx. This may
> be a stupid question however one would have guessed that BE would support
> ICT but apparently it doesn't even though there's nothing preventing you
> from setting them up... I'm just trying to prevent my sales guys from
> selling BE and IPCCx and then finding out that Cisco has no intentions of
> supporting that setup. Thanks!
> Ted
>
>
>
>
> On 10/18/07, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > from the PM: "IPCC Express is not included in the CUCMBE."
> >
> > /Wes
> >
> > Ted Nugent wrote:
> >
> >  I'm not able to find any info on the support of CM 6.x Business Edition
> > and UCCX? Does anyone know if this is a supported combo with the way that BE
> > is being marketed? I'm sure it will work however I need to make sure that
> > its supported or not?
> > TIA
> > Ted
> >
> > ------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
-------------- next part --------------
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URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20071018/e6a4876e/attachment.html 

From tednugent73 at gmail.com  Thu Oct 18 19:44:56 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 18 Oct 2007 19:44:56 -0400
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
In-Reply-To: <22e6380b0710181639m18976651y66c8afc3b8b1323e@mail.gmail.com>
References: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>
	<4717CC28.70806@cisco.com>
	<22e6380b0710181437y51f08477w9b4180d34ab7c888@mail.gmail.com>
	<4717D30C.5060705@cisco.com>
	<22e6380b0710181639m18976651y66c8afc3b8b1323e@mail.gmail.com>
Message-ID: <22e6380b0710181644h7bf4d3e0xf35412f2952e1ed7@mail.gmail.com>

Damn keyboard freaked out... To continue...

Ok sorry... that was actually a clarifcation to you and the Product
Manager... His response led me to beleive that he thought i was dumb
ass asking if IPCC was included IN BE?
"IPCC Express is not included in the CUCMBE"

So judging from your previous response it looks like that was just a typo
and its actually not supported becuase of Device wieght running CM and UC on
a MCS7828. Is that the gist?
Sorry for the confusion.. and Thanks




On 10/18/07, Ted Nugent <tednugent73 at gmail.com> wrote:
>
> Ok sorry... that was actually a clarifcation to you and the Product
> Manager... He response led me to beleive that he thought i was asking if
> IPCC was included IN BE?
>
> On 10/18/07, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > Ted,
> >
> > Good clarification.  It is a bit fuzzy but IPCCX was clear because of
> > the extra device weight on the server.
> >
> > /Wes
> >
> > Ted Nugent wrote:
> >
> > Thanks for looking into that Wes however my question was more is
> > integrating IPCCx with CMBE supported on 2 separate platforms, CM on one
> > server and IPCCE on another. I know CUCM-BE does not include IPCCx. This may
> > be a stupid question however one would have guessed that BE would support
> > ICT but apparently it doesn't even though there's nothing preventing you
> > from setting them up... I'm just trying to prevent my sales guys from
> > selling BE and IPCCx and then finding out that Cisco has no intentions of
> > supporting that setup. Thanks!
> > Ted
> >
> >
> >
> >
> > On 10/18/07, Wes Sisk <wsisk at cisco.com> wrote:
> > >
> > > from the PM: "IPCC Express is not included in the CUCMBE."
> > >
> > > /Wes
> > >
> > > Ted Nugent wrote:
> > >
> > >  I'm not able to find any info on the support of CM 6.x Business
> > > Edition and UCCX? Does anyone know if this is a supported combo with the way
> > > that BE is being marketed? I'm sure it will work however I need to make sure
> > > that its supported or not?
> > > TIA
> > > Ted
> > >
> > > ------------------------------
> > >
> > > _______________________________________________
> > > cisco-voip mailing listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> >
>
-------------- next part --------------
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URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20071018/721e4ef0/attachment.html 

From asobihoudai at yahoo.com  Thu Oct 18 20:16:03 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 18 Oct 2007 17:16:03 -0700 (PDT)
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>
Message-ID: <835082.21017.qm@web57203.mail.re3.yahoo.com>

I think it's the T302 timeout underneath enterprise
parameters. There is documentation on the timers
somewhere...I don't remember where I saw it
unfortunately.

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> If you have a large number of route patterns the CCM
> service can time
> out when starting
> 
> There is a service parameter to increase the
> timeout, but I forgot
> what it was...
> 
> 
> Jonathan
> 
> On 10/18/07, Ogden, Brandon
> <Brandon.Ogden at amersports.com> wrote:
> > LOL.. I got it resolved.. Didn't have to reboot..
> and before you say
> > so.. yes I am an idiot.. LOL  The Call Manager
> Service had not started
> > for some reason.. I restarted the service and it
> popped right up, all is
> > well now.  Thanks all!
> >
> > Brandon
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ogden, Brandon
> > Sent: Thursday, October 18, 2007 10:18 AM
> > To: Jonathan Charles
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Registration to
> Subscriber
> >
> > I did not do a reboot of the publisher, just the
> sub.
> >
> > Brandon Ogden
> > Amer Sports
> > 801-624-7529
> > www.amersports.com
> > http://www.linkedin.com/in/brandonogden
> >
> > -----Original Message-----
> > From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > Sent: Thursday, October 18, 2007 10:14 AM
> > To: Ogden, Brandon
> > Cc: Scott Voll; cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Registration to
> Subscriber
> >
> > Did you reboot the cluster after the rebuild?
> >
> >
> > Jonathan
> >
> > On 10/18/07, Ogden, Brandon
> <Brandon.Ogden at amersports.com> wrote:
> > >
> > >
> > >
> > >
> > > I just went in and looked at that and my
> subscriber is not showing up
> > as an
> > > available server in the Call Manager Group.  Any
> ideas?  It appears to
> > have
> > > replicated correctly, and I was able to use it
> to modify a Directory
> > Number.
> > >
> > >
> > >
> > >
> > >
> > > Brandon
> > >
> > >  ________________________________
> > >
> > >
> > > From: Scott Voll [mailto:svoll.voip at gmail.com]
> > >  Sent: Thursday, October 18, 2007 8:54 AM
> > >  To: Ogden, Brandon
> > >  Cc: cisco-voip at puck.nether.net
> > >  Subject: Re: [cisco-voip] Registration to
> Subscriber
> > >
> > >
> > >
> > >
> > >
> > > did you change the order of your servers so it
> registers with the sub?
> > >
> > >
> > >
> > >
> > >
> > > Scott
> > >
> > >
> > >
> > >
> > > On 10/18/07, Ogden, Brandon
> <Brandon.Ogden at amersports.com> wrote:
> > >
> > >
> > >
> > > I rebuilt my subscriber box last night, it
> seemed to go well.  I was
> > > wondering how long it should take for the phones
> to register with the
> > > Subscriber rather than the pub.  As of this
> morning, they are still
> > > registered with the pub.  Thanks!
> > >
> > >
> > >
> > >
> > >
> > > Brandon
> > >
> > >
> > >
> > >
> > >  _______________________________________________
> > >  cisco-voip mailing list
> > >  cisco-voip at puck.nether.net
> > > 
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> > This e-mail may contain confidential and/or
> privileged information. If
> > you are not the intended recipient (or have
> received this e-mail in
> > error)
> > please notify the sender immediately and destroy
> this e-mail. Any
> > unauthorized copying, disclosure or distribution
> of the material in this
> > e-mail is strictly forbidden.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________
Pinpoint customers who are looking for what you sell. 
http://searchmarketing.yahoo.com/

From asobihoudai at yahoo.com  Thu Oct 18 20:23:48 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 18 Oct 2007 17:23:48 -0700 (PDT)
Subject: [cisco-voip] CCME GUI adding phones "no phone to add"
In-Reply-To: <5d093f9a0710181612i6f003565q4128e7fdc24865a1@mail.gmail.com>
Message-ID: <15563.19244.qm@web57204.mail.re3.yahoo.com>

Yes Jonathan, GUI-using CCME users do exist.

For clarification, all major configuration was done
through CLI. The system was handed over to the
customer so they would be able to perform
adds/moves/changes easily without having to get
someone with AAA access involved.

I think the GUI works pretty well for end users. I'm
fairly satisfied with it...so long as there's no
aaa authentication config-commands in the router.

Paul
--- Jonathan Charles <jonvoip at gmail.com> wrote:

> People are actually out there configuring CCME via
> the GUI????
> 
> Wow.
> 
> 
> Jonathan
> 
> On 10/17/07, Paul Choi <asobihoudai at yahoo.com>
> wrote:
> > FYI, for anyone that is encountering an error
> dialog
> > of "no phones to add" from the CCME GUI, you have
> to
> > go into the box and under telephony-service mode,
> turn
> > auto-reg-ephone back on.
> >
> > hope that helps someone!
> > Paul
> >
> >
> >      
>
____________________________________________________________________________________
> > Shape Yahoo! in your own image.  Join our Network
> Research Panel today!  
>
http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 



       
____________________________________________________________________________________
Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. 
http://mobile.yahoo.com/go?refer=1GNXIC

From erickbee at gmail.com  Thu Oct 18 20:51:04 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Thu, 18 Oct 2007 19:51:04 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <04B6D8CDD22FD34993C47E33E21F82EF88DAA8@rachael-reborn.letnet.net>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
	<5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
	<D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>
	<5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>
	<021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca>
	<5d093f9a0710181122l6d58b2b7h84967895def7494f@mail.gmail.com>
	<02ef01c811bb$c5a21730$56196883@cfs.uoguelph.ca>
	<04B6D8CDD22FD34993C47E33E21F82EF88DAA8@rachael-reborn.letnet.net>
Message-ID: <f4445faf0710181751n2017e1e4jd004e244ada681eb@mail.gmail.com>

The service recovery option will recover if the service was started,
but if it doesn't start on boot then that is not going to kick in
since it was never started and running yet.

I've seen the ccm service not start on reboots when people have
applied M$ patches to the server directly instead of the Cisco OS
patches from cisco.com. I haven't seen the service not start since 3.3
days due to timeouts or initialization timers being to short. You may
want to check under call manager service parameters and see if all the
Initialization timers are set to at least the suggested value. I had
one place a long time ago update from 3.x something to 4.0x and it
kept the older 3.x settings which was less then the suggested timer
value for 4.x so that caused the ccm service not to start. Once the
timer was set to the suggested value for 4.x the service starts fine
on reboots.


On 10/18/07, Johnson, Ken <kenjohnson at letu.edu> wrote:
> Actually we've had this problem sporadically with our subscriber since
> some 4.2(X) patch with the CCM service not consistently starting on
> bootup (even with the service recovery processes set). It always starts
> okay manually once the monitor emails remind us of the issue - and it
> never affects the publisher, nor does it affect any other service or
> write anything useful in the error log.
>
> Since it never fails while running, only fails to come up with a reboot
> or other CCM restart which almost never happens - even though TAC could
> certainly fix it for us, we ignored it and focused on more pressing
> cases figuring it would be a non-issue with the wipe-and-reload CM6.0
> process....
>
> Ken Johnson
>    Mgr. Network Services,
>    Information Technology
>    LeTourneau University
> _________________________________
>  E-mail: kenjohnson at letu.edu
>  Helpdesk: (903) 233-3500
>  Phone: (903) 233-3520
>  Web: http://www.letu.edu/infotech/
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
> Sent: Thursday, October 18, 2007 2:19 PM
> To: Jonathan Charles
> Cc: Ogden, Brandon; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Registration to Subscriber
>
> gotta love the consistency... ;)
>
> ----- Original Message -----
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Cc: "Ogden, Brandon" <Brandon.Ogden at amersports.com>;
> <cisco-voip at puck.nether.net>
> Sent: Thursday, October 18, 2007 2:22 PM
> Subject: Re: [cisco-voip] Registration to Subscriber
>
>
> > It depends
> >
> > I have seen the service take forever on 100 patterns and start
> > immediately on 1000 patterns...
> >
> >
> > Jonathan
> >
> > On 10/18/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >> what's "a large number of route patterns" ?
> >>
> >> 100? 1,000? 10,000?
> >>
> >> ----- Original Message -----
> >> From: "Jonathan Charles" <jonvoip at gmail.com>
> >> To: "Ogden, Brandon" <Brandon.Ogden at amersports.com>
> >> Cc: <cisco-voip at puck.nether.net>
> >> Sent: Thursday, October 18, 2007 2:14 PM
> >> Subject: Re: [cisco-voip] Registration to Subscriber
> >>
> >>
> >> > If you have a large number of route patterns the CCM service can
> >> > time out when starting
> >> >
> >> > There is a service parameter to increase the timeout, but I forgot
> >> > what it was...
> >> >
> >> >
> >> > Jonathan
> >> >
> >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >> >> LOL.. I got it resolved.. Didn't have to reboot.. and before you
> >> >> say so.. yes I am an idiot.. LOL  The Call Manager Service had not
>
> >> >> started for some reason.. I restarted the service and it popped
> >> >> right up, all is well now.  Thanks all!
> >> >>
> >> >> Brandon
> >> >>
> >> >> -----Original Message-----
> >> >> From: cisco-voip-bounces at puck.nether.net
> >> >> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden,
> >> >> Brandon
> >> >> Sent: Thursday, October 18, 2007 10:18 AM
> >> >> To: Jonathan Charles
> >> >> Cc: cisco-voip at puck.nether.net
> >> >> Subject: Re: [cisco-voip] Registration to Subscriber
> >> >>
> >> >> I did not do a reboot of the publisher, just the sub.
> >> >>
> >> >> Brandon Ogden
> >> >> Amer Sports
> >> >> 801-624-7529
> >> >> www.amersports.com
> >> >> http://www.linkedin.com/in/brandonogden
> >> >>
> >> >> -----Original Message-----
> >> >> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> >> >> Sent: Thursday, October 18, 2007 10:14 AM
> >> >> To: Ogden, Brandon
> >> >> Cc: Scott Voll; cisco-voip at puck.nether.net
> >> >> Subject: Re: [cisco-voip] Registration to Subscriber
> >> >>
> >> >> Did you reboot the cluster after the rebuild?
> >> >>
> >> >>
> >> >> Jonathan
> >> >>
> >> >> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > I just went in and looked at that and my subscriber is not
> >> >> > showing up
> >> >> as an
> >> >> > available server in the Call Manager Group.  Any ideas?  It
> >> >> > appears to
> >> >> have
> >> >> > replicated correctly, and I was able to use it to modify a
> >> >> > Directory
> >> >> Number.
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > Brandon
> >> >> >
> >> >> >  ________________________________
> >> >> >
> >> >> >
> >> >> > From: Scott Voll [mailto:svoll.voip at gmail.com]
> >> >> >  Sent: Thursday, October 18, 2007 8:54 AM
> >> >> >  To: Ogden, Brandon
> >> >> >  Cc: cisco-voip at puck.nether.net
> >> >> >  Subject: Re: [cisco-voip] Registration to Subscriber
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > did you change the order of your servers so it registers with
> >> >> > the sub?
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > Scott
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com>
> wrote:
> >> >> >
> >> >> >
> >> >> >
> >> >> > I rebuilt my subscriber box last night, it seemed to go well.  I
>
> >> >> > was wondering how long it should take for the phones to register
>
> >> >> > with the Subscriber rather than the pub.  As of this morning,
> >> >> > they are still registered with the pub.  Thanks!
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > Brandon
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >  _______________________________________________
> >> >> >  cisco-voip mailing list
> >> >> >  cisco-voip at puck.nether.net
> >> >> >  https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > cisco-voip mailing list
> >> >> > cisco-voip at puck.nether.net
> >> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >> >
> >> >>
> >> >> This e-mail may contain confidential and/or privileged
> >> >> information. If you are not the intended recipient (or have
> >> >> received this e-mail in
> >> >> error)
> >> >> please notify the sender immediately and destroy this e-mail. Any
> >> >> unauthorized copying, disclosure or distribution of the material
> >> >> in this e-mail is strictly forbidden.
> >> >> _______________________________________________
> >> >> cisco-voip mailing list
> >> >> cisco-voip at puck.nether.net
> >> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >>
> >> > _______________________________________________
> >> > cisco-voip mailing list
> >> > cisco-voip at puck.nether.net
> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >
> >>
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> --
> BEGIN-ANTISPAM-VOTING-LINKS
> ------------------------------------------------------
>
> Teach CanIt if this mail (ID 8245599) is spam:
> Spam:
> http://antispam.letu.edu/canit/b.php?i=8245599&m=482943aeebf8&c=s
> Not spam:
> http://antispam.letu.edu/canit/b.php?i=8245599&m=482943aeebf8&c=n
> Forget vote:
> http://antispam.letu.edu/canit/b.php?i=8245599&m=482943aeebf8&c=f
> ------------------------------------------------------
> END-ANTISPAM-VOTING-LINKS
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Thu Oct 18 22:00:22 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 21:00:22 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <835082.21017.qm@web57203.mail.re3.yahoo.com>
References: <5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>
	<835082.21017.qm@web57203.mail.re3.yahoo.com>
Message-ID: <5d093f9a0710181900u48b306b9v1be1049d16c7c325@mail.gmail.com>

T302 is the Interdigit Timeout...



Jonathan

On 10/18/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> I think it's the T302 timeout underneath enterprise
> parameters. There is documentation on the timers
> somewhere...I don't remember where I saw it
> unfortunately.
>
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
>
> > If you have a large number of route patterns the CCM
> > service can time
> > out when starting
> >
> > There is a service parameter to increase the
> > timeout, but I forgot
> > what it was...
> >
> >
> > Jonathan
> >
> > On 10/18/07, Ogden, Brandon
> > <Brandon.Ogden at amersports.com> wrote:
> > > LOL.. I got it resolved.. Didn't have to reboot..
> > and before you say
> > > so.. yes I am an idiot.. LOL  The Call Manager
> > Service had not started
> > > for some reason.. I restarted the service and it
> > popped right up, all is
> > > well now.  Thanks all!
> > >
> > > Brandon
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Ogden, Brandon
> > > Sent: Thursday, October 18, 2007 10:18 AM
> > > To: Jonathan Charles
> > > Cc: cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] Registration to
> > Subscriber
> > >
> > > I did not do a reboot of the publisher, just the
> > sub.
> > >
> > > Brandon Ogden
> > > Amer Sports
> > > 801-624-7529
> > > www.amersports.com
> > > http://www.linkedin.com/in/brandonogden
> > >
> > > -----Original Message-----
> > > From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > > Sent: Thursday, October 18, 2007 10:14 AM
> > > To: Ogden, Brandon
> > > Cc: Scott Voll; cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] Registration to
> > Subscriber
> > >
> > > Did you reboot the cluster after the rebuild?
> > >
> > >
> > > Jonathan
> > >
> > > On 10/18/07, Ogden, Brandon
> > <Brandon.Ogden at amersports.com> wrote:
> > > >
> > > >
> > > >
> > > >
> > > > I just went in and looked at that and my
> > subscriber is not showing up
> > > as an
> > > > available server in the Call Manager Group.  Any
> > ideas?  It appears to
> > > have
> > > > replicated correctly, and I was able to use it
> > to modify a Directory
> > > Number.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Brandon
> > > >
> > > >  ________________________________
> > > >
> > > >
> > > > From: Scott Voll [mailto:svoll.voip at gmail.com]
> > > >  Sent: Thursday, October 18, 2007 8:54 AM
> > > >  To: Ogden, Brandon
> > > >  Cc: cisco-voip at puck.nether.net
> > > >  Subject: Re: [cisco-voip] Registration to
> > Subscriber
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > did you change the order of your servers so it
> > registers with the sub?
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Scott
> > > >
> > > >
> > > >
> > > >
> > > > On 10/18/07, Ogden, Brandon
> > <Brandon.Ogden at amersports.com> wrote:
> > > >
> > > >
> > > >
> > > > I rebuilt my subscriber box last night, it
> > seemed to go well.  I was
> > > > wondering how long it should take for the phones
> > to register with the
> > > > Subscriber rather than the pub.  As of this
> > morning, they are still
> > > > registered with the pub.  Thanks!
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Brandon
> > > >
> > > >
> > > >
> > > >
> > > >  _______________________________________________
> > > >  cisco-voip mailing list
> > > >  cisco-voip at puck.nether.net
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > > This e-mail may contain confidential and/or
> > privileged information. If
> > > you are not the intended recipient (or have
> > received this e-mail in
> > > error)
> > > please notify the sender immediately and destroy
> > this e-mail. Any
> > > unauthorized copying, disclosure or distribution
> > of the material in this
> > > e-mail is strictly forbidden.
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
>
> ____________________________________________________________________________________
> Pinpoint customers who are looking for what you sell.
> http://searchmarketing.yahoo.com/
>

From Timothy_Frazee at adp.com  Thu Oct 18 22:40:53 2007
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Thu, 18 Oct 2007 21:40:53 -0500
Subject: [cisco-voip] cucm 6 and TAPS
Message-ID: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>

Group,

 

From some of the reading I have seen, it looks like I still need an
external CRS server for TAPS to operate in 6. I see a "TAPS Service" in
the serviceability page.

 

Can anyone give me a firm answer?

 

If it truly is "all in one" then can anyone point me to a doc that shows
how to config it?

 

Thanks all.

 

 

 

 

 

Timothy Frazee 

Network Solutions

Automatic Data Processing, Inc.



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From tednugent73 at gmail.com  Thu Oct 18 22:46:16 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 18 Oct 2007 22:46:16 -0400
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>

confirmed... taps services is required for CM to talk to the taps.asp on the
CRS server

On 10/18/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote:
>
>  Group,
>
>
>
> From some of the reading I have seen, it looks like I still need an
> external CRS server for TAPS to operate in 6. I see a "TAPS Service" in the
> serviceability page.
>
>
>
> Can anyone give me a firm answer?
>
>
>
> If it truly is "all in one" then can anyone point me to a doc that shows
> how to config it?
>
>
>
> Thanks all.
>
>
>
>
>
>
>
>
>
>
>
> Timothy Frazee
>
> Network Solutions
>
> Automatic Data Processing, Inc.
>
> ------------------------------
> This message and any attachments are intended only for the use of the
> addressee and may contain information that is privileged and confidential.
> If the reader of the message is not the intended recipient or an authorized
> representative of the intended recipient, you are hereby notified that any
> dissemination of this communication is strictly prohibited. If you have
> received this communication in error, please notify us immediately by e-mail
> and delete the message and any attachments from your system.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Timothy_Frazee at adp.com  Thu Oct 18 22:50:13 2007
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Thu, 18 Oct 2007 21:50:13 -0500
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>
Message-ID: <723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>

So if I want to use TAPS to install a bulk amount of phones, I need a
minimum of two servers? My publisher and a CRS server.

 

I know this is a dumb question, but there is no "co-res" type of thing
for crs and callmanager on a 6 environment?

 

Thanks for any answers.

 

 

 

 

From: Ted Nugent [mailto:tednugent73 at gmail.com] 
Sent: Thursday, October 18, 2007 9:46 PM
To: Frazee, Timothy
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] cucm 6 and TAPS

 

confirmed... taps services is required for CM to talk to the taps.asp on
the CRS server

On 10/18/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote: 

Group,

 

From some of the reading I have seen, it looks like I still need an
external CRS server for TAPS to operate in 6. I see a "TAPS Service" in
the serviceability page.

 

Can anyone give me a firm answer?

 

If it truly is "all in one" then can anyone point me to a doc that shows
how to config it?

 

Thanks all.

 

 

 

 

 

Timothy Frazee 

Network Solutions

Automatic Data Processing, Inc.

________________________________

This message and any attachments are intended only for the use of the
addressee and may contain information that is privileged and
confidential. If the reader of the message is not the intended recipient
or an authorized representative of the intended recipient, you are
hereby notified that any dissemination of this communication is strictly
prohibited. If you have received this communication in error, please
notify us immediately by e-mail and delete the message and any
attachments from your system. 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

 



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From tednugent73 at gmail.com  Thu Oct 18 23:02:40 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 18 Oct 2007 23:02:40 -0400
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <22e6380b0710182002p30265d7do404b0f2d9b144842@mail.gmail.com>

I use VMWare with CRS and either spawn it on a customer server temporarily
or use my laptop. But yes it needs to be separate unfortunately



On 10/18/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote:
>
>  So if I want to use TAPS to install a bulk amount of phones, I need a
> minimum of two servers? My publisher and a CRS server.
>
>
>
> I know this is a dumb question, but there is no "co-res" type of thing for
> crs and callmanager on a 6 environment?
>
>
>
> Thanks for any answers.
>
>
>
>
>
>
>
>
>
> *From:* Ted Nugent [mailto:tednugent73 at gmail.com]
> *Sent:* Thursday, October 18, 2007 9:46 PM
> *To:* Frazee, Timothy
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] cucm 6 and TAPS
>
>
>
> confirmed... taps services is required for CM to talk to the taps.asp on
> the CRS server
>
> On 10/18/07, *Frazee, Timothy* <Timothy_Frazee at adp.com> wrote:
>
> Group,
>
>
>
> From some of the reading I have seen, it looks like I still need an
> external CRS server for TAPS to operate in 6. I see a "TAPS Service" in the
> serviceability page.
>
>
>
> Can anyone give me a firm answer?
>
>
>
> If it truly is "all in one" then can anyone point me to a doc that shows
> how to config it?
>
>
>
> Thanks all.
>
>
>
>
>
>
>
>
>
>
>
> Timothy Frazee
>
> Network Solutions
>
> Automatic Data Processing, Inc.
>  ------------------------------
>
> This message and any attachments are intended only for the use of the
> addressee and may contain information that is privileged and confidential.
> If the reader of the message is not the intended recipient or an authorized
> representative of the intended recipient, you are hereby notified that any
> dissemination of this communication is strictly prohibited. If you have
> received this communication in error, please notify us immediately by e-mail
> and delete the message and any attachments from your system.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> ------------------------------
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From Timothy_Frazee at adp.com  Thu Oct 18 23:08:10 2007
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Thu, 18 Oct 2007 22:08:10 -0500
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <22e6380b0710182002p30265d7do404b0f2d9b144842@mail.gmail.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710182002p30265d7do404b0f2d9b144842@mail.gmail.com>
Message-ID: <723FA17C6AA4C4428CDF572DCAEE08D405445C4D@EXCHANGEDS01.ds.ad.adp.com>

Does a CUCM 6 media kit come with some type of "Extended Services" or
some type of IPCC?

 

From: Ted Nugent [mailto:tednugent73 at gmail.com] 
Sent: Thursday, October 18, 2007 10:03 PM
To: Frazee, Timothy
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] cucm 6 and TAPS

 

I use VMWare with CRS and either spawn it on a customer server
temporarily or use my laptop. But yes it needs to be separate
unfortunately



 

On 10/18/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote: 

So if I want to use TAPS to install a bulk amount of phones, I need a
minimum of two servers? My publisher and a CRS server.

 

I know this is a dumb question, but there is no "co-res" type of thing
for crs and callmanager on a 6 environment?

 

Thanks for any answers.

 

 

 

 

From: Ted Nugent [mailto:tednugent73 at gmail.com ] 
Sent: Thursday, October 18, 2007 9:46 PM
To: Frazee, Timothy
Cc: cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] cucm 6 and TAPS

 

confirmed... taps services is required for CM to talk to the taps.asp on
the CRS server

On 10/18/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote: 

Group,

 

From some of the reading I have seen, it looks like I still need an
external CRS server for TAPS to operate in 6. I see a "TAPS Service" in
the serviceability page.

 

Can anyone give me a firm answer?

 

If it truly is "all in one" then can anyone point me to a doc that shows
how to config it?

 

Thanks all.

 

 

 

 

 

Timothy Frazee 

Network Solutions

Automatic Data Processing, Inc.

________________________________

This message and any attachments are intended only for the use of the
addressee and may contain information that is privileged and
confidential. If the reader of the message is not the intended recipient
or an authorized representative of the intended recipient, you are
hereby notified that any dissemination of this communication is strictly
prohibited. If you have received this communication in error, please
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_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

 

________________________________

This message and any attachments are intended only for the use of the
addressee and may contain information that is privileged and
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or an authorized representative of the intended recipient, you are
hereby notified that any dissemination of this communication is strictly
prohibited. If you have received this communication in error, please
notify us immediately by e-mail and delete the message and any
attachments from your system. 

 



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From jonvoip at gmail.com  Fri Oct 19 00:31:14 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 18 Oct 2007 23:31:14 -0500
Subject: [cisco-voip] CCME GUI adding phones "no phone to add"
In-Reply-To: <15563.19244.qm@web57204.mail.re3.yahoo.com>
References: <5d093f9a0710181612i6f003565q4128e7fdc24865a1@mail.gmail.com>
	<15563.19244.qm@web57204.mail.re3.yahoo.com>
Message-ID: <5d093f9a0710182131x472b0a8fi57312e5043d4168c@mail.gmail.com>

The GUI is for most purposes, useless.

Granted, I don't admin CME/CUE, I just stage and deploy... so, the CLI
is where all my config is done.

You can't configure dial-peers, you can't overlay numbers... it is
really quite limited.



Jonathan

On 10/18/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> Yes Jonathan, GUI-using CCME users do exist.
>
> For clarification, all major configuration was done
> through CLI. The system was handed over to the
> customer so they would be able to perform
> adds/moves/changes easily without having to get
> someone with AAA access involved.
>
> I think the GUI works pretty well for end users. I'm
> fairly satisfied with it...so long as there's no
> aaa authentication config-commands in the router.
>
> Paul
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
>
> > People are actually out there configuring CCME via
> > the GUI????
> >
> > Wow.
> >
> >
> > Jonathan
> >
> > On 10/17/07, Paul Choi <asobihoudai at yahoo.com>
> > wrote:
> > > FYI, for anyone that is encountering an error
> > dialog
> > > of "no phones to add" from the CCME GUI, you have
> > to
> > > go into the box and under telephony-service mode,
> > turn
> > > auto-reg-ephone back on.
> > >
> > > hope that helps someone!
> > > Paul
> > >
> > >
> > >
> >
> ____________________________________________________________________________________
> > > Shape Yahoo! in your own image.  Join our Network
> > Research Panel today!
> >
> http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
>
>
>
>
> ____________________________________________________________________________________
> Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more.
> http://mobile.yahoo.com/go?refer=1GNXIC
>

From Richard.vanBerkel at telindus.nl  Fri Oct 19 03:34:55 2007
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Fri, 19 Oct 2007 09:34:55 +0200
Subject: [cisco-voip] Disadvantage of using Unified Messaging instead of
	Voicemail only
References: <f84a38d30710051054q48eb6ddl9069c33ee839f56a@mail.gmail.com>
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157EABE@tnl1mx1.telindus.intra>

Hi,

 

Can somebody give me some big disadvantages why I should choose for
voicemail only instead of UM.

 

Regards

 

Richard

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From Richard.vanBerkel at telindus.nl  Fri Oct 19 03:59:36 2007
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Fri, 19 Oct 2007 09:59:36 +0200
Subject: [cisco-voip] reporting options
References: <f84a38d30710051054q48eb6ddl9069c33ee839f56a@mail.gmail.com> 
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157EAC4@tnl1mx1.telindus.intra>

Hi Guys,

 

Is there a list of reporting possibilities in CCX 4.1

 

Regards

Richard

<skype:richardvanberkel?call> 

<http://www.skype.com/go/download>  

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From asobihoudai at yahoo.com  Fri Oct 19 08:23:11 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 19 Oct 2007 05:23:11 -0700 (PDT)
Subject: [cisco-voip] CCME GUI adding phones "no phone to add"
In-Reply-To: <5d093f9a0710182131x472b0a8fi57312e5043d4168c@mail.gmail.com>
Message-ID: <448191.19011.qm@web57201.mail.re3.yahoo.com>

I'd disagree. For customers that have no way of
getting into the configuration, I'd say that using the
GUI is their best (and only) way of administering the
phone system themselves. I just happen to have a lot
of these customers.

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> The GUI is for most purposes, useless.
> 
> Granted, I don't admin CME/CUE, I just stage and
> deploy... so, the CLI
> is where all my config is done.
> 
> You can't configure dial-peers, you can't overlay
> numbers... it is
> really quite limited.
> 
> 
> 
> Jonathan
> 
> On 10/18/07, Paul Choi <asobihoudai at yahoo.com>
> wrote:
> > Yes Jonathan, GUI-using CCME users do exist.
> >
> > For clarification, all major configuration was
> done
> > through CLI. The system was handed over to the
> > customer so they would be able to perform
> > adds/moves/changes easily without having to get
> > someone with AAA access involved.
> >
> > I think the GUI works pretty well for end users.
> I'm
> > fairly satisfied with it...so long as there's no
> > aaa authentication config-commands in the router.
> >
> > Paul
> > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > > People are actually out there configuring CCME
> via
> > > the GUI????
> > >
> > > Wow.
> > >
> > >
> > > Jonathan
> > >
> > > On 10/17/07, Paul Choi <asobihoudai at yahoo.com>
> > > wrote:
> > > > FYI, for anyone that is encountering an error
> > > dialog
> > > > of "no phones to add" from the CCME GUI, you
> have
> > > to
> > > > go into the box and under telephony-service
> mode,
> > > turn
> > > > auto-reg-ephone back on.
> > > >
> > > > hope that helps someone!
> > > > Paul
> > > >
> > > >
> > > >
> > >
> >
>
____________________________________________________________________________________
> > > > Shape Yahoo! in your own image.  Join our
> Network
> > > Research Panel today!
> > >
> >
>
http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7
> > > >
> > > >
> > > >
> _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> >
> >
> >
> >
> >
>
____________________________________________________________________________________
> > Take the Internet to Go: Yahoo!Go puts the
> Internet in your pocket: mail, news, photos & more.
> > http://mobile.yahoo.com/go?refer=1GNXIC
> >
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From patrick.diener at gmail.com  Fri Oct 19 08:27:36 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Fri, 19 Oct 2007 14:27:36 +0200
Subject: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
Message-ID: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>

I'm trying to put together a rough step by step procedure to do a
disaster recovery for a cucm 5.1(2) cluster and there are some stuff I
can't figure out how it should/will work...

I based the steps on the DRS Admin Guide for Version 5.0 (5.1 not out yet?!?)

Important: I try to do a restore on a new server hw => new first node MAC

Step 1:
perform fresh install of cucm sw
!! set same hostname as before !!

Step 2:
perform restore using DRS and restart server (worked like a charm in the lab...)

form now on I'm stuck...

Step 3:
Licensing?!?
all activated services will not start cause the .lic files are not
valid for the new server hw (at least thats what happened in the
lab..)
=> will TAC send me a new .lic equal to the old one?
=> do I need to contact licensing at cisco.com?
=> any experience how long it will take to get the new .lic files?

Step 4:
move some phones to the restored publisher (assuming callmanager
service is running) and test all features needed
=> will the phones register if security has been configured? I assume
the new callmanager, tftp and capf certificates will not match the
certificates stored in the phones ctl file (right?) how do I get out
of this mess without touching each and every phone to delete the ctl
file or perform a factory reset?

Step 5:
restore sub (or subs)


all comments, experiences and remarks are appreciated!

cheers
Patrick

From cheesevoice at gmail.com  Fri Oct 19 09:03:35 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Fri, 19 Oct 2007 08:03:35 -0500
Subject: [cisco-voip] OT: Creating your own site to site PRI's?
In-Reply-To: <a1b07ea60710181259m63a3a4adq96ffe705ff7da7f6@mail.gmail.com>
References: <a1b07ea60710181259m63a3a4adq96ffe705ff7da7f6@mail.gmail.com>
Message-ID: <84b74aec0710190603j13836896y26f4cdfd51cc151@mail.gmail.com>

Paradyne makes ( or used to) Net-2-Net extenders, you can get T1, dry pair,
and PRI versions.
I have used the Dry Pair and T1 versions of those ...not recalling the PRI
version, but I know they made them back in the 90's.

When phones were TDM and days were filled with wheat beer and billiard...not
certification studies and IP translations...but I am not bitter...no ... not
bitter


On 10/18/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:

> I need to come up with a way to create my own PRI's between two
> locations.  These will have to be stand alone, PRI's not related or
> controlled by CallManager.  Its to tie a remote office PBX shelf unit
> into the main office PBX.
>
> See the call center still runs on an older PBX, we utilize CallManager
> for everything else.  The remote office will be relocating soon and
> well we have alot of fiber between the new location and the main
> office, so ideally I would prefer to come up with our own PRI solution
> rather then have to pay for and rely on BS/AT&T for PRIs.
>
> Any thoughts or suggestions?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jason.aarons at us.didata.com  Fri Oct 19 09:12:15 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 19 Oct 2007 09:12:15 -0400
Subject: [cisco-voip] Unity Connection 2.0(1) asking users to change
	passwords
Message-ID: <C1FE15183DA37645BC0633BC604E44F007294364@USNAEXCH.na.didata.local>

Last night I went into CUAdmin and changed the Recommended Voice Mail
Authentication Rule,

Password expires = 180 days

Changed to

Password expires = Never Expires

 

Several people came in the morning and it was asking them to change
their password, any ideas. System has been in production 20 days. Bug
maybe?

 

 




-----------------------------------------
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From kenjohnson at letu.edu  Fri Oct 19 10:03:45 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Fri, 19 Oct 2007 09:03:45 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <f4445faf0710181751n2017e1e4jd004e244ada681eb@mail.gmail.com>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int>
	<5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
	<D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>
	<5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>
	<021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca>
	<5d093f9a0710181122l6d58b2b7h84967895def7494f@mail.gmail.com>
	<02ef01c811bb$c5a21730$56196883@cfs.uoguelph.ca>
	<04B6D8CDD22FD34993C47E33E21F82EF88DAA8@rachael-reborn.letnet.net>
	<f4445faf0710181751n2017e1e4jd004e244ada681eb@mail.gmail.com>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EFD0B839@rachael-reborn.letnet.net>

Hm - that's interesting, I might look at those timing settings.

Not only do we not apply MS patches, we don't actually apply individual
Cisco OS patches to the CM's anymore - we only apply the Cisco OS
service packs. Keeping up with the individual Cisco OS patches got to
where it didn't seem worth the effort - especially for a CM environment
that is 100% isolated from anything but the phones and 3 management
stations :-)

So if I hadn't had this problem through a number of OS service packs I
might think it was the lack of an individual Cisco patch I hadn't
applied, but since it continued after the most recent OS service packs I
expect not.  I'll probably be content to wait for our 6.0 upgrade :-)

Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
 E-mail: kenjohnson at letu.edu
 Helpdesk: (903) 233-3500
 Phone: (903) 233-3520
 Web: http://www.letu.edu/infotech/

-----Original Message-----
From: Erick Bergquist [mailto:erickbee at gmail.com] 
Sent: Thursday, October 18, 2007 7:51 PM
To: Johnson, Ken
Cc: Lelio Fulgenzi; Jonathan Charles; Ogden, Brandon;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Registration to Subscriber

The service recovery option will recover if the service was started,
but if it doesn't start on boot then that is not going to kick in
since it was never started and running yet.

I've seen the ccm service not start on reboots when people have
applied M$ patches to the server directly instead of the Cisco OS
patches from cisco.com. I haven't seen the service not start since 3.3
days due to timeouts or initialization timers being to short. You may
want to check under call manager service parameters and see if all the
Initialization timers are set to at least the suggested value. I had
one place a long time ago update from 3.x something to 4.0x and it
kept the older 3.x settings which was less then the suggested timer
value for 4.x so that caused the ccm service not to start. Once the
timer was set to the suggested value for 4.x the service starts fine
on reboots.


On 10/18/07, Johnson, Ken <kenjohnson at letu.edu> wrote:
> Actually we've had this problem sporadically with our subscriber since
> some 4.2(X) patch with the CCM service not consistently starting on
> bootup (even with the service recovery processes set). It always
starts
> okay manually once the monitor emails remind us of the issue - and it
> never affects the publisher, nor does it affect any other service or
> write anything useful in the error log.
>
> Since it never fails while running, only fails to come up with a
reboot
> or other CCM restart which almost never happens - even though TAC
could
> certainly fix it for us, we ignored it and focused on more pressing
> cases figuring it would be a non-issue with the wipe-and-reload CM6.0
> process....
>
> Ken Johnson
>    Mgr. Network Services,
>    Information Technology
>    LeTourneau University
> _________________________________
>  E-mail: kenjohnson at letu.edu
>  Helpdesk: (903) 233-3500
>  Phone: (903) 233-3520
>  Web: http://www.letu.edu/infotech/
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio
Fulgenzi
> Sent: Thursday, October 18, 2007 2:19 PM
> To: Jonathan Charles
> Cc: Ogden, Brandon; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Registration to Subscriber
>
> gotta love the consistency... ;)
>
> ----- Original Message -----
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Cc: "Ogden, Brandon" <Brandon.Ogden at amersports.com>;
> <cisco-voip at puck.nether.net>
> Sent: Thursday, October 18, 2007 2:22 PM
> Subject: Re: [cisco-voip] Registration to Subscriber
>
>
> > It depends
> >
> > I have seen the service take forever on 100 patterns and start
> > immediately on 1000 patterns...
> >
> >
> > Jonathan
> >
> > On 10/18/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >> what's "a large number of route patterns" ?
> >>
> >> 100? 1,000? 10,000?
> >>
> >> ----- Original Message -----
> >> From: "Jonathan Charles" <jonvoip at gmail.com>
> >> To: "Ogden, Brandon" <Brandon.Ogden at amersports.com>
> >> Cc: <cisco-voip at puck.nether.net>
> >> Sent: Thursday, October 18, 2007 2:14 PM
> >> Subject: Re: [cisco-voip] Registration to Subscriber
> >>
> >>
> >> > If you have a large number of route patterns the CCM service can
> >> > time out when starting
> >> >
> >> > There is a service parameter to increase the timeout, but I
forgot
> >> > what it was...
> >> >
> >> >
> >> > Jonathan
> >> >
> >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >> >> LOL.. I got it resolved.. Didn't have to reboot.. and before you
> >> >> say so.. yes I am an idiot.. LOL  The Call Manager Service had
not
>
> >> >> started for some reason.. I restarted the service and it popped
> >> >> right up, all is well now.  Thanks all!
> >> >>
> >> >> Brandon
> >> >>
> >> >> -----Original Message-----
> >> >> From: cisco-voip-bounces at puck.nether.net
> >> >> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden,
> >> >> Brandon
> >> >> Sent: Thursday, October 18, 2007 10:18 AM
> >> >> To: Jonathan Charles
> >> >> Cc: cisco-voip at puck.nether.net
> >> >> Subject: Re: [cisco-voip] Registration to Subscriber
> >> >>
> >> >> I did not do a reboot of the publisher, just the sub.
> >> >>
> >> >> Brandon Ogden
> >> >> Amer Sports
> >> >> 801-624-7529
> >> >> www.amersports.com
> >> >> http://www.linkedin.com/in/brandonogden
> >> >>
> >> >> -----Original Message-----
> >> >> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> >> >> Sent: Thursday, October 18, 2007 10:14 AM
> >> >> To: Ogden, Brandon
> >> >> Cc: Scott Voll; cisco-voip at puck.nether.net
> >> >> Subject: Re: [cisco-voip] Registration to Subscriber
> >> >>
> >> >> Did you reboot the cluster after the rebuild?
> >> >>
> >> >>
> >> >> Jonathan
> >> >>
> >> >> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com>
wrote:
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > I just went in and looked at that and my subscriber is not
> >> >> > showing up
> >> >> as an
> >> >> > available server in the Call Manager Group.  Any ideas?  It
> >> >> > appears to
> >> >> have
> >> >> > replicated correctly, and I was able to use it to modify a
> >> >> > Directory
> >> >> Number.
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > Brandon
> >> >> >
> >> >> >  ________________________________
> >> >> >
> >> >> >
> >> >> > From: Scott Voll [mailto:svoll.voip at gmail.com]
> >> >> >  Sent: Thursday, October 18, 2007 8:54 AM
> >> >> >  To: Ogden, Brandon
> >> >> >  Cc: cisco-voip at puck.nether.net
> >> >> >  Subject: Re: [cisco-voip] Registration to Subscriber
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > did you change the order of your servers so it registers with
> >> >> > the sub?
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > Scott
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com>
> wrote:
> >> >> >
> >> >> >
> >> >> >
> >> >> > I rebuilt my subscriber box last night, it seemed to go well.
I
>
> >> >> > was wondering how long it should take for the phones to
register
>
> >> >> > with the Subscriber rather than the pub.  As of this morning,
> >> >> > they are still registered with the pub.  Thanks!
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > Brandon
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >  _______________________________________________
> >> >> >  cisco-voip mailing list
> >> >> >  cisco-voip at puck.nether.net
> >> >> >  https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > cisco-voip mailing list
> >> >> > cisco-voip at puck.nether.net
> >> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >> >
> >> >>
> >> >> This e-mail may contain confidential and/or privileged
> >> >> information. If you are not the intended recipient (or have
> >> >> received this e-mail in
> >> >> error)
> >> >> please notify the sender immediately and destroy this e-mail.
Any
> >> >> unauthorized copying, disclosure or distribution of the material
> >> >> in this e-mail is strictly forbidden.
> >> >> _______________________________________________
> >> >> cisco-voip mailing list
> >> >> cisco-voip at puck.nether.net
> >> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >>
> >> > _______________________________________________
> >> > cisco-voip mailing list
> >> > cisco-voip at puck.nether.net
> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >
> >>
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> --
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


-- 
BEGIN-ANTISPAM-VOTING-LINKS
------------------------------------------------------

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Spam:
http://antispam.letu.edu/canit/b.php?i=8255423&m=869444ceaabe&c=s
Not spam:
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Forget vote:
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END-ANTISPAM-VOTING-LINKS


From rkulagow at gmail.com  Fri Oct 19 10:23:08 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Fri, 19 Oct 2007 09:23:08 -0500
Subject: [cisco-voip] call manager express blocking outbound caller ID
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F84D692@CNTRRA20-XCS01.telkom.co.za>
References: <472E195E692EE74BAE4E7B2880A78C1F84D692@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <4718BDCC.9000600@gmail.com>

You could do this with two different route patterns (which is how we do it)

The first route pattern on CCM, where you want to block caller ID, would 
be set for "restricted" in the route pattern on the Calling line ID 
presentation.

Lets say you use "8" to get an external line.  On CCM you'd have "8.!" 
(or whatever) as the route pattern

On the router (h.323 dial-peers), it would be:

dial-peer voice 1000 pots
destination-pattern 8T
clid restrict
port 0/1/0:15

The second route pattern could be "*8.!" on CCM, and on the router you'd 
have

dial-peer voice 1001 pots
destination-patter *8T
port 0/1/0:15

So in dial-peer 1001, we don't have the clid restrict command, so it 
_will_ send caller ID.

From wsisk at cisco.com  Fri Oct 19 10:32:10 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 19 Oct 2007 10:32:10 -0400
Subject: [cisco-voip] CUCM-BE 6.x support for UCCX?
In-Reply-To: <4717D30C.5060705@cisco.com>
References: <22e6380b0710181139n7424cccek837ddb2d86ea96d3@mail.gmail.com>		<4717CC28.70806@cisco.com>	<22e6380b0710181437y51f08477w9b4180d34ab7c888@mail.gmail.com>
	<4717D30C.5060705@cisco.com>
Message-ID: <4718BFEA.6070707@cisco.com>

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From Timothy_Frazee at adp.com  Fri Oct 19 10:33:45 2007
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Fri, 19 Oct 2007 09:33:45 -0500
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <535625f70710190630j2671e49bs4d827d09278dc3d8@mail.gmail.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>
	<535625f70710190630j2671e49bs4d827d09278dc3d8@mail.gmail.com>
Message-ID: <723FA17C6AA4C4428CDF572DCAEE08D405445E35@EXCHANGEDS01.ds.ad.adp.com>

I wasn't sure if the TAPS service had been converted into a stand-alone
service built into BAT. No CRS server needed.

 

Hence, the dumb question.

 

From: Voice Noob [mailto:voicenoob at gmail.com] 
Sent: Friday, October 19, 2007 8:31 AM
To: Frazee, Timothy
Subject: Re: [cisco-voip] cucm 6 and TAPS

 

Well I don't see how there could be a "co-res" type of thing considering
CRS runs on Windows and CCM 6 is running on Linux. 

On 10/18/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote: 

So if I want to use TAPS to install a bulk amount of phones, I need a
minimum of two servers? My publisher and a CRS server.

 

I know this is a dumb question, but there is no "co-res" type of thing
for crs and callmanager on a 6 environment?

 

Thanks for any answers.

 

 

 

 

From: Ted Nugent [mailto:tednugent73 at gmail.com ] 
Sent: Thursday, October 18, 2007 9:46 PM
To: Frazee, Timothy
Cc: cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] cucm 6 and TAPS

 

confirmed... taps services is required for CM to talk to the taps.asp on
the CRS server

On 10/18/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote: 

Group,

 

From some of the reading I have seen, it looks like I still need an
external CRS server for TAPS to operate in 6. I see a "TAPS Service" in
the serviceability page.

 

Can anyone give me a firm answer?

 

If it truly is "all in one" then can anyone point me to a doc that shows
how to config it?

 

Thanks all.

 

 

 

 

 

Timothy Frazee 

Network Solutions

Automatic Data Processing, Inc.

________________________________

This message and any attachments are intended only for the use of the
addressee and may contain information that is privileged and
confidential. If the reader of the message is not the intended recipient
or an authorized representative of the intended recipient, you are
hereby notified that any dissemination of this communication is strictly
prohibited. If you have received this communication in error, please
notify us immediately by e-mail and delete the message and any
attachments from your system. 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

 

________________________________

This message and any attachments are intended only for the use of the
addressee and may contain information that is privileged and
confidential. If the reader of the message is not the intended recipient
or an authorized representative of the intended recipient, you are
hereby notified that any dissemination of this communication is strictly
prohibited. If you have received this communication in error, please
notify us immediately by e-mail and delete the message and any
attachments from your system. 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

 



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From svoll.voip at gmail.com  Fri Oct 19 10:48:48 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 19 Oct 2007 07:48:48 -0700
Subject: [cisco-voip] Disadvantage of using Unified Messaging instead of
	Voicemail only
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157EABE@tnl1mx1.telindus.intra>
References: <f84a38d30710051054q48eb6ddl9069c33ee839f56a@mail.gmail.com>
	<B42A9A0E2A14B943A3B7EA2FA7A0D8860157EABE@tnl1mx1.telindus.intra>
Message-ID: <f84a38d30710190748n60d2956ybcbf2c0b7864c033@mail.gmail.com>

unity connections is not based on M$.  ;-)  best I could think of.

Scott


On 10/19/07, Richard van Berkel <Richard.vanBerkel at telindus.nl> wrote:
>
>   Hi,
>
>
>
> Can somebody give me some big disadvantages why I should choose for
> voicemail only instead of UM.
>
>
>
> Regards
>
>
>
> Richard
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From svoll.voip at gmail.com  Fri Oct 19 10:52:49 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 19 Oct 2007 07:52:49 -0700
Subject: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
In-Reply-To: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>
References: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>
Message-ID: <f84a38d30710190752r7ba7ec13l3ff19927c81427d@mail.gmail.com>

for licensing you are correct.  you will need cisco licensing to change the
mac address to the new hardware.

I'm guessing you have a couple hours involved in this.  you might be able to
push it a little faster having a TAC engineer help, but that would have to
be confirmed by wes or ryan or mike.

Scott

PS.  No clue on the scurity stuff.


On 10/19/07, Patrick Diener <patrick.diener at gmail.com> wrote:
>
> I'm trying to put together a rough step by step procedure to do a
> disaster recovery for a cucm 5.1(2) cluster and there are some stuff I
> can't figure out how it should/will work...
>
> I based the steps on the DRS Admin Guide for Version 5.0 (5.1 not out
> yet?!?)
>
> Important: I try to do a restore on a new server hw => new first node MAC
>
> Step 1:
> perform fresh install of cucm sw
> !! set same hostname as before !!
>
> Step 2:
> perform restore using DRS and restart server (worked like a charm in the
> lab...)
>
> form now on I'm stuck...
>
> Step 3:
> Licensing?!?
> all activated services will not start cause the .lic files are not
> valid for the new server hw (at least thats what happened in the
> lab..)
> => will TAC send me a new .lic equal to the old one?
> => do I need to contact licensing at cisco.com?
> => any experience how long it will take to get the new .lic files?
>
> Step 4:
> move some phones to the restored publisher (assuming callmanager
> service is running) and test all features needed
> => will the phones register if security has been configured? I assume
> the new callmanager, tftp and capf certificates will not match the
> certificates stored in the phones ctl file (right?) how do I get out
> of this mess without touching each and every phone to delete the ctl
> file or perform a factory reset?
>
> Step 5:
> restore sub (or subs)
>
>
> all comments, experiences and remarks are appreciated!
>
> cheers
> Patrick
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From tednugent73 at gmail.com  Fri Oct 19 11:00:45 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Fri, 19 Oct 2007 11:00:45 -0400
Subject: [cisco-voip] Disadvantage of using Unified Messaging instead of
	Voicemail only
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157EABE@tnl1mx1.telindus.intra>
References: <f84a38d30710051054q48eb6ddl9069c33ee839f56a@mail.gmail.com>
	<B42A9A0E2A14B943A3B7EA2FA7A0D8860157EABE@tnl1mx1.telindus.intra>
Message-ID: <22e6380b0710190800i481980e4g2a9d6f787cebc262@mail.gmail.com>

the best reason i can think of is cost.. UM licencing is obviously more then
VM only

no schema extension required in your AD for VM only if its not part of your
domain

Thats about all i got...





On 10/19/07, Richard van Berkel <Richard.vanBerkel at telindus.nl> wrote:
>
>   Hi,
>
>
>
> Can somebody give me some big disadvantages why I should choose for
> voicemail only instead of UM.
>
>
>
> Regards
>
>
>
> Richard
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From pedro.nogueira at sicnet.pt  Fri Oct 19 11:19:00 2007
From: pedro.nogueira at sicnet.pt (Pedro Nogueira)
Date: Fri, 19 Oct 2007 16:19:00 +0100
Subject: [cisco-voip] how to change Span to PC Port configuration in a 7941
	phone
Message-ID: <A805E226FF74D74E90D5BE19D664A9A0137F25@neptuno.atminformatica.pt>

Hello folks,


Does anybody knows how to change the Span to PC Port configuration in a 7941 phone - No we cant do it through the web conf page since the option does not appear on it.

On the phone configuration i cant access it too, since the option appear but is disabled (not accessible), so i cant modify it.

Does anybody knows why does this happen and how to modify this setting ?

My Call Manager is a 4.2.3 version.

it seems a really stupid question but i really appreciate any help on it, since i cant put the supervisor monitor option of CRS running without this option on.

Really need help here,any thoughts or suggestions would be really appreciated !!!
 

Thanks in advanced,
regards,

Peter



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From akoul at dataways.gr  Fri Oct 19 11:30:03 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri, 19 Oct 2007 18:30:03 +0300
Subject: [cisco-voip] UC520 one way audio
Message-ID: <4718CD7B.2080600@dataways.gr>

Hi all,
i have a UC520 running 124-11.XW2 and 7940s running 7.2(T0.23).
The UC520 has 1 BRI VIC.

Sometimes, during a call of an inside ipphone user with a user outside 
the UC520 (after 5 minutes or so) the outside user stops hearing the 
ipphone user but the ipphone user continues to hear the other party.
The only solution is to hang up.
What do you suggest i should debug and how? It seems really random...
I have to possibility to log anything i want in a server.

Thanks
Anthony

From ErasmuE4 at telkom.co.za  Fri Oct 19 11:49:37 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Fri, 19 Oct 2007 17:49:37 +0200
Subject: [cisco-voip] call manager express blocking outbound caller ID
References: <472E195E692EE74BAE4E7B2880A78C1F84D692@CNTRRA20-XCS01.telkom.co.za>
	<4718BDCC.9000600@gmail.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D69C@CNTRRA20-XCS01.telkom.co.za>


thanks
 
 
souns like a good way of doing it -- will try it out

________________________________

From: Robert Kulagowski [mailto:rkulagow at gmail.com]
Sent: Fri 2007-10-19 16:23
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] call manager express blocking outbound
caller ID



You could do this with two different route patterns (which is how we
do it)

The first route pattern on CCM, where you want to block caller ID,
would
be set for "restricted" in the route pattern on the Calling line ID
presentation.

Lets say you use "8" to get an external line.  On CCM you'd have "8.!"
(or whatever) as the route pattern

On the router (h.323 dial-peers), it would be:

dial-peer voice 1000 pots
destination-pattern 8T
clid restrict
port 0/1/0:15

The second route pattern could be "*8.!" on CCM, and on the router
you'd
have

dial-peer voice 1001 pots
destination-patter *8T
port 0/1/0:15

So in dial-peer 1001, we don't have the clid restrict command, so it
_will_ send caller ID.




~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail is subject to the Telkom SA electronic communication legal notice, available at :
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From Brandon.Ogden at amersports.com  Fri Oct 19 11:53:19 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Fri, 19 Oct 2007 08:53:19 -0700
Subject: [cisco-voip] Phone Registration Question
References: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com><483125.82354.qm@web57205.mail.re3.yahoo.com><54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com><C1FE15183DA37645BC0633BC604E44F0071030AF@USNAEXCH.na.didata.local>
	<54DD8B39132EFA4293A5C0565004E5460A05FA34@SBEMAIL1.au.baesystems.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D99CBA@seas7633.am.amersports.int>

Strange Registration question... I have the last three phones that I am
trying to register; they are phones that were not connected when I
brought my publisher on line.  I am having a problem where it appears
the phone is pulling an IP Address, but is not registering.  The status
on the phone indicates a DHCP Timeout.   If I leave the phone connected
overnight, the next morning when I come in... it is up and running
great! 

 

I can reset and delete and recreate any other phones that are existing
with no problems.   

 

Any ideas? 

Brandon

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From jonvoip at gmail.com  Fri Oct 19 12:19:27 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 19 Oct 2007 11:19:27 -0500
Subject: [cisco-voip] Phone Registration Question
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D99CBA@seas7633.am.amersports.int>
References: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com>
	<483125.82354.qm@web57205.mail.re3.yahoo.com>
	<54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>
	<C1FE15183DA37645BC0633BC604E44F0071030AF@USNAEXCH.na.didata.local>
	<54DD8B39132EFA4293A5C0565004E5460A05FA34@SBEMAIL1.au.baesystems.com>
	<D646304AFC55DB4EBB92329BCC800A00D99CBA@seas7633.am.amersports.int>
Message-ID: <5d093f9a0710190919n6643264x1d4b7c6f9cf11395@mail.gmail.com>

Could be a voice vlan issue... is it getting an ip address for the
default tftp server?

Also, if your CCMs are listed by name and are not in DNS, it could be
taking that long to fail to the default tftp server.



Jonathan

On 10/19/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>
>
>
> Strange Registration question... I have the last three phones that I am
> trying to register; they are phones that were not connected when I brought
> my publisher on line.  I am having a problem where it appears the phone is
> pulling an IP Address, but is not registering.  The status on the phone
> indicates a DHCP Timeout.   If I leave the phone connected overnight, the
> next morning when I come in? it is up and running great!
>
>
>
> I can reset and delete and recreate any other phones that are existing with
> no problems.
>
>
>
> Any ideas?
>
> Brandon
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Fri Oct 19 12:23:03 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 19 Oct 2007 11:23:03 -0500
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <04B6D8CDD22FD34993C47E33E21F82EFD0B839@rachael-reborn.letnet.net>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int>
	<D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int>
	<5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com>
	<021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca>
	<5d093f9a0710181122l6d58b2b7h84967895def7494f@mail.gmail.com>
	<02ef01c811bb$c5a21730$56196883@cfs.uoguelph.ca>
	<04B6D8CDD22FD34993C47E33E21F82EF88DAA8@rachael-reborn.letnet.net>
	<f4445faf0710181751n2017e1e4jd004e244ada681eb@mail.gmail.com>
	<04B6D8CDD22FD34993C47E33E21F82EFD0B839@rachael-reborn.letnet.net>
Message-ID: <5d093f9a0710190923x1fe51d18x92a0f8fbad0d5ef7@mail.gmail.com>

The Cisco OS patches are not just for security... some of them are to
resolve issues with the OS, web server and other key services that CCM
uses.

For example, Daylight Savings Time requires OS patching...

I wouldn't be so quick to not apply patches... though I would
definitely wait for 3 weeks or so after Cisco releases a patch to
ensure it doesn't blow up your system (if it still on Cisco's site 3
weeks after, in all likelihood, it is a stable patch).



Jonathan

On 10/19/07, Johnson, Ken <kenjohnson at letu.edu> wrote:
> Hm - that's interesting, I might look at those timing settings.
>
> Not only do we not apply MS patches, we don't actually apply individual
> Cisco OS patches to the CM's anymore - we only apply the Cisco OS
> service packs. Keeping up with the individual Cisco OS patches got to
> where it didn't seem worth the effort - especially for a CM environment
> that is 100% isolated from anything but the phones and 3 management
> stations :-)
>
> So if I hadn't had this problem through a number of OS service packs I
> might think it was the lack of an individual Cisco patch I hadn't
> applied, but since it continued after the most recent OS service packs I
> expect not.  I'll probably be content to wait for our 6.0 upgrade :-)
>
> Ken Johnson
>    Mgr. Network Services,
>    Information Technology
>    LeTourneau University
> _________________________________
>  E-mail: kenjohnson at letu.edu
>  Helpdesk: (903) 233-3500
>  Phone: (903) 233-3520
>  Web: http://www.letu.edu/infotech/
>
> -----Original Message-----
> From: Erick Bergquist [mailto:erickbee at gmail.com]
> Sent: Thursday, October 18, 2007 7:51 PM
> To: Johnson, Ken
> Cc: Lelio Fulgenzi; Jonathan Charles; Ogden, Brandon;
> cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Registration to Subscriber
>
> The service recovery option will recover if the service was started,
> but if it doesn't start on boot then that is not going to kick in
> since it was never started and running yet.
>
> I've seen the ccm service not start on reboots when people have
> applied M$ patches to the server directly instead of the Cisco OS
> patches from cisco.com. I haven't seen the service not start since 3.3
> days due to timeouts or initialization timers being to short. You may
> want to check under call manager service parameters and see if all the
> Initialization timers are set to at least the suggested value. I had
> one place a long time ago update from 3.x something to 4.0x and it
> kept the older 3.x settings which was less then the suggested timer
> value for 4.x so that caused the ccm service not to start. Once the
> timer was set to the suggested value for 4.x the service starts fine
> on reboots.
>
>
> On 10/18/07, Johnson, Ken <kenjohnson at letu.edu> wrote:
> > Actually we've had this problem sporadically with our subscriber since
> > some 4.2(X) patch with the CCM service not consistently starting on
> > bootup (even with the service recovery processes set). It always
> starts
> > okay manually once the monitor emails remind us of the issue - and it
> > never affects the publisher, nor does it affect any other service or
> > write anything useful in the error log.
> >
> > Since it never fails while running, only fails to come up with a
> reboot
> > or other CCM restart which almost never happens - even though TAC
> could
> > certainly fix it for us, we ignored it and focused on more pressing
> > cases figuring it would be a non-issue with the wipe-and-reload CM6.0
> > process....
> >
> > Ken Johnson
> >    Mgr. Network Services,
> >    Information Technology
> >    LeTourneau University
> > _________________________________
> >  E-mail: kenjohnson at letu.edu
> >  Helpdesk: (903) 233-3500
> >  Phone: (903) 233-3520
> >  Web: http://www.letu.edu/infotech/
> >
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio
> Fulgenzi
> > Sent: Thursday, October 18, 2007 2:19 PM
> > To: Jonathan Charles
> > Cc: Ogden, Brandon; cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Registration to Subscriber
> >
> > gotta love the consistency... ;)
> >
> > ----- Original Message -----
> > From: "Jonathan Charles" <jonvoip at gmail.com>
> > To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> > Cc: "Ogden, Brandon" <Brandon.Ogden at amersports.com>;
> > <cisco-voip at puck.nether.net>
> > Sent: Thursday, October 18, 2007 2:22 PM
> > Subject: Re: [cisco-voip] Registration to Subscriber
> >
> >
> > > It depends
> > >
> > > I have seen the service take forever on 100 patterns and start
> > > immediately on 1000 patterns...
> > >
> > >
> > > Jonathan
> > >
> > > On 10/18/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > >> what's "a large number of route patterns" ?
> > >>
> > >> 100? 1,000? 10,000?
> > >>
> > >> ----- Original Message -----
> > >> From: "Jonathan Charles" <jonvoip at gmail.com>
> > >> To: "Ogden, Brandon" <Brandon.Ogden at amersports.com>
> > >> Cc: <cisco-voip at puck.nether.net>
> > >> Sent: Thursday, October 18, 2007 2:14 PM
> > >> Subject: Re: [cisco-voip] Registration to Subscriber
> > >>
> > >>
> > >> > If you have a large number of route patterns the CCM service can
> > >> > time out when starting
> > >> >
> > >> > There is a service parameter to increase the timeout, but I
> forgot
> > >> > what it was...
> > >> >
> > >> >
> > >> > Jonathan
> > >> >
> > >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> > >> >> LOL.. I got it resolved.. Didn't have to reboot.. and before you
> > >> >> say so.. yes I am an idiot.. LOL  The Call Manager Service had
> not
> >
> > >> >> started for some reason.. I restarted the service and it popped
> > >> >> right up, all is well now.  Thanks all!
> > >> >>
> > >> >> Brandon
> > >> >>
> > >> >> -----Original Message-----
> > >> >> From: cisco-voip-bounces at puck.nether.net
> > >> >> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ogden,
> > >> >> Brandon
> > >> >> Sent: Thursday, October 18, 2007 10:18 AM
> > >> >> To: Jonathan Charles
> > >> >> Cc: cisco-voip at puck.nether.net
> > >> >> Subject: Re: [cisco-voip] Registration to Subscriber
> > >> >>
> > >> >> I did not do a reboot of the publisher, just the sub.
> > >> >>
> > >> >> Brandon Ogden
> > >> >> Amer Sports
> > >> >> 801-624-7529
> > >> >> www.amersports.com
> > >> >> http://www.linkedin.com/in/brandonogden
> > >> >>
> > >> >> -----Original Message-----
> > >> >> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> > >> >> Sent: Thursday, October 18, 2007 10:14 AM
> > >> >> To: Ogden, Brandon
> > >> >> Cc: Scott Voll; cisco-voip at puck.nether.net
> > >> >> Subject: Re: [cisco-voip] Registration to Subscriber
> > >> >>
> > >> >> Did you reboot the cluster after the rebuild?
> > >> >>
> > >> >>
> > >> >> Jonathan
> > >> >>
> > >> >> On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com>
> wrote:
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> > I just went in and looked at that and my subscriber is not
> > >> >> > showing up
> > >> >> as an
> > >> >> > available server in the Call Manager Group.  Any ideas?  It
> > >> >> > appears to
> > >> >> have
> > >> >> > replicated correctly, and I was able to use it to modify a
> > >> >> > Directory
> > >> >> Number.
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> > Brandon
> > >> >> >
> > >> >> >  ________________________________
> > >> >> >
> > >> >> >
> > >> >> > From: Scott Voll [mailto:svoll.voip at gmail.com]
> > >> >> >  Sent: Thursday, October 18, 2007 8:54 AM
> > >> >> >  To: Ogden, Brandon
> > >> >> >  Cc: cisco-voip at puck.nether.net
> > >> >> >  Subject: Re: [cisco-voip] Registration to Subscriber
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> > did you change the order of your servers so it registers with
> > >> >> > the sub?
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> > Scott
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> > On 10/18/07, Ogden, Brandon <Brandon.Ogden at amersports.com>
> > wrote:
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> > I rebuilt my subscriber box last night, it seemed to go well.
> I
> >
> > >> >> > was wondering how long it should take for the phones to
> register
> >
> > >> >> > with the Subscriber rather than the pub.  As of this morning,
> > >> >> > they are still registered with the pub.  Thanks!
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> > Brandon
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >
> > >> >> >  _______________________________________________
> > >> >> >  cisco-voip mailing list
> > >> >> >  cisco-voip at puck.nether.net
> > >> >> >  https://puck.nether.net/mailman/listinfo/cisco-voip
> > >> >> >
> > >> >> >
> > >> >> > _______________________________________________
> > >> >> > cisco-voip mailing list
> > >> >> > cisco-voip at puck.nether.net
> > >> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >> >> >
> > >> >>
> > >> >> This e-mail may contain confidential and/or privileged
> > >> >> information. If you are not the intended recipient (or have
> > >> >> received this e-mail in
> > >> >> error)
> > >> >> please notify the sender immediately and destroy this e-mail.
> Any
> > >> >> unauthorized copying, disclosure or distribution of the material
> > >> >> in this e-mail is strictly forbidden.
> > >> >> _______________________________________________
> > >> >> cisco-voip mailing list
> > >> >> cisco-voip at puck.nether.net
> > >> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >> >>
> > >> > _______________________________________________
> > >> > cisco-voip mailing list
> > >> > cisco-voip at puck.nether.net
> > >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >> >
> > >>
> > >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> > --
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> --
> BEGIN-ANTISPAM-VOTING-LINKS
> ------------------------------------------------------
>
> Teach CanIt if this mail (ID 8255423) is spam:
> Spam:
> http://antispam.letu.edu/canit/b.php?i=8255423&m=869444ceaabe&c=s
> Not spam:
> http://antispam.letu.edu/canit/b.php?i=8255423&m=869444ceaabe&c=n
> Forget vote:
> http://antispam.letu.edu/canit/b.php?i=8255423&m=869444ceaabe&c=f
> ------------------------------------------------------
> END-ANTISPAM-VOTING-LINKS
>
>

From jgrace at adcapnet.com  Fri Oct 19 12:32:47 2007
From: jgrace at adcapnet.com (James Grace)
Date: Fri, 19 Oct 2007 12:32:47 -0400
Subject: [cisco-voip] IE on xp has delay for CM 6.0
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E8BF@mail3.adcapnetsvc.local>

?
for some reason my IE Browser has a delay when im navigating thru the administration pages.  But went I use a windows 2003 server IE browser its fine.  I've upgraded to 7 and down grade to 6.0 sp 2.  Have anybody else seen this.  By the way my browser works fine on everything else.  
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
recipient is prohibited. If you received this message in error, please contact the sender and delete the
material from all computers."

This original email was sent to the internet for delivery at 19 Oct 2007 12:35:26 -0400
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From svoll.voip at gmail.com  Fri Oct 19 12:50:03 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 19 Oct 2007 09:50:03 -0700
Subject: [cisco-voip] how to change Span to PC Port configuration in a
	7941 phone
In-Reply-To: <A805E226FF74D74E90D5BE19D664A9A0137F25@neptuno.atminformatica.pt>
References: <A805E226FF74D74E90D5BE19D664A9A0137F25@neptuno.atminformatica.pt>
Message-ID: <f84a38d30710190950s5787f7c3ja8eb67d4deb2ab9e@mail.gmail.com>

on the ccmadmin phone device it should be at the bottom.  I'm running 4.1 so
maybe something changed in 4.2 but I wouldn't think so.

Scott


On 10/19/07, Pedro Nogueira <pedro.nogueira at sicnet.pt> wrote:
>
>  Hello folks,
>
>
> Does anybody knows how to change the Span to PC Port configuration in a
> 7941 phone - No we cant do it through the web conf page since the option
> does not appear on it.
>
> On the phone configuration i cant access it too, since the option appear
> but is disabled (not accessible), so i cant modify it.
>
> Does anybody knows why does this happen and how to modify this setting ?
>
> My Call Manager is a 4.2.3 version.
>
> it seems a really stupid question but i really appreciate any help on it,
> since i cant put the supervisor monitor option of CRS running without this
> option on.
>
> Really need help here,any thoughts or suggestions would be really
> appreciated !!!
>
>
> Thanks in advanced,
> regards,
>
> Peter
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From svoll.voip at gmail.com  Fri Oct 19 12:52:42 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 19 Oct 2007 09:52:42 -0700
Subject: [cisco-voip] Phone Registration Question
In-Reply-To: <5d093f9a0710190919n6643264x1d4b7c6f9cf11395@mail.gmail.com>
References: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com>
	<483125.82354.qm@web57205.mail.re3.yahoo.com>
	<54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>
	<C1FE15183DA37645BC0633BC604E44F0071030AF@USNAEXCH.na.didata.local>
	<54DD8B39132EFA4293A5C0565004E5460A05FA34@SBEMAIL1.au.baesystems.com>
	<D646304AFC55DB4EBB92329BCC800A00D99CBA@seas7633.am.amersports.int>
	<5d093f9a0710190919n6643264x1d4b7c6f9cf11395@mail.gmail.com>
Message-ID: <f84a38d30710190952t7aca97afy74c07a94fce179d9@mail.gmail.com>

are these phone on a vlan that doesn't have DNS access.

I've seen issues with DNS could NOT resolve the CMPUB.

Scott


On 10/19/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Could be a voice vlan issue... is it getting an ip address for the
> default tftp server?
>
> Also, if your CCMs are listed by name and are not in DNS, it could be
> taking that long to fail to the default tftp server.
>
>
>
> Jonathan
>
> On 10/19/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >
> >
> >
> >
> > Strange Registration question... I have the last three phones that I am
> > trying to register; they are phones that were not connected when I
> brought
> > my publisher on line.  I am having a problem where it appears the phone
> is
> > pulling an IP Address, but is not registering.  The status on the phone
> > indicates a DHCP Timeout.   If I leave the phone connected overnight,
> the
> > next morning when I come in? it is up and running great!
> >
> >
> >
> > I can reset and delete and recreate any other phones that are existing
> with
> > no problems.
> >
> >
> >
> > Any ideas?
> >
> > Brandon
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From wsisk at cisco.com  Fri Oct 19 12:55:47 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 19 Oct 2007 12:55:47 -0400
Subject: [cisco-voip] Registration to Subscriber
In-Reply-To: <04B6D8CDD22FD34993C47E33E21F82EF88DAA8@rachael-reborn.letnet.net>
References: <5c72afa80710172026x1cde3272of8bfc30ca6204ae5@mail.gmail.com><5c72afa80710172146l70246230x40f8d32eadd4c2a8@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D5A3BB@seas7633.am.amersports.int><f84a38d30710180753n22d15e34o78911f190dd8fbc3@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D99511@seas7633.am.amersports.int><5d093f9a0710180914s61bb4ef5v1cc8f5ab424be979@mail.gmail.com><D646304AFC55DB4EBB92329BCC800A00D9954D@seas7633.am.amersports.int><D646304AFC55DB4EBB92329BCC800A00D99676@seas7633.am.amersports.int><5d093f9a0710181114x35631f72x233c8f619f9be68@mail.gmail.com><021001c811b3$2ad48c90$56196883@cfs.uoguelph.ca><5d093f9a0710181122l6d58b2b7h84967895def7494f@mail.gmail.com>	<02ef01c811bb$c5a21730$56196883@cfs.uoguelph.ca>
	<04B6D8CDD22FD34993C47E33E21F82EF88DAA8@rachael-reborn.letnet.net>
Message-ID: <4718E193.1010001@cisco.com>

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From wsisk at cisco.com  Fri Oct 19 13:03:14 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 19 Oct 2007 13:03:14 -0400
Subject: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
In-Reply-To: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>
References: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>
Message-ID: <4718E352.5070004@cisco.com>

Hi Patrick,

Good stuff.  Most of this is finally covered in CM513 docs:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_installation_guide09186a00808be1dc.html



Patrick Diener wrote:
> I'm trying to put together a rough step by step procedure to do a
> disaster recovery for a cucm 5.1(2) cluster and there are some stuff I
> can't figure out how it should/will work...
>
> I based the steps on the DRS Admin Guide for Version 5.0 (5.1 not out yet?!?)
>
> Important: I try to do a restore on a new server hw => new first node MAC
>
> Step 1:
> perform fresh install of cucm sw
> !! set same hostname as before !!
>   
ws: same hostname, ip, and passwords
> Step 2:
> perform restore using DRS and restart server (worked like a charm in the lab...)
>
> form now on I'm stuck...
>   
ws: you restored the pub only, correct? and rebooted?  after you 
reinstall sub will need to DRS restore it as well.
> Step 3:
> Licensing?!?
> all activated services will not start cause the .lic files are not
> valid for the new server hw (at least thats what happened in the
> lab..)
> => will TAC send me a new .lic equal to the old one?
> => do I need to contact licensing at cisco.com?
> => any experience how long it will take to get the new .lic files?
>   
ws: yep, new license files if replacing the pub. covered in the doc, 
licensing at cisco.com
> Step 4:
> move some phones to the restored publisher (assuming callmanager
> service is running) and test all features needed
> => will the phones register if security has been configured? I assume
> the new callmanager, tftp and capf certificates will not match the
> certificates stored in the phones ctl file (right?) how do I get out
> of this mess without touching each and every phone to delete the ctl
> file or perform a factory reset?
>   
ws: this comes later actually. check the doc, you will create new CTL 
files using same USB token.
> Step 5:
> restore sub (or subs)
>   
step5: reinstall subs, then restore subs, then perform post-replace 
checklist.
>
> all comments, experiences and remarks are appreciated!
>
> cheers
> Patrick
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Fri Oct 19 13:04:11 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 19 Oct 2007 13:04:11 -0400
Subject: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
In-Reply-To: <f84a38d30710190752r7ba7ec13l3ff19927c81427d@mail.gmail.com>
References: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>
	<f84a38d30710190752r7ba7ec13l3ff19927c81427d@mail.gmail.com>
Message-ID: <4718E38B.4070708@cisco.com>

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From wsisk at cisco.com  Fri Oct 19 13:05:18 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 19 Oct 2007 13:05:18 -0400
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <723FA17C6AA4C4428CDF572DCAEE08D405445C4D@EXCHANGEDS01.ds.ad.adp.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>	<22e6380b0710182002p30265d7do404b0f2d9b144842@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445C4D@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <4718E3CE.20403@cisco.com>

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From wsisk at cisco.com  Fri Oct 19 13:06:05 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 19 Oct 2007 13:06:05 -0400
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <723FA17C6AA4C4428CDF572DCAEE08D405445E35@EXCHANGEDS01.ds.ad.adp.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>	<535625f70710190630j2671e49bs4d827d09278dc3d8@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445E35@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <4718E3FD.2020501@cisco.com>

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From mattjhughes at gmail.com  Fri Oct 19 13:11:13 2007
From: mattjhughes at gmail.com (Matthew J. Hughes)
Date: Fri, 19 Oct 2007 10:11:13 -0700
Subject: [cisco-voip] Push IP phone service to phone
Message-ID: <ea211ea30710191011l43fff97am9c082cec23fd8a0e@mail.gmail.com>

First post here, new to the group :)

I am attempting to  gather the physical location of phones on campus for use
with emergency responder.  I want the phone user to be prompted with a IP
Phone Service on their 79xx phones, that asks them to verify their physical
location (Room Number, Building Name)  I don't want this to be burdensome so
I only want it to come up every time a phone reboots (It wont move locations
unless its unplugged).   To accomplish this  I ran into these roadblocks

1.  IDLE URL will certainly force them to see it, but will bug them far to
much. (although if i set the idle timer long enough it could come up only
once a day.)
2. I can push messages to their screen, but don't think i can push a XML
service.

I am thinking I would have to use AXL/SOAP to add/remove idle URL,  OR
somehow as part of the boot process force them to a XML service.

any thoughts?  I am running CCM 6.0 and CER

Thanks in advance,
Matt
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From Ruben.Montes at eu.didata.com  Fri Oct 19 13:27:56 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Fri, 19 Oct 2007 19:27:56 +0200
Subject: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
References: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>
	<4718E352.5070004@cisco.com>
Message-ID: <FA233791F398B848B978464507739E8FDA77CE@EUBEBRUSVEX1.eu.didata.local>

Hello,

I had to do a Publisher replacement from a MCS7825 to a MCS7835 last
week. We had two additional subscribers. I followed step by step the
guide Wes has mentioned and all went fine (the whole change took us 9
hours).

The main problem is that you MUST reinstall also all the subscribers
from the cluster: this was confirmed by TAC and by the customer's SE
cisco engineer.

If you have any additional question about this just drop me an email...

Regards,

Ruben

-----Mensaje original-----
De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Wes Sisk
Enviado el: viernes, 19 de octubre de 2007 19:03
Para: Patrick Diener
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)

Hi Patrick,

Good stuff.  Most of this is finally covered in CM513 docs:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_installation_g
uide09186a00808be1dc.html



Patrick Diener wrote:
> I'm trying to put together a rough step by step procedure to do a
> disaster recovery for a cucm 5.1(2) cluster and there are some stuff I
> can't figure out how it should/will work...
>
> I based the steps on the DRS Admin Guide for Version 5.0 (5.1 not out
yet?!?)
>
> Important: I try to do a restore on a new server hw => new first node
MAC
>
> Step 1:
> perform fresh install of cucm sw
> !! set same hostname as before !!
>   
ws: same hostname, ip, and passwords
> Step 2:
> perform restore using DRS and restart server (worked like a charm in
the lab...)
>
> form now on I'm stuck...
>   
ws: you restored the pub only, correct? and rebooted?  after you 
reinstall sub will need to DRS restore it as well.
> Step 3:
> Licensing?!?
> all activated services will not start cause the .lic files are not
> valid for the new server hw (at least thats what happened in the
> lab..)
> => will TAC send me a new .lic equal to the old one?
> => do I need to contact licensing at cisco.com?
> => any experience how long it will take to get the new .lic files?
>   
ws: yep, new license files if replacing the pub. covered in the doc, 
licensing at cisco.com
> Step 4:
> move some phones to the restored publisher (assuming callmanager
> service is running) and test all features needed
> => will the phones register if security has been configured? I assume
> the new callmanager, tftp and capf certificates will not match the
> certificates stored in the phones ctl file (right?) how do I get out
> of this mess without touching each and every phone to delete the ctl
> file or perform a factory reset?
>   
ws: this comes later actually. check the doc, you will create new CTL 
files using same USB token.
> Step 5:
> restore sub (or subs)
>   
step5: reinstall subs, then restore subs, then perform post-replace 
checklist.
>
> all comments, experiences and remarks are appreciated!
>
> cheers
> Patrick
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jonvoip at gmail.com  Fri Oct 19 14:04:14 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 19 Oct 2007 13:04:14 -0500
Subject: [cisco-voip] CUE Versions and upgrades...
Message-ID: <5d093f9a0710191104l22475af8v4b0b9882f3f10601@mail.gmail.com>

I am doing my first CUE upgrade next week... customer is hyper
high-profile (e.g. my CEO will hear about it if this goes bad)...

Anyway, I am looking for a stable version of CUE software to upgrade
to, they are probably at an older version of 2.3 (haven't gotten all
the info yet)...

Was curious:

Which version of CUE software is stable? I see that 3.0.2 was released
four days ago, so let's stay away from that... but is 3.0.1 stable?

Any hilarious bugs (like CME 3.3 where random phones would just ring
for no reason) I should be paranoid about?



Jonathan

From jvw at medicineforthedefense.com  Fri Oct 19 14:27:20 2007
From: jvw at medicineforthedefense.com (Weigand, John V.)
Date: Fri, 19 Oct 2007 14:27:20 -0400
Subject: [cisco-voip] 7941G-GE/3rd Party Phone System
In-Reply-To: <5d093f9a0710191104l22475af8v4b0b9882f3f10601@mail.gmail.com>
References: <5d093f9a0710191104l22475af8v4b0b9882f3f10601@mail.gmail.com>
Message-ID: <1E196438D2BA76448E5A077C8C99329D065B2E5B@lmiexmb01.corp.medicineforthedefense.com>

Serious Medicine for the Defense R 

Is there any way to use a 7941G-GE with a 3rd party phone system (i.e.
not Callmanager?) I've seen there's non-Callmanager SIP firmware for the
7940, but I'm not seeing any for the 7941, or a way to configure it to
use a 3rd party system.

Thanks!


John V. Weigand
Help Desk Support

Litigation Management, Inc.
300 Allen-Bradley Drive
Suite 200
Mayfield Heights, OH 44124





Tel: 
Fax: 440-484-2020
Cell: 
email: jvw at medicineforthedefense.com 




PRIVILEGE AND CONFIDENTIALITY NOTICE 

The information in this electronic mail is intended for the named recipients only. It may contain privileged and confidential material and may be protected under law by the Health Insurance Portability and Accountability Act. Any use of this information by anyone other than the intended receiver is prohibited. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, copying, or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately by replying to this electronic e-mail or by calling (800) 778-5424. Please delete it from your computer. Thank you.

From mattjhughes at gmail.com  Fri Oct 19 14:42:45 2007
From: mattjhughes at gmail.com (Matthew J. Hughes)
Date: Fri, 19 Oct 2007 11:42:45 -0700
Subject: [cisco-voip] 7941G-GE/3rd Party Phone System
In-Reply-To: <1E196438D2BA76448E5A077C8C99329D065B2E5B@lmiexmb01.corp.medicineforthedefense.com>
References: <5d093f9a0710191104l22475af8v4b0b9882f3f10601@mail.gmail.com>
	<1E196438D2BA76448E5A077C8C99329D065B2E5B@lmiexmb01.corp.medicineforthedefense.com>
Message-ID: <ea211ea30710191142x5f4356b1v25bc8717f6244c1e@mail.gmail.com>

Hope this helps

Here is the link to the v8 version of the SIP firmware for 79x1 phones

http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a0080621246.html

excerpt from
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx


The v8.0 software for the 7941/7961 has these significant differences over
the v8.0 software for the 7940/7960:
The built-in web server previously only in the non-SIP images is now in the
v8.0 SIP builds

   - The phone does not appear to support the old style config files, all
   configs for this phone need to be in XML
   - The phone does not appear to support telnet, but instead supports
   SSH-2.0, although this appears to be undocumented on CCO. At this
   stage unsure what the default user/password is
   - The config file name is therefore now SEP<mac-address>.cnf.xml
   - Upgrading the firmware load involves the phone uploading 5 signed
   binary files: jar41sip.8-0-1-18.sbn, cnu41.3-1-1-15.sbn,
   apps41.1-1-1-15.sbn, dsp41.1-1-1-15.sbn, and
cvm41sip.8-0-1-18.sbn(this was for release load version 8-0-2SR1-0-1
which is the latest as of
   early April 2006). These phones can be upgraded and the protocol support
   changed via the normal means of adding
<loadInformation>term41.default</loadInformation>
   to the .cfg.xml file



On 10/19/07, Weigand, John V. <jvw at medicineforthedefense.com> wrote:
>
> Serious Medicine for the Defense R
>
> Is there any way to use a 7941G-GE with a 3rd party phone system (i.e.
> not Callmanager?) I've seen there's non-Callmanager SIP firmware for the
> 7940, but I'm not seeing any for the 7941, or a way to configure it to
> use a 3rd party system.
>
> Thanks!
>
>
> John V. Weigand
> Help Desk Support
>
> Litigation Management, Inc.
> 300 Allen-Bradley Drive
> Suite 200
> Mayfield Heights, OH 44124
>
>
>
>
>
> Tel:
> Fax: 440-484-2020
> Cell:
> email: jvw at medicineforthedefense.com
>
>
>
>
> PRIVILEGE AND CONFIDENTIALITY NOTICE
>
> The information in this electronic mail is intended for the named
> recipients only. It may contain privileged and confidential material and may
> be protected under law by the Health Insurance Portability and
> Accountability Act. Any use of this information by anyone other than the
> intended receiver is prohibited. If the reader of this message is not the
> intended recipient, you are hereby notified that any dissemination,
> distribution, copying, or other use of this message or its attachments is
> strictly prohibited. If you have received this message in error, please
> notify the sender immediately by replying to this electronic e-mail or by
> calling (800) 778-5424. Please delete it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Brandon.Ogden at amersports.com  Fri Oct 19 15:03:14 2007
From: Brandon.Ogden at amersports.com (Ogden, Brandon)
Date: Fri, 19 Oct 2007 12:03:14 -0700
Subject: [cisco-voip] Phone Registration Question
In-Reply-To: <5d093f9a0710190919n6643264x1d4b7c6f9cf11395@mail.gmail.com>
References: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com>
	<483125.82354.qm@web57205.mail.re3.yahoo.com>
	<54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>
	<C1FE15183DA37645BC0633BC604E44F0071030AF@USNAEXCH.na.didata.local>
	<54DD8B39132EFA4293A5C0565004E5460A05FA34@SBEMAIL1.au.baesystems.com>
	<D646304AFC55DB4EBB92329BCC800A00D99CBA@seas7633.am.amersports.int>
	<5d093f9a0710190919n6643264x1d4b7c6f9cf11395@mail.gmail.com>
Message-ID: <D646304AFC55DB4EBB92329BCC800A00D99EDE@seas7633.am.amersports.int>

I just added the CCM's to the DNS and it did not seem to solve the
problem...  It does not appear they are getting the TFTP server ip's.
This phone I am working on was working fine yesterday, was unplugged
overnight, and now wont register.  But if I leave it overnight, I can
all but promise it will be registered tomorrow.  VLAN's etc. has not
changed.  I have several hundred phones registered, but any new phone is
taking forever. 

Brandon 





From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, October 19, 2007 10:19 AM
To: Ogden, Brandon
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Registration Question

Could be a voice vlan issue... is it getting an ip address for the
default tftp server?

Also, if your CCMs are listed by name and are not in DNS, it could be
taking that long to fail to the default tftp server.



Jonathan

On 10/19/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
>
>
>
>
> Strange Registration question... I have the last three phones that I
am
> trying to register; they are phones that were not connected when I
brought
> my publisher on line.  I am having a problem where it appears the
phone is
> pulling an IP Address, but is not registering.  The status on the
phone
> indicates a DHCP Timeout.   If I leave the phone connected overnight,
the
> next morning when I come in... it is up and running great!
>
>
>
> I can reset and delete and recreate any other phones that are existing
with
> no problems.
>
>
>
> Any ideas?
>
> Brandon
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Fri Oct 19 15:34:01 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 19 Oct 2007 14:34:01 -0500
Subject: [cisco-voip] Phone Registration Question
In-Reply-To: <D646304AFC55DB4EBB92329BCC800A00D99EDE@seas7633.am.amersports.int>
References: <84b74aec0710110601l6a6f94dcy162dfb4e68178e1b@mail.gmail.com>
	<483125.82354.qm@web57205.mail.re3.yahoo.com>
	<54DD8B39132EFA4293A5C0565004E5460A05F866@SBEMAIL1.au.baesystems.com>
	<C1FE15183DA37645BC0633BC604E44F0071030AF@USNAEXCH.na.didata.local>
	<54DD8B39132EFA4293A5C0565004E5460A05FA34@SBEMAIL1.au.baesystems.com>
	<D646304AFC55DB4EBB92329BCC800A00D99CBA@seas7633.am.amersports.int>
	<5d093f9a0710190919n6643264x1d4b7c6f9cf11395@mail.gmail.com>
	<D646304AFC55DB4EBB92329BCC800A00D99EDE@seas7633.am.amersports.int>
Message-ID: <5d093f9a0710191234v2cb9669ex74f88ae8ea4b29e3@mail.gmail.com>

OK, the IP address that they are getting, take a look at the DHCP
scope (go to the DHCP server (first line of the network config on the
phone)).

Does option 150 exist on this scope? Does it point to your CCM Publisher?


Jonathan

On 10/19/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> I just added the CCM's to the DNS and it did not seem to solve the
> problem...  It does not appear they are getting the TFTP server ip's.
> This phone I am working on was working fine yesterday, was unplugged
> overnight, and now wont register.  But if I leave it overnight, I can
> all but promise it will be registered tomorrow.  VLAN's etc. has not
> changed.  I have several hundred phones registered, but any new phone is
> taking forever.
>
> Brandon
>
>
>
>
>
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Friday, October 19, 2007 10:19 AM
> To: Ogden, Brandon
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Phone Registration Question
>
> Could be a voice vlan issue... is it getting an ip address for the
> default tftp server?
>
> Also, if your CCMs are listed by name and are not in DNS, it could be
> taking that long to fail to the default tftp server.
>
>
>
> Jonathan
>
> On 10/19/07, Ogden, Brandon <Brandon.Ogden at amersports.com> wrote:
> >
> >
> >
> >
> > Strange Registration question... I have the last three phones that I
> am
> > trying to register; they are phones that were not connected when I
> brought
> > my publisher on line.  I am having a problem where it appears the
> phone is
> > pulling an IP Address, but is not registering.  The status on the
> phone
> > indicates a DHCP Timeout.   If I leave the phone connected overnight,
> the
> > next morning when I come in... it is up and running great!
> >
> >
> >
> > I can reset and delete and recreate any other phones that are existing
> with
> > no problems.
> >
> >
> >
> > Any ideas?
> >
> > Brandon
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From slz at baycix.de  Fri Oct 19 14:59:05 2007
From: slz at baycix.de (Sascha Lenz)
Date: Fri, 19 Oct 2007 20:59:05 +0200
Subject: [cisco-voip] 7941G-GE/3rd Party Phone System
In-Reply-To: <1E196438D2BA76448E5A077C8C99329D065B2E5B@lmiexmb01.corp.medicineforthedefense.com>
References: <5d093f9a0710191104l22475af8v4b0b9882f3f10601@mail.gmail.com>
	<1E196438D2BA76448E5A077C8C99329D065B2E5B@lmiexmb01.corp.medicineforthedefense.com>
Message-ID: <4718FE79.5060301@baycix.de>

Hi,

Weigand, John V. schrieb:
> Serious Medicine for the Defense R 
> 
> Is there any way to use a 7941G-GE with a 3rd party phone system (i.e.
> not Callmanager?) I've seen there's non-Callmanager SIP firmware for the
> 7940, but I'm not seeing any for the 7941, or a way to configure it to
> use a 3rd party system.

try

http://www.voip-info.org/wiki/view/Standalone+Cisco+7941%252F7961+without+a+local+PBX

for SIP (complicated, doesn't always work as expected)

or if you want to run Asterisk anyways, there's always chan_skinny / 
chan_sccp

http://www.voip-info.org/wiki/view/Asterisk+SCCP+channels
http://www.voip-info.org/wiki/view/chan_sccp2
(follow the links and use your prefered search enginge, too)

note: Asterisk is not a full solution as a Callmanager replacement, and 
chan_sccp lacks some development power for a while now :-/
Basic Phone-stuff works quite fine though.
I personally prefer to use Cisco Phones with CM/CME though whenever 
possible :-)


-- 
========================================================================
= Sascha Lenz                  SLZ-RIPE          slz at baycix.de         =
= Network Operations                                                   =
= BayCIX GmbH, Landshut                  * PGP public Key on demand *  =
========================================================================

From cheesevoice at gmail.com  Fri Oct 19 16:25:38 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Fri, 19 Oct 2007 15:25:38 -0500
Subject: [cisco-voip] 7941G-GE/3rd Party Phone System
In-Reply-To: <1E196438D2BA76448E5A077C8C99329D065B2E5B@lmiexmb01.corp.medicineforthedefense.com>
References: <5d093f9a0710191104l22475af8v4b0b9882f3f10601@mail.gmail.com>
	<1E196438D2BA76448E5A077C8C99329D065B2E5B@lmiexmb01.corp.medicineforthedefense.com>
Message-ID: <84b74aec0710191325v2defcb2l361f5e41ea12fd11@mail.gmail.com>

last job was with a VOIP wholesale provider running Sylantro system, used
polycom and cisco phones...SIP based...

On 10/19/07, Weigand, John V. <jvw at medicineforthedefense.com> wrote:
>
> Serious Medicine for the Defense R
>
> Is there any way to use a 7941G-GE with a 3rd party phone system (i.e.
> not Callmanager?) I've seen there's non-Callmanager SIP firmware for the
> 7940, but I'm not seeing any for the 7941, or a way to configure it to
> use a 3rd party system.
>
> Thanks!
>
>
> John V. Weigand
> Help Desk Support
>
> Litigation Management, Inc.
> 300 Allen-Bradley Drive
> Suite 200
> Mayfield Heights, OH 44124
>
>
>
>
>
> Tel:
> Fax: 440-484-2020
> Cell:
> email: jvw at medicineforthedefense.com
>
>
>
>
> PRIVILEGE AND CONFIDENTIALITY NOTICE
>
> The information in this electronic mail is intended for the named
> recipients only. It may contain privileged and confidential material and may
> be protected under law by the Health Insurance Portability and
> Accountability Act. Any use of this information by anyone other than the
> intended receiver is prohibited. If the reader of this message is not the
> intended recipient, you are hereby notified that any dissemination,
> distribution, copying, or other use of this message or its attachments is
> strictly prohibited. If you have received this message in error, please
> notify the sender immediately by replying to this electronic e-mail or by
> calling (800) 778-5424. Please delete it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From asobihoudai at yahoo.com  Fri Oct 19 16:29:07 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 19 Oct 2007 13:29:07 -0700 (PDT)
Subject: [cisco-voip] CUE Versions and upgrades...
In-Reply-To: <5d093f9a0710191104l22475af8v4b0b9882f3f10601@mail.gmail.com>
Message-ID: <561424.5609.qm@web57215.mail.re3.yahoo.com>

Jonathan,

I am using 3.0.1 at three customer offices and it is
working quite handily. There was one instance where a
user would receive a voicemail and not be able to hear
it but a reboot of CUE solved it quickly. I have not
heard of any other major issues since then.

Any version older than 2.3.4 MUST be upgraded to at
least 2.3.4 for DST patches.

Paul
--- Jonathan Charles <jonvoip at gmail.com> wrote:

> I am doing my first CUE upgrade next week...
> customer is hyper
> high-profile (e.g. my CEO will hear about it if this
> goes bad)...
> 
> Anyway, I am looking for a stable version of CUE
> software to upgrade
> to, they are probably at an older version of 2.3
> (haven't gotten all
> the info yet)...
> 
> Was curious:
> 
> Which version of CUE software is stable? I see that
> 3.0.2 was released
> four days ago, so let's stay away from that... but
> is 3.0.1 stable?
> 
> Any hilarious bugs (like CME 3.3 where random phones
> would just ring
> for no reason) I should be paranoid about?
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From svoll.voip at gmail.com  Fri Oct 19 17:40:25 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 19 Oct 2007 14:40:25 -0700
Subject: [cisco-voip] Push IP phone service to phone
In-Reply-To: <ea211ea30710191011l43fff97am9c082cec23fd8a0e@mail.gmail.com>
References: <ea211ea30710191011l43fff97am9c082cec23fd8a0e@mail.gmail.com>
Message-ID: <f84a38d30710191440s4d0a66b8sda594dace1429f57@mail.gmail.com>

first off..... Welcome to the Puck list!

Are you running CER downt to the room location?  Is this a policy?  Are you
going to have a ERL / ELIN per room?  seems a little over kill.  or is it
just for the response personel?

Scott


On 10/19/07, Matthew J. Hughes <mattjhughes at gmail.com> wrote:
>
> First post here, new to the group :)
>
> I am attempting to  gather the physical location of phones on campus for
> use with emergency responder.  I want the phone user to be prompted with a
> IP Phone Service on their 79xx phones, that asks them to verify their
> physical location (Room Number, Building Name)  I don't want this to be
> burdensome so I only want it to come up every time a phone reboots (It wont
> move locations unless its unplugged).   To accomplish this  I ran into these
> roadblocks
>
> 1.  IDLE URL will certainly force them to see it, but will bug them far to
> much. (although if i set the idle timer long enough it could come up only
> once a day.)
> 2. I can push messages to their screen, but don't think i can push a XML
> service.
>
> I am thinking I would have to use AXL/SOAP to add/remove idle URL,  OR
> somehow as part of the boot process force them to a XML service.
>
> any thoughts?  I am running CCM 6.0 and CER
>
> Thanks in advance,
> Matt
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From mattjhughes at gmail.com  Fri Oct 19 18:20:45 2007
From: mattjhughes at gmail.com (Matthew J. Hughes)
Date: Fri, 19 Oct 2007 15:20:45 -0700
Subject: [cisco-voip] Fwd:  Push IP phone service to phone
In-Reply-To: <f84a38d30710191440s4d0a66b8sda594dace1429f57@mail.gmail.com>
References: <ea211ea30710191011l43fff97am9c082cec23fd8a0e@mail.gmail.com>
	<f84a38d30710191440s4d0a66b8sda594dace1429f57@mail.gmail.com>
Message-ID: <ea211ea30710191520q3a875f58t4a1cdca4e39ff06b@mail.gmail.com>

The University  this is for currently maintains manual records down to the
room level, so anything less seems like a step back to them (conjures images
of stroke victims grasping for life behind desk, while emergency workers
search in vain)   All phones have a DID so the PD gets a call back number
automatically with that, the main thrust for us with CER was to have fine
grain ERLs at a room  by room level.(which isn't how it was designed to work
by my understanding)   Personally a per room ERL seems a bit much, but rules
are getting tighter as to how accurate (within 100m) your info is.  Also the
Police/Dispatch are on campus so they had a say in the decision.    I would
love to map ports on switches with that granularity to ERLs but that dosent
seem pratical. The best I could figure is to have users declair thier ERL
everytime the phone boots.


---------- Forwarded message ----------
From: Scott Voll <svoll.voip at gmail.com>
Date: Oct 19, 2007 2:40 PM
Subject: Re: [cisco-voip] Push IP phone service to phone
To: "Matthew J. Hughes" <mattjhughes at gmail.com>
Cc: cisco-voip at puck.nether.net


first off..... Welcome to the Puck list!

Are you running CER downt to the room location?  Is this a policy?  Are you
going to have a ERL / ELIN per room?  seems a little over kill.  or is it
just for the response personel?

Scott


 On 10/19/07, Matthew J. Hughes <mattjhughes at gmail.com> wrote:

>  First post here, new to the group :)
>
> I am attempting to  gather the physical location of phones on campus for
> use with emergency responder.  I want the phone user to be prompted with a
> IP Phone Service on their 79xx phones, that asks them to verify their
> physical location (Room Number, Building Name)  I don't want this to be
> burdensome so I only want it to come up every time a phone reboots (It wont
> move locations unless its unplugged).   To accomplish this  I ran into these
> roadblocks
>
> 1.  IDLE URL will certainly force them to see it, but will bug them far to
> much. (although if i set the idle timer long enough it could come up only
> once a day.)
> 2. I can push messages to their screen, but don't think i can push a XML
> service.
>
> I am thinking I would have to use AXL/SOAP to add/remove idle URL,  OR
> somehow as part of the boot process force them to a XML service.
>
> any thoughts?  I am running CCM 6.0 and CER
>
> Thanks in advance,
> Matt
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jonvoip at gmail.com  Fri Oct 19 20:05:25 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 19 Oct 2007 19:05:25 -0500
Subject: [cisco-voip] CUE Versions and upgrades...
In-Reply-To: <561424.5609.qm@web57215.mail.re3.yahoo.com>
References: <5d093f9a0710191104l22475af8v4b0b9882f3f10601@mail.gmail.com>
	<561424.5609.qm@web57215.mail.re3.yahoo.com>
Message-ID: <5d093f9a0710191705x53551a33k9ee15d7d1de1f057@mail.gmail.com>

Cool, I will go with 3.0.1


J

On 10/19/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> Jonathan,
>
> I am using 3.0.1 at three customer offices and it is
> working quite handily. There was one instance where a
> user would receive a voicemail and not be able to hear
> it but a reboot of CUE solved it quickly. I have not
> heard of any other major issues since then.
>
> Any version older than 2.3.4 MUST be upgraded to at
> least 2.3.4 for DST patches.
>
> Paul
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
>
> > I am doing my first CUE upgrade next week...
> > customer is hyper
> > high-profile (e.g. my CEO will hear about it if this
> > goes bad)...
> >
> > Anyway, I am looking for a stable version of CUE
> > software to upgrade
> > to, they are probably at an older version of 2.3
> > (haven't gotten all
> > the info yet)...
> >
> > Was curious:
> >
> > Which version of CUE software is stable? I see that
> > 3.0.2 was released
> > four days ago, so let's stay away from that... but
> > is 3.0.1 stable?
> >
> > Any hilarious bugs (like CME 3.3 where random phones
> > would just ring
> > for no reason) I should be paranoid about?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>

From wsisk at cisco.com  Fri Oct 19 21:32:12 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 19 Oct 2007 21:32:12 -0400
Subject: [cisco-voip] Push IP phone service to phone
In-Reply-To: <ea211ea30710191011l43fff97am9c082cec23fd8a0e@mail.gmail.com>
References: <ea211ea30710191011l43fff97am9c082cec23fd8a0e@mail.gmail.com>
Message-ID: <47195A9C.6020305@cisco.com>

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From jozjan at cnl.tuke.sk  Sat Oct 20 06:38:56 2007
From: jozjan at cnl.tuke.sk (Jozef Janitor)
Date: Sat, 20 Oct 2007 13:38:56 +0300
Subject: [cisco-voip] CCM 5 or 6 or vmware player
In-Reply-To: <475fda330710181028p4ae6e6c2saaa84c5bf2b0de30@mail.gmail.com>
References: <475fda330710181028p4ae6e6c2saaa84c5bf2b0de30@mail.gmail.com>
Message-ID: <010c01c81305$6bcc3b50$4364b1f0$@tuke.sk>

Hello,

 

I have tried it on Vmware Workstation on my laptop but it was damn slow :(

 

All the best,

   Jozef Janitor

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick
Sent: Thursday, October 18, 2007 8:28 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM 5 or 6 or vmware player

 

I know that ccm 5 has been well documented about installing it on vmware esc
and gsx server but hasd anyone installed it on vmware player on a pc?

 

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From cshaffer at gmail.com  Sat Oct 20 07:40:15 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Sat, 20 Oct 2007 07:40:15 -0400
Subject: [cisco-voip] Home user
In-Reply-To: <f84a38d30710171105n6977175by2725263a1bc4afc@mail.gmail.com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>	<AB0A35F2486F81468971AA68558BE719073BF05C@smieln01winm01.micoanet.com>	<6A059BC4-F184-4563-B117-1D28BCDAFFA9@jerkys.org>
	<f84a38d30710171105n6977175by2725263a1bc4afc@mail.gmail.com>
Message-ID: <002901c8130d$fc836030$f58a2090$@com>

I tried searching on this on cisco.com. Do you have a part number or a more
direct name that may help? Is this something that is production ready or is
it still beta quality?

 

Thanks

 

Curt

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Wednesday, October 17, 2007 2:05 PM
To: Jerky
Cc: Linsemier, Matthew; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Home user

 

no Cisco CPE required.

 

ip phone

  |

internet connection

  |

  --------------------------- phone proxy

  |                               |

  |                                |

FW / router                  |

  |                                |

internal network ------ voice network

 

basically you save the money of a cisco CPE by getting the phone proxy.  let
the end users VPN in with the client for data purposes or use Citrix to get
around VPN all together.

 

the phone proxy has a north / South interface so the only thing going
through it is the authenticated voice traffic.

 

hope that's understandable.

 

scott

 



 

On 10/17/07, Jerky <lists at jerkys.org> wrote: 

so it would be more like this: 


 

Cisco 871

     |

DSL CABLE

     |

Internet

     |

T1 Connection (Serial0/0/0)

                    |

   _____ 3800 _____ 

  |                                  |

ethernet 0/0          ethernet 0/1

|                                    |

PIX/ASA                    3800 (Cisco 871 VPN's terminate here)

|                                    |

LAN(computers)       LAN (Voice)


 


 

Homefully my crude diagram makes sense. Do your home users have access to
any data on the computer network side. Or is the 87x VPNs solely for getting
to the voice network If users access things on the "computer" side would you
have a separate tunnel setup just for that? 


 

Thanks for so much helping enlighten me. It's been very helpful.


 

jeff


 


 


 

On Oct 17, 2007, at 10:19 AM, Linsemier, Matthew wrote:





In our environment we utilize PIX firewalls (still have to upgrade to ASA's)
to handle our firewall needs and then use the 3800 series router just to
terminate the DMVPN home users.  They are deployed in parallel and sit
behind a perimeter screening router (another 3800 series router).  We shied
away from using the PIX for the simple fact that while it would preserve QoS
markings, we couldn't do any remarking or shaping in the device.  Maybe this
has changed in the ASA, but I don't think you have the control like you do
in IOS (such as qos pre-classify, shaping, policing,  etc.).  Depending on
how many tunnels you plan on using, you could use a router much smaller than
a 3800 series to terminate the end nodes. 

 

On the home user end we have the Cisco 871/877 routers configured to support
wired and wireless connections using three VLANS.  We have a VLAN configured
for corporate connectivity, one VLAN configured as a voice VLAN, and then a
VLAN configured for untrusted traffic.  One Ethernet port on the router
provides connectivity to the corporate and voice VLANS, while the remaining
three are configured as untrusted.  Similarly with Wireless, we extend PEAP
authentication from the headquarters and authenticate users to the corporate
VLAN, and use a WPA-PSK to secure the untrusted connections.  This way the
users plug in their phone, then their laptop/docking station to port 0, and
any other home devices can be connected to port 1-3 or use the wireless
WPA-PSK network and be logically segregated (using ACL's) from any data on
the corporate network.  This way we can also control QoS and mark down all
traffic that enters the router from the untrusted network.  So when said
employees son or daughter starts downing a 2 gig torrent from a home PC,
they don't kill the voice or impact the corporate workflow.  Eventually we
will be implementing 802.1x on the corporate port for additional security,
but have had mixed results of getting it to work with Windows XP.


Hope this helps. 

 

Matt

 

 

From: Jerky [mailto:lists at jerkys.org] 
Sent: Tuesday, October 16, 2007 6:32 PM 
To: Linsemier, Matthew
Cc: Curt Shaffer; cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] Home user

 

This has been kicked around for a while since we moved to CallManager but
not much thought has been given to it. I'm trying to understand how your
hardware is setup. How would it look, similar to one of these? 

 

87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845
<--Ethernet--> LAN

 

or

 

87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <---> ASA
or PIX Firewall <--Ethernet--> LAN

 

Is the 3800 used for all your firewalling needs in lieu of something like an
ASA or PIX? Sonicwall's are currently in place and haven't worked very well
for the remote users it was tested with. The Sonicwalls we have don't have
anything similar to what the 871's seem to have in regards to vlans and
packet tagging. We would probably kick the Sonicwalls out if something else
would work better. 

 

jeff

 

On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:

 

We currently have about 40 production remote home teleworkers that have been
deployed using Cisco 871/877 wireless routers and a 7960 phones.  We are
using a Cisco 3845 series router at the head-end so that we can control QoS
tagging on the egress / ingress points of both sides of the VPN tunnel.  We
are using a phase 2 DMVPN solution dual-homed to two sites to provide secure
redundant connectivity. 

  

It took me a bit to tweak my router configurations (I started on Cisco
831/837 routers) to get the results that we wanted, but all and all our
users are happy.  There is the occasional jitter and packet loss (it is the
Internet mind you) but g.729 is working quite well coupled with business
cable and DSL services.

  

If you have any other questions, feel free to ask. 

  

Matt 

  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Curt Shaffer
Sent: Monday, October 15, 2007 6:37 PM
To: cisco-voip at puck.nether.net 
Subject: [cisco-voip] Home user

 

I was wondering want everyone out there is using for the situation where you
have someone on your CCM or CCME that has 1 phone at a home office.
Something tells me an ASA is overkill and I haven't found solid information
that any of the 87x routers support tagging QoS of packets going through the
VPN tunnel. We would obviously like to have QoS in place even though it's
not respected at their ISP just to make sure the VPN/Voice packets are
leaving their routers first as a best effort to get some quality. 

 

Thanks

 

 

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Neither this information block, the typed name of the sender, nor anything
else in this message is intended to constitute an electronic signature
unless a specific statement to the contrary is included in this message. If
you have received this communication in error, please immediately contact me
and delete this communication from your computer. Thank you. 

  _____  

_______________________________________________

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 <mailto:cisco-voip at puck.nether.net> cisco-voip at puck.nether.net 

 <https://puck.nether.net/mailman/listinfo/cisco-voip>
https://puck.nether.net/mailman/listinfo/cisco-voip 

 

  _____  

CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may be protected
by one or more legal privileges. It is intended solely for the use of the
addressee identified above. If you are not the intended recipient, any use,
disclosure, copying or distribution of this communication is UNAUTHORIZED.
Neither this information block, the typed name of the sender, nor anything
else in this message is intended to constitute an electronic signature
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you have received this communication in error, please immediately contact me
and delete this communication from your computer. Thank you. 

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_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip 

 

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From tednugent73 at gmail.com  Sat Oct 20 10:03:54 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Sat, 20 Oct 2007 10:03:54 -0400
Subject: [cisco-voip] CCM 5 or 6 or vmware player
In-Reply-To: <010c01c81305$6bcc3b50$4364b1f0$@tuke.sk>
References: <475fda330710181028p4ae6e6c2saaa84c5bf2b0de30@mail.gmail.com>
	<010c01c81305$6bcc3b50$4364b1f0$@tuke.sk>
Message-ID: <22e6380b0710200703h4e4ce270k2f60d73c9bdb9cc0@mail.gmail.com>

I'm running CUCM-BE 6.0 on my laptop, IBM T60 3GB of RAM, it does ok.
ccmadmin is slow but useable. Unfortnately thats workstation also, I've not
tried Player yets...




On 10/20/07, Jozef Janitor <jozjan at cnl.tuke.sk> wrote:
>
>  Hello,
>
>
>
> I have tried it on Vmware Workstation on my laptop but it was damn slow :(
>
>
>
> All the best,
>
>    Jozef Janitor
>
>
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Nick
> *Sent:* Thursday, October 18, 2007 8:28 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] CCM 5 or 6 or vmware player
>
>
>
> I know that ccm 5 has been well documented about installing it on vmware
> esc and gsx server but hasd anyone installed it on vmware player on a pc?
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Andrew.Riley at getronics.com  Sat Oct 20 16:50:42 2007
From: Andrew.Riley at getronics.com (Riley, Andrew)
Date: Sun, 21 Oct 2007 06:50:42 +1000
Subject: [cisco-voip] CCM 5 or 6 or vmware player
Message-ID: <ec7401c8135a$ecb53ae5$3c84a8c0@europe.unity>

It works well under vmware server on ubuntu. Min 1.5gig ram to the guest os.

-----Original Message-----
From: "Ted Nugent" <tednugent73 at gmail.com>
To: "Jozef Janitor" <jozjan at cnl.tuke.sk>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Sent: 21/10/07 00:05
Subject: Re: [cisco-voip] CCM 5 or 6 or vmware player

I'm running CUCM-BE 6.0 on my laptop, IBM T60 3GB of RAM, it does ok.
ccmadmin is slow but useable. Unfortnately thats workstation also, I've not
tried Player yets...




On 10/20/07, Jozef Janitor <jozjan at cnl.tuke.sk> wrote:
>
>  Hello,
>
>
>
> I have tried it on Vmware Workstation on my laptop but it was damn slow :(
>
>
>
> All the best,
>
>    Jozef Janitor
>
>
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Nick
> *Sent:* Thursday, October 18, 2007 8:28 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] CCM 5 or 6 or vmware player
>
>
>
> I know that ccm 5 has been well documented about installing it on vmware
> esc and gsx server but hasd anyone installed it on vmware player on a pc?
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From lists at jerkys.org  Sun Oct 21 04:29:19 2007
From: lists at jerkys.org (Jerky)
Date: Sun, 21 Oct 2007 03:29:19 -0500
Subject: [cisco-voip] Home user
In-Reply-To: <002901c8130d$fc836030$f58a2090$@com>
References: <000601c80f7b$ec317f40$c4947dc0$@com>	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>	<AB0A35F2486F81468971AA68558BE719073BF05C@smieln01winm01.micoanet.com>	<6A059BC4-F184-4563-B117-1D28BCDAFFA9@jerkys.org>
	<f84a38d30710171105n6977175by2725263a1bc4afc@mail.gmail.com>
	<002901c8130d$fc836030$f58a2090$@com>
Message-ID: <680945A7-7C6A-4EDC-B180-5D9116E86CE8@jerkys.org>

Think this is what you are looking for:

http://www.cisco.com/en/US/products/ps7057/ 
products_data_sheet0900aecd80546906.html

jeff

On Oct 20, 2007, at 6:40 AM, Curt Shaffer wrote:

> I tried searching on this on cisco.com. Do you have a part number  
> or a more direct name that may help? Is this something that is  
> production ready or is it still beta quality?
>
>
>
> Thanks
>
>
>
> Curt
>
>
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net] On Behalf Of Scott Voll
> Sent: Wednesday, October 17, 2007 2:05 PM
> To: Jerky
> Cc: Linsemier, Matthew; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Home user
>
>
>
> no Cisco CPE required.
>
>
>
> ip phone
>
>   |
>
> internet connection
>
>   |
>
>   --------------------------- phone proxy
>
>   |                               |
>
>   |                                |
>
> FW / router                  |
>
>   |                                |
>
> internal network ------ voice network
>
>
>
> basically you save the money of a cisco CPE by getting the phone  
> proxy.  let the end users VPN in with the client for data purposes  
> or use Citrix to get around VPN all together.
>
>
>
> the phone proxy has a north / South interface so the only thing  
> going through it is the authenticated voice traffic.
>
>
>
> hope that's understandable.
>
>
>
> scott
>
>
>
>
>
>
>
> On 10/17/07, Jerky <lists at jerkys.org> wrote:
>
> so it would be more like this:
>
>
>
>
> Cisco 871
>
>      |
>
> DSL CABLE
>
>      |
>
> Internet
>
>      |
>
> T1 Connection (Serial0/0/0)
>
>                     |
>
>    _____ 3800 _____
>
>   |                                  |
>
> ethernet 0/0          ethernet 0/1
>
> |                                    |
>
> PIX/ASA                    3800 (Cisco 871 VPN's terminate here)
>
> |                                    |
>
> LAN(computers)       LAN (Voice)
>
>
>
>
>
>
>
> Homefully my crude diagram makes sense. Do your home users have  
> access to any data on the computer network side. Or is the 87x VPNs  
> solely for getting to the voice network If users access things on  
> the "computer" side would you have a separate tunnel setup just for  
> that?
>
>
>
>
> Thanks for so much helping enlighten me. It's been very helpful.
>
>
>
>
> jeff
>
>
>
>
>
>
>
>
>
>
> On Oct 17, 2007, at 10:19 AM, Linsemier, Matthew wrote:
>
>
>
>
> In our environment we utilize PIX firewalls (still have to upgrade  
> to ASA's) to handle our firewall needs and then use the 3800 series  
> router just to terminate the DMVPN home users.  They are deployed  
> in parallel and sit behind a perimeter screening router (another  
> 3800 series router).  We shied away from using the PIX for the  
> simple fact that while it would preserve QoS markings, we couldn't  
> do any remarking or shaping in the device.  Maybe this has changed  
> in the ASA, but I don't think you have the control like you do in  
> IOS (such as qos pre-classify, shaping, policing,  etc.).   
> Depending on how many tunnels you plan on using, you could use a  
> router much smaller than a 3800 series to terminate the end nodes.
>
>
>
> On the home user end we have the Cisco 871/877 routers configured  
> to support wired and wireless connections using three VLANS.  We  
> have a VLAN configured for corporate connectivity, one VLAN  
> configured as a voice VLAN, and then a VLAN configured for  
> untrusted traffic.  One Ethernet port on the router provides  
> connectivity to the corporate and voice VLANS, while the remaining  
> three are configured as untrusted.  Similarly with Wireless, we  
> extend PEAP authentication from the headquarters and authenticate  
> users to the corporate VLAN, and use a WPA-PSK to secure the  
> untrusted connections.  This way the users plug in their phone,  
> then their laptop/docking station to port 0, and any other home  
> devices can be connected to port 1-3 or use the wireless WPA-PSK  
> network and be logically segregated (using ACL's) from any data on  
> the corporate network.  This way we can also control QoS and mark  
> down all traffic that enters the router from the untrusted  
> network.  So when said employees son or daughter starts downing a 2  
> gig torrent from a home PC, they don't kill the voice or impact the  
> corporate workflow.  Eventually we will be implementing 802.1x on  
> the corporate port for additional security, but have had mixed  
> results of getting it to work with Windows XP.
>
>
> Hope this helps.
>
>
>
> Matt
>
>
>
>
>
> From: Jerky [mailto:lists at jerkys.org]
> Sent: Tuesday, October 16, 2007 6:32 PM
> To: Linsemier, Matthew
> Cc: Curt Shaffer; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Home user
>
>
>
> This has been kicked around for a while since we moved to  
> CallManager but not much thought has been given to it. I'm trying  
> to understand how your hardware is setup. How would it look,  
> similar to one of these?
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
> <--Ethernet--> LAN
>
>
>
> or
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845  
> <---> ASA or PIX Firewall <--Ethernet--> LAN
>
>
>
> Is the 3800 used for all your firewalling needs in lieu of  
> something like an ASA or PIX? Sonicwall's are currently in place  
> and haven't worked very well for the remote users it was tested  
> with. The Sonicwalls we have don't have anything similar to what  
> the 871's seem to have in regards to vlans and packet tagging. We  
> would probably kick the Sonicwalls out if something else would work  
> better.
>
>
>
> jeff
>
>
>
> On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
>
>
>
> We currently have about 40 production remote home teleworkers that  
> have been deployed using Cisco 871/877 wireless routers and a 7960  
> phones.  We are using a Cisco 3845 series router at the head-end so  
> that we can control QoS tagging on the egress / ingress points of  
> both sides of the VPN tunnel.  We are using a phase 2 DMVPN  
> solution dual-homed to two sites to provide secure redundant  
> connectivity.
>
>
>
> It took me a bit to tweak my router configurations (I started on  
> Cisco 831/837 routers) to get the results that we wanted, but all  
> and all our users are happy.  There is the occasional jitter and  
> packet loss (it is the Internet mind you) but g.729 is working  
> quite well coupled with business cable and DSL services.
>
>
>
> If you have any other questions, feel free to ask.
>
>
>
> Matt
>
>
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net ] On Behalf Of Curt Shaffer
> Sent: Monday, October 15, 2007 6:37 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Home user
>
>
>
> I was wondering want everyone out there is using for the situation  
> where you have someone on your CCM or CCME that has 1 phone at a  
> home office. Something tells me an ASA is overkill and I haven't  
> found solid information that any of the 87x routers support tagging  
> QoS of packets going through the VPN tunnel. We would obviously  
> like to have QoS in place even though it's not respected at their  
> ISP just to make sure the VPN/Voice packets are leaving their  
> routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>
> CONFIDENTIALITY STATEMENT
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> protected by one or more legal privileges. It is intended solely  
> for the use of the addressee identified above. If you are not the  
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>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be  
> protected by one or more legal privileges. It is intended solely  
> for the use of the addressee identified above. If you are not the  
> intended recipient, any use, disclosure, copying or distribution of  
> this communication is UNAUTHORIZED. Neither this information block,  
> the typed name of the sender, nor anything else in this message is  
> intended to constitute an electronic signature unless a specific  
> statement to the contrary is included in this message. If you have  
> received this communication in error, please immediately contact me  
> and delete this communication from your computer. Thank you.
>
>
>
>
>
>
>
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>

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From erickbee at gmail.com  Sun Oct 21 23:22:26 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Sun, 21 Oct 2007 22:22:26 -0500
Subject: [cisco-voip] MGCP FXO port and sending no digits in CCM 5.x
In-Reply-To: <f4445faf0710040903l1b0027dcn4625203e478a85ad@mail.gmail.com>
References: <f4445faf0710040903l1b0027dcn4625203e478a85ad@mail.gmail.com>
Message-ID: <f4445faf0710212022t5b4c5b29o3ce9f82a175fe2c4@mail.gmail.com>

FYI,

This issue appears to be bug CSCsi19351.

On 10/4/07, Erick Bergquist <erickbee at gmail.com> wrote:
> Is there another trick to doing this in CCM 5.x then 4.x? In 4.x I
> would set up the route pattern as say 1234. and do a PreDot digit
> strip. I tried this in CCM 5.x for a MGCP FXO port going to a paging
> system and it did not work, said unknown number and gave reorder tone
> when route pattern was 1234.  but 1234 works and connects but sends
> 1234 out port which doesn't work with the paging system.  I made it
> H323 but was wondering how to just have a FXO port go offhook and send
> no digits with MGCP in 5.x.
>

From pedro.nogueira at sicnet.pt  Mon Oct 22 05:00:45 2007
From: pedro.nogueira at sicnet.pt (Pedro Nogueira)
Date: Mon, 22 Oct 2007 10:00:45 +0100
Subject: [cisco-voip] how to change Span to PC Port configuration in a 7941
	phone
Message-ID: <A805E226FF74D74E90D5BE19D664A9A0137F26@neptuno.atminformatica.pt>

Hello,

i know where to change it in the web pages, my problem is that it dont appear in the admin conf page (CCM), neither in the   
phone admin page, in the 7911 model it appears, but in the 7941 or 7961 model, dont.
Any sugestions or ideas ?  

Pedro Nogueira
Dep. T?cnico
Sicnet


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From patrick.diener at gmail.com  Mon Oct 22 08:57:05 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Mon, 22 Oct 2007 14:57:05 +0200
Subject: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
In-Reply-To: <FA233791F398B848B978464507739E8FDA77CE@EUBEBRUSVEX1.eu.didata.local>
References: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>
	<4718E352.5070004@cisco.com>
	<FA233791F398B848B978464507739E8FDA77CE@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <c0deb5b80710220557h3b5bd7d2w67c4eb3d70f43212@mail.gmail.com>

thanks a lot for all the good responses, you really helped a lot!!!

considering the new information I changed my step by step process to
the following:

Step 1: Gather necessary stuff
- contact cisco licensing to get new .lic files for the replacement publisher hw
- make sure the USB tokens needed to change the CTL File are accessible

Step 2: Install CUCM on replacement hw
- make sure to use same hostname, ip address and passwords as the
server you like to replace
- add .lic files acquired in Step 1
- check weather sufficient DLUs and node licenses are now available

Step 3: DRS
- recover the latest backup using the disaster recovery system

Step 4: Reinstall Subscribers
- a fresh install is necessary for all subscribers in the cluster
- after the reinstall use DRS to recovery lost data on the Sub. (I
assume activated services, enterprise parameters, etc.)

Step 5: Update CTL file and LSCs
- use CTL Client and USB tokens to update the CTL File with the new
Certificates of all CallManager nodes, the TFTP Servers and CAPF
- if used, update the LSCs (cause CAPF cert. has been changed(??))

=> as I see it telephony service will be interrupted during sub.
rebuild. if security is configured tel. service will not be back up
until step 5 is finished...

all this is also covered in the doc wes posted!

Regards
Patrick


On 10/19/07, Ruben Montes (Europe) <Ruben.Montes at eu.didata.com> wrote:
> Hello,
>
> I had to do a Publisher replacement from a MCS7825 to a MCS7835 last
> week. We had two additional subscribers. I followed step by step the
> guide Wes has mentioned and all went fine (the whole change took us 9
> hours).
>
> The main problem is that you MUST reinstall also all the subscribers
> from the cluster: this was confirmed by TAC and by the customer's SE
> cisco engineer.
>
> If you have any additional question about this just drop me an email...
>
> Regards,
>
> Ruben
>
> -----Mensaje original-----
> De: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] En nombre de Wes Sisk
> Enviado el: viernes, 19 de octubre de 2007 19:03
> Para: Patrick Diener
> CC: cisco-voip at puck.nether.net
> Asunto: Re: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
>
> Hi Patrick,
>
> Good stuff.  Most of this is finally covered in CM513 docs:
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_installation_g
> uide09186a00808be1dc.html
>
>
>
> Patrick Diener wrote:
> > I'm trying to put together a rough step by step procedure to do a
> > disaster recovery for a cucm 5.1(2) cluster and there are some stuff I
> > can't figure out how it should/will work...
> >
> > I based the steps on the DRS Admin Guide for Version 5.0 (5.1 not out
> yet?!?)
> >
> > Important: I try to do a restore on a new server hw => new first node
> MAC
> >
> > Step 1:
> > perform fresh install of cucm sw
> > !! set same hostname as before !!
> >
> ws: same hostname, ip, and passwords
> > Step 2:
> > perform restore using DRS and restart server (worked like a charm in
> the lab...)
> >
> > form now on I'm stuck...
> >
> ws: you restored the pub only, correct? and rebooted?  after you
> reinstall sub will need to DRS restore it as well.
> > Step 3:
> > Licensing?!?
> > all activated services will not start cause the .lic files are not
> > valid for the new server hw (at least thats what happened in the
> > lab..)
> > => will TAC send me a new .lic equal to the old one?
> > => do I need to contact licensing at cisco.com?
> > => any experience how long it will take to get the new .lic files?
> >
> ws: yep, new license files if replacing the pub. covered in the doc,
> licensing at cisco.com
> > Step 4:
> > move some phones to the restored publisher (assuming callmanager
> > service is running) and test all features needed
> > => will the phones register if security has been configured? I assume
> > the new callmanager, tftp and capf certificates will not match the
> > certificates stored in the phones ctl file (right?) how do I get out
> > of this mess without touching each and every phone to delete the ctl
> > file or perform a factory reset?
> >
> ws: this comes later actually. check the doc, you will create new CTL
> files using same USB token.
> > Step 5:
> > restore sub (or subs)
> >
> step5: reinstall subs, then restore subs, then perform post-replace
> checklist.
> >
> > all comments, experiences and remarks are appreciated!
> >
> > cheers
> > Patrick
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From wsisk at cisco.com  Mon Oct 22 09:58:01 2007
From: wsisk at cisco.com (Wesley Sisk)
Date: Mon, 22 Oct 2007 09:58:01 -0400 (EDT)
Subject: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
In-Reply-To: <c0deb5b80710220557h3b5bd7d2w67c4eb3d70f43212@mail.gmail.com>
References: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com>
	<4718E352.5070004@cisco.com>
	<FA233791F398B848B978464507739E8FDA77CE@EUBEBRUSVEX1.eu.didata.local>
	<c0deb5b80710220557h3b5bd7d2w67c4eb3d70f43212@mail.gmail.com>
Message-ID: <Pine.GSO.4.64.0710220956390.28870@clubhouse-12.cisco.com>

Patrick,

Nice compilation.  You users will have dialtone after subscriber install. 
Technically you could wait until next maint window to redo 
security/capf/tls.  existing phones will register to the cluster. 
However, you will not be able to add new phones.


From ealeatherman at gmail.com  Mon Oct 22 10:48:17 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 22 Oct 2007 10:48:17 -0400
Subject: [cisco-voip] CM 5 Upgrade - My Fast Dials
Message-ID: <94a1afde0710220748s41161048va702d7dda2fa88a8@mail.gmail.com>

Hi folks,

Hitting one last snag (I hope the last one) in my upgrade plans for ccm 4.1to
5.1. Our problem was the change to My Fast Dials outlined here:
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_tech_note09186a008088d0dc.shtml
Fast dials and address book now live in the Directories button. This in and
of itself isn't a problem, but it prompts for userID and PIN before you can
use it.

Requiring our users to log into the phone in order to use the fast dial
service isn't acceptable in our environment. If they had to do that, they
would A) not use the feature any longer, and B) complain, a lot. Up until
now we haven't provided them access to the CCMUser page, we've just
provisioned the service for them so in most cases user's wont even know what
their PIN is.

Solution was a different service URL that another cisco customer told us
about that functions very similar to the old service, allowing you to add
the userID and PIN as parameters and pass them along inside the URL.
Unfortunately, the parameter name for user ID is different. When I go into
the IP Phone services in CM5 and change the service URL, I also must rename
that parameter, which clears out the user ID value in all the subscriptions.

So now my task is to find a way to re-insert all those user ID values onto
the correct phones. I have this information available in a CSV file, but I
can't figure out how to make BAT read this in and update the existing
settings. The only thing I can do is specify device names in a file and then
fill in the update phone web page form.. however that only allows you to add
new services rather than update existing ones. I would just re-add the
service, but the web page form doesn't allow for arbitrary parameter values
per phone, in other words I'd be specifying the same UserID and PIN to all
the phones.

Is there no way to update via a csv file or am I missing it somewhere? I'm
not familiar at all with AXL but is that going to be my only option here?

Maybe I'm making this harder than it needs to be.

Thanks!
Ed


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From voicenoob at gmail.com  Mon Oct 22 11:01:24 2007
From: voicenoob at gmail.com (Voice Noob)
Date: Mon, 22 Oct 2007 10:01:24 -0500
Subject: [cisco-voip] Sip Dialer with Outlook
Message-ID: <535625f70710220801t7e807dbew82118f0be7e4b2a@mail.gmail.com>

I was wondering if anyone had any good experiences with a free SIP
dialer for Outlook? We have used the Cisco TSP in the past and
sometimes it works but most of the time it does not. If anyone has any
opinions about this please pass them on.
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From manoj.kalpage at gmail.com  Mon Oct 22 11:52:54 2007
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Tue, 23 Oct 2007 00:52:54 +0900
Subject: [cisco-voip] CCM 5 or 6 or vmware player
In-Reply-To: <475fda330710181028p4ae6e6c2saaa84c5bf2b0de30@mail.gmail.com>
References: <475fda330710181028p4ae6e6c2saaa84c5bf2b0de30@mail.gmail.com>
Message-ID: <bbe09c1d0710220852u2f8f20a0w4a4597cc2d4cbe29@mail.gmail.com>

Nick, We mention here that ccm5 installation vmware 5 is well
documented. Could you send me any link which has step by step
information.

Best regrds,

On 10/19/07, Nick <csvoip at googlemail.com> wrote:
> I know that ccm 5 has been well documented about installing it on vmware esc
> and gsx server but hasd anyone installed it on vmware player on a pc?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From dustin.s.fowler at gmail.com  Mon Oct 22 12:43:49 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Mon, 22 Oct 2007 12:43:49 -0400
Subject: [cisco-voip] CCM 6.0 Question
Message-ID: <471cd346.111f400a.74aa.fffff1a2@mx.google.com>

 

All,

 

Does anyone know of a setting to present a tone to allow an internal caller
to be aware that the person they are trying to call that the party is on
another call.

 

Thank you,

 

Dustin

 

 

 

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From Ruben.Montes at eu.didata.com  Mon Oct 22 12:42:26 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Mon, 22 Oct 2007 18:42:26 +0200
Subject: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)
References: <c0deb5b80710190527g1e547cfdl49fecddb8cc23b19@mail.gmail.com><4718E352.5070004@cisco.com><FA233791F398B848B978464507739E8FDA77CE@EUBEBRUSVEX1.eu.didata.local><c0deb5b80710220557h3b5bd7d2w67c4eb3d70f43212@mail.gmail.com>
	<Pine.GSO.4.64.0710220956390.28870@clubhouse-12.cisco.com>
Message-ID: <FA233791F398B848B978464507739E8FDA7B69@EUBEBRUSVEX1.eu.didata.local>

Hello,

Just my 5 cents... In my case, it was necessary to update the CTL in
order to have all the phones registered again in the cluster...

Regards,

Ruben

-----Mensaje original-----
De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Wesley Sisk
Enviado el: lunes, 22 de octubre de 2007 15:58
Para: Patrick Diener
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Disaster Recovery Procedure CUCM 5.1(2)

Patrick,

Nice compilation.  You users will have dialtone after subscriber
install. 
Technically you could wait until next maint window to redo 
security/capf/tls.  existing phones will register to the cluster. 
However, you will not be able to add new phones.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From dustin.s.fowler at gmail.com  Mon Oct 22 13:57:14 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Mon, 22 Oct 2007 13:57:14 -0400
Subject: [cisco-voip]  CCM 6.0 Question
Message-ID: <471ce47a.0e1f400a.24b8.fffffc9e@mx.google.com>

 

That is great. LMAO. Speaking more of a call waiting tone instead. Telling
the caller that the called is on the phone. Maybe even a screen
notification. 

 

From: Voice Noob [mailto:voicenoob at gmail.com] 
Sent: Monday, October 22, 2007 12:58 PM
To: Dustin S Fowler
Subject: Re: [cisco-voip] CCM 6.0 Question

 

Busy tone. 

:)

 

On 10/22/07, Dustin S Fowler <dustin.s.fowler at gmail.com> wrote: 

 

All,

 

Does anyone know of a setting to present a tone to allow an internal caller
to be aware that the person they are trying to call that the party is on
another call.

 

Thank you,

 

Dustin

 

 

 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net  <mailto:cisco-voip at puck.nether.net> 
https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From jonvoip at gmail.com  Mon Oct 22 14:35:01 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 22 Oct 2007 13:35:01 -0500
Subject: [cisco-voip] Files not backed up by BARS
Message-ID: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>

Just checking here, besides TFTP path stuff (ringtones, backgrounds,
etc.) what else is not backed up by BARS 4.0(8)

We have a dying publisher and are backing up data before the rebuild
and want to make sure we have everything...



Jonathan

From jgrace at adcapnet.com  Mon Oct 22 14:41:55 2007
From: jgrace at adcapnet.com (James Grace)
Date: Mon, 22 Oct 2007 14:41:55 -0400
Subject: [cisco-voip] cisco 7921  wpa setting
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E8C2@mail3.adcapnetsvc.local>

?
Is I possible to set up wpa on an 7921 wireless phone.  I heard it was.  But i dont see how.  
 
James Grace 
 
<http://www.adcapnet.com/>  
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
recipient is prohibited. If you received this message in error, please contact the sender and delete the
material from all computers."

This original email was sent to the internet for delivery at 22 Oct 2007 14:42:13 -0400
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From MillerS at DicksteinShapiro.COM  Mon Oct 22 14:49:14 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 22 Oct 2007 14:49:14 -0400
Subject: [cisco-voip] Files not backed up by BARS
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
Message-ID: <421CAA1A5300B84A87D200D70998578263DF23@DCEX2.DSMO.COM>

How old is your Pub? 


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Monday, October 22, 2007 2:35 PM
To: cisco voip list
Subject: [cisco-voip] Files not backed up by BARS

Just checking here, besides TFTP path stuff (ringtones, backgrounds,
etc.) what else is not backed up by BARS 4.0(8)

We have a dying publisher and are backing up data before the rebuild
and want to make sure we have everything...



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


--------------------------------------------------------
This e-mail message and any attached files are confidential
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named above. This communication may contain material 
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to the intended recipient, you have received this 
communication in error, and any review, use, dissemination, 
forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
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==============================================================================


From lelio at uoguelph.ca  Mon Oct 22 14:51:40 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 22 Oct 2007 14:51:40 -0400
Subject: [cisco-voip] Files not backed up by BARS
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
Message-ID: <02fd01c814dc$950a4ce0$56196883@cfs.uoguelph.ca>

why wouldnt' the tftp stuff be backed up?

----- Original Message ----- 
From: "Jonathan Charles" <jonvoip at gmail.com>
To: "cisco voip list" <cisco-voip at puck.nether.net>
Sent: Monday, October 22, 2007 2:35 PM
Subject: [cisco-voip] Files not backed up by BARS


> Just checking here, besides TFTP path stuff (ringtones, backgrounds,
> etc.) what else is not backed up by BARS 4.0(8)
> 
> We have a dying publisher and are backing up data before the rebuild
> and want to make sure we have everything...
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From CarlosOrtiz at bayviewfinancial.com  Mon Oct 22 14:52:43 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Mon, 22 Oct 2007 14:52:43 -0400
Subject: [cisco-voip] Call static
Message-ID: <OF1B7BC7BE.D9BF5343-ON8525737C.006730BC-8525737C.0067B456@bftg.com>

I am getting reports from a remote site of sporadic static/one way voice 
on calls.  These calls are through a gateway that is local at the site 
with a PRI.   I tried making about 8-10 test calls and every single call 
was fine for me.  The controller T1 1/0 is clean.

Total Data (last 24 hours)
     0 Line Code Violations, 0 Path Code Violations,
     0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
     0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail 
Secs


Anyone seen this before?

Carlos
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From matt at saskin.net  Mon Oct 22 14:54:47 2007
From: matt at saskin.net (Matthew Saskin)
Date: Mon, 22 Oct 2007 14:54:47 -0400
Subject: [cisco-voip] cisco 7921  wpa setting
In-Reply-To: <49AFE548BC7F7E49A1A4A2F28AF1914901E8C2@mail3.adcapnetsvc.local>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E8C2@mail3.adcapnetsvc.local>
Message-ID: <471CF1F7.1080308@saskin.net>

I believe you have to configure wpa via the desktop config app rather 
than on the phone itself - at least this was the case with the 7920's.

-matt

James Grace wrote:
> ?
> Is I possible to set up wpa on an 7921 wireless phone.  I heard it was.  
> But i dont see how. 
>  
> James Grace
>  
> <http://www.adcapnet.com/> 
> 
> Disclaimer For ADCAP Network Systems
> 
> "The information transmitted is intended only for the person or entity to which it is addressed and may
> contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
> or taking of any action in reliance upon this information by persons or entities other than the intended
> recipient is prohibited. If you received this message in error, please contact the sender and delete the
> material from all computers."
> 
> This original email was sent to the internet for delivery at 22 Oct 2007 14:42:13 -0400
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From jonvoip at gmail.com  Mon Oct 22 15:01:29 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 22 Oct 2007 14:01:29 -0500
Subject: [cisco-voip] Files not backed up by BARS
In-Reply-To: <421CAA1A5300B84A87D200D70998578263DF23@DCEX2.DSMO.COM>
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
	<421CAA1A5300B84A87D200D70998578263DF23@DCEX2.DSMO.COM>
Message-ID: <5d093f9a0710221201w29d47692q6cd71c519cbc435b@mail.gmail.com>

It is a 7835... DL 380 G3

BARS is 4.0(8)


J

On 10/22/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
> How old is your Pub?
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> MillerS at dicksteinshapiro.com
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Monday, October 22, 2007 2:35 PM
> To: cisco voip list
> Subject: [cisco-voip] Files not backed up by BARS
>
> Just checking here, besides TFTP path stuff (ringtones, backgrounds,
> etc.) what else is not backed up by BARS 4.0(8)
>
> We have a dying publisher and are backing up data before the rebuild
> and want to make sure we have everything...
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential
> and are intended solely for the use of the addressee(s)
> named above. This communication may contain material
> protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
> responsible for delivering this confidential communication
> to the intended recipient, you have received this
> communication in error, and any review, use, dissemination,
> forwarding, printing, copying, or other distribution of
> this e-mail message and any attached files is strictly
> prohibited. Dickstein Shapiro reserves the right to monitor
> any communication that is created, received, or sent on its
> network.  If you have received this confidential
> communication in error, please notify the sender
> immediately by reply e-mail message and permanently delete
> the original message.
>
> To reply to our email administrator directly, send an email
> to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
>
> ==============================================================================
>
>

From jonvoip at gmail.com  Mon Oct 22 15:01:47 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 22 Oct 2007 14:01:47 -0500
Subject: [cisco-voip] Files not backed up by BARS
In-Reply-To: <02fd01c814dc$950a4ce0$56196883@cfs.uoguelph.ca>
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
	<02fd01c814dc$950a4ce0$56196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0710221201v14a9196cy44addda6b1a7e12a@mail.gmail.com>

Dunno, I seem to remember it not being backed up...


J

On 10/22/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> why wouldnt' the tftp stuff be backed up?
>
> ----- Original Message -----
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "cisco voip list" <cisco-voip at puck.nether.net>
> Sent: Monday, October 22, 2007 2:35 PM
> Subject: [cisco-voip] Files not backed up by BARS
>
>
> > Just checking here, besides TFTP path stuff (ringtones, backgrounds,
> > etc.) what else is not backed up by BARS 4.0(8)
> >
> > We have a dying publisher and are backing up data before the rebuild
> > and want to make sure we have everything...
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From jonvoip at gmail.com  Mon Oct 22 15:08:21 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 22 Oct 2007 14:08:21 -0500
Subject: [cisco-voip] Call static
In-Reply-To: <OF1B7BC7BE.D9BF5343-ON8525737C.006730BC-8525737C.0067B456@bftg.com>
References: <OF1B7BC7BE.D9BF5343-ON8525737C.006730BC-8525737C.0067B456@bftg.com>
Message-ID: <5d093f9a0710221208p6b55443fwf7cc6133e597f4a6@mail.gmail.com>

One way audio is always (seriously, always) a routing problem.

The static could be occurring on a leg after the T1 is terminated at
telco or on the PRI switch itself, or on the other end of the call...





Jonathan

On 10/22/07, CarlosOrtiz at bayviewfinancial.com
<CarlosOrtiz at bayviewfinancial.com> wrote:
>
> I am getting reports from a remote site of sporadic static/one way voice on
> calls.  These calls are through a gateway that is local at the site with a
> PRI.   I tried making about 8-10 test calls and every single call was fine
> for me.  The controller T1 1/0 is clean.
>
> Total Data (last 24 hours)
>      0 Line Code Violations, 0 Path Code Violations,
>      0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
>      0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
>
>
> Anyone seen this before?
>
> Carlos
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From lists at jerkys.org  Mon Oct 22 15:18:00 2007
From: lists at jerkys.org (Jerky)
Date: Mon, 22 Oct 2007 14:18:00 -0500
Subject: [cisco-voip] cisco 7921  wpa setting
In-Reply-To: <471CF1F7.1080308@saskin.net>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E8C2@mail3.adcapnetsvc.local>
	<471CF1F7.1080308@saskin.net>
Message-ID: <E470D8C5-5371-4801-B963-125F4811AE22@jerkys.org>

I don't have access to a 7921 but I have played around with WPA on  
the 7920. I don't know if this helps since I haven't seen the 7921 UI  
but on the 7920 it was Menu -> Profiles -> Network Profiles  From  
there you can modify an existing one or make a new profile. Once  
you've chosen a profile 802.11b Configuration is the area you're  
looking for.

jeff


On Oct 22, 2007, at 1:54 PM, Matthew Saskin wrote:

> I believe you have to configure wpa via the desktop config app rather
> than on the phone itself - at least this was the case with the 7920's.
>
> -matt
>
> James Grace wrote:
>> ?
>> Is I possible to set up wpa on an 7921 wireless phone.  I heard it  
>> was.
>> But i dont see how.
>>
>> James Grace
>>
>> <http://www.adcapnet.com/>
>>
>> Disclaimer For ADCAP Network Systems
>>
>> "The information transmitted is intended only for the person or  
>> entity to which it is addressed and may
>> contain confidential and/or privileged material. Any review,  
>> retransmission, dissemination or other use of,
>> or taking of any action in reliance upon this information by  
>> persons or entities other than the intended
>> recipient is prohibited. If you received this message in error,  
>> please contact the sender and delete the
>> material from all computers."
>>
>> This original email was sent to the internet for delivery at 22  
>> Oct 2007 14:42:13 -0400
>>
>>
>> --------------------------------------------------------------------- 
>> ---
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jeff at ocjtech.us  Mon Oct 22 15:43:51 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Mon, 22 Oct 2007 14:43:51 -0500
Subject: [cisco-voip] Files not backed up by BARS
In-Reply-To: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
Message-ID: <1193082231.7445.3.camel@lt21223.campus.dmacc.edu>

On Mon, 2007-10-22 at 13:35 -0500, Jonathan Charles wrote:
> Just checking here, besides TFTP path stuff (ringtones, backgrounds,
> etc.) what else is not backed up by BARS 4.0(8)
> 
> We have a dying publisher and are backing up data before the rebuild
> and want to make sure we have everything...

I'd recommend making a full backup of the filesystem using Windows
Backup, in addition to manually copying any files.  It's been awhile,
but ISTR that Windows Backup is available in the Cisco default Windows
install.  I'd also recommend setting aside one disk in the RAID mirror
for emergency purposes.

Jeff

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From jonvoip at gmail.com  Mon Oct 22 16:11:09 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 22 Oct 2007 15:11:09 -0500
Subject: [cisco-voip] Files not backed up by BARS
In-Reply-To: <1193082231.7445.3.camel@lt21223.campus.dmacc.edu>
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
	<1193082231.7445.3.camel@lt21223.campus.dmacc.edu>
Message-ID: <5d093f9a0710221311n3b328f27xac8e71116640523b@mail.gmail.com>

There is NO RAID mirror

If there were, i would just replace it...



Jonathan

On 10/22/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote:
> On Mon, 2007-10-22 at 13:35 -0500, Jonathan Charles wrote:
> > Just checking here, besides TFTP path stuff (ringtones, backgrounds,
> > etc.) what else is not backed up by BARS 4.0(8)
> >
> > We have a dying publisher and are backing up data before the rebuild
> > and want to make sure we have everything...
>
> I'd recommend making a full backup of the filesystem using Windows
> Backup, in addition to manually copying any files.  It's been awhile,
> but ISTR that Windows Backup is available in the Cisco default Windows
> install.  I'd also recommend setting aside one disk in the RAID mirror
> for emergency purposes.
>
> Jeff
>
>
>

From MLinsemier at apcapital.com  Mon Oct 22 16:12:40 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Mon, 22 Oct 2007 16:12:40 -0400
Subject: [cisco-voip] Files not backed up by BARS
In-Reply-To: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
Message-ID: <AB0A35F2486F81468971AA68558BE7190743FA89@smieln01winm01.micoanet.com>

You also may want to do a bug scrub for issues with backups and restores
in your current version BARS 4.0(8) compared to the current version
4.0(13).  I think that I restored a publisher using 4.0(8) here a few
months ago and didn't have to many issues other than I had to use the
exact same version to do the restore.

I always back up any UCXX scripts, ringtones, prompts (wav files), etc
by hand just because they are easy to just stick on a file share.

Matt

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Monday, October 22, 2007 2:35 PM
To: cisco voip list
Subject: [cisco-voip] Files not backed up by BARS

Just checking here, besides TFTP path stuff (ringtones, backgrounds,
etc.) what else is not backed up by BARS 4.0(8)

We have a dying publisher and are backing up data before the rebuild
and want to make sure we have everything...



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may
be protected by one or more legal privileges. It is intended
solely for the use of the addressee identified above. If you
are not the intended recipient, any use, disclosure, copying
or distribution of this communication is UNAUTHORIZED. Neither
this information block, the typed name of the sender, nor
anything else in this message is intended to constitute an
electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
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From baryonyx5 at gmail.com  Mon Oct 22 16:32:58 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Mon, 22 Oct 2007 15:32:58 -0500
Subject: [cisco-voip] Unity Connections to CME using G.729
Message-ID: <a749d0620710221332r15a74befqf503e3ca28a0cb63@mail.gmail.com>

In a multiple CME to centralized Unity Connections deployment, is it
possible to use G.729 as the codec between UC and CME?  I don't see anything
on CCO either way.

In a lab configuration using UC 1.2.1 sr2 and CME 4.0 (12.4(9)T) I
configured UC to advertise only G.729a and used the codec ephone command to
specify G.729 on the voicemail ephones but all calls to UC were connected
using G.711.  I'm not sure if that's working be design or not.

-- 
kris seraphine
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From rkulagow at gmail.com  Mon Oct 22 17:05:58 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Mon, 22 Oct 2007 16:05:58 -0500
Subject: [cisco-voip] 7970 AUX port pinout?
Message-ID: <471D10B6.1010707@gmail.com>

I'm trying to connect to the AUX port; I'm assuming that it's a standard 
serial connection, but I can't find a pinout.


From bcarter at sentinel.com  Mon Oct 22 22:36:02 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 22 Oct 2007 21:36:02 -0500
Subject: [cisco-voip] Different ring types for internal/external calls
In-Reply-To: <643A3A543DB5224499BC3D73473AEC27869E63@NYEXMB2.shearman.com>
References: <NYEXM4domP0l0HQFDMz00017f35@NYEXMB2.shearman.com>
	<643A3A543DB5224499BC3D73473AEC27869E63@NYEXMB2.shearman.com>
Message-ID: <6A0744639B86DD459433108DBF101D26019FE50B@MAIL01.sentinel.com>

On the inverse.  Recently upgraded customer from 4.1 to 5.1.  They don't
like different rings for OnNet/OffNet.  Could this be fixed by changing
the Call Classification to OnNet for all PSTN calls ?


________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Stillman
Sent: Thursday, October 18, 2007 3:54 PM
To: Wes Sisk <wsisk at cisco.com>@SHSDOMAIN
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Different ring types for internal/external
calls


Genius Wes!
 
I did a hard shut no/shut of MGCP and it works!

________________________________

From: Wes Sisk <wsisk at cisco.com>@SHSDOMAIN 
Sent: Thursday, October 18, 2007 4:06 PM
To: Erik Stillman
Cc: Lelio Fulgenzi; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Different ring types for internal/external
calls


In your PSTN gateway configuration make sure the 'Call Classification'
is set to OffNet or use system default.
Make sure advanced CM service parameter "Call Classification" = OffNet.

/Wes

Erik Stillman wrote: 

	Thanks Lelio, but this request is for the phones to ring
differently
	when called from the PSTN.
	
	-Erik 
	
	-----Original Message-----
	From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
<mailto:lelio at uoguelph.ca> @SHSDOMAIN 
	Sent: Thursday, October 18, 2007 1:35 PM
	To: Erik Stillman; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Different ring types for
internal/external
	calls
	
	As long as you are using a CTI based autoattendant like IPCC
Express or
	Unity Express than this should be doable. There is an enterprise
or
	system parameter that differentiates external calls with a
double ring.
	
	If you are using Unity, then you are out of luck because it uses
SCCP
	and transfers calls as if a user is transferring them.
	
	----- Original Message -----
	From: "Erik Stillman" <Erik.Stillman at Shearman.com>
<mailto:Erik.Stillman at Shearman.com> 
	To: <cisco-voip at puck.nether.net>
<mailto:cisco-voip at puck.nether.net> 
	Sent: Thursday, October 18, 2007 1:12 PM
	Subject: [cisco-voip] Different ring types for internal/external
calls
	
	
	  

		Hey guys, I've got a hot one here. Request has come in
to
		    

	differentiate
	  

		calls by ring type.
		
		Is there anything I can do in CCM4.2 to achieve this?
		
		Any help would be great. Thanks!
		
		-Erik
		
		
		
		    

	
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From daniel.abelski at gmail.com  Tue Oct 23 01:55:36 2007
From: daniel.abelski at gmail.com (Daniel A. Abelski)
Date: Tue, 23 Oct 2007 11:55:36 +0600
Subject: [cisco-voip] =?koi8-r?b?79TXxdQ6ICBDQ00gNi4wIFF1ZXN0aW9u?=
In-Reply-To: <471ce47a.0e1f400a.24b8.fffffc9e@mx.google.com>
References: <471ce47a.0e1f400a.24b8.fffffc9e@mx.google.com>
Message-ID: <cd0147ab0710222255t4f179929i57ca658ea8071c65@mail.gmail.com>

Id like to know that too. Definity gives this "called-on-another-call
sound" and its convenient.

2007/10/22, Dustin S Fowler <dustin.s.fowler at gmail.com>:
>
>
> That is great. LMAO. Speaking more of a call waiting tone instead. Telling
> the caller that the called is on the phone. Maybe even a screen
> notification.
>
>
>
> From: Voice Noob [mailto:voicenoob at gmail.com]
> Sent: Monday, October 22, 2007 12:58 PM
> To: Dustin S Fowler
> Subject: Re: [cisco-voip] CCM 6.0 Question
>
>
>
> Busy tone.
>
> :)
>
>
>
> On 10/22/07, Dustin S Fowler <dustin.s.fowler at gmail.com> wrote:
>
>
>
> All,
>
>
>
> Does anyone know of a setting to present a tone to allow an internal caller
> to be aware that the person they are trying to call that the party is on
> another call.
>
>
>
> Thank you,
>
>
>
> Dustin
>
>
>
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net  <mailto:cisco-voip at puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>


-- 
Regards,                    |        ? ?????????,
Daniel A. Abelski        |        ????????? ?????? ?????????????

From erickbee at gmail.com  Tue Oct 23 02:04:37 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Tue, 23 Oct 2007 01:04:37 -0500
Subject: [cisco-voip] Files not backed up by BARS
In-Reply-To: <AB0A35F2486F81468971AA68558BE7190743FA89@smieln01winm01.micoanet.com>
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE7190743FA89@smieln01winm01.micoanet.com>
Message-ID: <f4445faf0710222304o68c80a07kcca3e08c719cb5aa@mail.gmail.com>

Last I looked (awhile ago), BARS backed up all the files in the
TFTTPath folders and stuck them in the tarball including any extra
non-CCM stuff you may have stuck there (IOS images, pictures of your
dog, etc).

On 10/22/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
> You also may want to do a bug scrub for issues with backups and restores
> in your current version BARS 4.0(8) compared to the current version
> 4.0(13).  I think that I restored a publisher using 4.0(8) here a few
> months ago and didn't have to many issues other than I had to use the
> exact same version to do the restore.
>
> I always back up any UCXX scripts, ringtones, prompts (wav files), etc
> by hand just because they are easy to just stick on a file share.
>
> Matt
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Monday, October 22, 2007 2:35 PM
> To: cisco voip list
> Subject: [cisco-voip] Files not backed up by BARS
>
> Just checking here, besides TFTP path stuff (ringtones, backgrounds,
> etc.) what else is not backed up by BARS 4.0(8)
>
> We have a dying publisher and are backing up data before the rebuild
> and want to make sure we have everything...
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may
> be protected by one or more legal privileges. It is intended
> solely for the use of the addressee identified above. If you
> are not the intended recipient, any use, disclosure, copying
> or distribution of this communication is UNAUTHORIZED. Neither
> this information block, the typed name of the sender, nor
> anything else in this message is intended to constitute an
> electronic signature unless a specific statement to the
> contrary is included in this message. If you have received this
> communication in error, please immediately contact me and delete
> this communication from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From erickbee at gmail.com  Tue Oct 23 02:10:05 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Tue, 23 Oct 2007 01:10:05 -0500
Subject: [cisco-voip] Delay when calling one extension - CCM 4.1(3)sr5b
Message-ID: <f4445faf0710222310j3ccb261em7d9b946c2d4bb0bd@mail.gmail.com>

Hi,

I am having an issue where we get the interdigit timeout delay when
dialing just one extension, 4222. We can dial 4223, 4224, 4225 without
delay. The CCM Detailed traces show potential matches exist but there
are no more matches anywhere to be found. I have checked the
NumPlan.CSV file and there are no translation patterns, no route
patterns, etc that also match. I have deleted the extension, put it
back and no change. I checked call park, meet me, and other numbers in
the system and nothing conflicts or overlaps this.

DNA doesn't show more then one match either.

Is there a way to find the other potential match from a trace file?

From adman at adman.net  Tue Oct 23 08:27:56 2007
From: adman at adman.net (Adam Blomfield)
Date: Tue, 23 Oct 2007 07:27:56 -0500
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <4718E3CE.20403@cisco.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710182002p30265d7do404b0f2d9b144842@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445C4D@EXCHANGEDS01.ds.ad.adp.com>
	<4718E3CE.20403@cisco.com>
Message-ID: <c6f32ce40710230527ub3a9e1bl2eb60f7a651f5c9e@mail.gmail.com>

I guess this free version is licensed for installation on a standalone
server, unlike the CM 4.x bundle? I have a customer that is taking advantage
of the bundled five seats on CM 4.1 but need to upgrade to 6.0 to get off
Windows 2000 to comply with their parent division's security policies. I
guess I will have to sell them an extra MCS to be able to install UCCX onto?

-Adam

On 10/19/07, Wes Sisk <wsisk at cisco.com> wrote:
>
>  CUCM6 does still come with the free 5 port version of IPCCX, now UCCX.
> /wes
>
> Frazee, Timothy wrote:
>
>  Does a CUCM 6 media kit come with some type of "Extended Services" or
> some type of IPCC?
>
>
>
> *From:* Ted Nugent [mailto:tednugent73 at gmail.com <tednugent73 at gmail.com>]
> *Sent:* Thursday, October 18, 2007 10:03 PM
> *To:* Frazee, Timothy
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] cucm 6 and TAPS
>
>
>
> I use VMWare with CRS and either spawn it on a customer server temporarily
> or use my laptop. But yes it needs to be separate unfortunately
>
>
>
>
>
> On 10/18/07, *Frazee, Timothy* <Timothy_Frazee at adp.com> wrote:
>
> So if I want to use TAPS to install a bulk amount of phones, I need a
> minimum of two servers? My publisher and a CRS server.
>
>
>
> I know this is a dumb question, but there is no "co-res" type of thing for
> crs and callmanager on a 6 environment?
>
>
>
> Thanks for any answers.
>
>
>
>
>
>
>
>
>
> *From:* Ted Nugent [mailto:tednugent73 at gmail.com ]
> *Sent:* Thursday, October 18, 2007 9:46 PM
> *To:* Frazee, Timothy
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] cucm 6 and TAPS
>
>
>
> confirmed... taps services is required for CM to talk to the taps.asp on
> the CRS server
>
> On 10/18/07, *Frazee, Timothy* <Timothy_Frazee at adp.com> wrote:
>
> Group,
>
>
>
> From some of the reading I have seen, it looks like I still need an
> external CRS server for TAPS to operate in 6. I see a "TAPS Service" in the
> serviceability page.
>
>
>
> Can anyone give me a firm answer?
>
>
>
> If it truly is "all in one" then can anyone point me to a doc that shows
> how to config it?
>
>
>
> Thanks all.
>
>
>
>
>
>
>
>
>
>
>
> Timothy Frazee
>
> Network Solutions
>
> Automatic Data Processing, Inc.
>   ------------------------------
>
> This message and any attachments are intended only for the use of the
> addressee and may contain information that is privileged and confidential.
> If the reader of the message is not the intended recipient or an authorized
> representative of the intended recipient, you are hereby notified that any
> dissemination of this communication is strictly prohibited. If you have
> received this communication in error, please notify us immediately by e-mail
> and delete the message and any attachments from your system.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>   ------------------------------
>
> This message and any attachments are intended only for the use of the
> addressee and may contain information that is privileged and confidential.
> If the reader of the message is not the intended recipient or an authorized
> representative of the intended recipient, you are hereby notified that any
> dissemination of this communication is strictly prohibited. If you have
> received this communication in error, please notify us immediately by e-mail
> and delete the message and any attachments from your system.
>
>
>  ------------------------------
> This message and any attachments are intended only for the use of the
> addressee and may contain information that is privileged and confidential.
> If the reader of the message is not the intended recipient or an authorized
> representative of the intended recipient, you are hereby notified that any
> dissemination of this communication is strictly prohibited. If you have
> received this communication in error, please notify us immediately by e-mail
> and delete the message and any attachments from your system.
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From voicenoob at gmail.com  Tue Oct 23 09:02:33 2007
From: voicenoob at gmail.com (Voice Noob)
Date: Tue, 23 Oct 2007 08:02:33 -0500
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <c6f32ce40710230527ub3a9e1bl2eb60f7a651f5c9e@mail.gmail.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>
	<22e6380b0710182002p30265d7do404b0f2d9b144842@mail.gmail.com>
	<723FA17C6AA4C4428CDF572DCAEE08D405445C4D@EXCHANGEDS01.ds.ad.adp.com>
	<4718E3CE.20403@cisco.com>
	<c6f32ce40710230527ub3a9e1bl2eb60f7a651f5c9e@mail.gmail.com>
Message-ID: <535625f70710230602g4550f9dx5ffd3f5ab53df059@mail.gmail.com>

Yes they will need to have another server, which I still think is windows
2000 unless UCCX is supported on the 2003 train. Also can Wes confirm that
you still get the bundled 5 seat license with 5.x / 6.x or if it is just for
using TAPS?

On 10/23/07, Adam Blomfield <adman at adman.net> wrote:
>
> I guess this free version is licensed for installation on a standalone
> server, unlike the CM 4.x bundle? I have a customer that is taking
> advantage of the bundled five seats on CM 4.1 but need to upgrade to 6.0to get off Windows 2000 to comply with their parent division's security
> policies. I guess I will have to sell them an extra MCS to be able to
> install UCCX onto?
>
> -Adam
>
> On 10/19/07, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > CUCM6 does still come with the free 5 port version of IPCCX, now UCCX.
> > /wes
> >
> > Frazee, Timothy wrote:
> >
> >  Does a CUCM 6 media kit come with some type of "Extended Services" or
> > some type of IPCC?
> >
> >
> >
> > *From:* Ted Nugent [mailto:tednugent73 at gmail.com <tednugent73 at gmail.com>]
> >
> > *Sent:* Thursday, October 18, 2007 10:03 PM
> > *To:* Frazee, Timothy
> > *Cc:* cisco-voip at puck.nether.net
> > *Subject:* Re: [cisco-voip] cucm 6 and TAPS
> >
> >
> >
> > I use VMWare with CRS and either spawn it on a customer server
> > temporarily or use my laptop. But yes it needs to be separate unfortunately
> >
> >
> >
> >
> >
> > On 10/18/07, *Frazee, Timothy* <Timothy_Frazee at adp.com> wrote:
> >
> > So if I want to use TAPS to install a bulk amount of phones, I need a
> > minimum of two servers? My publisher and a CRS server.
> >
> >
> >
> > I know this is a dumb question, but there is no "co-res" type of thing
> > for crs and callmanager on a 6 environment?
> >
> >
> >
> > Thanks for any answers.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > *From:* Ted Nugent [mailto:tednugent73 at gmail.com ]
> > *Sent:* Thursday, October 18, 2007 9:46 PM
> > *To:* Frazee, Timothy
> > *Cc:* cisco-voip at puck.nether.net
> > *Subject:* Re: [cisco-voip] cucm 6 and TAPS
> >
> >
> >
> > confirmed... taps services is required for CM to talk to the taps.asp on
> > the CRS server
> >
> > On 10/18/07, *Frazee, Timothy* <Timothy_Frazee at adp.com> wrote:
> >
> > Group,
> >
> >
> >
> > From some of the reading I have seen, it looks like I still need an
> > external CRS server for TAPS to operate in 6. I see a "TAPS Service" in the
> > serviceability page.
> >
> >
> >
> > Can anyone give me a firm answer?
> >
> >
> >
> > If it truly is "all in one" then can anyone point me to a doc that shows
> > how to config it?
> >
> >
> >
> > Thanks all.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Timothy Frazee
> >
> > Network Solutions
> >
> > Automatic Data Processing, Inc.
> >  ------------------------------
> >
> > This message and any attachments are intended only for the use of the
> > addressee and may contain information that is privileged and confidential.
> > If the reader of the message is not the intended recipient or an authorized
> > representative of the intended recipient, you are hereby notified that any
> > dissemination of this communication is strictly prohibited. If you have
> > received this communication in error, please notify us immediately by e-mail
> > and delete the message and any attachments from your system.
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >  ------------------------------
> >
> > This message and any attachments are intended only for the use of the
> > addressee and may contain information that is privileged and confidential.
> > If the reader of the message is not the intended recipient or an authorized
> > representative of the intended recipient, you are hereby notified that any
> > dissemination of this communication is strictly prohibited. If you have
> > received this communication in error, please notify us immediately by e-mail
> > and delete the message and any attachments from your system.
> >
> >
> >  ------------------------------
> > This message and any attachments are intended only for the use of the
> > addressee and may contain information that is privileged and confidential.
> > If the reader of the message is not the intended recipient or an authorized
> > representative of the intended recipient, you are hereby notified that any
> > dissemination of this communication is strictly prohibited. If you have
> > received this communication in error, please notify us immediately by e-mail
> > and delete the message and any attachments from your system.
> >
> > ------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
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> cisco-voip at puck.nether.net
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>
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From jonvoip at gmail.com  Tue Oct 23 09:33:04 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 23 Oct 2007 08:33:04 -0500
Subject: [cisco-voip] Files not backed up by BARS
In-Reply-To: <f4445faf0710222304o68c80a07kcca3e08c719cb5aa@mail.gmail.com>
References: <5d093f9a0710221135i43c52a15ub006de6304b77a8@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE7190743FA89@smieln01winm01.micoanet.com>
	<f4445faf0710222304o68c80a07kcca3e08c719cb5aa@mail.gmail.com>
Message-ID: <5d093f9a0710230633jc925569x56dbd45013dee70a@mail.gmail.com>

4.0(13) backed up everything... it worked like a champ.


J

On 10/23/07, Erick Bergquist <erickbee at gmail.com> wrote:
> Last I looked (awhile ago), BARS backed up all the files in the
> TFTTPath folders and stuck them in the tarball including any extra
> non-CCM stuff you may have stuck there (IOS images, pictures of your
> dog, etc).
>
> On 10/22/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
> > You also may want to do a bug scrub for issues with backups and restores
> > in your current version BARS 4.0(8) compared to the current version
> > 4.0(13).  I think that I restored a publisher using 4.0(8) here a few
> > months ago and didn't have to many issues other than I had to use the
> > exact same version to do the restore.
> >
> > I always back up any UCXX scripts, ringtones, prompts (wav files), etc
> > by hand just because they are easy to just stick on a file share.
> >
> > Matt
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > Charles
> > Sent: Monday, October 22, 2007 2:35 PM
> > To: cisco voip list
> > Subject: [cisco-voip] Files not backed up by BARS
> >
> > Just checking here, besides TFTP path stuff (ringtones, backgrounds,
> > etc.) what else is not backed up by BARS 4.0(8)
> >
> > We have a dying publisher and are backing up data before the rebuild
> > and want to make sure we have everything...
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > CONFIDENTIALITY STATEMENT
> > This communication and any attachments are CONFIDENTIAL and may
> > be protected by one or more legal privileges. It is intended
> > solely for the use of the addressee identified above. If you
> > are not the intended recipient, any use, disclosure, copying
> > or distribution of this communication is UNAUTHORIZED. Neither
> > this information block, the typed name of the sender, nor
> > anything else in this message is intended to constitute an
> > electronic signature unless a specific statement to the
> > contrary is included in this message. If you have received this
> > communication in error, please immediately contact me and delete
> > this communication from your computer. Thank you.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From cshaffer at gmail.com  Tue Oct 23 10:05:54 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Tue, 23 Oct 2007 10:05:54 -0400
Subject: [cisco-voip] "cracky MOH"
Message-ID: <005301c8157d$d52adaf0$3900490a@acg.arlington.local>

I have a client running CME and they have reported "cracky" MOH recently.
Every time I have called in (making them random) I have yet to replicate the
issue. Some initial research shows this to happen on Call Manager under
heavy loads. I'm wondering if the router is spiking for some reason but
knowing what I do about the company I can't see this as being an issue,
although I will probably set up some generic monitoring just in case. 

Has anyone out there heard of this phantom "cracky" MOH in CME? If so what
have you found to be the cause? As a note, this is a new occurrence. The
system has been up for over a year without anyone mentioning this.

Thanks in advance!

Curt


From cheesevoice at gmail.com  Tue Oct 23 10:42:57 2007
From: cheesevoice at gmail.com (Kevin Dunn)
Date: Tue, 23 Oct 2007 09:42:57 -0500
Subject: [cisco-voip] Remote analog phone Voice Quality
Message-ID: <84b74aec0710230742x6818f2a8j8ae21daf7c7ead4c@mail.gmail.com>

I have a remote office with only one analog phone, we have an FXS port to a
"black phone" and one FXO port for 911 calls.

The extesnion is 9227, the configuration is enclosed below...

When on a call there is "static" or background noise, and occasionally one
way transmission (typically the IP phone calling him can not hear the black
phone and the black phone can hear just fine).  Other times the call drops
all together.  Yet other times, everythign is just peachy and we can make
6-8 calls in a row that work great.  So I was thinking it might be a QoS
issue.  Any hints or leads are appreciated.

Here is the gateway configuration (H323, 2620 running 12.2.23F):

voice-port 1/0/0
 description FXS port to Office Phone
 station-id name Plant3_Office
 station-id number 9227
 caller-id enable

dial-peer voice 1 pots
 preference 1
 destination-pattern 9227
 port 1/0/0

Here is the QoS config:

class-map match-all voice-traffic
  match ip precedence 5
class-map match-all Call-Control
  match access-group 110
class-map match-all RTP
  match ip precedence 5

policy-map OutboundPolicy
  class Call-Control
   set cos 3
   set ip precedence 3
  class RTP
   set cos 5
  class class-default
   set cos 0
policy-map VOICE-POLICY
  class voice-traffic
    priority 480
  class Call-Control
   bandwidth 16
  class class-default
   fair-queue

I was wondering if the bandwidth 16 is enough for the voice call.  It is a
dry-pair T1 connection and only one phone on the site.
Any help is appreciated, and thank you in advnace

Kevin
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From asobihoudai at yahoo.com  Tue Oct 23 10:48:15 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 23 Oct 2007 07:48:15 -0700 (PDT)
Subject: [cisco-voip] "cracky MOH"
In-Reply-To: <005301c8157d$d52adaf0$3900490a@acg.arlington.local>
Message-ID: <284085.52933.qm@web57201.mail.re3.yahoo.com>

Curt,

Is every user experiencing this problem or is it an
isolate user?

Paul

--- Curt Shaffer <cshaffer at gmail.com> wrote:

> I have a client running CME and they have reported
> "cracky" MOH recently.
> Every time I have called in (making them random) I
> have yet to replicate the
> issue. Some initial research shows this to happen on
> Call Manager under
> heavy loads. I'm wondering if the router is spiking
> for some reason but
> knowing what I do about the company I can't see this
> as being an issue,
> although I will probably set up some generic
> monitoring just in case. 
> 
> Has anyone out there heard of this phantom "cracky"
> MOH in CME? If so what
> have you found to be the cause? As a note, this is a
> new occurrence. The
> system has been up for over a year without anyone
> mentioning this.
> 
> Thanks in advance!
> 
> Curt
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From cshaffer at gmail.com  Tue Oct 23 10:50:05 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Tue, 23 Oct 2007 10:50:05 -0400
Subject: [cisco-voip] "cracky MOH"
In-Reply-To: <284085.52933.qm@web57201.mail.re3.yahoo.com>
References: <005301c8157d$d52adaf0$3900490a@acg.arlington.local>
	<284085.52933.qm@web57201.mail.re3.yahoo.com>
Message-ID: <005a01c81584$00d22ea0$3900490a@acg.arlington.local>

Doesn't appear to be user specific.

> >-----Original Message-----
> >From: Paul Choi [mailto:asobihoudai at yahoo.com]
> >Sent: Tuesday, October 23, 2007 10:48 AM
> >To: Curt Shaffer; cisco-voip at puck.nether.net
> >Subject: Re: [cisco-voip] "cracky MOH"
> >
> >Curt,
> >
> >Is every user experiencing this problem or is it an
> >isolate user?
> >
> >Paul
> >
> >--- Curt Shaffer <cshaffer at gmail.com> wrote:
> >
> >> I have a client running CME and they have reported
> >> "cracky" MOH recently.
> >> Every time I have called in (making them random) I
> >> have yet to replicate the
> >> issue. Some initial research shows this to happen on
> >> Call Manager under
> >> heavy loads. I'm wondering if the router is spiking
> >> for some reason but
> >> knowing what I do about the company I can't see this
> >> as being an issue,
> >> although I will probably set up some generic
> >> monitoring just in case.
> >>
> >> Has anyone out there heard of this phantom "cracky"
> >> MOH in CME? If so what
> >> have you found to be the cause? As a note, this is a
> >> new occurrence. The
> >> system has been up for over a year without anyone
> >> mentioning this.
> >>
> >> Thanks in advance!
> >>
> >> Curt
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> >__________________________________________________
> >Do You Yahoo!?
> >Tired of spam?  Yahoo! Mail has the best spam protection around
> >http://mail.yahoo.com


From sweeper1 at gmail.com  Tue Oct 23 10:50:32 2007
From: sweeper1 at gmail.com (sweeper1)
Date: Tue, 23 Oct 2007 10:50:32 -0400
Subject: [cisco-voip] "cracky MOH"
In-Reply-To: <284085.52933.qm@web57201.mail.re3.yahoo.com>
References: <005301c8157d$d52adaf0$3900490a@acg.arlington.local>
	<284085.52933.qm@web57201.mail.re3.yahoo.com>
Message-ID: <cf6e8aa90710230750k18b03c2aidee65492fb965d23@mail.gmail.com>

FXOs or PRIs?

On 10/23/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> Curt,
>
> Is every user experiencing this problem or is it an
> isolate user?
>
> Paul
>
> --- Curt Shaffer <cshaffer at gmail.com> wrote:
>
> > I have a client running CME and they have reported
> > "cracky" MOH recently.
> > Every time I have called in (making them random) I
> > have yet to replicate the
> > issue. Some initial research shows this to happen on
> > Call Manager under
> > heavy loads. I'm wondering if the router is spiking
> > for some reason but
> > knowing what I do about the company I can't see this
> > as being an issue,
> > although I will probably set up some generic
> > monitoring just in case.
> >
> > Has anyone out there heard of this phantom "cracky"
> > MOH in CME? If so what
> > have you found to be the cause? As a note, this is a
> > new occurrence. The
> > system has been up for over a year without anyone
> > mentioning this.
> >
> > Thanks in advance!
> >
> > Curt
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From asobihoudai at yahoo.com  Tue Oct 23 10:54:00 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 23 Oct 2007 07:54:00 -0700 (PDT)
Subject: [cisco-voip] "cracky MOH"
In-Reply-To: <005301c8157d$d52adaf0$3900490a@acg.arlington.local>
Message-ID: <939344.16975.qm@web57210.mail.re3.yahoo.com>

Curt,

Is this the caller complaining or the callee?




--- Curt Shaffer <cshaffer at gmail.com> wrote:

> I have a client running CME and they have reported
> "cracky" MOH recently.
> Every time I have called in (making them random) I
> have yet to replicate the
> issue. Some initial research shows this to happen on
> Call Manager under
> heavy loads. I'm wondering if the router is spiking
> for some reason but
> knowing what I do about the company I can't see this
> as being an issue,
> although I will probably set up some generic
> monitoring just in case. 
> 
> Has anyone out there heard of this phantom "cracky"
> MOH in CME? If so what
> have you found to be the cause? As a note, this is a
> new occurrence. The
> system has been up for over a year without anyone
> mentioning this.
> 

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From cshaffer at gmail.com  Tue Oct 23 10:58:47 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Tue, 23 Oct 2007 10:58:47 -0400
Subject: [cisco-voip] "cracky MOH"
In-Reply-To: <cf6e8aa90710230750k18b03c2aidee65492fb965d23@mail.gmail.com>
References: <005301c8157d$d52adaf0$3900490a@acg.arlington.local>
	<284085.52933.qm@web57201.mail.re3.yahoo.com>
	<cf6e8aa90710230750k18b03c2aidee65492fb965d23@mail.gmail.com>
Message-ID: <005e01c81585$38167a50$3900490a@acg.arlington.local>

PRI

 

  _____  

From: sweeper1 [mailto:sweeper1 at gmail.com] 
Sent: Tuesday, October 23, 2007 10:51 AM
To: Paul Choi
Cc: Curt Shaffer; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] "cracky MOH"

 

FXOs or PRIs?

On 10/23/07, Paul Choi <asobihoudai at yahoo.com> wrote: 

Curt,

Is every user experiencing this problem or is it an
isolate user?

Paul

--- Curt Shaffer < cshaffer at gmail.com> wrote:

> I have a client running CME and they have reported
> "cracky" MOH recently.
> Every time I have called in (making them random) I 
> have yet to replicate the
> issue. Some initial research shows this to happen on
> Call Manager under
> heavy loads. I'm wondering if the router is spiking
> for some reason but
> knowing what I do about the company I can't see this 
> as being an issue,
> although I will probably set up some generic
> monitoring just in case.
>
> Has anyone out there heard of this phantom "cracky"
> MOH in CME? If so what 
> have you found to be the cause? As a note, this is a
> new occurrence. The
> system has been up for over a year without anyone
> mentioning this.
>
> Thanks in advance!
>
> Curt 
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From cshaffer at gmail.com  Tue Oct 23 10:58:58 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Tue, 23 Oct 2007 10:58:58 -0400
Subject: [cisco-voip] "cracky MOH"
In-Reply-To: <939344.16975.qm@web57210.mail.re3.yahoo.com>
References: <005301c8157d$d52adaf0$3900490a@acg.arlington.local>
	<939344.16975.qm@web57210.mail.re3.yahoo.com>
Message-ID: <006301c81585$3e9699f0$3900490a@acg.arlington.local>

caller

> >-----Original Message-----
> >From: Paul Choi [mailto:asobihoudai at yahoo.com]
> >Sent: Tuesday, October 23, 2007 10:54 AM
> >To: Curt Shaffer; cisco-voip at puck.nether.net
> >Subject: Re: [cisco-voip] "cracky MOH"
> >
> >Curt,
> >
> >Is this the caller complaining or the callee?
> >
> >
> >
> >
> >--- Curt Shaffer <cshaffer at gmail.com> wrote:
> >
> >> I have a client running CME and they have reported
> >> "cracky" MOH recently.
> >> Every time I have called in (making them random) I
> >> have yet to replicate the
> >> issue. Some initial research shows this to happen on
> >> Call Manager under
> >> heavy loads. I'm wondering if the router is spiking
> >> for some reason but
> >> knowing what I do about the company I can't see this
> >> as being an issue,
> >> although I will probably set up some generic
> >> monitoring just in case.
> >>
> >> Has anyone out there heard of this phantom "cracky"
> >> MOH in CME? If so what
> >> have you found to be the cause? As a note, this is a
> >> new occurrence. The
> >> system has been up for over a year without anyone
> >> mentioning this.
> >>
> >
> >__________________________________________________
> >Do You Yahoo!?
> >Tired of spam?  Yahoo! Mail has the best spam protection around
> >http://mail.yahoo.com


From asobihoudai at yahoo.com  Tue Oct 23 11:02:22 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 23 Oct 2007 08:02:22 -0700 (PDT)
Subject: [cisco-voip] "cracky MOH"
In-Reply-To: <006301c81585$3e9699f0$3900490a@acg.arlington.local>
Message-ID: <776151.31165.qm@web57213.mail.re3.yahoo.com>

That could be a myriad of issues IMHO. For all you
know, most everything that caller is listening to is
going to be crackly. If it was *internal*, then maybe
you'd have better luck working on such a problem.

Paul

--- Curt Shaffer <cshaffer at gmail.com> wrote:

> caller
> 
> > >-----Original Message-----
> > >From: Paul Choi [mailto:asobihoudai at yahoo.com]
> > >Sent: Tuesday, October 23, 2007 10:54 AM
> > >To: Curt Shaffer; cisco-voip at puck.nether.net
> > >Subject: Re: [cisco-voip] "cracky MOH"
> > >
> > >Curt,
> > >
> > >Is this the caller complaining or the callee?
> > >
> > >
> > >
> > >
> > >--- Curt Shaffer <cshaffer at gmail.com> wrote:
> > >
> > >> I have a client running CME and they have
> reported
> > >> "cracky" MOH recently.
> > >> Every time I have called in (making them
> random) I
> > >> have yet to replicate the
> > >> issue. Some initial research shows this to
> happen on
> > >> Call Manager under
> > >> heavy loads. I'm wondering if the router is
> spiking
> > >> for some reason but
> > >> knowing what I do about the company I can't see
> this
> > >> as being an issue,
> > >> although I will probably set up some generic
> > >> monitoring just in case.
> > >>
> > >> Has anyone out there heard of this phantom
> "cracky"
> > >> MOH in CME? If so what
> > >> have you found to be the cause? As a note, this
> is a
> > >> new occurrence. The
> > >> system has been up for over a year without
> anyone
> > >> mentioning this.
> > >>
> > >
> >
> >__________________________________________________
> > >Do You Yahoo!?
> > >Tired of spam?  Yahoo! Mail has the best spam
> protection around
> > >http://mail.yahoo.com
> 
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From svoll.voip at gmail.com  Tue Oct 23 11:29:31 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 23 Oct 2007 08:29:31 -0700
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <471D10B6.1010707@gmail.com>
References: <471D10B6.1010707@gmail.com>
Message-ID: <f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com>

WHY?  What are you doing with the AUX port that you need pin out?

Scott


On 10/22/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
>
> I'm trying to connect to the AUX port; I'm assuming that it's a standard
> serial connection, but I can't find a pinout.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From svoll.voip at gmail.com  Tue Oct 23 11:33:30 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 23 Oct 2007 08:33:30 -0700
Subject: [cisco-voip] Delay when calling one extension - CCM 4.1(3)sr5b
In-Reply-To: <f4445faf0710222310j3ccb261em7d9b946c2d4bb0bd@mail.gmail.com>
References: <f4445faf0710222310j3ccb261em7d9b946c2d4bb0bd@mail.gmail.com>
Message-ID: <f84a38d30710230833v168dfa12sd1c790ab124c8eee@mail.gmail.com>

have you looked at your DNA? Https://cmpub/dna <https://cmpub/dna>


It has to be matching something else.  I just had this problem with 911.  I
have a 9.1xxxxxxxxxx.  <- bad form but I was testing a new system and forgot
about it.

Scott


On 10/22/07, Erick Bergquist <erickbee at gmail.com> wrote:
>
> Hi,
>
> I am having an issue where we get the interdigit timeout delay when
> dialing just one extension, 4222. We can dial 4223, 4224, 4225 without
> delay. The CCM Detailed traces show potential matches exist but there
> are no more matches anywhere to be found. I have checked the
> NumPlan.CSV file and there are no translation patterns, no route
> patterns, etc that also match. I have deleted the extension, put it
> back and no change. I checked call park, meet me, and other numbers in
> the system and nothing conflicts or overlaps this.
>
> DNA doesn't show more then one match either.
>
> Is there a way to find the other potential match from a trace file?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From svoll.voip at gmail.com  Tue Oct 23 11:38:54 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 23 Oct 2007 08:38:54 -0700
Subject: [cisco-voip] "cracky MOH"
In-Reply-To: <776151.31165.qm@web57213.mail.re3.yahoo.com>
References: <006301c81585$3e9699f0$3900490a@acg.arlington.local>
	<776151.31165.qm@web57213.mail.re3.yahoo.com>
Message-ID: <f84a38d30710230838p5a040a93q7a1795a40bbcbf6c@mail.gmail.com>

if they are calling on a cell phone and get MOH then I could see this
happening.  Cell phones were never designed to get music.  they were
designed for voice.

Scott


On 10/23/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> That could be a myriad of issues IMHO. For all you
> know, most everything that caller is listening to is
> going to be crackly. If it was *internal*, then maybe
> you'd have better luck working on such a problem.
>
> Paul
>
> --- Curt Shaffer <cshaffer at gmail.com> wrote:
>
> > caller
> >
> > > >-----Original Message-----
> > > >From: Paul Choi [mailto:asobihoudai at yahoo.com]
> > > >Sent: Tuesday, October 23, 2007 10:54 AM
> > > >To: Curt Shaffer; cisco-voip at puck.nether.net
> > > >Subject: Re: [cisco-voip] "cracky MOH"
> > > >
> > > >Curt,
> > > >
> > > >Is this the caller complaining or the callee?
> > > >
> > > >
> > > >
> > > >
> > > >--- Curt Shaffer <cshaffer at gmail.com> wrote:
> > > >
> > > >> I have a client running CME and they have
> > reported
> > > >> "cracky" MOH recently.
> > > >> Every time I have called in (making them
> > random) I
> > > >> have yet to replicate the
> > > >> issue. Some initial research shows this to
> > happen on
> > > >> Call Manager under
> > > >> heavy loads. I'm wondering if the router is
> > spiking
> > > >> for some reason but
> > > >> knowing what I do about the company I can't see
> > this
> > > >> as being an issue,
> > > >> although I will probably set up some generic
> > > >> monitoring just in case.
> > > >>
> > > >> Has anyone out there heard of this phantom
> > "cracky"
> > > >> MOH in CME? If so what
> > > >> have you found to be the cause? As a note, this
> > is a
> > > >> new occurrence. The
> > > >> system has been up for over a year without
> > anyone
> > > >> mentioning this.
> > > >>
> > > >
> > >
> > >__________________________________________________
> > > >Do You Yahoo!?
> > > >Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> > > >http://mail.yahoo.com
> >
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From rkulagow at gmail.com  Tue Oct 23 11:51:27 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Tue, 23 Oct 2007 10:51:27 -0500
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com>
References: <471D10B6.1010707@gmail.com>
	<f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com>
Message-ID: <471E187F.2020201@gmail.com>

Scott Voll wrote:
> WHY?  What are you doing with the AUX port that you need pin out?

Because it's there?  Because older versions of the admin guide for the 
7970 tell you the pinout of the ethernet ports (gah!), but not the 
serial port?  Plain-old "Hrmm, what's that for?"?

See?  Lots of reasons!

From erickbee at gmail.com  Tue Oct 23 11:59:22 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Tue, 23 Oct 2007 10:59:22 -0500
Subject: [cisco-voip] Delay when calling one extension - CCM 4.1(3)sr5b
In-Reply-To: <f84a38d30710230833v168dfa12sd1c790ab124c8eee@mail.gmail.com>
References: <f4445faf0710222310j3ccb261em7d9b946c2d4bb0bd@mail.gmail.com>
	<f84a38d30710230833v168dfa12sd1c790ab124c8eee@mail.gmail.com>
Message-ID: <f4445faf0710230859pcc60dc0ie2b2ae6140eae161@mail.gmail.com>

Yep, DNA does not show any more potential matches other then the one
phone number.

I have 3 people comb through this and the numplan.csv and there are no
other matches.

Going to reboot the cluster next when we get an window.

On 10/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> have you looked at your DNA? Https://cmpub/dna
>
>
> It has to be matching something else.  I just had this problem with 911.  I
> have a 9.1xxxxxxxxxx.  <- bad form but I was testing a new system and forgot
> about it.
>
> Scott
>
>
> On 10/22/07, Erick Bergquist <erickbee at gmail.com> wrote:
> >
> > Hi,
> >
> > I am having an issue where we get the interdigit timeout delay when
> > dialing just one extension, 4222. We can dial 4223, 4224, 4225 without
> > delay. The CCM Detailed traces show potential matches exist but there
> > are no more matches anywhere to be found. I have checked the
> > NumPlan.CSV file and there are no translation patterns, no route
> > patterns, etc that also match. I have deleted the extension, put it
> > back and no change. I checked call park, meet me, and other numbers in
> > the system and nothing conflicts or overlaps this.
> >
> > DNA doesn't show more then one match either.
> >
> > Is there a way to find the other potential match from a trace file?
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From voicenoob at gmail.com  Tue Oct 23 12:02:38 2007
From: voicenoob at gmail.com (Voice Noob)
Date: Tue, 23 Oct 2007 11:02:38 -0500
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <471E187F.2020201@gmail.com>
References: <471D10B6.1010707@gmail.com>
	<f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com>
	<471E187F.2020201@gmail.com>
Message-ID: <535625f70710230902w73c4d474g6260caa0ad29cac7@mail.gmail.com>

"Because it's there"

WOW I wish I was an admin somewhere that didn't have anything better to do
with my day than look up pinouts. What a life.


On 10/23/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
>
> Scott Voll wrote:
> > WHY?  What are you doing with the AUX port that you need pin out?
>
> Because it's there?  Because older versions of the admin guide for the
> 7970 tell you the pinout of the ethernet ports (gah!), but not the
> serial port?  Plain-old "Hrmm, what's that for?"?
>
> See?  Lots of reasons!
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From aman.chugh at gmail.com  Tue Oct 23 12:05:53 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Tue, 23 Oct 2007 21:35:53 +0530
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <535625f70710230902w73c4d474g6260caa0ad29cac7@mail.gmail.com>
References: <471D10B6.1010707@gmail.com>
	<f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com>
	<471E187F.2020201@gmail.com>
	<535625f70710230902w73c4d474g6260caa0ad29cac7@mail.gmail.com>
Message-ID: <d4c697ad0710230905v284d45ci80255e256deb8883@mail.gmail.com>

Yes , it was there.

You can use it to telnet into the switch in the phone and do some fancy
stuff like advanced troubleshooting and even reset the phone to defaults, I
had done it to bring a dead 7970 alive.

Aman


On 10/23/07, Voice Noob <voicenoob at gmail.com> wrote:
>
> "Because it's there"
>
> WOW I wish I was an admin somewhere that didn't have anything better to do
> with my day than look up pinouts. What a life.
>
>
>  On 10/23/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
> >
> > Scott Voll wrote:
> > > WHY?  What are you doing with the AUX port that you need pin out?
> >
> > Because it's there?  Because older versions of the admin guide for the
> > 7970 tell you the pinout of the ethernet ports (gah!), but not the
> > serial port?  Plain-old "Hrmm, what's that for?"?
> >
> > See?  Lots of reasons!
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From rratliff at cisco.com  Tue Oct 23 12:07:20 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 23 Oct 2007 12:07:20 -0400
Subject: [cisco-voip] how to change Span to PC Port configuration in a
	7941 phone
In-Reply-To: <A805E226FF74D74E90D5BE19D664A9A0137F26@neptuno.atminformatica.pt>
References: <A805E226FF74D74E90D5BE19D664A9A0137F26@neptuno.atminformatica.pt>
Message-ID: <71760F9C-CE88-447D-A797-4B1AF29F88EC@cisco.com>

I found that 4.2(3) with no SRs did not show the "span to PC port"  
for the 41/61 phones but did for the 7970s.  After upgrading to SR3  
it is fixed.

-Ryan

On Oct 22, 2007, at 5:00 AM, Pedro Nogueira wrote:

Hello,

i know where to change it in the web pages, my problem is that it  
dont appear in the admin conf page (CCM), neither in the
phone admin page, in the 7911 model it appears, but in the 7941 or  
7961 model, dont.
Any sugestions or ideas ?

Pedro Nogueira
Dep. T?cnico
Sicnet



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From rratliff at cisco.com  Tue Oct 23 12:09:00 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 23 Oct 2007 12:09:00 -0400
Subject: [cisco-voip] Delay when calling one extension - CCM 4.1(3)sr5b
In-Reply-To: <f4445faf0710230859pcc60dc0ie2b2ae6140eae161@mail.gmail.com>
References: <f4445faf0710222310j3ccb261em7d9b946c2d4bb0bd@mail.gmail.com>
	<f84a38d30710230833v168dfa12sd1c790ab124c8eee@mail.gmail.com>
	<f4445faf0710230859pcc60dc0ie2b2ae6140eae161@mail.gmail.com>
Message-ID: <47750682-4E00-4D69-A7BB-C323C3DEA3BE@cisco.com>

Try changing the CM service parameter "digit analysis complexity" to  
"TranslationAndAlternatePatternAnalysis" and then look at the CCM  
trace when you make the call.  This should give you info on the  
potential match.

-Ryan

On Oct 23, 2007, at 11:59 AM, Erick Bergquist wrote:

Yep, DNA does not show any more potential matches other then the one
phone number.

I have 3 people comb through this and the numplan.csv and there are no
other matches.

Going to reboot the cluster next when we get an window.

On 10/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> have you looked at your DNA? Https://cmpub/dna
>
>
> It has to be matching something else.  I just had this problem with  
> 911.  I
> have a 9.1xxxxxxxxxx.  <- bad form but I was testing a new system  
> and forgot
> about it.
>
> Scott
>
>
> On 10/22/07, Erick Bergquist <erickbee at gmail.com> wrote:
>>
>> Hi,
>>
>> I am having an issue where we get the interdigit timeout delay when
>> dialing just one extension, 4222. We can dial 4223, 4224, 4225  
>> without
>> delay. The CCM Detailed traces show potential matches exist but there
>> are no more matches anywhere to be found. I have checked the
>> NumPlan.CSV file and there are no translation patterns, no route
>> patterns, etc that also match. I have deleted the extension, put it
>> back and no change. I checked call park, meet me, and other  
>> numbers in
>> the system and nothing conflicts or overlaps this.
>>
>> DNA doesn't show more then one match either.
>>
>> Is there a way to find the other potential match from a trace file?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From wsisk at cisco.com  Tue Oct 23 12:19:19 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 23 Oct 2007 12:19:19 -0400
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <471D10B6.1010707@gmail.com>
References: <471D10B6.1010707@gmail.com>
Message-ID: <471E1F07.1080900@cisco.com>

it's the same as the 7960 console port pinout.  7960 console cable will 
work with it.  that doc/pinout has been posted out here a few times. /wes

Robert Kulagowski wrote:
> I'm trying to connect to the AUX port; I'm assuming that it's a standard 
> serial connection, but I can't find a pinout.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From rsingleton at morsco.com  Tue Oct 23 12:13:14 2007
From: rsingleton at morsco.com (Robert)
Date: Tue, 23 Oct 2007 11:13:14 -0500
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <535625f70710230902w73c4d474g6260caa0ad29cac7@mail.gmail.com>
References: <471D10B6.1010707@gmail.com>
	<f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com>
	<471E187F.2020201@gmail.com>
	<535625f70710230902w73c4d474g6260caa0ad29cac7@mail.gmail.com>
Message-ID: <1193155994.15226.30.camel@robert.morsco.com>

On Tue, 2007-10-23 at 11:02 -0500, Voice Noob wrote:

> WOW I wish I was an admin somewhere that didn't have anything better
> to do with my day than look up pinouts. What a life.

As an admin, I'm expected to find better, faster and cheaper ways to do
things and ignorance of the hardware at my fingertips does not further
that task.

Sadly, I do possess a lot of ignorance about the hardware at my
fingertips, but I do what I can... :)

Robert



From kenjohnson at letu.edu  Tue Oct 23 13:22:29 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Tue, 23 Oct 2007 12:22:29 -0500
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <1193155994.15226.30.camel@robert.morsco.com>
References: <471D10B6.1010707@gmail.com><f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com><471E187F.2020201@gmail.com><535625f70710230902w73c4d474g6260caa0ad29cac7@mail.gmail.com>
	<1193155994.15226.30.camel@robert.morsco.com>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EFD0BD6A@rachael-reborn.letnet.net>

For instance - constructing a longer 7914 cable (whether supported ot
not) can provide some flexibility for users. When such cables aren't
straight-through, (which actually I think the 14 cable might be, just
can't recall from the last time we made one) knowledge of how the serial
ports are used *can* prove useful on at least sporadic occasion.

While random fiddling with things may seem like a waste, and sometimes
may be, that's up to the employer to decide. As has been noted, asking
questions about apparently unused or underusued features on equipment
can (though does not always) provide a ROI.

Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
Cisco IP Telecommunications Users Group
http://www.ciptug.org
_________________________________

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Sent: Tuesday, October 23, 2007 11:13 AM
To: Cisco Voip Mailing list
Subject: Re: [cisco-voip] 7970 AUX port pinout?

On Tue, 2007-10-23 at 11:02 -0500, Voice Noob wrote:

> WOW I wish I was an admin somewhere that didn't have anything better
> to do with my day than look up pinouts. What a life.

As an admin, I'm expected to find better, faster and cheaper ways to do
things and ignorance of the hardware at my fingertips does not further
that task.

Sadly, I do possess a lot of ignorance about the hardware at my
fingertips, but I do what I can... :)

Robert


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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Spam:
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From tim.reimers at asheville.k12.nc.us  Tue Oct 23 13:31:57 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue, 23 Oct 2007 13:31:57 -0400
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <471E187F.2020201@gmail.com>
References: <471D10B6.1010707@gmail.com><f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com>
	<471E187F.2020201@gmail.com>
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1AF7C@webmail.asheville.k12.nc.us>

The good old reason....

Because it's there.

And who knows --- you might even pull off something like this:
http://www.trygve.com/furbeowulf.html


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Kulagowski
Sent: Tuesday, October 23, 2007 11:51 AM
To: Scott Voll
Cc: Cisco Voip Mailing list
Subject: Re: [cisco-voip] 7970 AUX port pinout?

Scott Voll wrote:
> WHY?  What are you doing with the AUX port that you need pin out?

Because it's there?  Because older versions of the admin guide for the 
7970 tell you the pinout of the ethernet ports (gah!), but not the 
serial port?  Plain-old "Hrmm, what's that for?"?

See?  Lots of reasons!
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From svoll.voip at gmail.com  Tue Oct 23 14:08:37 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 23 Oct 2007 11:08:37 -0700
Subject: [cisco-voip] Unity VMO
Message-ID: <f84a38d30710231108y3d4e6063t9caf275d8e92fc1e@mail.gmail.com>

I'm running Unity 4.0.5 until I upgrade later this year.

we run UM and use the VMO.  This works great on Office 2k3 or lower.  Is
there support for Office 2k7 or do I have to upgrade to Unity 5 to get
support.

TIA

Scott
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From MLinsemier at apcapital.com  Tue Oct 23 14:18:50 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 23 Oct 2007 14:18:50 -0400
Subject: [cisco-voip] Unity VMO
In-Reply-To: <f84a38d30710231108y3d4e6063t9caf275d8e92fc1e@mail.gmail.com>
References: <f84a38d30710231108y3d4e6063t9caf275d8e92fc1e@mail.gmail.com>
Message-ID: <AB0A35F2486F81468971AA68558BE719074B4C68@smieln01winm01.micoanet.com>

I'm using the latest VMO with Vista, Office 2007, and Unity 4.2(1) and
it seems to be working fine for me.

 

Matt

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Tuesday, October 23, 2007 2:09 PM
To: cisco voip list
Subject: [cisco-voip] Unity VMO

 

I'm running Unity 4.0.5 until I upgrade later this year.

 

we run UM and use the VMO.  This works great on Office 2k3 or lower.  Is
there support for Office 2k7 or do I have to upgrade to Unity 5 to get
support.

 

TIA

 

Scott


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From rkulagow at gmail.com  Tue Oct 23 15:06:07 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Tue, 23 Oct 2007 14:06:07 -0500
Subject: [cisco-voip] Daylight Savings Time issues...
Message-ID: <471E461F.7040705@gmail.com>

Well, it's getting to be that time of the year again.

Has anyone noticed that 
http://www.cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml#voice_products

was updated earlier this month, and that CM and phone loads are affected 
for the switch back from DST?  Maybe I'm reading it wrong, but it looks 
like I've got another upgrade cycle to go through.

Yay.

From rkulagow at gmail.com  Tue Oct 23 15:06:17 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Tue, 23 Oct 2007 14:06:17 -0500
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <471E1F07.1080900@cisco.com>
References: <471D10B6.1010707@gmail.com> <471E1F07.1080900@cisco.com>
Message-ID: <471E4629.80203@gmail.com>

Wes Sisk wrote:
> it's the same as the 7960 console port pinout.  7960 console cable will 
> work with it.  that doc/pinout has been posted out here a few times. /wes

I must be using the wrong search terms over at gossamer-threads.com, 
because "7960 console cable", "console cable", "7960 rs232" came up with 
nada.

Any pointers / better search terms?  Is the port 9600 n-8-1?  Does this 
work with SCCP?

From wsisk at cisco.com  Tue Oct 23 15:16:32 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 23 Oct 2007 15:16:32 -0400
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <471E4629.80203@gmail.com>
References: <471D10B6.1010707@gmail.com> <471E1F07.1080900@cisco.com>
	<471E4629.80203@gmail.com>
Message-ID: <471E4890.3030301@cisco.com>

http://www.cisco.com/en/US/customer/tech/tk652/tk701/technologies_configuration_guide_chapter09186a00800eadf9.html#67513

/Wes

Robert Kulagowski wrote:
> Wes Sisk wrote:
>> it's the same as the 7960 console port pinout.  7960 console cable 
>> will work with it.  that doc/pinout has been posted out here a few 
>> times. /wes
>
> I must be using the wrong search terms over at gossamer-threads.com, 
> because "7960 console cable", "console cable", "7960 rs232" came up 
> with nada.
>
> Any pointers / better search terms?  Is the port 9600 n-8-1?  Does 
> this work with SCCP?

From wsisk at cisco.com  Tue Oct 23 15:18:07 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 23 Oct 2007 15:18:07 -0400
Subject: [cisco-voip] cucm 6 and TAPS
In-Reply-To: <535625f70710230602g4550f9dx5ffd3f5ab53df059@mail.gmail.com>
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>	
	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>	
	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>	
	<22e6380b0710182002p30265d7do404b0f2d9b144842@mail.gmail.com>	
	<723FA17C6AA4C4428CDF572DCAEE08D405445C4D@EXCHANGEDS01.ds.ad.adp.com>	
	<4718E3CE.20403@cisco.com>	
	<c6f32ce40710230527ub3a9e1bl2eb60f7a651f5c9e@mail.gmail.com>
	<535625f70710230602g4550f9dx5ffd3f5ab53df059@mail.gmail.com>
Message-ID: <471E48EF.7070202@cisco.com>

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From wsisk at cisco.com  Tue Oct 23 15:22:56 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 23 Oct 2007 15:22:56 -0400
Subject: [cisco-voip] Different ring types for internal/external calls
In-Reply-To: <6A0744639B86DD459433108DBF101D26019FE50B@MAIL01.sentinel.com>
References: <NYEXM4domP0l0HQFDMz00017f35@NYEXMB2.shearman.com>
	<643A3A543DB5224499BC3D73473AEC27869E63@NYEXMB2.shearman.com>
	<6A0744639B86DD459433108DBF101D26019FE50B@MAIL01.sentinel.com>
Message-ID: <471E4A10.4050400@cisco.com>

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From Erik.Stillman at Shearman.com  Mon Oct 22 17:11:06 2007
From: Erik.Stillman at Shearman.com (Erik Stillman)
Date: Mon, 22 Oct 2007 17:11:06 -0400
Subject: [cisco-voip] SRST
In-Reply-To: <NYEXM40V261ZAbhRpXh0001998d@NYEXMB2.shearman.com>
Message-ID: <643A3A543DB5224499BC3D73473AEC2792160D@NYEXMB2.shearman.com>

Hey guys, is it possible to test one single phone on SRST while the rest
of the site is registered to CallManager?

I can get the test registered with the gateway, but it won't dial out. 
Does this have something got do with the PRI being MGCP controlled?

My goal is to do some testing without having to take the site down. 

All input appreciated!

Many thanks,
Erik

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net@SHSDOMAIN On Behalf Of
cisco-voip-request at puck.nether.net
Sent: Monday, October 22, 2007 4:13 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 56, Issue 135

Send cisco-voip mailing list submissions to
	cisco-voip at puck.nether.net

To subscribe or unsubscribe via the World Wide Web, visit
	https://puck.nether.net/mailman/listinfo/cisco-voip
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You can reach the person managing the list at
	cisco-voip-owner at puck.nether.net

When replying, please edit your Subject line so it is more specific than
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***********************************************************************

This communication is not intended to be used, and cannot be used,
by the recipient or any other person for the purpose of avoiding United
States federal tax penalties that may be imposed on the recipient or
such other person. In addition, if any United States federal tax
advice contained in this communication is used or referred to in
promoting, marketing or recommending any corporation, partnership
or other entity, investment plan, concept, structure or arrangement
(which should be assumed to be the case by a recipient or other
person who is not our client with respect to the subject matter of the
communication), then (i) such tax advice should be construed as written
to support the promotion or marketing of the transactions or matters
addressed by the advice and (ii) the recipient or other person should
seek advice based on the recipient or other person's particular
circumstances from an independent tax advisor. For further information,
please go to http://www.shearman.com/disclaimer/tax_disclosure.html

***********************************************************************

This transmittal and/or attachments may be a confidential
attorney-client communication or may otherwise be privileged or
confidential. If you are not the intended recipient, you are hereby
notified that you have received this transmittal in error; any review,
dissemination, distribution or copying of this transmittal is strictly
prohibited. If you have received this transmittal and/or attachments in
error, please notify us immediately by reply or by telephone (call us
collect at +1 212-848-8400) and immediately delete this message and
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Shearman & Sterling LLP is a limited liability partnership organized in
the United States under the laws of the State of Delaware, which laws
limit the personal liability of partners.

From Andrew.Riley at getronics.com  Tue Oct 23 15:28:52 2007
From: Andrew.Riley at getronics.com (Riley, Andrew)
Date: Wed, 24 Oct 2007 05:28:52 +1000
Subject: [cisco-voip] SRST
In-Reply-To: <643A3A543DB5224499BC3D73473AEC2792160D@NYEXMB2.shearman.com>
References: <NYEXM40V261ZAbhRpXh0001998d@NYEXMB2.shearman.com>
	<643A3A543DB5224499BC3D73473AEC2792160D@NYEXMB2.shearman.com>
Message-ID: <A49371C6FF101B46BD3E3A8C89985F02E42AA1@excausy200.australia.unity>

Put an ACL in the network to force the one phone to register to SRST.

If you are doing MGCP, then the ISDN interface is under the control of
CCM and won't work with the dial peers.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Stillman
Sent: Tuesday, 23 October 2007 7:11 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SRST

Hey guys, is it possible to test one single phone on SRST while the rest
of the site is registered to CallManager?

I can get the test registered with the gateway, but it won't dial out. 
Does this have something got do with the PRI being MGCP controlled?

My goal is to do some testing without having to take the site down. 

All input appreciated!

Many thanks,
Erik

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net@SHSDOMAIN On Behalf Of
cisco-voip-request at puck.nether.net
Sent: Monday, October 22, 2007 4:13 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 56, Issue 135

Send cisco-voip mailing list submissions to
	cisco-voip at puck.nether.net

To subscribe or unsubscribe via the World Wide Web, visit
	https://puck.nether.net/mailman/listinfo/cisco-voip
or, via email, send a message with subject or body 'help' to
	cisco-voip-request at puck.nether.net

You can reach the person managing the list at
	cisco-voip-owner at puck.nether.net

When replying, please edit your Subject line so it is more specific than
"Re: Contents of cisco-voip digest..."


***********************************************************************

This communication is not intended to be used, and cannot be used,
by the recipient or any other person for the purpose of avoiding United
States federal tax penalties that may be imposed on the recipient or
such other person. In addition, if any United States federal tax
advice contained in this communication is used or referred to in
promoting, marketing or recommending any corporation, partnership
or other entity, investment plan, concept, structure or arrangement
(which should be assumed to be the case by a recipient or other
person who is not our client with respect to the subject matter of the
communication), then (i) such tax advice should be construed as written
to support the promotion or marketing of the transactions or matters
addressed by the advice and (ii) the recipient or other person should
seek advice based on the recipient or other person's particular
circumstances from an independent tax advisor. For further information,
please go to http://www.shearman.com/disclaimer/tax_disclosure.html

***********************************************************************

This transmittal and/or attachments may be a confidential
attorney-client communication or may otherwise be privileged or
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notified that you have received this transmittal in error; any review,
dissemination, distribution or copying of this transmittal is strictly
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error, please notify us immediately by reply or by telephone (call us
collect at +1 212-848-8400) and immediately delete this message and
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Shearman & Sterling LLP is a limited liability partnership organized in
the United States under the laws of the State of Delaware, which laws
limit the personal liability of partners.
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From Erik.Stillman at Shearman.com  Tue Oct 23 15:30:42 2007
From: Erik.Stillman at Shearman.com (Erik Stillman)
Date: Tue, 23 Oct 2007 15:30:42 -0400
Subject: [cisco-voip] SRST
In-Reply-To: <NYEXM4Cla2kdlV7wGHc0001a302@NYEXMB2.shearman.com>
Message-ID: <643A3A543DB5224499BC3D73473AEC27921827@NYEXMB2.shearman.com>

Thanks Andrew, it's exactly what I did.


Anyway to get those dial-peers working with like an FXO? 

-----Original Message-----
From: "Riley, Andrew" <Andrew.Riley at getronics.com>@SHSDOMAIN 
Sent: Tuesday, October 23, 2007 3:29 PM
To: Erik Stillman; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] SRST

Put an ACL in the network to force the one phone to register to SRST.

If you are doing MGCP, then the ISDN interface is under the control of
CCM and won't work with the dial peers.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Stillman
Sent: Tuesday, 23 October 2007 7:11 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SRST

Hey guys, is it possible to test one single phone on SRST while the rest
of the site is registered to CallManager?

I can get the test registered with the gateway, but it won't dial out. 
Does this have something got do with the PRI being MGCP controlled?

My goal is to do some testing without having to take the site down. 

All input appreciated!

Many thanks,
Erik

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net@SHSDOMAIN On Behalf Of
cisco-voip-request at puck.nether.net
Sent: Monday, October 22, 2007 4:13 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 56, Issue 135

Send cisco-voip mailing list submissions to
	cisco-voip at puck.nether.net

To subscribe or unsubscribe via the World Wide Web, visit
	https://puck.nether.net/mailman/listinfo/cisco-voip
or, via email, send a message with subject or body 'help' to
	cisco-voip-request at puck.nether.net

You can reach the person managing the list at
	cisco-voip-owner at puck.nether.net

When replying, please edit your Subject line so it is more specific than
"Re: Contents of cisco-voip digest..."


***********************************************************************

This communication is not intended to be used, and cannot be used, by
the recipient or any other person for the purpose of avoiding United
States federal tax penalties that may be imposed on the recipient or
such other person. In addition, if any United States federal tax advice
contained in this communication is used or referred to in promoting,
marketing or recommending any corporation, partnership or other entity,
investment plan, concept, structure or arrangement (which should be
assumed to be the case by a recipient or other person who is not our
client with respect to the subject matter of the communication), then
(i) such tax advice should be construed as written to support the
promotion or marketing of the transactions or matters addressed by the
advice and (ii) the recipient or other person should seek advice based
on the recipient or other person's particular circumstances from an
independent tax advisor. For further information, please go to
http://www.shearman.com/disclaimer/tax_disclosure.html

***********************************************************************

This transmittal and/or attachments may be a confidential
attorney-client communication or may otherwise be privileged or
confidential. If you are not the intended recipient, you are hereby
notified that you have received this transmittal in error; any review,
dissemination, distribution or copying of this transmittal is strictly
prohibited. If you have received this transmittal and/or attachments in
error, please notify us immediately by reply or by telephone (call us
collect at +1 212-848-8400) and immediately delete this message and all
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Shearman & Sterling LLP is a limited liability partnership organized in
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limit the personal liability of partners.
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip


From wsisk at cisco.com  Tue Oct 23 15:42:01 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 23 Oct 2007 15:42:01 -0400
Subject: [cisco-voip] SRST
In-Reply-To: <643A3A543DB5224499BC3D73473AEC2792160D@NYEXMB2.shearman.com>
References: <643A3A543DB5224499BC3D73473AEC2792160D@NYEXMB2.shearman.com>
Message-ID: <471E4E89.9010707@cisco.com>

apply an ACL that blocks sccp and both mgcp ports from the remote site 
to CM.
default ports:
sccp: tcp 2000
MGCP            2427-listen/ka            UDP
                        2428-q931 backhaul        TCP

/Wes

Erik Stillman wrote:
> Hey guys, is it possible to test one single phone on SRST while the rest
> of the site is registered to CallManager?
>
> I can get the test registered with the gateway, but it won't dial out. 
> Does this have something got do with the PRI being MGCP controlled?
>
> My goal is to do some testing without having to take the site down. 
>
> All input appreciated!
>
> Many thanks,
> Erik
>
>  
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net@SHSDOMAIN On Behalf Of
> cisco-voip-request at puck.nether.net
> Sent: Monday, October 22, 2007 4:13 PM
> To: cisco-voip at puck.nether.net
> Subject: cisco-voip Digest, Vol 56, Issue 135
>
> Send cisco-voip mailing list submissions to
> 	cisco-voip at puck.nether.net
>
> To subscribe or unsubscribe via the World Wide Web, visit
> 	https://puck.nether.net/mailman/listinfo/cisco-voip
> or, via email, send a message with subject or body 'help' to
> 	cisco-voip-request at puck.nether.net
>
> You can reach the person managing the list at
> 	cisco-voip-owner at puck.nether.net
>
> When replying, please edit your Subject line so it is more specific than
> "Re: Contents of cisco-voip digest..."
>
>
> ***********************************************************************
>
> This communication is not intended to be used, and cannot be used,
> by the recipient or any other person for the purpose of avoiding United
> States federal tax penalties that may be imposed on the recipient or
> such other person. In addition, if any United States federal tax
> advice contained in this communication is used or referred to in
> promoting, marketing or recommending any corporation, partnership
> or other entity, investment plan, concept, structure or arrangement
> (which should be assumed to be the case by a recipient or other
> person who is not our client with respect to the subject matter of the
> communication), then (i) such tax advice should be construed as written
> to support the promotion or marketing of the transactions or matters
> addressed by the advice and (ii) the recipient or other person should
> seek advice based on the recipient or other person's particular
> circumstances from an independent tax advisor. For further information,
> please go to http://www.shearman.com/disclaimer/tax_disclosure.html
>
> ***********************************************************************
>
> This transmittal and/or attachments may be a confidential
> attorney-client communication or may otherwise be privileged or
> confidential. If you are not the intended recipient, you are hereby
> notified that you have received this transmittal in error; any review,
> dissemination, distribution or copying of this transmittal is strictly
> prohibited. If you have received this transmittal and/or attachments in
> error, please notify us immediately by reply or by telephone (call us
> collect at +1 212-848-8400) and immediately delete this message and
> all its attachments.
>
> Shearman & Sterling LLP is a limited liability partnership organized in
> the United States under the laws of the State of Delaware, which laws
> limit the personal liability of partners.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From cisco1.charles at gmail.com  Tue Oct 23 15:47:08 2007
From: cisco1.charles at gmail.com (charles george)
Date: Tue, 23 Oct 2007 15:47:08 -0400
Subject: [cisco-voip] ICM NFR SOFTWARE
Message-ID: <9552cdf0710231247x674e7c93l8a5f017aa839139c@mail.gmail.com>

Hello,

Can I purchase from anyone on tis site this part number ICME-NPSENT-DART=


-- 
Regards,
Charles
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From pat-cv at wcyv.com  Tue Oct 23 17:28:13 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Tue, 23 Oct 2007 17:28:13 -0400
Subject: [cisco-voip] Unity VMO
In-Reply-To: <AB0A35F2486F81468971AA68558BE719074B4C68@smieln01winm01.micoanet.com>
References: <f84a38d30710231108y3d4e6063t9caf275d8e92fc1e@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE719074B4C68@smieln01winm01.micoanet.com>
Message-ID: <471E676D.9040009@wcyv.com>

Check the docs, this one covers pretty much all possible combinations:

http://www.cisco.com/en/US/docs/voice_ip_comm/unity/compatibility/matrix/cuclientmtx.html


-------- Original Message  --------
From: "Linsemier, Matthew" <MLinsemier at apcapital.com>
To: "Scott Voll" <svoll.voip at gmail.com>, "cisco voip list" 
<cisco-voip at puck.nether.net>
Subject: Re:[cisco-voip] Unity VMO
Date: 10/23/2007 2:18 PM

> I'm using the latest VMO with Vista, Office 2007, and Unity 4.2(1) and
> it seems to be working fine for me.
> 
>  
> 
> Matt
> 
>  
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
> Sent: Tuesday, October 23, 2007 2:09 PM
> To: cisco voip list
> Subject: [cisco-voip] Unity VMO
> 
>  
> 
> I'm running Unity 4.0.5 until I upgrade later this year.
> 
>  
> 
> we run UM and use the VMO.  This works great on Office 2k3 or lower.  Is
> there support for Office 2k7 or do I have to upgrade to Unity 5 to get
> support.
> 
>  
> 
> TIA
> 
>  
> 
> Scott
> 
> 
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may
> be protected by one or more legal privileges. It is intended
> solely for the use of the addressee identified above. If you
> are not the intended recipient, any use, disclosure, copying
> or distribution of this communication is UNAUTHORIZED. Neither
> this information block, the typed name of the sender, nor
> anything else in this message is intended to constitute an
> electronic signature unless a specific statement to the
> contrary is included in this message. If you have received this
> communication in error, please immediately contact me and delete
> this communication from your computer. Thank you.
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From erickbee at gmail.com  Tue Oct 23 17:44:23 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Tue, 23 Oct 2007 16:44:23 -0500
Subject: [cisco-voip] Delay when calling one extension - CCM 4.1(3)sr5b
In-Reply-To: <47750682-4E00-4D69-A7BB-C323C3DEA3BE@cisco.com>
References: <f4445faf0710222310j3ccb261em7d9b946c2d4bb0bd@mail.gmail.com>
	<f84a38d30710230833v168dfa12sd1c790ab124c8eee@mail.gmail.com>
	<f4445faf0710230859pcc60dc0ie2b2ae6140eae161@mail.gmail.com>
	<47750682-4E00-4D69-A7BB-C323C3DEA3BE@cisco.com>
Message-ID: <f4445faf0710231444l39b3b528pff67a93bb621dd52@mail.gmail.com>

Ryan,

Ok, I'll keep that in mind for next time. Someone else rebooted the
servers early this morning which cleared the issue. Thanks.

On 10/23/07, Ryan Ratliff <rratliff at cisco.com> wrote:
> Try changing the CM service parameter "digit analysis complexity" to
> "TranslationAndAlternatePatternAnalysis" and then look at the CCM
> trace when you make the call.  This should give you info on the
> potential match.
>
> -Ryan
>
> On Oct 23, 2007, at 11:59 AM, Erick Bergquist wrote:
>
> Yep, DNA does not show any more potential matches other then the one
> phone number.
>
> I have 3 people comb through this and the numplan.csv and there are no
> other matches.
>
> Going to reboot the cluster next when we get an window.
>
> On 10/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > have you looked at your DNA? Https://cmpub/dna
> >
> >
> > It has to be matching something else.  I just had this problem with
> > 911.  I
> > have a 9.1xxxxxxxxxx.  <- bad form but I was testing a new system
> > and forgot
> > about it.
> >
> > Scott
> >
> >
> > On 10/22/07, Erick Bergquist <erickbee at gmail.com> wrote:
> >>
> >> Hi,
> >>
> >> I am having an issue where we get the interdigit timeout delay when
> >> dialing just one extension, 4222. We can dial 4223, 4224, 4225
> >> without
> >> delay. The CCM Detailed traces show potential matches exist but there
> >> are no more matches anywhere to be found. I have checked the
> >> NumPlan.CSV file and there are no translation patterns, no route
> >> patterns, etc that also match. I have deleted the extension, put it
> >> back and no change. I checked call park, meet me, and other
> >> numbers in
> >> the system and nothing conflicts or overlaps this.
> >>
> >> DNA doesn't show more then one match either.
> >>
> >> Is there a way to find the other potential match from a trace file?
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From svoll.voip at gmail.com  Tue Oct 23 17:59:37 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 23 Oct 2007 14:59:37 -0700
Subject: [cisco-voip] Unity VMO
In-Reply-To: <471E676D.9040009@wcyv.com>
References: <f84a38d30710231108y3d4e6063t9caf275d8e92fc1e@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE719074B4C68@smieln01winm01.micoanet.com>
	<471E676D.9040009@wcyv.com>
Message-ID: <f84a38d30710231459q664c9656v5c0bfc4a8aa0df3a@mail.gmail.com>

Thanks all.. It ended up being a DCOM issue.

Scott


On 10/23/07, Pat Hayes <pat-cv at wcyv.com> wrote:
>
> Check the docs, this one covers pretty much all possible combinations:
>
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/unity/compatibility/matrix/cuclientmtx.html
>
>
> -------- Original Message  --------
> From: "Linsemier, Matthew" <MLinsemier at apcapital.com>
> To: "Scott Voll" <svoll.voip at gmail.com>, "cisco voip list"
> <cisco-voip at puck.nether.net>
> Subject: Re:[cisco-voip] Unity VMO
> Date: 10/23/2007 2:18 PM
>
> > I'm using the latest VMO with Vista, Office 2007, and Unity 4.2(1) and
> > it seems to be working fine for me.
> >
> >
> >
> > Matt
> >
> >
> >
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
> > Sent: Tuesday, October 23, 2007 2:09 PM
> > To: cisco voip list
> > Subject: [cisco-voip] Unity VMO
> >
> >
> >
> > I'm running Unity 4.0.5 until I upgrade later this year.
> >
> >
> >
> > we run UM and use the VMO.  This works great on Office 2k3 or lower.  Is
> > there support for Office 2k7 or do I have to upgrade to Unity 5 to get
> > support.
> >
> >
> >
> > TIA
> >
> >
> >
> > Scott
> >
> >
> > CONFIDENTIALITY STATEMENT
> > This communication and any attachments are CONFIDENTIAL and may
> > be protected by one or more legal privileges. It is intended
> > solely for the use of the addressee identified above. If you
> > are not the intended recipient, any use, disclosure, copying
> > or distribution of this communication is UNAUTHORIZED. Neither
> > this information block, the typed name of the sender, nor
> > anything else in this message is intended to constitute an
> > electronic signature unless a specific statement to the
> > contrary is included in this message. If you have received this
> > communication in error, please immediately contact me and delete
> > this communication from your computer. Thank you.
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From erickbee at gmail.com  Tue Oct 23 18:06:42 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Tue, 23 Oct 2007 17:06:42 -0500
Subject: [cisco-voip] SRST
In-Reply-To: <471E4E89.9010707@cisco.com>
References: <643A3A543DB5224499BC3D73473AEC2792160D@NYEXMB2.shearman.com>
	<471E4E89.9010707@cisco.com>
Message-ID: <f4445faf0710231506u2f3c9d50ud756e16de40ebe6d@mail.gmail.com>

Keep in mind, that the ACL will tear down the MGCP connection
effecting all calls over the PRI . You also want to make sure MGCP
Fallback is configured and working so the voice-ports fallback to H323
mode (seperate function then SRST).

On 10/23/07, Wes Sisk <wsisk at cisco.com> wrote:
> apply an ACL that blocks sccp and both mgcp ports from the remote site
> to CM.
> default ports:
> sccp: tcp 2000
> MGCP            2427-listen/ka            UDP
>                         2428-q931 backhaul        TCP
>
> /Wes
>
> Erik Stillman wrote:
> > Hey guys, is it possible to test one single phone on SRST while the rest
> > of the site is registered to CallManager?
> >
> > I can get the test registered with the gateway, but it won't dial out.
> > Does this have something got do with the PRI being MGCP controlled?
> >
> > My goal is to do some testing without having to take the site down.
> >
> > All input appreciated!
> >
> > Many thanks,
> > Erik
> >
> >
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net@SHSDOMAIN On Behalf Of
> > cisco-voip-request at puck.nether.net
> > Sent: Monday, October 22, 2007 4:13 PM
> > To: cisco-voip at puck.nether.net
> > Subject: cisco-voip Digest, Vol 56, Issue 135
> >
> > Send cisco-voip mailing list submissions to
> >       cisco-voip at puck.nether.net
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From rleetun at co.boulder.co.us  Tue Oct 23 18:15:15 2007
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue, 23 Oct 2007 16:15:15 -0600
Subject: [cisco-voip] Connecting a Avaya G700 to a CUCM 4.1.3 environment
Message-ID: <963CE94787005149A52C17F33A028E120A3209BD@EVS1B.boco.co.boulder.co.us>

Hi,

Has anyone connected an Avaya G700 Gateway to a Cisco 4.1.3 environment?
If so, what are the steps?

Thanks much.

Rob
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From mattjhughes at gmail.com  Wed Oct 24 02:35:20 2007
From: mattjhughes at gmail.com (Matthew J. Hughes)
Date: Tue, 23 Oct 2007 23:35:20 -0700
Subject: [cisco-voip] Stumped with errors with AXL call using asp
Message-ID: <ea211ea30710232335n5dbf1d4ax77866fcfd2d926ba@mail.gmail.com>

HI guys,
I am running CCM6 and trying to run a test call to
https://myserver:8443/axl
i am using the API included text2base64 function to hash my username and
pass.
I keep getting a msxml parser error, I am thinking this is coming from my
webserver xml parser and not the ccm server?

msxml3.dll error 'c00ce505'
System error: -1072896763.

I'm pulling my hair out as I have been at this for 2 days now, I use to know
my way around ASP, but its been a while.
Anyone have experience with AXL on CCM6 that can shed some light?
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From pokkali at gmail.com  Wed Oct 24 05:21:55 2007
From: pokkali at gmail.com (Mohan Pokkali)
Date: Wed, 24 Oct 2007 10:21:55 +0100
Subject: [cisco-voip] External Callers can not leave long voice messages -
	UCM 6.0.1 Business Edition
References: <A56BD242744BF540B15D1672E0E3E87501DB2161@LON-INT-MSGVS01.internal.vancocorp.com>
	<043501c7faaa$0753df70$c313a8c0@hq.cct>
	<d4c697ad0709190351q596e7012x58bd1763a2b76900@mail.gmail.com> 
Message-ID: <019801c8161f$52b3c710$1f12fea9@hq.cct>

 

Hi All

 

Problem:                       Unity Connection (Part of Business Edition). 

 

External callers while leaving voice messages get cut of randomly after
sometime. There is no pattern. Some get cut off after 30 seconds, some after
35 seconds and some after 1 minute and 20 second. This happens across the
whole system affecting all users. There is no problem with internal callers.


 

Other Information:          MGCP Voice Gateway connecting to PSTN.

 

Thanks in advance for your help.

 

Mohan

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From Richard.vanBerkel at telindus.nl  Wed Oct 24 05:23:56 2007
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Wed, 24 Oct 2007 11:23:56 +0200
Subject: [cisco-voip] CCX4.1 changing Ip address
References: <723FA17C6AA4C4428CDF572DCAEE08D405445C3C@EXCHANGEDS01.ds.ad.adp.com>	<22e6380b0710181946h5e2535f7q71b14f9c5713d61f@mail.gmail.com>	<723FA17C6AA4C4428CDF572DCAEE08D405445C40@EXCHANGEDS01.ds.ad.adp.com>	<22e6380b0710182002p30265d7do404b0f2d9b144842@mail.gmail.com>	<723FA17C6AA4C4428CDF572DCAEE08D405445C4D@EXCHANGEDS01.ds.ad.adp.com>	<4718E3CE.20403@cisco.com>	<c6f32ce40710230527ub3a9e1bl2eb60f7a651f5c9e@mail.gmail.com><535625f70710230602g4550f9dx5ffd3f5ab53df059@mail.gmail.com>
	<471E48EF.7070202@cisco.com>
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157EC32@tnl1mx1.telindus.intra>

Hi,

 

Can anyone tell me were I can find the IPUpate.exe utility.

So I can change the Ip address of the CCX4.1 server.

 

I found a document that states the file should be in the following
directory:

\\CRS IP address\DESKTOP_CFG\Util folder.

 

 

Only this folder isn't there :-)

 

Groeten Richard,

Mailto:Richard.vanBerkel at Telindus.nl
<mailto:Richard.vanBerkel at Telindus.nl> 

  

 

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From pat-cv at wcyv.com  Wed Oct 24 07:44:04 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Wed, 24 Oct 2007 07:44:04 -0400
Subject: [cisco-voip] External Callers can not leave long voice messages
 -	UCM 6.0.1 Business Edition
In-Reply-To: <019801c8161f$52b3c710$1f12fea9@hq.cct>
References: <A56BD242744BF540B15D1672E0E3E87501DB2161@LON-INT-MSGVS01.internal.vancocorp.com>	<043501c7faaa$0753df70$c313a8c0@hq.cct>	<d4c697ad0709190351q596e7012x58bd1763a2b76900@mail.gmail.com>
	<019801c8161f$52b3c710$1f12fea9@hq.cct>
Message-ID: <471F3004.60109@wcyv.com>

Does the called user receive the message with the first however many 
seconds? Or do they get nothing? If they get nothing, verify the RTP 
path through your network and confirm that MGCP is bound to the 
interface that it should be. Packet captures help here. If they do get 
the first part of the message, check out:

http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K88174040

-------- Original Message  --------
From: "Mohan Pokkali" <pokkali at gmail.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] External Callers can not leave long voice 
messages -	UCM 6.0.1 Business Edition
Date: 10/24/2007 5:21 AM

>  
> 
> Hi All
> 
>  
> 
> Problem:                       Unity Connection (Part of Business Edition). 
> 
>  
> 
> External callers while leaving voice messages get cut of randomly after
> sometime. There is no pattern. Some get cut off after 30 seconds, some after
> 35 seconds and some after 1 minute and 20 second. This happens across the
> whole system affecting all users. There is no problem with internal callers.
> 
> 
>  
> 
> Other Information:          MGCP Voice Gateway connecting to PSTN.
> 
>  
> 
> Thanks in advance for your help.
> 
>  
> 
> Mohan
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From CarlosOrtiz at bayviewfinancial.com  Wed Oct 24 08:23:02 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 24 Oct 2007 08:23:02 -0400
Subject: [cisco-voip] CCX4.1 changing Ip address
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157EC32@tnl1mx1.telindus.intra>
Message-ID: <OF68D0E29F.BF24E7C9-ON8525737E.0043C18E-8525737E.00440CB0@bftg.com>

I did this a few months back but I followed the instructions in the CRS 
4.0X admin guide.

Chapter - Managing the Cisco CRS System > Cisco CRS Utilities section

This also requires you to update the IP address on all the agent PC's as 
well.

Carlos




"Richard van Berkel" <Richard.vanBerkel at telindus.nl> 
Sent by: cisco-voip-bounces at puck.nether.net
10/24/2007 05:23 AM

To
"cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] CCX4.1 changing Ip address






Hi,
 
Can anyone tell me were I can find the IPUpate.exe utility.
So I can change the Ip address of the CCX4.1 server.
 
I found a document that states the file should be in the following 
directory:
\\CRS IP address\DESKTOP_CFG\Util folder.
 
 
Only this folder isn?t there J
 
Groeten Richard,
Mailto:Richard.vanBerkel at Telindus.nl
 
 _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From wsisk at cisco.com  Wed Oct 24 09:45:06 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 24 Oct 2007 13:45:06 -0000
Subject: [cisco-voip] ccm 5.0(4) --> 5.1(2)
In-Reply-To: <20070916121832.87330@gmx.net>
References: <20070916121832.87330@gmx.net>
Message-ID: <33B14F10-BA8A-4849-B4F9-6B5CBA086201@cisco.com>

download the upgrade file.
either burn to DVD or put on SFTP server
login to the platform page (http://server/iptplatform)
use upgrade menu.

it is this process:
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/ 
products_upgrade_guides09186a0080792e84.html#wp38710



/Wes

On Sep 16, 2007, at 8:18 AM, Olafsen Her grmpf wrote:

General question about upgrading 5.0(4) to 5.1(2),

I have read the upgrade guide (http://www.cisco.com/en/US/products/sw/ 
voicesw/ps556/products_upgrade_guides09186a0080792e84.html#wp43371 )  
but I just don't know where to start to be honest, which patch to  
download or if I need the DVD's and reboot and do the upgrade install  
thing ...... ? clueless-

Couldn?t even find documentation on the ?Using Cisco Unified  
CallManager Upgrade Utility? ?
  there seems to be a lot of documentation when upgrading from 4.x  
but from 5.0(x) though ): !?

could someone help me with the right steps to perform an upgrade, I  
am not even sure which patches to download?

any help much appreciated and sorry for the n00b post, I am kind of  
new to the CCM VoIP world of ccm/ucm


Olaf


-- 
GMX FreeMail: 1 GB Postfach, 5 E-Mail-Adressen, 10 Free SMS.
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_______________________________________________
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From Hesham.Moussa at vanco.co.uk  Wed Oct 24 09:54:17 2007
From: Hesham.Moussa at vanco.co.uk (Hesham Moussa)
Date: Wed, 24 Oct 2007 14:54:17 +0100
Subject: [cisco-voip] 7911 and overlays on a CME system
Message-ID: <876C8F9957792243B670CAEBDA19C6D101FEF6DF@LON-INT-MSGVS01.internal.vancocorp.com>

Hi all,

 

Does the 7911 support overlay buttons?

 

I am trying to setup a broadcast group using a very crude method of
multiple DNs with the same number and assigning these multiple DNs to
the 7911 but am worried that it doesn't support overlays at all.

 

Any pointers or confirmation as to whether yay or nay appreciated.

 

Cheers,

 

H



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From SCASPER at mtb.com  Wed Oct 24 09:53:42 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Wed, 24 Oct 2007 09:53:42 -0400
Subject: [cisco-voip]  CCM  Upgrade - Best Practices
In-Reply-To: <ea211ea30710232335n5dbf1d4ax77866fcfd2d926ba@mail.gmail.com>
References: <ea211ea30710232335n5dbf1d4ax77866fcfd2d926ba@mail.gmail.com>
Message-ID: <471F1626.9874.0039.0@mtb.com>

 
 Looking for some advice for the best way to do Call Manager upgrades to minimize system disruptions. From a recent ES doc:
 
"You can minimize call-processing interruptions if you register all devices
to servers that are running the same version of Cisco CallManager during the
entire upgrade process; i.e., you register all devices to the backup Cisco
CallManager servers or the primary Cisco CallManager servers, but not to
both types of servers."
 
 I understand what the ES doc is saying what I am not sure about is the best practice to ensure that all devices stay registered on the backup or primary servers during the upgrade. Should you manipulate your Call Manager groups during the upgrade? 
 
 For instance all of our devices are registered to Subscribers A, B, C, & D. We have two back up subscribers X & Y. Do we:
 
1. Upgrade the Publisher.
2. Upgrade X & Y
3. Change our Call Manager Groups so X & Y subs are the primary.
4. Reset all devices so they re-register to X & Y.
5. Upgrade A, B, C, & D.
6. Change Call Manager groups so A, B, C, & D are primary again.
7. Reset devices.

 
 
Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

************************************
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From daniel.argueta at comcast.net  Wed Oct 24 11:09:30 2007
From: daniel.argueta at comcast.net (Daniel Argueta)
Date: Wed, 24 Oct 2007 08:09:30 -0700
Subject: [cisco-voip] External Callers can not leave long voice messages
	-UCM 6.0.1 Business Edition
References: <A56BD242744BF540B15D1672E0E3E87501DB2161@LON-INT-MSGVS01.internal.vancocorp.com><043501c7faaa$0753df70$c313a8c0@hq.cct><d4c697ad0709190351q596e7012x58bd1763a2b76900@mail.gmail.com>
	<019801c8161f$52b3c710$1f12fea9@hq.cct>
Message-ID: <DDC439FD4BB54558BBB3045902F83F86@main>

Something similar happened to me...my fix was rather simple though...it started happening after we re-arranged the network infrastructure and I noticed the speed/duplex on the server were hardset....the switch was not...so we had a lot of packets drop and a lot of collisions which caused this same thing to happen........it's one simple thing to check..

  ----- Original Message ----- 
  From: Mohan Pokkali 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, October 24, 2007 2:21 AM
  Subject: [cisco-voip] External Callers can not leave long voice messages -UCM 6.0.1 Business Edition




  Hi All



  Problem:                       Unity Connection (Part of Business Edition). 



  External callers while leaving voice messages get cut of randomly after sometime. There is no pattern. Some get cut off after 30 seconds, some after 35 seconds and some after 1 minute and 20 second. This happens across the whole system affecting all users. There is no problem with internal callers. 



  Other Information:          MGCP Voice Gateway connecting to PSTN.



  Thanks in advance for your help.



  Mohan



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From pokkali at gmail.com  Wed Oct 24 10:57:07 2007
From: pokkali at gmail.com (Mohan Pokkali)
Date: Wed, 24 Oct 2007 15:57:07 +0100
Subject: [cisco-voip] External Callers can not leave long voice messages
	-	UCM 6.0.1 Business Edition
In-Reply-To: <471F3004.60109@wcyv.com>
References: <A56BD242744BF540B15D1672E0E3E87501DB2161@LON-INT-MSGVS01.internal.vancocorp.com>	<043501c7faaa$0753df70$c313a8c0@hq.cct>	<d4c697ad0709190351q596e7012x58bd1763a2b76900@mail.gmail.com>
	<019801c8161f$52b3c710$1f12fea9@hq.cct> <471F3004.60109@wcyv.com>
Message-ID: <01d901c8164e$26578830$1f12fea9@hq.cct>

Hi Pat

They do get the first part of the message. I believe I found the answer from
the URL. I just updated the MGCP Gateway. Let's see if it resolves the
problem. I am conducting extensive testing now.

Thanks for your help.

Mohan

-----Original Message-----
From: Pat Hayes [mailto:pat-cv at wcyv.com] 
Sent: 24 October 2007 12:44
To: Mohan Pokkali
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] External Callers can not leave long voice messages
- UCM 6.0.1 Business Edition

Does the called user receive the message with the first however many 
seconds? Or do they get nothing? If they get nothing, verify the RTP 
path through your network and confirm that MGCP is bound to the 
interface that it should be. Packet captures help here. If they do get 
the first part of the message, check out:

http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K88174040

-------- Original Message  --------
From: "Mohan Pokkali" <pokkali at gmail.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] External Callers can not leave long voice 
messages -	UCM 6.0.1 Business Edition
Date: 10/24/2007 5:21 AM

>  
> 
> Hi All
> 
>  
> 
> Problem:                       Unity Connection (Part of Business
Edition). 
> 
>  
> 
> External callers while leaving voice messages get cut of randomly after
> sometime. There is no pattern. Some get cut off after 30 seconds, some
after
> 35 seconds and some after 1 minute and 20 second. This happens across the
> whole system affecting all users. There is no problem with internal
callers.
> 
> 
>  
> 
> Other Information:          MGCP Voice Gateway connecting to PSTN.
> 
>  
> 
> Thanks in advance for your help.
> 
>  
> 
> Mohan
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From rratliff at cisco.com  Wed Oct 24 11:44:06 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 24 Oct 2007 11:44:06 -0400
Subject: [cisco-voip] CCM  Upgrade - Best Practices
In-Reply-To: <471F1626.9874.0039.0@mtb.com>
References: <ea211ea30710232335n5dbf1d4ax77866fcfd2d926ba@mail.gmail.com>
	<471F1626.9874.0039.0@mtb.com>
Message-ID: <3C45A46A-6C47-4FBE-814E-DD84085D367E@cisco.com>

The easiest way to ensure phones don't register to servers during the  
upgrade is to disable or set to manual startup the CallManager  
service on the servers being upgraded.   That way you can do the  
upgrade, reboot the server, and take a look at things before having  
phones fail back to it.

-Ryan

On Oct 24, 2007, at 9:53 AM, STEVEN CASPER wrote:


  Looking for some advice for the best way to do Call Manager  
upgrades to minimize system disruptions. From a recent ES doc:

"You can minimize call-processing interruptions if you register all  
devices
to servers that are running the same version of Cisco CallManager  
during the
entire upgrade process; i.e., you register all devices to the backup  
Cisco
CallManager servers or the primary Cisco CallManager servers, but not to
both types of servers."

  I understand what the ES doc is saying what I am not sure about is  
the best practice to ensure that all devices stay registered on the  
backup or primary servers during the upgrade. Should you manipulate  
your Call Manager groups during the upgrade?

  For instance all of our devices are registered to Subscribers A, B,  
C, & D. We have two back up subscribers X & Y. Do we:

1. Upgrade the Publisher.
2. Upgrade X & Y
3. Change our Call Manager Groups so X & Y subs are the primary.
4. Reset all devices so they re-register to X & Y.
5. Upgrade A, B, C, & D.
6. Change Call Manager groups so A, B, C, & D are primary again.
7. Reset devices.



Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026
************************************
This email may contain privileged and/or confidential information  
that is intended solely for the use of the addressee. If you are not  
the intended recipient or entity, you are strictly prohibited from  
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error, please contact the sender immediately and destroy the material  
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https://puck.nether.net/mailman/listinfo/cisco-voip

From jgrace at adcapnet.com  Wed Oct 24 12:23:27 2007
From: jgrace at adcapnet.com (James Grace)
Date: Wed, 24 Oct 2007 12:23:27 -0400
Subject: [cisco-voip] issue with Xlate timeout for sip on asa
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E8DA@mail3.adcapnetsvc.local>

?
we are setting up sip connection through our ASA.  and for some reason the connections is the asa drops, and we have to make a outbound call from an ipphone to bring the connection back up.  Now we set the Xlante timeout for 72 hrs.  But this is still happening.  sometime less then a minute.  Has anyone ran into this.
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
recipient is prohibited. If you received this message in error, please contact the sender and delete the
material from all computers."

This original email was sent to the internet for delivery at 24 Oct 2007 12:22:07 -0400
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From rkulagow at gmail.com  Wed Oct 24 12:28:26 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Wed, 24 Oct 2007 11:28:26 -0500
Subject: [cisco-voip] 7975G telephone features
Message-ID: <471F72AA.1090502@gmail.com>

According to 
http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0900aecd80699b31.shtml

The 7975G features 8 line buttons and "KEM"

In the comparison table, it says that the 7970 and 7971 have 6 line 
keys.  I'm looking at the 7970 on my desk, and it sure looks like I've 
got 8 line keys.  What are they talking about?

What's "KEM"?

Also:  still no Bluetooth headset support?

From spackett at gmail.com  Wed Oct 24 12:34:05 2007
From: spackett at gmail.com (Stu Packett)
Date: Wed, 24 Oct 2007 09:34:05 -0700
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <471E461F.7040705@gmail.com>
References: <471E461F.7040705@gmail.com>
Message-ID: <feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>

I just read this field notice and it mentions to load 4.1(3)DevPack 57.1,
but I can't find it under the Downloads section.  The latest version I see
is 4.1(56).  Am I the only one experiencing this?

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_field_notice09186a00807e02b1.shtml

On 10/23/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
>
> Well, it's getting to be that time of the year again.
>
> Has anyone noticed that
>
> http://www.cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml#voice_products
>
> was updated earlier this month, and that CM and phone loads are affected
> for the switch back from DST?  Maybe I'm reading it wrong, but it looks
> like I've got another upgrade cycle to go through.
>
> Yay.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From svoll.voip at gmail.com  Wed Oct 24 12:57:09 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 24 Oct 2007 09:57:09 -0700
Subject: [cisco-voip] 7975G telephone features
In-Reply-To: <471F72AA.1090502@gmail.com>
References: <471F72AA.1090502@gmail.com>
Message-ID: <f84a38d30710240957k7070df9ucffdb4054cc76542@mail.gmail.com>

my 7970 is touch screen...... so I don't know what they are trying to say.
Wes, who needs to redo this doc?

Scott


On 10/24/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
>
> According to
>
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0900aecd80699b31.shtml
>
> The 7975G features 8 line buttons and "KEM"
>
> In the comparison table, it says that the 7970 and 7971 have 6 line
> keys.  I'm looking at the 7970 on my desk, and it sure looks like I've
> got 8 line keys.  What are they talking about?
>
> What's "KEM"?
>
> Also:  still no Bluetooth headset support?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From daniel.argueta at comcast.net  Wed Oct 24 14:04:10 2007
From: daniel.argueta at comcast.net (Daniel Argueta)
Date: Wed, 24 Oct 2007 11:04:10 -0700
Subject: [cisco-voip] Error when running DMA to upgrade from 4.1.3 to 5.1.x
Message-ID: <7DCA946BCD2746BBB122C545DB1A04B1@main>

I'm receiving the following error when running the DMA process...it happens during the validation....I see it in the DirExport_Error.log:

10/24/2007  09:33:05:  Error: IO::Socket::INET: connect: Unknown error. ErrorMessage: Failed to connect to reference Host:Port=ldap://DomainDNSZones.vistastaff.com/DC=DomainDnsZones,DC=Vistastaff,DC=com->host_port

i've re-ran the customerdirectoryplugin and have been unable to successfully change this to what it should be......any Ideas where it might be getting the "DomainDNSZones" from? this isn't ours.....

TIA

-Daniel

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From kenjohnson at letu.edu  Wed Oct 24 13:09:25 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Wed, 24 Oct 2007 12:09:25 -0500
Subject: [cisco-voip] 7975G telephone features
In-Reply-To: <f84a38d30710240957k7070df9ucffdb4054cc76542@mail.gmail.com>
References: <471F72AA.1090502@gmail.com>
	<f84a38d30710240957k7070df9ucffdb4054cc76542@mail.gmail.com>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EFD0C00D@rachael-reborn.letnet.net>

That's funny - yeah, our 70s have 8 lines and a touchscreen - that's an
interesting error.

 

KEM is the sidecar I believe - maybe it stands for Keypad Expansion
Module or something...

 

Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
Cisco IP Telecommunications Users Group

http://www.ciptug.org

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Wednesday, October 24, 2007 11:57 AM
To: Robert Kulagowski
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7975G telephone features

 

my 7970 is touch screen...... so I don't know what they are trying to
say.  Wes, who needs to redo this doc?

 

Scott

 

On 10/24/07, Robert Kulagowski <rkulagow at gmail.com> wrote: 

According to
http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0
900aecd80699b31.shtml

The 7975G features 8 line buttons and "KEM"

In the comparison table, it says that the 7970 and 7971 have 6 line 
keys.  I'm looking at the 7970 on my desk, and it sure looks like I've
got 8 line keys.  What are they talking about?

What's "KEM"?

Also:  still no Bluetooth headset support?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

 

________________________________


Spam <http://antispam.letu.edu/canit/b.php?i=8451049&m=70a3134c72b0&c=s>

Not spam
<http://antispam.letu.edu/canit/b.php?i=8451049&m=70a3134c72b0&c=n> 
Forget previous vote
<http://antispam.letu.edu/canit/b.php?i=8451049&m=70a3134c72b0&c=f> 

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From wsisk at cisco.com  Wed Oct 24 13:40:21 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 24 Oct 2007 13:40:21 -0400
Subject: [cisco-voip] Error when running DMA to upgrade from 4.1.3 to
 5.1.x
In-Reply-To: <7DCA946BCD2746BBB122C545DB1A04B1@main>
References: <7DCA946BCD2746BBB122C545DB1A04B1@main>
Message-ID: <471F8385.3040905@cisco.com>

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From cisco.voip at verizon.net  Wed Oct 24 13:39:59 2007
From: cisco.voip at verizon.net (cisco.voip)
Date: Wed, 24 Oct 2007 13:39:59 -0400
Subject: [cisco-voip] Change phone numbers bat
References: <mailman.3080.1193245774.13771.cisco-voip@puck.nether.net>
Message-ID: <000a01c81665$f1542270$661b1b0a@cs.myharris.net>

I need to change my local CCM phones so the phone numbers are 555-1XXX from
555-2XXX.
Can the bulk admin tool do this in one step- I have like 600 phones need to
change?

Tx



From wsisk at cisco.com  Wed Oct 24 14:09:54 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 24 Oct 2007 14:09:54 -0400
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>
References: <471E461F.7040705@gmail.com>
	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>
Message-ID: <471F8A72.6020007@cisco.com>

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From SCASPER at mtb.com  Wed Oct 24 14:09:59 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Wed, 24 Oct 2007 14:09:59 -0400
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>
References: <471E461F.7040705@gmail.com>
	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>
Message-ID: <471F5236.9874.0039.0@mtb.com>

4.1.3sr5d contains device pack 57.1

>>> "Stu Packett" <spackett at gmail.com> 10/24/2007 12:34 PM >>>
I just read this field notice and it mentions to load 4.1(3)DevPack 57.1, but I can't find it under the Downloads section.  The latest version I see is 4.1(56).  Am I the only one experiencing this? 

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_field_notice09186a00807e02b1.shtml 

On 10/23/07, Robert Kulagowski <rkulagow at gmail.com> wrote:

Well, it's getting to be that time of the year again.

Has anyone noticed that
http://www.cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml#voice_products 

was updated earlier this month, and that CM and phone loads are affected
for the switch back from DST?  Maybe I'm reading it wrong, but it looks
like I've got another upgrade cycle to go through. 

Yay.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 



************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee.  If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission.  If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy.  This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
There are risks associated with the use of electronic transmission.  The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
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From svoll.voip at gmail.com  Wed Oct 24 14:18:18 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 24 Oct 2007 11:18:18 -0700
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <471F5236.9874.0039.0@mtb.com>
References: <471E461F.7040705@gmail.com>
	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>
	<471F5236.9874.0039.0@mtb.com>
Message-ID: <f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com>

so if your running 4.1.3sr5d and OS 4.4asr11 I should be good?

I'm upgrading tonight as I'm still on 4.4asr7 and 4.1.3es100.

Scott

PS.  any issues with Unity 4.0.5 or IPCCx 4.0.1?


On 10/24/07, STEVEN CASPER <SCASPER at mtb.com> wrote:
>
> 4.1.3sr5d contains device pack 57.1
>
> >>> "Stu Packett" <spackett at gmail.com> 10/24/2007 12:34 PM >>>
> I just read this field notice and it mentions to load 4.1(3)DevPack 57.1,
> but I can't find it under the Downloads section.  The latest version I see
> is 4.1(56).  Am I the only one experiencing this?
>
> http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_field_notice09186a00807e02b1.shtml
>
>
> On 10/23/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
> >
> > Well, it's getting to be that time of the year again.
> >
> > Has anyone noticed that
> > http://www.cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml#voice_products
> >
> >
> > was updated earlier this month, and that CM and phone loads are affected
> > for the switch back from DST?  Maybe I'm reading it wrong, but it looks
> > like I've got another upgrade cycle to go through.
> >
> > Yay.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> ************************************
> This email may contain privileged and/or confidential information that is
> intended solely for the use of the addressee. If you are not the intended
> recipient or entity, you are strictly prohibited from disclosing, copying,
> distributing or using any of the information contained in the transmission.
> If you received this communication in error, please contact the sender
> immediately and destroy the material in its entirety, whether electronic or
> hard copy. This communication may contain nonpublic personal information
> about consumers subject to the restrictions of the Gramm-Leach-Bliley Act
> and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or
> disclose such information for any purpose other than to provide the services
> for which you are receiving the information.
> There are risks associated with the use of electronic transmission. The
> sender of this information does not control the method of transmittal or
> service providers and assumes no duty or obligation for the security,
> receipt, or third party interception of this transmission.
> ************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jason.aarons at us.didata.com  Wed Oct 24 15:24:39 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 24 Oct 2007 15:24:39 -0400
Subject: [cisco-voip] Secondary DNs only have 7 ring choices?
Message-ID: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>

Customer asked why their line 2, line 3, etc only have 7 ring choices in
Settings > Ring Type > X when their default ring has 25 choices?

 

I said "because it is what it is" (works on my 3 year old all the
time).....what's the real reason?

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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From matt at saskin.net  Wed Oct 24 15:30:50 2007
From: matt at saskin.net (Matthew Saskin)
Date: Wed, 24 Oct 2007 15:30:50 -0400
Subject: [cisco-voip] Secondary DNs only have 7 ring choices?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>
Message-ID: <471F9D6A.9010009@saskin.net>

DistinctiveRingList.xml

-matt

Jason Aarons (US) wrote:
> Customer asked why their line 2, line 3, etc only have 7 ring choices in 
> Settings > Ring Type > X when their default ring has 25 choices?
> 
>  
> 
> I said ?because it is what it is? (works on my 3 year old all the 
> time)?..what?s the real reason?
> 
>  
> 
> ------------------------------------------------------------------------
> 
> * Disclaimer: This e-mail communication and any attachments may contain 
> confidential and privileged information and is for use by the designated 
> addressee(s) named above only. If you are not the intended addressee, 
> you are hereby notified that you have received this communication in 
> error and that any use or reproduction of this email or its contents is 
> strictly prohibited and may be unlawful. If you have received this 
> communication in error, please notify us immediately by replying to this 
> message and deleting it from your computer. Thank you. *
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From jason.aarons at us.didata.com  Wed Oct 24 15:55:11 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 24 Oct 2007 15:55:11 -0400
Subject: [cisco-voip] Secondary DNs only have 7 ring choices?
In-Reply-To: <471F9D6A.9010009@saskin.net>
References: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>
	<471F9D6A.9010009@saskin.net>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0073844AB@USNAEXCH.na.didata.local>

I recall how to modify the .xml file in CallManager 4.1, how do I modify
the DistinctiveRingList.xml in 5.1.2?

-----Original Message-----
From: Matthew Saskin [mailto:matt at saskin.net] 
Sent: Wednesday, October 24, 2007 3:31 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?

DistinctiveRingList.xml

-matt

Jason Aarons (US) wrote:
> Customer asked why their line 2, line 3, etc only have 7 ring choices
in 
> Settings > Ring Type > X when their default ring has 25 choices?
> 
>  
> 
> I said "because it is what it is" (works on my 3 year old all the 
> time).....what's the real reason?
> 
>  
> 
>
------------------------------------------------------------------------
> 
> * Disclaimer: This e-mail communication and any attachments may
contain 
> confidential and privileged information and is for use by the
designated 
> addressee(s) named above only. If you are not the intended addressee, 
> you are hereby notified that you have received this communication in 
> error and that any use or reproduction of this email or its contents
is 
> strictly prohibited and may be unlawful. If you have received this 
> communication in error, please notify us immediately by replying to
this 
> message and deleting it from your computer. Thank you. *
> 
> 
>
------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From rkulagow at gmail.com  Wed Oct 24 15:56:19 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Wed, 24 Oct 2007 14:56:19 -0500
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com>
References: <471E461F.7040705@gmail.com>	
	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>	
	<471F5236.9874.0039.0@mtb.com>
	<f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com>
Message-ID: <471FA363.904@gmail.com>

Scott Voll wrote:
> so if your running 4.1.3sr5d and OS 4.4asr11 I should be good? 
>  
> I'm upgrading tonight as I'm still on 4.4asr7 and 4.1.3es100.
>  
> Scott
>  
> PS.  any issues with Unity 4.0.5 or IPCCx 4.0.1?

Any opinions on running Win2K in the GMT timezone with no DST 
adjustments at all?

I'm thinking that the impact would be:
1)  System clock in toolbar is wrong.
2)  My logs are going to be 6 hours off relative to wall clock time.

What's going to be the impact on the phones though?  If I say that a 
phone is in the CST timezone, am I still going to have the wrong time on 
2007-10-28?


From asobihoudai at yahoo.com  Wed Oct 24 16:13:22 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 24 Oct 2007 13:13:22 -0700 (PDT)
Subject: [cisco-voip] Secondary DNs only have 7 ring choices?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0073844AB@USNAEXCH.na.didata.local>
Message-ID: <176350.72873.qm@web57202.mail.re3.yahoo.com>

Is there no way to just edit one and upload it via
sftp/scp?

Paul
--- "Jason Aarons (US)" <jason.aarons at us.didata.com>
wrote:

> I recall how to modify the .xml file in CallManager
> 4.1, how do I modify
> the DistinctiveRingList.xml in 5.1.2?
> 
> -----Original Message-----
> From: Matthew Saskin [mailto:matt at saskin.net] 
> Sent: Wednesday, October 24, 2007 3:31 PM
> To: Jason Aarons (US)
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Secondary DNs only have 7
> ring choices?
> 
> DistinctiveRingList.xml
> 
> -matt
> 
> Jason Aarons (US) wrote:
> > Customer asked why their line 2, line 3, etc only
> have 7 ring choices
> in 
> > Settings > Ring Type > X when their default ring
> has 25 choices?
> > 
> >  
> > 
> > I said "because it is what it is" (works on my 3
> year old all the 
> > time).....what's the real reason?
> > 
> >  
> > 
> >
>
------------------------------------------------------------------------
> > 
> > * Disclaimer: This e-mail communication and any
> attachments may
> contain 
> > confidential and privileged information and is for
> use by the
> designated 
> > addressee(s) named above only. If you are not the
> intended addressee, 
> > you are hereby notified that you have received
> this communication in 
> > error and that any use or reproduction of this
> email or its contents
> is 
> > strictly prohibited and may be unlawful. If you
> have received this 
> > communication in error, please notify us
> immediately by replying to
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> > message and deleting it from your computer. Thank
> you. *
> > 
> > 
> >
>
------------------------------------------------------------------------
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> -----------------------------------------
> Disclaimer:
> 
> This e-mail communication and any attachments may
> contain
> confidential and privileged information and is for
> use by the
> designated addressee(s) named above only.  If you
> are not the
> intended addressee, you are hereby notified that you
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> this communication in error and that any use or
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> this email or its contents is strictly prohibited
> and may be
> unlawful.  If you have received this communication
> in error, please
> notify us immediately by replying to this message
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> 


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From jason.aarons at us.didata.com  Wed Oct 24 16:34:15 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 24 Oct 2007 16:34:15 -0400
Subject: [cisco-voip] CCM 5.1.2 offhook picks up any line
References: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local><471F9D6A.9010009@saskin.net>
	<C1FE15183DA37645BC0633BC604E44F0073844AB@USNAEXCH.na.didata.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E906@USNAEXCH.na.didata.local>

If my Line 2 rings I want to be forced when going offhook to press Line 2, isn't that a service parameter? Instead if I get a call on Line 2 today when I go offhook I get connected to Line 2. 
 
Pretty much all my service parameters are default.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Wed 10/24/2007 3:55 PM
To: Matthew Saskin
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?



I recall how to modify the .xml file in CallManager 4.1, how do I modify
the DistinctiveRingList.xml in 5.1.2?

-----Original Message-----
From: Matthew Saskin [mailto:matt at saskin.net]
Sent: Wednesday, October 24, 2007 3:31 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?

DistinctiveRingList.xml

-matt

Jason Aarons (US) wrote:
> Customer asked why their line 2, line 3, etc only have 7 ring choices
in
> Settings > Ring Type > X when their default ring has 25 choices?
>
> 
>
> I said "because it is what it is" (works on my 3 year old all the
> time).....what's the real reason?
>
> 
>
>
------------------------------------------------------------------------
>
> * Disclaimer: This e-mail communication and any attachments may
contain
> confidential and privileged information and is for use by the
designated
> addressee(s) named above only. If you are not the intended addressee,
> you are hereby notified that you have received this communication in
> error and that any use or reproduction of this email or its contents
is
> strictly prohibited and may be unlawful. If you have received this
> communication in error, please notify us immediately by replying to
this
> message and deleting it from your computer. Thank you. *
>
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



-----------------------------------------
Disclaimer:

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https://puck.nether.net/mailman/listinfo/cisco-voip





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Disclaimer:

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From svoll.voip at gmail.com  Wed Oct 24 16:43:17 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 24 Oct 2007 13:43:17 -0700
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <471FA363.904@gmail.com>
References: <471E461F.7040705@gmail.com>
	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>
	<471F5236.9874.0039.0@mtb.com>
	<f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com>
	<471FA363.904@gmail.com>
Message-ID: <f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>

someone can correct me if I'm wrong but the system clock on the call manager
is where the phones get there time, right?

Scott


On 10/24/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
>
> Scott Voll wrote:
> > so if your running 4.1.3sr5d and OS 4.4asr11 I should be good?
> >
> > I'm upgrading tonight as I'm still on 4.4asr7 and 4.1.3es100.
> >
> > Scott
> >
> > PS.  any issues with Unity 4.0.5 or IPCCx 4.0.1?
>
> Any opinions on running Win2K in the GMT timezone with no DST
> adjustments at all?
>
> I'm thinking that the impact would be:
> 1)  System clock in toolbar is wrong.
> 2)  My logs are going to be 6 hours off relative to wall clock time.
>
> What's going to be the impact on the phones though?  If I say that a
> phone is in the CST timezone, am I still going to have the wrong time on
> 2007-10-28?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From svoll.voip at gmail.com  Wed Oct 24 16:45:19 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 24 Oct 2007 13:45:19 -0700
Subject: [cisco-voip] CCM 5.1.2 offhook picks up any line
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E906@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>
	<471F9D6A.9010009@saskin.net>
	<C1FE15183DA37645BC0633BC604E44F0073844AB@USNAEXCH.na.didata.local>
	<C1FE15183DA37645BC0633BC604E44F013E906@USNAEXCH.na.didata.local>
Message-ID: <f84a38d30710241345h3200bbaj963a6066a976b5fd@mail.gmail.com>

do a find on primary line.  it has that in the service parameter but I don't
remember the whole term.

Scott


On 10/24/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  If my Line 2 rings I want to be forced when going offhook to press Line
> 2, isn't that a service parameter? Instead if I get a call on Line 2 today
> when I go offhook I get connected to Line 2.
>
> Pretty much all my service parameters are default.
>
>  ------------------------------
>  *From:* cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
> *Sent:* Wed 10/24/2007 3:55 PM
> *To:* Matthew Saskin
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Secondary DNs only have 7 ring choices?
>
>
>
> I recall how to modify the .xml file in CallManager 4.1, how do I modify
> the DistinctiveRingList.xml in 5.1.2?
>
> -----Original Message-----
> From: Matthew Saskin [mailto:matt at saskin.net <matt at saskin.net>]
> Sent: Wednesday, October 24, 2007 3:31 PM
> To: Jason Aarons (US)
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?
>
> DistinctiveRingList.xml
>
> -matt
>
> Jason Aarons (US) wrote:
> > Customer asked why their line 2, line 3, etc only have 7 ring choices
> in
> > Settings > Ring Type > X when their default ring has 25 choices?
> >
> >
> >
> > I said "because it is what it is" (works on my 3 year old all the
> > time).....what's the real reason?
> >
> >
> >
> >
> ------------------------------------------------------------------------
> >
> > * Disclaimer: This e-mail communication and any attachments may
> contain
> > confidential and privileged information and is for use by the
> designated
> > addressee(s) named above only. If you are not the intended addressee,
> > you are hereby notified that you have received this communication in
> > error and that any use or reproduction of this email or its contents
> is
> > strictly prohibited and may be unlawful. If you have received this
> > communication in error, please notify us immediately by replying to
> this
> > message and deleting it from your computer. Thank you. *
> >
> >
> >
> ------------------------------------------------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck..nether.net/mailman/listinfo/cisco-voip<https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck..nether.net/mailman/listinfo/cisco-voip<https://puck.nether.net/mailman/listinfo/cisco-voip>
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
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>
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From rkulagow at gmail.com  Wed Oct 24 16:54:50 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Wed, 24 Oct 2007 15:54:50 -0500
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>
References: <471E461F.7040705@gmail.com>	
	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>	
	<471F5236.9874.0039.0@mtb.com>	
	<f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com>	
	<471FA363.904@gmail.com>
	<f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>
Message-ID: <471FB11A.4080305@gmail.com>

Scott Voll wrote:
> someone can correct me if I'm wrong but the system clock on the call 
> manager is where the phones get there time, right?

Right, but the device pool allows you to choose a date/time group.  If 
the DTG is appropriate, then the clock on the phone should be accurate. 
  I'm labbing something now where I'm going to move the server to GMT 
and then advance the clock to 2007-10-28 at 0158 to see what happens.

I'll do the same with 2007-11-04 at 0158.

I'm just not sure which combination of MCS OS, CM SR and phone load is 
required to ensure that CM knows that DST in NA ends on 2007-11-04 this 
year.  (Running 4.1.3sr2 with the DST patches we needed back in March)

Whew.  That's a lot of acronyms.

From rduric at uoguelph.ca  Wed Oct 24 17:16:21 2007
From: rduric at uoguelph.ca (Rasim Duric)
Date: Wed, 24 Oct 2007 17:16:21 -0400
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>
References: <471E461F.7040705@gmail.com><feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com><471F5236.9874.0039.0@mtb.com><f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com><471FA363.904@gmail.com>
	<f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>
Message-ID: <BD89624EE83BD349BD31A08B453DE24401DB7B24@uogexb201.cfs.uoguelph.ca>

The phones will get their time from the CCM DB, TypeTimeZone Table, CCM
4.1 

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Wednesday, October 24, 2007 4:43 PM
To: Robert Kulagowski
Cc: Cisco Voip Mailing list
Subject: Re: [cisco-voip] Daylight Savings Time issues...

 

someone can correct me if I'm wrong but the system clock on the call
manager is where the phones get there time, right?

 

Scott

 

On 10/24/07, Robert Kulagowski <rkulagow at gmail.com> wrote: 

Scott Voll wrote:
> so if your running 4.1.3sr5d and OS 4.4asr11 I should be good?
>
> I'm upgrading tonight as I'm still on 4.4asr7 and 4.1.3es100.
>
> Scott
>
> PS.  any issues with Unity 4.0.5 or IPCCx 4.0.1?

Any opinions on running Win2K in the GMT timezone with no DST
adjustments at all?

I'm thinking that the impact would be: 
1)  System clock in toolbar is wrong.
2)  My logs are going to be 6 hours off relative to wall clock time.

What's going to be the impact on the phones though?  If I say that a
phone is in the CST timezone, am I still going to have the wrong time on

2007-10-28?

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From wsisk at cisco.com  Wed Oct 24 17:17:32 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 24 Oct 2007 17:17:32 -0400
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>
References: <471E461F.7040705@gmail.com>	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>	<471F5236.9874.0039.0@mtb.com>	<f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com>	<471FA363.904@gmail.com>
	<f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>
Message-ID: <471FB66C.8070406@cisco.com>

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From svoll.voip at gmail.com  Wed Oct 24 17:32:35 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 24 Oct 2007 14:32:35 -0700
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <471FB11A.4080305@gmail.com>
References: <471E461F.7040705@gmail.com>
	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>
	<471F5236.9874.0039.0@mtb.com>
	<f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com>
	<471FA363.904@gmail.com>
	<f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>
	<471FB11A.4080305@gmail.com>
Message-ID: <f84a38d30710241432r52945ff9wed527fc8e2e0854d@mail.gmail.com>

now who would be a little cranked at the goverment if they decided to go
back to the old Daylight savings time?  LOL

On 10/24/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
>
> Scott Voll wrote:
> > someone can correct me if I'm wrong but the system clock on the call
> > manager is where the phones get there time, right?
>
> Right, but the device pool allows you to choose a date/time group.  If
> the DTG is appropriate, then the clock on the phone should be accurate.
> I'm labbing something now where I'm going to move the server to GMT
> and then advance the clock to 2007-10-28 at 0158 to see what happens.
>
> I'll do the same with 2007-11-04 at 0158.
>
> I'm just not sure which combination of MCS OS, CM SR and phone load is
> required to ensure that CM knows that DST in NA ends on 2007-11-04 this
> year.  (Running 4.1.3sr2 with the DST patches we needed back in March)
>
> Whew.  That's a lot of acronyms.
>
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From jason.aarons at us.didata.com  Wed Oct 24 17:51:03 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 24 Oct 2007 17:51:03 -0400
Subject: [cisco-voip] Unity Connection 2.0.1 callback feature after
	listening to message?
References: <fc.004c4fb706f48974004c4fb706f48974.6f4913f@pediatrix.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E908@USNAEXCH.na.didata.local>



With our old Merian Mail on the Nortel System, after listening to a voice message we had the option to call back the person that left you that message.  Do we have that option on this system?  I checked the Unity Connection 2.0 Voicemail user guide but didn't find anything.  It only provides the option to reply back with a message but not through a call.




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From craig at staffin.org  Wed Oct 24 17:53:58 2007
From: craig at staffin.org (Craig Staffin)
Date: Wed, 24 Oct 2007 16:53:58 -0500
Subject: [cisco-voip] Daylight Savings Time issues...
In-Reply-To: <f84a38d30710241432r52945ff9wed527fc8e2e0854d@mail.gmail.com>
References: <471E461F.7040705@gmail.com>
	<feb44fca0710240934j1074ae12g5e65fa0b75af7e9d@mail.gmail.com>
	<471F5236.9874.0039.0@mtb.com>
	<f84a38d30710241118w7efbfd9dw44ae9e70926a6c89@mail.gmail.com>
	<471FA363.904@gmail.com>
	<f84a38d30710241343q747d7accjc243feb2bc3bcadd@mail.gmail.com>
	<471FB11A.4080305@gmail.com>
	<f84a38d30710241432r52945ff9wed527fc8e2e0854d@mail.gmail.com>
Message-ID: <9b76d88c0710241453w4797e176t3e921f9b8c589ba@mail.gmail.com>

Come on Scott anyone in the consulting world would love it :)

They saved millions of dollars in energy cost but it also cost companies
millians of dollars to update all of there old PBX's, windows boxes, novell
servers, apps and so on and so on.

Craig


On 10/24/07, Scott Voll <svoll.voip at gmail.com> wrote:
>
> now who would be a little cranked at the goverment if they decided to go
> back to the old Daylight savings time?  LOL
>
> On 10/24/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
> >
> > Scott Voll wrote:
> > > someone can correct me if I'm wrong but the system clock on the call
> > > manager is where the phones get there time, right?
> >
> > Right, but the device pool allows you to choose a date/time group.  If
> > the DTG is appropriate, then the clock on the phone should be accurate.
> > I'm labbing something now where I'm going to move the server to GMT
> > and then advance the clock to 2007-10-28 at 0158 to see what happens.
> >
> > I'll do the same with 2007-11-04 at 0158.
> >
> > I'm just not sure which combination of MCS OS, CM SR and phone load is
> > required to ensure that CM knows that DST in NA ends on 2007-11-04 this
> > year.  (Running 4.1.3sr2 with the DST patches we needed back in March)
> >
> > Whew.  That's a lot of acronyms.
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003
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From pat-cv at wcyv.com  Wed Oct 24 18:02:41 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Wed, 24 Oct 2007 18:02:41 -0400
Subject: [cisco-voip] Unity Connection 2.0.1 callback feature
 after	listening to message?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E908@USNAEXCH.na.didata.local>
References: <fc.004c4fb706f48974004c4fb706f48974.6f4913f@pediatrix.com>
	<C1FE15183DA37645BC0633BC604E44F013E908@USNAEXCH.na.didata.local>
Message-ID: <471FC101.7070407@wcyv.com>

That'd be 'Live Reply':

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/2x/user_mac/guide/2xcucmac050.html#wp1056117

-------- Original Message  --------
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Connection 2.0.1 callback feature after 
listening to message?
Date: 10/24/2007 5:51 PM

> 
> With our old Merian Mail on the Nortel System, after listening to a voice message we had the option to call back the person that left you that message.  Do we have that option on this system?  I checked the Unity Connection 2.0 Voicemail user guide but didn't find anything.  It only provides the option to reply back with a message but not through a call.
> 
> 
> 
> 
> -----------------------------------------
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> 
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> designated addressee(s) named above only.  If you are not the
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> https://puck.nether.net/mailman/listinfo/cisco-voip

From asobihoudai at yahoo.com  Wed Oct 24 18:05:28 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 24 Oct 2007 15:05:28 -0700 (PDT)
Subject: [cisco-voip] any Unity 5 + Exchange 2007 updates?
In-Reply-To: <471FC101.7070407@wcyv.com>
Message-ID: <706130.81328.qm@web57213.mail.re3.yahoo.com>

I remember someone doing a Unity 5 + Exchange 2007
installation. I was wondering how that turned out. The
consensus I've been hearing is that it's supported BUT
don't ignore the 100 or so caveats that come along
with it.

Would anyone care to share their experiences
installing these two applications together?

Paul

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From Joe.Pollere at us.didata.com  Wed Oct 24 18:10:16 2007
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Wed, 24 Oct 2007 18:10:16 -0400
Subject: [cisco-voip] CCM 5.1.2 offhook picks up any line
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E906@USNAEXCH.na.didata.local>
Message-ID: <C65033A817B5734392E55CB9DF4461040DC6928D@USNAEXCH.na.didata.local>

Jason,

 

Check the "always use prime line" CCM Service Parameter

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Wednesday, October 24, 2007 4:34 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM 5.1.2 offhook picks up any line

 

If my Line 2 rings I want to be forced when going offhook to press Line
2, isn't that a service parameter? Instead if I get a call on Line 2
today when I go offhook I get connected to Line 2. 

 

Pretty much all my service parameters are default.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Wed 10/24/2007 3:55 PM
To: Matthew Saskin
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?

I recall how to modify the .xml file in CallManager 4.1, how do I modify
the DistinctiveRingList.xml in 5.1.2?

-----Original Message-----
From: Matthew Saskin [mailto:matt at saskin.net]
Sent: Wednesday, October 24, 2007 3:31 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?

DistinctiveRingList.xml

-matt

Jason Aarons (US) wrote:
> Customer asked why their line 2, line 3, etc only have 7 ring choices
in
> Settings > Ring Type > X when their default ring has 25 choices?
>
> 
>
> I said "because it is what it is" (works on my 3 year old all the
> time).....what's the real reason?
>
> 
>
>
------------------------------------------------------------------------
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From svoll.voip at gmail.com  Wed Oct 24 18:12:55 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 24 Oct 2007 15:12:55 -0700
Subject: [cisco-voip] any Unity 5 + Exchange 2007 updates?
In-Reply-To: <706130.81328.qm@web57213.mail.re3.yahoo.com>
References: <471FC101.7070407@wcyv.com>
	<706130.81328.qm@web57213.mail.re3.yahoo.com>
Message-ID: <f84a38d30710241512w1565b883p65f20f7b26116d91@mail.gmail.com>

I'm in the process of getting ready.

I just emailed Licensing the MAC address and licenses needed and now there
asking for the PAK.  How is the proper way to get the license files I need
for the upgrade?

Scott


On 10/24/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> I remember someone doing a Unity 5 + Exchange 2007
> installation. I was wondering how that turned out. The
> consensus I've been hearing is that it's supported BUT
> don't ignore the 100 or so caveats that come along
> with it.
>
> Would anyone care to share their experiences
> installing these two applications together?
>
> Paul
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
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From PRuttman at foley.com  Wed Oct 24 18:16:17 2007
From: PRuttman at foley.com (Ruttman, Pete G.)
Date: Wed, 24 Oct 2007 17:16:17 -0500
Subject: [cisco-voip] 7970s will not register
Message-ID: <5DB17A1560B7EE4186747DF43A6D4C6CCA0531@WIMKEEX07.foleylaw.com>

We have a couple 7970 phones that will not register with callmanager.  A
previous engineer told me that the there was a bug that caused the phone
to not register if they were not connected to callmanager for an
extended period.  We have had the phones offline now for years and I
just tried to plug them in and they will not register.  I tried to reset
the phone to factory based on the directions below:

Unplug the power cable from the phone and then plug it back in. The
phone begins its power up cycle. 
While the phone powers up, and before the Speaker button flashes on and
off, press #. Each line button flashes on and off in sequence. 
Press 123456789*0#. You can press a key twice in a row, but if you press
the keys out of sequence, the factory reset does not take place. Now the
lights should change color from amber to red.
Press 1 in order to maintain the current network configuration settings
for the phone when the phone resets.

Does anyone have any other suggestions to fix these?

Pete



The preceding email message may be confidential or protected by the attorney-client privilege. It is not intended for transmission to, or receipt by, any unauthorized persons.  If you have received this message in error, please (i) do not read it, (ii) reply to the sender that you received the message in error, and (iii) erase or destroy the message.  Legal advice contained in the preceding message is solely for the benefit of the Foley & Lardner LLP client(s) represented by the Firm in the particular matter that is the subject of this message, and may not be relied upon by any other party.      

  
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From jason.aarons at us.didata.com  Wed Oct 24 18:48:24 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 24 Oct 2007 18:48:24 -0400
Subject: [cisco-voip] CCMAdmin in 5.1
Message-ID: <C1FE15183DA37645BC0633BC604E44F007384779@USNAEXCH.na.didata.local>

I bought a 24" monitor thinking I would no longer have to scroll to the
right in CCMAdmin to see items on the right side, what is the required
screen size/resolution to not have to right scroll in CCMAdmin? 40"?

 




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From jason.aarons at us.didata.com  Wed Oct 24 18:58:46 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 24 Oct 2007 18:58:46 -0400
Subject: [cisco-voip] Unity Connection 2.0.1 callback feature
	after	listening to message?
References: <fc.004c4fb706f48974004c4fb706f48974.6f4913f@pediatrix.com>
	<C1FE15183DA37645BC0633BC604E44F013E908@USNAEXCH.na.didata.local>
	<471FC101.7070407@wcyv.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E90C@USNAEXCH.na.didata.local>

Is Live Reply a licensed feature?
 
Users with voice mailboxes	 LicSubscribersMax	 500	 483	 17 	
Users with IMAP access to voice messages	 LicIMAPSubscribersMax	 10	 1	 9 	
Cisco Unity Inbox users	 LicVMISubscribersMax	 10	 1	 9 	
TTS and ASR (advanced) users	 LicAdvancedUserMax	 10	 0	 10 	
Voice ports	 LicVoicePortsMax	 16	 16	 0 	
Text-to-speech sessions	 LicRealspeakSessionsMax	 16	 0	 16 	
Voice-recognition sessions	 LicUnityVoiceRecSessionsMax	

________________________________

From: Pat Hayes [mailto:pat-cv at wcyv.com]
Sent: Wed 10/24/2007 6:02 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity Connection 2.0.1 callback feature after listening to message?



That'd be 'Live Reply':

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/2x/user_mac/guide/2xcucmac050.html#wp1056117

-------- Original Message  --------
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Connection 2.0.1 callback feature after
listening to message?
Date: 10/24/2007 5:51 PM

>
> With our old Merian Mail on the Nortel System, after listening to a voice message we had the option to call back the person that left you that message.  Do we have that option on this system?  I checked the Unity Connection 2.0 Voicemail user guide but didn't find anything.  It only provides the option to reply back with a message but not through a call.
>
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip





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From jason.aarons at us.didata.com  Wed Oct 24 19:12:08 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 24 Oct 2007 19:12:08 -0400
Subject: [cisco-voip] CCMAdmin in 5.1
References: <C1FE15183DA37645BC0633BC604E44F007384779@USNAEXCH.na.didata.local>
	<5CC6FF682D32164BA5F2E830ABDF146001BFB9DB@SNEXCHANGE02.stifelnet.stifel.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E90D@USNAEXCH.na.didata.local>

I'm good now and happier at 1440x900, I must have been crazy with the big fonts.
 
Is there a URL I can send the customer about this screen size thing ?

________________________________

From: Gibson, Brian (Jersey City) [mailto:gibsonb at stifel.com]
Sent: Wed 10/24/2007 6:59 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CCMAdmin in 5.1



Screen size doesn't matter.  Screen resolution is more important.  

 

With my 19" monitor on 1280 resolution I have no problems seeing the whole screen.

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (US)
Sent: Wednesday, October 24, 2007 6:48 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCMAdmin in 5.1

 

I bought a 24" monitor thinking I would no longer have to scroll to the right in CCMAdmin to see items on the right side, what is the required screen size/resolution to not have to right scroll in CCMAdmin? 40"?

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 

********************************************************************************

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Associates are subject to review by Stifel Nicolaus. Stifel Nicolaus

may retain and reproduce electronic messages for state, federal, or

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IMPORTANT: Please do not use e-mail to request or authorize the

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From pat-cv at wcyv.com  Wed Oct 24 19:34:52 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Wed, 24 Oct 2007 19:34:52 -0400
Subject: [cisco-voip] Unity Connection 2.0.1 callback feature
 after	listening to message?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E90C@USNAEXCH.na.didata.local>
References: <fc.004c4fb706f48974004c4fb706f48974.6f4913f@pediatrix.com>
	<C1FE15183DA37645BC0633BC604E44F013E908@USNAEXCH.na.didata.local>
	<471FC101.7070407@wcyv.com>
	<C1FE15183DA37645BC0633BC604E44F013E90C@USNAEXCH.na.didata.local>
Message-ID: <471FD69C.1000206@wcyv.com>

I don't think so, but give it a try. The UC COS page won't let you 
enable something you're not licensed for.

-------- Original Message  --------
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
To: "Pat Hayes" <pat-cv at wcyv.com>
Cc: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Connection 2.0.1 callback feature after   
 listening to message?
Date: 10/24/2007 6:58 PM
> Is Live Reply a licensed feature?
>  
> Users with voice mailboxes	 LicSubscribersMax	 500	 483	 17 	
> Users with IMAP access to voice messages	 LicIMAPSubscribersMax	 10	 1	 9 	
> Cisco Unity Inbox users	 LicVMISubscribersMax	 10	 1	 9 	
> TTS and ASR (advanced) users	 LicAdvancedUserMax	 10	 0	 10 	
> Voice ports	 LicVoicePortsMax	 16	 16	 0 	
> Text-to-speech sessions	 LicRealspeakSessionsMax	 16	 0	 16 	
> Voice-recognition sessions	 LicUnityVoiceRecSessionsMax	
>
> ________________________________
>
> From: Pat Hayes [mailto:pat-cv at wcyv.com]
> Sent: Wed 10/24/2007 6:02 PM
> To: Jason Aarons (US)
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Unity Connection 2.0.1 callback feature after listening to message?
>
>
>
> That'd be 'Live Reply':
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/connection/2x/user_mac/guide/2xcucmac050.html#wp1056117
>
> -------- Original Message  --------
> From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Unity Connection 2.0.1 callback feature after
> listening to message?
> Date: 10/24/2007 5:51 PM
>
>   
>> With our old Merian Mail on the Nortel System, after listening to a voice message we had the option to call back the person that left you that message.  Do we have that option on this system?  I checked the Unity Connection 2.0 Voicemail user guide but didn't find anything.  It only provides the option to reply back with a message but not through a call.
>>
>>
>>
>>
>> -----------------------------------------
>> Disclaimer:
>>
>> This e-mail communication and any attachments may contain
>> confidential and privileged information and is for use by the
>> designated addressee(s) named above only.  If you are not the
>> intended addressee, you are hereby notified that you have received
>> this communication in error and that any use or reproduction of
>> this email or its contents is strictly prohibited and may be
>> unlawful.  If you have received this communication in error, please
>> notify us immediately by replying to this message and deleting it
>> from your computer. Thank you.
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>     
>
>
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
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>   



From tednugent73 at gmail.com  Wed Oct 24 19:48:31 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Wed, 24 Oct 2007 19:48:31 -0400
Subject: [cisco-voip] CCMAdmin in 5.1
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F007384779@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F007384779@USNAEXCH.na.didata.local>
Message-ID: <22e6380b0710241648x27721c19p9d82cf9a244e555f@mail.gmail.com>

1280 appears to be the magic number, anything less and you're panning..

On 10/24/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  I bought a 24" monitor thinking I would no longer have to scroll to the
> right in CCMAdmin to see items on the right side, what is the required
> screen size/resolution to not have to right scroll in CCMAdmin? 40"?
>
>
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From mthompson729 at gmail.com  Wed Oct 24 19:57:10 2007
From: mthompson729 at gmail.com (Michael Thompson)
Date: Wed, 24 Oct 2007 19:57:10 -0400
Subject: [cisco-voip] Home user
In-Reply-To: <680945A7-7C6A-4EDC-B180-5D9116E86CE8@jerkys.org>
References: <000601c80f7b$ec317f40$c4947dc0$@com>
	<AB0A35F2486F81468971AA68558BE719073BE9A9@smieln01winm01.micoanet.com>
	<513E3090-22BD-47A1-91EB-D4044AF17754@jerkys.org>
	<AB0A35F2486F81468971AA68558BE719073BF05C@smieln01winm01.micoanet.com>
	<6A059BC4-F184-4563-B117-1D28BCDAFFA9@jerkys.org>
	<f84a38d30710171105n6977175by2725263a1bc4afc@mail.gmail.com>
	<002901c8130d$fc836030$f58a2090$@com>
	<680945A7-7C6A-4EDC-B180-5D9116E86CE8@jerkys.org>
Message-ID: <363f21ee0710241657g701619e2s77e60f03e16af260@mail.gmail.com>

keep in mind, this proxy is just relaying the traffic.  It is NOT addressing
any QoS issues with SoHo work.

On 10/21/07, Jerky <lists at jerkys.org> wrote:
>
>  Think this is what you are looking for:
>
>
>
> http://www.cisco.com/en/US/products/ps7057/products_data_sheet0900aecd80546906.html
>
>
> jeff
>
>  On Oct 20, 2007, at 6:40 AM, Curt Shaffer wrote:
>
>  I tried searching on this on cisco.com. Do you have a part number or a
> more direct name that may help? Is this something that is production ready
> or is it still beta quality?
>
>
>
> Thanks
>
>
>
> Curt
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [
> mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether.net>]
> *On Behalf Of *Scott Voll
> *Sent:* Wednesday, October 17, 2007 2:05 PM
> *To:* Jerky
> *Cc:* Linsemier, Matthew; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Home user
>
>
>
> no Cisco CPE required.
>
>
>
> ip phone
>
>   |
>
> internet connection
>
>   |
>
>   --------------------------- phone proxy
>
>   |                               |
>
>   |                                |
>
> FW / router                  |
>
>   |                                |
>
> internal network ------ voice network
>
>
>
> basically you save the money of a cisco CPE by getting the phone proxy.
> let the end users VPN in with the client for data purposes or use Citrix to
> get around VPN all together.
>
>
>
> the phone proxy has a north / South interface so the only thing going
> through it is the authenticated voice traffic.
>
>
>
> hope that's understandable.
>
>
>
> scott
>
>
>
>
>
>
>
> On 10/17/07, *Jerky* <lists at jerkys.org> wrote:
>
> so it would be more like this:
>
>
>
>
> Cisco 871
>
>      |
>
> DSL CABLE
>
>      |
>
> Internet
>
>      |
>
> T1 Connection (Serial0/0/0)
>
>                     |
>
>    _____ 3800 _____
>
>   |                                  |
>
> ethernet 0/0          ethernet 0/1
>
> |                                    |
>
> PIX/ASA                    3800 (Cisco 871 VPN's terminate here)
>
> |                                    |
>
> LAN(computers)       LAN (Voice)
>
>
>
>
>
>
>
> Homefully my crude diagram makes sense. Do your home users have access to
> any data on the computer network side. Or is the 87x VPNs solely for getting
> to the voice network If users access things on the "computer" side would you
> have a separate tunnel setup just for that?
>
>
>
>
> Thanks for so much helping enlighten me. It's been very helpful.
>
>
>
>
> jeff
>
>
>
>
>
>
>
>
>
>
> On Oct 17, 2007, at 10:19 AM, Linsemier, Matthew wrote:
>
>
>
>  In our environment we utilize PIX firewalls (still have to upgrade to
> ASA's) to handle our firewall needs and then use the 3800 series router just
> to terminate the DMVPN home users.  They are deployed in parallel and sit
> behind a perimeter screening router (another 3800 series router).  We shied
> away from using the PIX for the simple fact that while it would preserve QoS
> markings, we couldn't do any remarking or shaping in the device.  Maybe this
> has changed in the ASA, but I don't think you have the control like you do
> in IOS (such as qos pre-classify, shaping, policing,  etc.).  Depending on
> how many tunnels you plan on using, you could use a router much smaller than
> a 3800 series to terminate the end nodes.
>
>
>
> On the home user end we have the Cisco 871/877 routers configured to
> support wired and wireless connections using three VLANS.  We have a VLAN
> configured for corporate connectivity, one VLAN configured as a voice VLAN,
> and then a VLAN configured for untrusted traffic.  One Ethernet port on the
> router provides connectivity to the corporate and voice VLANS, while the
> remaining three are configured as untrusted.  Similarly with Wireless, we
> extend PEAP authentication from the headquarters and authenticate users to
> the corporate VLAN, and use a WPA-PSK to secure the untrusted connections.
> This way the users plug in their phone, then their laptop/docking station to
> port 0, and any other home devices can be connected to port 1-3 or use the
> wireless WPA-PSK network and be logically segregated (using ACL's) from any
> data on the corporate network.  This way we can also control QoS and mark
> down all traffic that enters the router from the untrusted network.  So when
> said employees son or daughter starts downing a 2 gig torrent from a home
> PC, they don't kill the voice or impact the corporate workflow.  Eventually
> we will be implementing 802.1x on the corporate port for additional
> security, but have had mixed results of getting it to work with Windows XP.
>
>
> Hope this helps.
>
>
>
> Matt
>
>
>
>
>
> *From:* Jerky [mailto:lists at jerkys.org <lists at jerkys.org>]
> *Sent:* Tuesday, October 16, 2007 6:32 PM
> *To:* Linsemier, Matthew
> *Cc: *Curt Shaffer; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Home user
>
>
>
> This has been kicked around for a while since we moved to CallManager but
> not much thought has been given to it. I'm trying to understand how your
> hardware is setup. How would it look, similar to one of these?
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845
> <--Ethernet--> LAN
>
>
>
> or
>
>
>
> 87x router <---DSL or Cable---> INTERNET <--T1 connection---> 3845 <--->
> ASA or PIX Firewall <--Ethernet--> LAN
>
>
>
> Is the 3800 used for all your firewalling needs in lieu of something like
> an ASA or PIX? Sonicwall's are currently in place and haven't worked very
> well for the remote users it was tested with. The Sonicwalls we have don't
> have anything similar to what the 871's seem to have in regards to vlans and
> packet tagging. We would probably kick the Sonicwalls out if something else
> would work better.
>
>
>
> jeff
>
>
>
> On Oct 16, 2007, at 8:16 AM, Linsemier, Matthew wrote:
>
>
>
> We currently have about 40 production remote home teleworkers that have
> been deployed using Cisco 871/877 wireless routers and a 7960 phones.  We
> are using a Cisco 3845 series router at the head-end so that we can control
> QoS tagging on the egress / ingress points of both sides of the VPN tunnel.
> We are using a phase 2 DMVPN solution dual-homed to two sites to provide
> secure redundant connectivity.
>
>
>
> It took me a bit to tweak my router configurations (I started on Cisco
> 831/837 routers) to get the results that we wanted, but all and all our
> users are happy.  There is the occasional jitter and packet loss (it is the
> Internet mind you) but g.729 is working quite well coupled with business
> cable and DSL services.
>
>
>
> If you have any other questions, feel free to ask.
>
>
>
> Matt
>
>
>
> *From: *cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net
> <cisco-voip-bounces at puck.nether.net>] *On Behalf Of *Curt Shaffer
> *Sent:* Monday, October 15, 2007 6:37 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Home user
>
>
>
> I was wondering want everyone out there is using for the situation where
> you have someone on your CCM or CCME that has 1 phone at a home office.
> Something tells me an ASA is overkill and I haven't found solid information
> that any of the 87x routers support tagging QoS of packets going through the
> VPN tunnel. We would obviously like to have QoS in place even though it's
> not respected at their ISP just to make sure the VPN/Voice packets are
> leaving their routers first as a best effort to get some quality.
>
>
>
> Thanks
>
>
>
>
>  ------------------------------
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be
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From tednugent73 at gmail.com  Wed Oct 24 19:59:34 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Wed, 24 Oct 2007 19:59:34 -0400
Subject: [cisco-voip] any Unity 5 + Exchange 2007 updates?
In-Reply-To: <f84a38d30710241512w1565b883p65f20f7b26116d91@mail.gmail.com>
References: <471FC101.7070407@wcyv.com>
	<706130.81328.qm@web57213.mail.re3.yahoo.com>
	<f84a38d30710241512w1565b883p65f20f7b26116d91@mail.gmail.com>
Message-ID: <22e6380b0710241659o5ccaeeccu4f141ff81e5828f@mail.gmail.com>

You need to order the upgrade .lic 4.x to 5.x using your UCSS or SAU, they
send you the media and a PAK to register to get an upgrade .lic. Once you
get the 4 to 5 .lic you can use your existing 4.x licenses. I also tried to
contact licensing and got shot down and instructed to go this path. It sucks
if you don;t know this up front you need to wait for them to send you the
damn PAK.

Unity/exchange 2007 went fine for me, with the exception of the exchange
2007 config its almost exactly like a win/exch 2k3 UM install. Nothing
special.



On 10/24/07, Scott Voll <svoll.voip at gmail.com> wrote:
>
> I'm in the process of getting ready.
>
> I just emailed Licensing the MAC address and licenses needed and now there
> asking for the PAK.  How is the proper way to get the license files I need
> for the upgrade?
>
> Scott
>
>
>  On 10/24/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> >
> > I remember someone doing a Unity 5 + Exchange 2007
> > installation. I was wondering how that turned out. The
> > consensus I've been hearing is that it's supported BUT
> > don't ignore the 100 or so caveats that come along
> > with it.
> >
> > Would anyone care to share their experiences
> > installing these two applications together?
> >
> > Paul
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From mthompson729 at gmail.com  Wed Oct 24 20:22:53 2007
From: mthompson729 at gmail.com (Michael Thompson)
Date: Wed, 24 Oct 2007 20:22:53 -0400
Subject: [cisco-voip] SRST
In-Reply-To: <643A3A543DB5224499BC3D73473AEC27921827@NYEXMB2.shearman.com>
References: <NYEXM4Cla2kdlV7wGHc0001a302@NYEXMB2.shearman.com>
	<643A3A543DB5224499BC3D73473AEC27921827@NYEXMB2.shearman.com>
Message-ID: <363f21ee0710241722l18605820p173845743980d594@mail.gmail.com>

Erik,

    The only TRUE way to simulate it is to bring down MGCP and truly enter
the DEFAULT applicaiton state (H.323 / SRST).  If all you're trying to test
is the ability to write H.323 dial peers then you can do just that.  The
problem is that unless you break MGCP, it doesn't go into fallback state
from a system perspective.  In the MGCP fallback config segment, it refers
to Application Alternate DEFAULT (differs from version to version on the
syntax).  That's the mode you go into after MGCP dies.  Once you're there,
the MGCP dial peers go down and your H.323 dial peers take over (including
your PRI local association).  That's going to be the only real way to test
if SRST is going to work.

MAKE SURE you configure the Applicaiton Alternate DEFAULT.  failure to do so
will give VERY eratic results (personal experience) and HOURS of lost bliss
banging your head against the wall.

my 2 pennies.

On 10/23/07, Erik Stillman <Erik.Stillman at shearman.com> wrote:
>
> Thanks Andrew, it's exactly what I did.
>
>
> Anyway to get those dial-peers working with like an FXO?
>
> -----Original Message-----
> From: "Riley, Andrew" <Andrew.Riley at getronics.com>@SHSDOMAIN
> Sent: Tuesday, October 23, 2007 3:29 PM
> To: Erik Stillman; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] SRST
>
> Put an ACL in the network to force the one phone to register to SRST.
>
> If you are doing MGCP, then the ISDN interface is under the control of
> CCM and won't work with the dial peers.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Stillman
> Sent: Tuesday, 23 October 2007 7:11 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] SRST
>
> Hey guys, is it possible to test one single phone on SRST while the rest
> of the site is registered to CallManager?
>
> I can get the test registered with the gateway, but it won't dial out.
> Does this have something got do with the PRI being MGCP controlled?
>
> My goal is to do some testing without having to take the site down.
>
> All input appreciated!
>
> Many thanks,
> Erik
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net@SHSDOMAIN On Behalf Of
> cisco-voip-request at puck.nether.net
> Sent: Monday, October 22, 2007 4:13 PM
> To: cisco-voip at puck.nether.net
> Subject: cisco-voip Digest, Vol 56, Issue 135
>
> Send cisco-voip mailing list submissions to
>        cisco-voip at puck.nether.net
>
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>
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>
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From mthompson729 at gmail.com  Wed Oct 24 20:53:00 2007
From: mthompson729 at gmail.com (Michael Thompson)
Date: Wed, 24 Oct 2007 20:53:00 -0400
Subject: [cisco-voip] 7970 AUX port pinout?
In-Reply-To: <535625f70710230902w73c4d474g6260caa0ad29cac7@mail.gmail.com>
References: <471D10B6.1010707@gmail.com>
	<f84a38d30710230829y742b473fs44c783a851abcd7d@mail.gmail.com>
	<471E187F.2020201@gmail.com>
	<535625f70710230902w73c4d474g6260caa0ad29cac7@mail.gmail.com>
Message-ID: <363f21ee0710241753x1e5f8712g8a9187cef6001fa5@mail.gmail.com>

Noob,
    If the best use of your time on this site is to berate someone's
questions, then I suggest you don't waste any more of your time here.

People are here to ask questions, whether earth shattering or inane.  If you
don't have constructive input, then don't output.


On 10/23/07, Voice Noob <voicenoob at gmail.com> wrote:
>
> "Because it's there"
>
> WOW I wish I was an admin somewhere that didn't have anything better to do
> with my day than look up pinouts. What a life.
>
>
>  On 10/23/07, Robert Kulagowski <rkulagow at gmail.com> wrote:
> >
> > Scott Voll wrote:
> > > WHY?  What are you doing with the AUX port that you need pin out?
> >
> > Because it's there?  Because older versions of the admin guide for the
> > 7970 tell you the pinout of the ethernet ports (gah!), but not the
> > serial port?  Plain-old "Hrmm, what's that for?"?
> >
> > See?  Lots of reasons!
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From michael.pereira at us.didata.com  Wed Oct 24 22:12:26 2007
From: michael.pereira at us.didata.com (Michael Pereira (US))
Date: Wed, 24 Oct 2007 22:12:26 -0400
Subject: [cisco-voip] CCM 5.1.2 offhook picks up any line
References: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local><471F9D6A.9010009@saskin.net><C1FE15183DA37645BC0633BC604E44F0073844AB@USNAEXCH.na.didata.local>
	<C1FE15183DA37645BC0633BC604E44F013E906@USNAEXCH.na.didata.local>
Message-ID: <C65033A817B5734392E55CB9DF446104080D6E67@USNAEXCH.na.didata.local>

Service Params -> Call Manager -> Clusterwide Params -> Always Use Prime Line = True


________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Wed 10/24/2007 4:34 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM 5.1.2 offhook picks up any line


If my Line 2 rings I want to be forced when going offhook to press Line 2, isn't that a service parameter? Instead if I get a call on Line 2 today when I go offhook I get connected to Line 2. 
 
Pretty much all my service parameters are default.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Wed 10/24/2007 3:55 PM
To: Matthew Saskin
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?



I recall how to modify the .xml file in CallManager 4.1, how do I modify
the DistinctiveRingList.xml in 5.1.2?

-----Original Message-----
From: Matthew Saskin [mailto:matt at saskin.net]
Sent: Wednesday, October 24, 2007 3:31 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?

DistinctiveRingList.xml

-matt

Jason Aarons (US) wrote:
> Customer asked why their line 2, line 3, etc only have 7 ring choices
in
> Settings > Ring Type > X when their default ring has 25 choices?
>
> 
>
> I said "because it is what it is" (works on my 3 year old all the
> time).....what's the real reason?
>
> 
>
>
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From wsisk at cisco.com  Wed Oct 24 22:59:32 2007
From: wsisk at cisco.com (Wesley Sisk)
Date: Wed, 24 Oct 2007 22:59:32 -0400 (EDT)
Subject: [cisco-voip] 7970s will not register
In-Reply-To: <5DB17A1560B7EE4186747DF43A6D4C6CCA0531@WIMKEEX07.foleylaw.com>
References: <5DB17A1560B7EE4186747DF43A6D4C6CCA0531@WIMKEEX07.foleylaw.com>
Message-ID: <Pine.GSO.4.64.0710242253140.29065@clubhouse-7.cisco.com>

Hi Pete,

That defect is only applicable if phones are powered up but lack IP 
connectivity to CM for an extended period of time.  The workaround is to 
power the phone down/up.

If that didn't do it you're in another state. Possibilities:
1. network issue - are the phones getting DHCP option 150 or DNS 
resolution for CiscoCM1 to find their TFTP server?  Do the phones get 
valid IP's, subnet mask, defeault gateway?  Can you ping the phones from 
the CM server.  Is there any firewall or ACL that would block IP 
connectivity?  On the CM server if your run 'netstat -an | findstr 2000' 
do you see the IP of the phones in the output?  If not, your phones are 
attempting to contact CM and the problem is closer to the phones.

2. dead phones - have these phones every registered successfully?  Are 
they running SIP loads?  Do the icons and graphics on the LCD's look 
differnt from any of your other phones?  Can you point your web brower to 
the IP of the phone?  Are there any 'interesting' messages in the status 
logs from the phone web page?

There are many components between 'phone' and 'cm' that may inhibit 
registration.  We need a bit more info on what is working to figure out 
what is not working.

/Wes

On Wed, 24 Oct 2007, Ruttman, Pete G. wrote:

> We have a couple 7970 phones that will not register with callmanager.  A
> previous engineer told me that the there was a bug that caused the phone
> to not register if they were not connected to callmanager for an
> extended period.  We have had the phones offline now for years and I
> just tried to plug them in and they will not register.  I tried to reset
> the phone to factory based on the directions below:
>
> Unplug the power cable from the phone and then plug it back in. The
> phone begins its power up cycle.
> While the phone powers up, and before the Speaker button flashes on and
> off, press #. Each line button flashes on and off in sequence.
> Press 123456789*0#. You can press a key twice in a row, but if you press
> the keys out of sequence, the factory reset does not take place. Now the
> lights should change color from amber to red.
> Press 1 in order to maintain the current network configuration settings
> for the phone when the phone resets.
>
> Does anyone have any other suggestions to fix these?
>
> Pete
>
>
>
> The preceding email message may be confidential or protected by the attorney-client privilege. It is not intended for transmission to, or receipt by, any unauthorized persons.  If you have received this message in error, please (i) do not read it, (ii) reply to the sender that you received the message in error, and (iii) erase or destroy the message.  Legal advice contained in the preceding message is solely for the benefit of the Foley & Lardner LLP client(s) represented by the Firm in the particular matter that is the subject of this message, and may not be relied upon by any other party.
>
>
> Internal Revenue Service regulations require that certain types of written advice include a disclaimer. To the extent the preceding message contains advice relating to a Federal tax issue, unless expressly stated otherwise the advice is not intended or written to be used, and it cannot be used by the recipient or any other taxpayer, for the purpose of avoiding Federal tax penalties, and was not written to support the promotion or marketing of any transaction or matter discussed herein.
>
>

From jsteinberg at gmail.com  Thu Oct 25 01:07:18 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 25 Oct 2007 01:07:18 -0400
Subject: [cisco-voip] CCMAdmin in 5.1
In-Reply-To: <22e6380b0710241648x27721c19p9d82cf9a244e555f@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F007384779@USNAEXCH.na.didata.local>
	<22e6380b0710241648x27721c19p9d82cf9a244e555f@mail.gmail.com>
Message-ID: <f1b65f880710242207o7055f4fbyc872096461932c2f@mail.gmail.com>

just fyi....thankfully, the scrolling issue is resolved in CM6.   with
cm6 ccmadmin will resize according to whatever resolution you are
using.

On 10/24/07, Ted Nugent <tednugent73 at gmail.com> wrote:
> 1280 appears to be the magic number, anything less and you're panning..
>
>
> On 10/24/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> >
> >
> >
> >
> > I bought a 24" monitor thinking I would no longer have to scroll to the
> right in CCMAdmin to see items on the right side, what is the required
> screen size/resolution to not have to right scroll in CCMAdmin? 40"?
> >
> >
> >
> >
> > ________________________________
>
> >
> >
> >
> > Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
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> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
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From CarlosOrtiz at bayviewfinancial.com  Thu Oct 25 08:50:18 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 25 Oct 2007 08:50:18 -0400
Subject: [cisco-voip] Change phone numbers bat
In-Reply-To: <000a01c81665$f1542270$661b1b0a@cs.myharris.net>
Message-ID: <OF299651BD.71944195-ON8525737F.004669E3-8525737F.00468650@bftg.com>

I think you should be able to accomplish this by using the Add lines 
feature in BAT.  I added a second line on about 50 phones and noticed that 
there is an "override the existing configuration" option. I assume that 
this means if line 1 exists on my phone and is currently DN 2xxx you 
overwrite with the DN 1xxx in my BAT file.

To use this add lines feature you need to have the MAC address of the 
phones.  There are many options to export this data from the Call Mgr 
database.

Note that after you fill in the BAT file with the needed information and 
export the file to a CSV file that there is a blank line at the bottom of 
the file that needs to be removed.  If not the import fails. This is Call 
Mgr 4.1.3 sr5b so not sure if it differs on other versions.

Note that the line index field indicates which line you are adding ( 
changing?? ) on the phone

I would test this on a couple of phones first to verify and then do the 
bulk edit.


Carlos




"cisco.voip" <cisco.voip at verizon.net> 
Sent by: cisco-voip-bounces at puck.nether.net
10/24/2007 01:39 PM

To
<cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] Change phone numbers bat






I need to change my local CCM phones so the phone numbers are 555-1XXX 
from
555-2XXX.
Can the bulk admin tool do this in one step- I have like 600 phones need 
to
change?

Tx


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From erickbee at gmail.com  Thu Oct 25 08:52:38 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Thu, 25 Oct 2007 07:52:38 -0500
Subject: [cisco-voip] Secondary DNs only have 7 ring choices?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0073844AB@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>
	<471F9D6A.9010009@saskin.net>
	<C1FE15183DA37645BC0633BC604E44F0073844AB@USNAEXCH.na.didata.local>
Message-ID: <f4445faf0710250552v5e16eacvfbd8e84a56641d12@mail.gmail.com>

You need to use a TFTP client (TFTPD32, etc) to download the file from
the server, then modify it as your used to. Then to put the new file
back, SSH into the Call Manager and issue a 'file delete tftp
DistinctiveRingList.xml' command to delete the old file. Then you go
into the OS Admin (IPT Platform) web page and go under Software ->
TFTP and upload the new XML file. I've found (at least on the 5.1
versions I've done this on) that if you use the web page to upload the
file and it already exists it puts it there with Upper or lowercase at
front, etc and you have 2 files which are the same name just with
upper/lower case differences.

Afterwards, you may need to restart the Cisco TFTP service if it is
caching the old file.

On 10/24/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> I recall how to modify the .xml file in CallManager 4.1, how do I modify
> the DistinctiveRingList.xml in 5.1.2?
>
> -----Original Message-----
> From: Matthew Saskin [mailto:matt at saskin.net]
> Sent: Wednesday, October 24, 2007 3:31 PM
> To: Jason Aarons (US)
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?
>
> DistinctiveRingList.xml
>
> -matt
>
> Jason Aarons (US) wrote:
> > Customer asked why their line 2, line 3, etc only have 7 ring choices
> in
> > Settings > Ring Type > X when their default ring has 25 choices?
> >
> >
> >
> > I said "because it is what it is" (works on my 3 year old all the
> > time).....what's the real reason?
> >
> >
> >
> >
> ------------------------------------------------------------------------
> >
> > * Disclaimer: This e-mail communication and any attachments may
> contain
> > confidential and privileged information and is for use by the
> designated
> > addressee(s) named above only. If you are not the intended addressee,
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> > error and that any use or reproduction of this email or its contents
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> > strictly prohibited and may be unlawful. If you have received this
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> >
> >
> ------------------------------------------------------------------------
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>
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>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
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From Richard.vanBerkel at telindus.nl  Thu Oct 25 09:12:01 2007
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Thu, 25 Oct 2007 15:12:01 +0200
Subject: [cisco-voip] any Unity 5 + Exchange 2007 updates?
References: <471FC101.7070407@wcyv.com><706130.81328.qm@web57213.mail.re3.yahoo.com>
	<f84a38d30710241512w1565b883p65f20f7b26116d91@mail.gmail.com>
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157ECC2@tnl1mx1.telindus.intra>

I have it working in our lab.

It's not really a problem to install. 

 

No problem so far.

 

Groeten Richard,

Mailto:Richard.vanBerkel at Telindus.nl
<mailto:Richard.vanBerkel at Telindus.nl> 

  

<skype:richardvanberkel?call> 

<http://www.skype.com/go/download>  

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: donderdag 25 oktober 2007 0:13
To: Paul Choi
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any Unity 5 + Exchange 2007 updates?
Importance: High

 

I'm in the process of getting ready.  

 

I just emailed Licensing the MAC address and licenses needed and now
there asking for the PAK.  How is the proper way to get the license
files I need for the upgrade?

 

Scott

 

On 10/24/07, Paul Choi <asobihoudai at yahoo.com> wrote: 

I remember someone doing a Unity 5 + Exchange 2007
installation. I was wondering how that turned out. The 
consensus I've been hearing is that it's supported BUT
don't ignore the 100 or so caveats that come along
with it.

Would anyone care to share their experiences
installing these two applications together? 

Paul

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From SCASPER at mtb.com  Thu Oct 25 09:20:06 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Thu, 25 Oct 2007 09:20:06 -0400
Subject: [cisco-voip]  7921 Question
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157ECC2@tnl1mx1.telindus.intra>
References: <471FC101.7070407@wcyv.com><706130.81328.qm@web57213.mail.re3.yahoo.com>
	<f84a38d30710241512w1565b883p65f20f7b26116d91@mail.gmail.com>
	<B42A9A0E2A14B943A3B7EA2FA7A0D8860157ECC2@tnl1mx1.telindus.intra>
Message-ID: <47205FC6.9874.0039.0@mtb.com>

 I know this may be dumb question but I am being told that the use of a 7921 does not require the use of a Cisco Access Point. Is that true?
 
 
Thanks,

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee.  If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission.  If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy.  This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
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From matt at saskin.net  Thu Oct 25 09:38:29 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 25 Oct 2007 09:38:29 -0400
Subject: [cisco-voip] 7921 Question
In-Reply-To: <47205FC6.9874.0039.0@mtb.com>
References: <471FC101.7070407@wcyv.com><706130.81328.qm@web57213.mail.re3.yahoo.com>	<f84a38d30710241512w1565b883p65f20f7b26116d91@mail.gmail.com>	<B42A9A0E2A14B943A3B7EA2FA7A0D8860157ECC2@tnl1mx1.telindus.intra>
	<47205FC6.9874.0039.0@mtb.com>
Message-ID: <47209C55.8060902@saskin.net>

I would assume that you shouldn't need a Cisco AP.  I have two 7920's at 
home running just fine on Linksys AP's.

-matt

STEVEN CASPER wrote:
>  I know this may be dumb question but I am being told that the use of a 
> 7921 does not require the use of a Cisco Access Point. Is that true?
>  
>  
> Thanks,
> 
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
> 
> ************************************
> This email may contain privileged and/or confidential information that 
> is intended solely for the use of the addressee. If you are not the 
> intended recipient or entity, you are strictly prohibited from 
> disclosing, copying, distributing or using any of the information 
> contained in the transmission. If you received this communication in 
> error, please contact the sender immediately and destroy the material in 
> its entirety, whether electronic or hard copy. This communication may 
> contain nonpublic personal information about consumers subject to the 
> restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. 
> You may not directly or indirectly reuse or disclose such information 
> for any purpose other than to provide the services for which you are 
> receiving the information.
> There are risks associated with the use of electronic transmission. The 
> sender of this information does not control the method of transmittal or 
> service providers and assumes no duty or obligation for the security, 
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> ************************************
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> 
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From wsisk at cisco.com  Thu Oct 25 09:47:18 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 09:47:18 -0400
Subject: [cisco-voip] cm5.1 and 6.0 'SRs' on Cisco.com
Message-ID: <47209E66.1040704@cisco.com>

For CM3.x and 4.x days Cisco had SR releases every 1-2 months to make 
fixes available on Cisco.com.  There was no similar process for CM5.x or 
6.x until now.  Every 4 weeks Cisco will now publish an 'SR' for 5.1 and 
6.0 to this location:
http://www.cisco.com/cgi-bin/tablebuild.pl/CUCM-ES5X
http://www.cisco.com/cgi-bin/tablebuild.pl/CUCM-ES6X

The 'SR' comes in the form of an engineering special that has been 
tested for 4 weeks.  This is the same evaluation used for 3.x and 4.x 
"SRs" but without the name change.

/Wes

From l.durso at gmail.com  Thu Oct 25 10:01:03 2007
From: l.durso at gmail.com (=?utf-8?B?TGVvbmFyZG8gRCdVcnNv?=)
Date: Thu, 25 Oct 2007 14:01:03 +0000
Subject: [cisco-voip] Callmanager cluster in Wan
Message-ID: <929274954-1193320798-cardhu_decombobulator_blackberry.rim.net-1398273527-@bxe006.bisx.produk.on.blackberry>

Hi there

A customer of mine have two site: one in EU, one in USA with two different clusters of callmanager. Actually there is a trunk that permits to call between. 
I have users that want to have the same extension mobility profile on each side. 
I have evaluated that with a latency of 120ms it is not possible to have 1 centralized ccm cluster infrastructure with 1pub and 2sub in EU and 1 sub in USA. But what happens if i have all ccm server in a single location EU and a single voice gateway with SRST in USA and the  ip phones that directly point to ccm cluster in EU?

Regards
Leo 
---
Leonardo D'Urso
l.durso at gmail.com


From SCASPER at mtb.com  Thu Oct 25 09:59:29 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Thu, 25 Oct 2007 09:59:29 -0400
Subject: [cisco-voip] 7921 Question
In-Reply-To: <47209C55.8060902@saskin.net>
References: <471FC101.7070407@wcyv.com><706130.81328.qm@web57213.mail.re3.yahoo.com>
	<f84a38d30710241512w1565b883p65f20f7b26116d91@mail.gmail.com>
	<B42A9A0E2A14B943A3B7EA2FA7A0D8860157ECC2@tnl1mx1.telindus.intra>
	<47205FC6.9874.0039.0@mtb.com> <47209C55.8060902@saskin.net>
Message-ID: <47206901.9874.0039.0@mtb.com>

Ok,  thanks, how do you connect back to your Call Manager....VPN?
 
Steve

>>> "Matthew Saskin" <matt at saskin.net> 10/25/2007 9:38 AM >>>
I would assume that you shouldn't need a Cisco AP.  I have two 7920's at 
home running just fine on Linksys AP's.

-matt

STEVEN CASPER wrote:
>  I know this may be dumb question but I am being told that the use of a 
> 7921 does not require the use of a Cisco Access Point. Is that true?
>  
>  
> Thanks,
> 
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
> 
> ************************************
> This email may contain privileged and/or confidential information that 
> is intended solely for the use of the addressee. If you are not the 
> intended recipient or entity, you are strictly prohibited from 
> disclosing, copying, distributing or using any of the information 
> contained in the transmission. If you received this communication in 
> error, please contact the sender immediately and destroy the material in 
> its entirety, whether electronic or hard copy. This communication may 
> contain nonpublic personal information about consumers subject to the 
> restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. 
> You may not directly or indirectly reuse or disclose such information 
> for any purpose other than to provide the services for which you are 
> receiving the information.
> There are risks associated with the use of electronic transmission. The 
> sender of this information does not control the method of transmittal or 
> service providers and assumes no duty or obligation for the security, 
> receipt, or third party interception of this transmission.
> ************************************
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip 





************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee.  If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission.  If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy.  This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
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From cisco at adman.net  Thu Oct 25 10:04:58 2007
From: cisco at adman.net (Adam)
Date: Thu, 25 Oct 2007 09:04:58 -0500
Subject: [cisco-voip] Upgrade Hardware Question
Message-ID: <c6f32ce40710250704o36ae9352q1a2fa06772f58f25@mail.gmail.com>

I have a customer that has the following environment:

* CallMananger 4.1(3) running on two identical MCS-7835-H1-IPC1. IPCC 4.0(4)
also runs coresident on the Publisher.
* Unity UM 4.0(3) running in a failover configuration on a as the primary
MCS-7835H-3.0-ECS1 and a UNITY-SVR7837-1A as the secondary.

We would like migrate to CallManager 6.0 and Unity 4.2, while replacing the
Unity hardware and ordering a new standalone UCCX server at the same time.
In order to have the least impact on up time we thought it would be ideal to
install the CallManagers onto new hardware. What would be the best way to
achieve this. I know that I should order an ECS1 server for the new Unity
boxes, but these are probably the servers that I will actually end up using
for the CallManagers. I would like to retask the two MCS-7835-H1-IPC1
servers as the Unity servers after the CallManagers have been upgraded, but
I'm not sure where I'm left from a licensing standpoint. Will I be able to
obtain and use a Unity platform preparation disk on these servers? Would
these be TAC supported? Any insight is appreciated.

Thanks,
Adam
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From manoj.kalpage at gmail.com  Thu Oct 25 10:19:30 2007
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Thu, 25 Oct 2007 23:19:30 +0900
Subject: [cisco-voip] Cannot login to Callmanager 5.0 admin page
Message-ID: <bbe09c1d0710250719x783bab50v1a27ead04d60d773@mail.gmail.com>

Hi All,
I am having very strange issue with my newly installed Call Manager
5.0 server. It doesn't accept my credentials for call manager
administration page bellow.
http://192.168.0.10/ccmadmin.
BUT I can login to the URL http://192.168.0.10/iptflatform and server
CLI. I have reset the password using
SET PASSWORD ADMIN and SET PASSWORD SECURITY but still it keep saying
"Login failed. Please try again". I am just wondering I can create
another username and password for ccmadmin using CLI prompt. Any
information would be greatly appreciated.

Thank you.
MK

From bcarter at sentinel.com  Thu Oct 25 10:32:29 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 25 Oct 2007 09:32:29 -0500
Subject: [cisco-voip] Cannot login to Callmanager 5.0 admin page
In-Reply-To: <bbe09c1d0710250719x783bab50v1a27ead04d60d773@mail.gmail.com>
References: <bbe09c1d0710250719x783bab50v1a27ead04d60d773@mail.gmail.com>
Message-ID: <6A0744639B86DD459433108DBF101D26019FE767@MAIL01.sentinel.com>

The user name on http://x.x.x.x/cmadmin is ccmadministrator 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
Sent: Thursday, October 25, 2007 9:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cannot login to Callmanager 5.0 admin page

Hi All,
I am having very strange issue with my newly installed Call Manager 5.0
server. It doesn't accept my credentials for call manager administration
page bellow.
http://192.168.0.10/ccmadmin.
BUT I can login to the URL http://192.168.0.10/iptflatform and server
CLI. I have reset the password using SET PASSWORD ADMIN and SET PASSWORD
SECURITY but still it keep saying "Login failed. Please try again". I am
just wondering I can create another username and password for ccmadmin
using CLI prompt. Any information would be greatly appreciated.

Thank you.
MK
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From wsisk at cisco.com  Thu Oct 25 10:38:17 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 10:38:17 -0400
Subject: [cisco-voip] Cannot login to Callmanager 5.0 admin page
In-Reply-To: <6A0744639B86DD459433108DBF101D26019FE767@MAIL01.sentinel.com>
References: <bbe09c1d0710250719x783bab50v1a27ead04d60d773@mail.gmail.com>
	<6A0744639B86DD459433108DBF101D26019FE767@MAIL01.sentinel.com>
Message-ID: <4720AA59.6020305@cisco.com>

And in early version of 5.0 it was case sensitive: CCMAdministrator

Carter, Bill wrote:
> The user name on http://x.x.x.x/cmadmin is ccmadministrator 
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
> Sent: Thursday, October 25, 2007 9:20 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Cannot login to Callmanager 5.0 admin page
>
> Hi All,
> I am having very strange issue with my newly installed Call Manager 5.0
> server. It doesn't accept my credentials for call manager administration
> page bellow.
> http://192.168.0.10/ccmadmin.
> BUT I can login to the URL http://192.168.0.10/iptflatform and server
> CLI. I have reset the password using SET PASSWORD ADMIN and SET PASSWORD
> SECURITY but still it keep saying "Login failed. Please try again". I am
> just wondering I can create another username and password for ccmadmin
> using CLI prompt. Any information would be greatly appreciated.
>
> Thank you.
> MK
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Thu Oct 25 10:43:02 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 10:43:02 -0400
Subject: [cisco-voip] Callmanager cluster in Wan
In-Reply-To: <929274954-1193320798-cardhu_decombobulator_blackberry.rim.net-1398273527-@bxe006.bisx.produk.on.blackberry>
References: <929274954-1193320798-cardhu_decombobulator_blackberry.rim.net-1398273527-@bxe006.bisx.produk.on.blackberry>
Message-ID: <4720AB76.7070903@cisco.com>

I've seen many deployments that use this model.  Noteworthy items:
1.register a phone and gw in the US to the EU servers and test the 
timing.  With high RTT cm<->gw and cm<->phone users may notice a delay 
in audio cut through when answering calls.  Faststart for h323 and 
optimized mgcp help some, but delays can still become an issue.  I will 
post a followup message with details when I locate them.
2. dialpan on EU servers will become significantly more complex hosting 
PSTN patterns for US and EU
3. Consider SRST for the US site, possibly using multiple SRST routers 
to allow all phones to register in the event of any network outgate 
between US and EU.
4. if you deploy 3rd gen ipphones (7941/7961/7911/7970/7971, etc.) be 
aware of upgrade issues due to TFTP over high RTT.  Strongly consider 
upgrading to CM version and phone versions that support peer to peer 
firmware sharing. (cm6.0 srnd page 3-15)

More details of this are covered in the SRND:
http://www.cisco.com/go/srnd
Your scenario is covered under "multi-site wan with centralized call 
processing' but depends on information in other sectsions as well.


/Wes

Leonardo D'Urso wrote:
> Hi there
>
> A customer of mine have two site: one in EU, one in USA with two different clusters of callmanager. Actually there is a trunk that permits to call between. 
> I have users that want to have the same extension mobility profile on each side. 
> I have evaluated that with a latency of 120ms it is not possible to have 1 centralized ccm cluster infrastructure with 1pub and 2sub in EU and 1 sub in USA. But what happens if i have all ccm server in a single location EU and a single voice gateway with SRST in USA and the  ip phones that directly point to ccm cluster in EU?
>
> Regards
> Leo 
> ---
> Leonardo D'Urso
> l.durso at gmail.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From manoj.kalpage at gmail.com  Thu Oct 25 10:50:29 2007
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Thu, 25 Oct 2007 23:50:29 +0900
Subject: [cisco-voip] Cannot login to Callmanager 5.0 admin page
In-Reply-To: <4720AA59.6020305@cisco.com>
References: <bbe09c1d0710250719x783bab50v1a27ead04d60d773@mail.gmail.com>
	<6A0744639B86DD459433108DBF101D26019FE767@MAIL01.sentinel.com>
	<4720AA59.6020305@cisco.com>
Message-ID: <bbe09c1d0710250750s3ad0a2i506e378e1beebb1e@mail.gmail.com>

Thank you all. it worked for ccmadministrator.
Best regards,
MK


On 10/25/07, Wes Sisk <wsisk at cisco.com> wrote:
> And in early version of 5.0 it was case sensitive: CCMAdministrator
>
> Carter, Bill wrote:
> > The user name on http://x.x.x.x/cmadmin is ccmadministrator
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
> > Sent: Thursday, October 25, 2007 9:20 AM
> > To: cisco-voip at puck.nether.net
> > Subject: [cisco-voip] Cannot login to Callmanager 5.0 admin page
> >
> > Hi All,
> > I am having very strange issue with my newly installed Call Manager 5.0
> > server. It doesn't accept my credentials for call manager administration
> > page bellow.
> > http://192.168.0.10/ccmadmin.
> > BUT I can login to the URL http://192.168.0.10/iptflatform and server
> > CLI. I have reset the password using SET PASSWORD ADMIN and SET PASSWORD
> > SECURITY but still it keep saying "Login failed. Please try again". I am
> > just wondering I can create another username and password for ccmadmin
> > using CLI prompt. Any information would be greatly appreciated.
> >
> > Thank you.
> > MK
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From matt at saskin.net  Thu Oct 25 10:58:06 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 25 Oct 2007 10:58:06 -0400
Subject: [cisco-voip] 7921 Question
In-Reply-To: <47206901.9874.0039.0@mtb.com>
References: <471FC101.7070407@wcyv.com><706130.81328.qm@web57213.mail.re3.yahoo.com>
	<f84a38d30710241512w1565b883p65f20f7b26116d91@mail.gmail.com>
	<B42A9A0E2A14B943A3B7EA2FA7A0D8860157ECC2@tnl1mx1.telindus.intra>
	<47205FC6.9874.0039.0@mtb.com> <47209C55.8060902@saskin.net>
	<47206901.9874.0039.0@mtb.com>
Message-ID: <4720AEFE.4020804@saskin.net>

No CallManager - these are home phones run on top of Asterisk, not a 
remote work phone :)

That said, yes, I've had users at home behind various devices (Pix 506, 
Cisco 871's, etc.) with phones tied to the office CallManager. 
Generally they have all had 794x phones instead of wireless however.

-matt

STEVEN CASPER wrote:
> Ok,  thanks, how do you connect back to your Call Manager....VPN?
>  
> Steve
> 
>  >>> "Matthew Saskin" <matt at saskin.net> 10/25/2007 9:38 AM >>>
> I would assume that you shouldn't need a Cisco AP.  I have two 7920's at
> home running just fine on Linksys AP's.
> 
> -matt
> 
> STEVEN CASPER wrote:
>  >  I know this may be dumb question but I am being told that the use of a
>  > 7921 does not require the use of a Cisco Access Point. Is that true?
>  > 
>  > 
>  > Thanks,
>  >
>  > Steve Casper
>  > Voice Technologies
>  > M&T Bank
>  > (410) 347-6026
>  >
>  > ************************************
>  > This email may contain privileged and/or confidential information that
>  > is intended solely for the use of the addressee. If you are not the
>  > intended recipient or entity, you are strictly prohibited from
>  > disclosing, copying, distributing or using any of the information
>  > contained in the transmission. If you received this communication in
>  > error, please contact the sender immediately and destroy the material in
>  > its entirety, whether electronic or hard copy. This communication may
>  > contain nonpublic personal information about consumers subject to the
>  > restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.
>  > You may not directly or indirectly reuse or disclose such information
>  > for any purpose other than to provide the services for which you are
>  > receiving the information.
>  > There are risks associated with the use of electronic transmission. The
>  > sender of this information does not control the method of transmittal or
>  > service providers and assumes no duty or obligation for the security,
>  > receipt, or third party interception of this transmission.
>  > ************************************
>  >
>  >
>  > ------------------------------------------------------------------------
>  >
>  > _______________________________________________
>  > cisco-voip mailing list
>  > cisco-voip at puck.nether.net
>  > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> 
> ************************************
> This email may contain privileged and/or confidential information that 
> is intended solely for the use of the addressee. If you are not the 
> intended recipient or entity, you are strictly prohibited from 
> disclosing, copying, distributing or using any of the information 
> contained in the transmission. If you received this communication in 
> error, please contact the sender immediately and destroy the material in 
> its entirety, whether electronic or hard copy. This communication may 
> contain nonpublic personal information about consumers subject to the 
> restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. 
> You may not directly or indirectly reuse or disclose such information 
> for any purpose other than to provide the services for which you are 
> receiving the information.
> There are risks associated with the use of electronic transmission. The 
> sender of this information does not control the method of transmittal or 
> service providers and assumes no duty or obligation for the security, 
> receipt, or third party interception of this transmission.
> ************************************
> 



From svoll.voip at gmail.com  Thu Oct 25 11:19:57 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 25 Oct 2007 08:19:57 -0700
Subject: [cisco-voip] cm5.1 and 6.0 'SRs' on Cisco.com
In-Reply-To: <47209E66.1040704@cisco.com>
References: <47209E66.1040704@cisco.com>
Message-ID: <f84a38d30710250819s1f664945i54212dece36e12ac@mail.gmail.com>

Thanks Wes for all the information you bring to us!

Scott


On 10/25/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> For CM3.x and 4.x days Cisco had SR releases every 1-2 months to make
> fixes available on Cisco.com.  There was no similar process for CM5.x or
> 6.x until now.  Every 4 weeks Cisco will now publish an 'SR' for 5.1 and
> 6.0 to this location:
> http://www.cisco.com/cgi-bin/tablebuild.pl/CUCM-ES5X
> http://www.cisco.com/cgi-bin/tablebuild.pl/CUCM-ES6X
>
> The 'SR' comes in the form of an engineering special that has been
> tested for 4 weeks.  This is the same evaluation used for 3.x and 4.x
> "SRs" but without the name change.
>
> /Wes
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From manoj.kalpage at gmail.com  Thu Oct 25 11:36:31 2007
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Fri, 26 Oct 2007 00:36:31 +0900
Subject: [cisco-voip] Callmanager upgrade 4.1 to 5.0(4)
Message-ID: <bbe09c1d0710250836h677cc2e8w9dbd9cd89a09d1ef@mail.gmail.com>

I am looking for good step by step upgrade guide for callmanager 4.1
to 5.0. Does anyone know?

Thanks
MK

From csvoip at googlemail.com  Thu Oct 25 11:39:59 2007
From: csvoip at googlemail.com (Nick)
Date: Thu, 25 Oct 2007 16:39:59 +0100
Subject: [cisco-voip] Arc Consoles losing connection
Message-ID: <475fda330710250839k66e43713r850e92d0243a8b11@mail.gmail.com>

I have an issue with arc consoles losing connection this is happening to a
number of consoles in the same site when I check the phones the debug
display on the phone it says
 *15:57:33 14: Name=SEP00146905158B Load= 7.0(3.0S) Last=CM-closed-TCP*
each time it loses connection, we have tried re patching to different
switchports and different switches and still seem to be getting the same
issue, could this be an isue with the phone load as it is very old firmware,
all other consoles on the same arc server and other phones in the same site
don't appear to be getting the issue althouh it may not be as noticeable as
on an Arc console.

Has anyone had this issue with Arc before or can make any suggestions on
what else to check
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From tednugent73 at gmail.com  Thu Oct 25 11:50:32 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 25 Oct 2007 11:50:32 -0400
Subject: [cisco-voip] Upgrade Hardware Question
In-Reply-To: <c6f32ce40710250704o36ae9352q1a2fa06772f58f25@mail.gmail.com>
References: <c6f32ce40710250704o36ae9352q1a2fa06772f58f25@mail.gmail.com>
Message-ID: <22e6380b0710250850l6dda53a2hcf503781245e4f97@mail.gmail.com>

Yes it will work but don't do it...! I've got a project starting next week
to fix exactly this problem. This particular customer took it upon
themselves last year to upgrade their entire IPT environment and in the
process they juggled all their servers around. They've since been having
chronic Unity issues that have been going on for months now (very likely
unrelated to the server swap). They've been working with TAC over the last
couple months and since TAC can't seem to figure out the issue they've asked
that all servers go back to the appropriate platform before they will
proceed with additional troubleshooting. This was obviously not TAC first
response, they will work with you the best that they can HOWEVER they can,
and in this case have, asked to retask the servers back to their appropriate
roles. IPC for CM, ECS for UNITY and CCX for IPCC.

My $0.02
Ted


On 10/25/07, Adam <cisco at adman.net> wrote:
>
> I have a customer that has the following environment:
>
> * CallMananger 4.1(3) running on two identical MCS-7835-H1-IPC1. IPCC 4.0(4)
> also runs coresident on the Publisher.
> * Unity UM 4.0 (3) running in a failover configuration on a as the primary
> MCS-7835H-3.0-ECS1 and a UNITY-SVR7837-1A as the secondary.
>
> We would like migrate to CallManager 6.0 and Unity 4.2, while replacing
> the Unity hardware and ordering a new standalone UCCX server at the same
> time. In order to have the least impact on up time we thought it would be
> ideal to install the CallManagers onto new hardware. What would be the best
> way to achieve this. I know that I should order an ECS1 server for the new
> Unity boxes, but these are probably the servers that I will actually end up
> using for the CallManagers. I would like to retask the two MCS-7835-H1-IPC1
> servers as the Unity servers after the CallManagers have been upgraded, but
> I'm not sure where I'm left from a licensing standpoint. Will I be able to
> obtain and use a Unity platform preparation disk on these servers? Would
> these be TAC supported? Any insight is appreciated.
>
> Thanks,
> Adam
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From wsisk at cisco.com  Thu Oct 25 12:01:03 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 12:01:03 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 to 5.0(4)
In-Reply-To: <bbe09c1d0710250836h677cc2e8w9dbd9cd89a09d1ef@mail.gmail.com>
References: <bbe09c1d0710250836h677cc2e8w9dbd9cd89a09d1ef@mail.gmail.com>
Message-ID: <4720BDBF.7080005@cisco.com>

Unless you have a very specific need for CM5.0 you would be much better 
off at 5.1.3.

Here is a complete upgrade checklist and guide:
http://www.cisco.com/iam/ucmproto/index.htm

/Wes

Manoj Kalpage wrote:
> I am looking for good step by step upgrade guide for callmanager 4.1
> to 5.0. Does anyone know?
>
> Thanks
> MK
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Thu Oct 25 12:03:35 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 12:03:35 -0400
Subject: [cisco-voip] Arc Consoles losing connection
In-Reply-To: <475fda330710250839k66e43713r850e92d0243a8b11@mail.gmail.com>
References: <475fda330710250839k66e43713r850e92d0243a8b11@mail.gmail.com>
Message-ID: <4720BE57.1040303@cisco.com>

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From jsteinberg at gmail.com  Thu Oct 25 12:25:38 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 25 Oct 2007 12:25:38 -0400
Subject: [cisco-voip] cm5.1 and 6.0 'SRs' on Cisco.com
In-Reply-To: <f84a38d30710250819s1f664945i54212dece36e12ac@mail.gmail.com>
References: <47209E66.1040704@cisco.com>
	<f84a38d30710250819s1f664945i54212dece36e12ac@mail.gmail.com>
Message-ID: <f1b65f880710250925r6fea6c08pb57661c91de91aa9@mail.gmail.com>

How do I know whether an ES is incorporated into a newer release of CM code?

For example, 5.1.2es 5.1.2.3104-1 was released on October 4th.  5.1.3
was released on Oct 1st.  Is it fair to assume then that the fixes in
5.1.2.3104-1 are not included in 5.1.3, or is it just that the files
were posted at different times?  (also, when I read the 5.1.3 release
notes it doesn't indicate when it branched from 5.1.2)

I'm sure that someone will correct me if I'm wrong but, I don't recall
any SR's being released on the 4.1.2 branch once 4.1.3 was released.

So I'm confused as to why development on 5.1.2 (via ES updates) would
be released concurrently with 5.1.3 versions.

I just want to make sure that if I apply an ES for 5.1.2 that I won't
gain any features/fixes that I might lose if I then upgrade to 5.1.3.
(I can understand losing features for a major upgrade from 5.1.3 to
6.x (just like 4.2 to 5.0)

This release cycle kinda seems a little more like the traditional CCM
ES concept than a SR. Am I reading this incorrectly?

Thanks.
Justin


On 10/25/07, Scott Voll <svoll.voip at gmail.com> wrote:
> Thanks Wes for all the information you bring to us!
>
> Scott
>
>
>
> On 10/25/07, Wes Sisk <wsisk at cisco.com> wrote:
> > For CM3.x and 4.x days Cisco had SR releases every 1-2 months to make
> > fixes available on Cisco.com.  There was no similar process for CM5.x or
> > 6.x until now.  Every 4 weeks Cisco will now publish an 'SR' for 5.1 and
> > 6.0 to this location:
> > http://www.cisco.com/cgi-bin/tablebuild.pl/CUCM-ES5X
> > http://www.cisco.com/cgi-bin/tablebuild.pl/CUCM-ES6X
> >
> > The 'SR' comes in the form of an engineering special that has been
> > tested for 4 weeks.  This is the same evaluation used for 3.x and 4.x
> > "SRs" but without the name change.
> >
> > /Wes
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From wsisk at cisco.com  Thu Oct 25 14:16:28 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 14:16:28 -0400
Subject: [cisco-voip] cm5.1 and 6.0 'SRs' on Cisco.com
In-Reply-To: <f1b65f880710250925r6fea6c08pb57661c91de91aa9@mail.gmail.com>
References: <47209E66.1040704@cisco.com>	
	<f84a38d30710250819s1f664945i54212dece36e12ac@mail.gmail.com>
	<f1b65f880710250925r6fea6c08pb57661c91de91aa9@mail.gmail.com>
Message-ID: <4720DD7C.4000007@cisco.com>

Hi Justin,

5.1.2es will include a subset but not all fixes from 5.1.3.  Later 
5.1.2es will contain fixes that are not present in 5.1.3.  Unfortunately 
there is not a clear answer on the break even point.

The timing of 5.1.2es release was unfortunate in relation to the 5.1.3 
release.  However, there is a large customer base who would rather stay 
on 5.1.2es rather than do complete upgrade to 5.1.3 to avoid the 'new 
feature's that come with 5.1.3.

The release process is most akin to the 4.x SR process.
An 'es' is produced internally.  It is tested and distributed to 
specific customers for 4 weeks.  Provided not major issues are raised in 
4 weeks the es is posted to Cisco.com for publicly available.  Only 1 es 
will post every 4 weeks.  Previously TAC was distributing ES via the SFA 
process several hundred times per month.  Original release schedule had 
maintenance release on Cisco.com once every 8-12 months.  This 
significantly shortens the release cycle for those who need additional 
fixes beyond the base release.

The basic ES process delivers code that is compiled, built as an 
installer and minimally tested for install and service start.  There is 
significant testing and exposure difference between this basic es 
process and and the ES that is posted to Cisco.com.  The ES that is 
posted to Cisco.com actually receives more testing than the SR's of 4.x.

/Wes

Justin Steinberg wrote:
> How do I know whether an ES is incorporated into a newer release of CM code?
>
> For example, 5.1.2es 5.1.2.3104-1 was released on October 4th.  5.1.3
> was released on Oct 1st.  Is it fair to assume then that the fixes in
> 5.1.2.3104-1 are not included in 5.1.3, or is it just that the files
> were posted at different times?  (also, when I read the 5.1.3 release
> notes it doesn't indicate when it branched from 5.1.2)
>
> I'm sure that someone will correct me if I'm wrong but, I don't recall
> any SR's being released on the 4.1.2 branch once 4.1.3 was released.
>
> So I'm confused as to why development on 5.1.2 (via ES updates) would
> be released concurrently with 5.1.3 versions.
>
> I just want to make sure that if I apply an ES for 5.1.2 that I won't
> gain any features/fixes that I might lose if I then upgrade to 5.1.3.
> (I can understand losing features for a major upgrade from 5.1.3 to
> 6.x (just like 4.2 to 5.0)
>
> This release cycle kinda seems a little more like the traditional CCM
> ES concept than a SR. Am I reading this incorrectly?
>
> Thanks.
> Justin
>
>
> On 10/25/07, Scott Voll <svoll.voip at gmail.com> wrote:
>   
>> Thanks Wes for all the information you bring to us!
>>
>> Scott
>>
>>
>>
>> On 10/25/07, Wes Sisk <wsisk at cisco.com> wrote:
>>     
>>> For CM3.x and 4.x days Cisco had SR releases every 1-2 months to make
>>> fixes available on Cisco.com.  There was no similar process for CM5.x or
>>> 6.x until now.  Every 4 weeks Cisco will now publish an 'SR' for 5.1 and
>>> 6.0 to this location:
>>> http://www.cisco.com/cgi-bin/tablebuild.pl/CUCM-ES5X
>>> http://www.cisco.com/cgi-bin/tablebuild.pl/CUCM-ES6X
>>>
>>> The 'SR' comes in the form of an engineering special that has been
>>> tested for 4 weeks.  This is the same evaluation used for 3.x and 4.x
>>> "SRs" but without the name change.
>>>
>>> /Wes
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>       
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>     

From jason.aarons at us.didata.com  Thu Oct 25 15:27:49 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 25 Oct 2007 15:27:49 -0400
Subject: [cisco-voip] DMS 250?
Message-ID: <C1FE15183DA37645BC0633BC604E44F0073C3DE2@USNAEXCH.na.didata.local>

What option should I set for a DMS-250?

 

 

1VG01(config)#isdn switch-type ?

  primary-4ess    Lucent 4ESS switch type for the U.S.

  primary-5ess    Lucent 5ESS switch type for the U.S.

  primary-dms100  Northern Telecom DMS-100 switch type for the U.S.

  primary-dpnss   DPNSS switch type for Europe

  primary-net5    NET5 switch type for UK, Europe, Asia and Australia

  primary-ni      National ISDN Switch type for the U.S.

  primary-ntt     NTT switch type for Japan

  primary-qsig    QSIG switch type

  primary-ts014   TS014 switch type for Australia (obsolete)

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
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From svoll.voip at gmail.com  Thu Oct 25 15:33:44 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 25 Oct 2007 12:33:44 -0700
Subject: [cisco-voip] OT -- Wireless to remote sites?
Message-ID: <f84a38d30710251233y4374391eobe39ab100fee48e7@mail.gmail.com>

is anyone using some kind of Wireless solution to a remote IP Telephony
site?

I have a site that will be $1 million to build fiber to so I'm looking for
options.  I'm thinking something with a licensed band or microwave.  anyone
have experience with something like this?  if so, what are you using?
uptime? fog or rain issues? latency? packet loss?

Thanks

Scott

PS.  you can unicast me off line.  Thanks
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From svoll.voip at gmail.com  Thu Oct 25 15:29:44 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 25 Oct 2007 12:29:44 -0700
Subject: [cisco-voip] Can over subscription cause echo?
Message-ID: <f84a38d30710251229o3319b6e2udf7bd8eca1427a93@mail.gmail.com>

I have a remote site that is complaining of echo progressively getting worse
on calls.  All Calls, IP Phone to IP Phone or IP Phone to PSTN.  Gets so bad
they have to hang up.

I'm looking at the connection between the remote site and the central site
and it's being fulling used.

Bandwidth is hard to this location as it's two Telco's running one DS3 ATM
to the other Telco frame-relay T1 capped at ~800 - 900 k.  I'm doing the
best I can with QoS but is it a possibllity this is causing the echo?

TIA

Scott
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From wsisk at cisco.com  Thu Oct 25 15:39:01 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 15:39:01 -0400
Subject: [cisco-voip] DMS 250?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0073C3DE2@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0073C3DE2@USNAEXCH.na.didata.local>
Message-ID: <4720F0D5.6050306@cisco.com>

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From wsisk at cisco.com  Thu Oct 25 15:42:57 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 15:42:57 -0400
Subject: [cisco-voip] Can over subscription cause echo?
In-Reply-To: <f84a38d30710251229o3319b6e2udf7bd8eca1427a93@mail.gmail.com>
References: <f84a38d30710251229o3319b6e2udf7bd8eca1427a93@mail.gmail.com>
Message-ID: <4720F1C1.4030700@cisco.com>

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From erickbee at gmail.com  Thu Oct 25 15:48:31 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Thu, 25 Oct 2007 14:48:31 -0500
Subject: [cisco-voip] Call on analog port gets conferenced in with Unity if
	picked up on 3rd-4th ring
Message-ID: <f4445faf0710251248u526a65afq40a4bf540d3f145e@mail.gmail.com>

I have a strange issue, when a user at a remote site picks up a call
on the last ring before it rolls to Voicemail the caller gets
conferenced in with the Unity greeting. The outside caller is talking
to the person and hears Unity in the background.

The call comes in on a H323 gateway FXO port to the IP Phone at the
local site with the same gateway. All calls are set for G729 and QoS
is good on both sides, but it is a MPLS WAN between. They have
multiple sites and this is only site having this problem.

If they answer the call before the last ring, everything is fine.

Call Manager version 4.1(3)sr4d, IOS on gateway is 12.4.3g.

From wsisk at cisco.com  Thu Oct 25 15:53:46 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 25 Oct 2007 15:53:46 -0400
Subject: [cisco-voip] Call on analog port gets conferenced in with Unity
 if	picked up on 3rd-4th ring
In-Reply-To: <f4445faf0710251248u526a65afq40a4bf540d3f145e@mail.gmail.com>
References: <f4445faf0710251248u526a65afq40a4bf540d3f145e@mail.gmail.com>
Message-ID: <4720F44A.4050005@cisco.com>

Most likely either Unity, phone, or router allocated the same UDP port 
for this call as a previous call and then did not properly teardown 
previous call.  Packet capture would show who's still streaming packets 
that shouldn't be.  /wes

Erick Bergquist wrote:
> I have a strange issue, when a user at a remote site picks up a call
> on the last ring before it rolls to Voicemail the caller gets
> conferenced in with the Unity greeting. The outside caller is talking
> to the person and hears Unity in the background.
>
> The call comes in on a H323 gateway FXO port to the IP Phone at the
> local site with the same gateway. All calls are set for G729 and QoS
> is good on both sides, but it is a MPLS WAN between. They have
> multiple sites and this is only site having this problem.
>
> If they answer the call before the last ring, everything is fine.
>
> Call Manager version 4.1(3)sr4d, IOS on gateway is 12.4.3g.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From erickbee at gmail.com  Thu Oct 25 15:59:59 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Thu, 25 Oct 2007 14:59:59 -0500
Subject: [cisco-voip] Call on analog port gets conferenced in with Unity
	if picked up on 3rd-4th ring
In-Reply-To: <4720F44A.4050005@cisco.com>
References: <f4445faf0710251248u526a65afq40a4bf540d3f145e@mail.gmail.com>
	<4720F44A.4050005@cisco.com>
Message-ID: <f4445faf0710251259u2e61f326xaf1a07df8bd90c65@mail.gmail.com>

Ok. It is a new unity message as it is the persons VM greeting that
starts to play and the persons phone was idle for awhile before the
call. I'll see what the phones web page, stream page saids for the
remote IP address is. It should be the gateways IP address.


On 10/25/07, Wes Sisk <wsisk at cisco.com> wrote:
> Most likely either Unity, phone, or router allocated the same UDP port
> for this call as a previous call and then did not properly teardown
> previous call.  Packet capture would show who's still streaming packets
> that shouldn't be.  /wes
>
> Erick Bergquist wrote:
> > I have a strange issue, when a user at a remote site picks up a call
> > on the last ring before it rolls to Voicemail the caller gets
> > conferenced in with the Unity greeting. The outside caller is talking
> > to the person and hears Unity in the background.
> >
> > The call comes in on a H323 gateway FXO port to the IP Phone at the
> > local site with the same gateway. All calls are set for G729 and QoS
> > is good on both sides, but it is a MPLS WAN between. They have
> > multiple sites and this is only site having this problem.
> >
> > If they answer the call before the last ring, everything is fine.
> >
> > Call Manager version 4.1(3)sr4d, IOS on gateway is 12.4.3g.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From jeff at ocjtech.us  Thu Oct 25 16:04:46 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Thu, 25 Oct 2007 15:04:46 -0500
Subject: [cisco-voip] OT -- Wireless to remote sites?
In-Reply-To: <f84a38d30710251233y4374391eobe39ab100fee48e7@mail.gmail.com>
References: <f84a38d30710251233y4374391eobe39ab100fee48e7@mail.gmail.com>
Message-ID: <1193342686.9977.29.camel@lt21223.campus.dmacc.edu>

On Thu, 2007-10-25 at 12:33 -0700, Scott Voll wrote:
> is anyone using some kind of Wireless solution to a remote IP
> Telephony site?
>  
> I have a site that will be $1 million to build fiber to so I'm looking
> for options.  I'm thinking something with a licensed band or
> microwave.  anyone have experience with something like this?  if so,
> what are you using?  uptime? fog or rain issues? latency? packet loss?

We rent a small amount of space in a building across the road from my
main campus.  We use Cisco 1310 wireless bridges to link the main campus
to the remote building.  The span is around 2000ft.  We had some initial
issues with the link that were resolved by replacing a bad power supply
on one of the wireless bridges.  Other than that the connection has been
pretty solid.  It's just "temporary" though - the people in that office
will be moving into a new building on campus due to open in 18 months or
so.  So you may want to look into using the unlicensed 802.11 bands.
With the appropriate antennas and a good line of sight it should work
pretty well.  I'd probably look seriously at a MPLS VPN too.

Jeff

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From lists at jerkys.org  Thu Oct 25 16:06:58 2007
From: lists at jerkys.org (Jerky)
Date: Thu, 25 Oct 2007 15:06:58 -0500
Subject: [cisco-voip] DMS 250?
In-Reply-To: <4720F0D5.6050306@cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F0073C3DE2@USNAEXCH.na.didata.local>
	<4720F0D5.6050306@cisco.com>
Message-ID: <1CC8C850-F714-4A8E-B781-72A07FEB201A@jerkys.org>

I can vouch for the DMS100 setting working with a 250. That's what  
our carrier uses.


On Oct 25, 2007, at 2:39 PM, Wes Sisk wrote:

> In IOS it's iffy.  IOS doesn't claim to 'support' dms250, but  
> dms100 usually works.
>
> If you're doing MGCP bh to cm: either dms100 or ni, depends on the  
> version of the upstream switch.  if the d-ch will come up and calls  
> setup and clear properly with ni you will be best served there.  if  
> not, go back to dms100 and keep an eye on the 'dms' service  
> parameters in CM and on the gw config page.
>
> /Wes
>
> Jason Aarons (US) wrote:
>> What option should I set for a DMS-250?
>>
>>
>>
>>
>>
>> 1VG01(config)#isdn switch-type ?
>>
>>   primary-4ess    Lucent 4ESS switch type for the U.S.
>>
>>   primary-5ess    Lucent 5ESS switch type for the U.S.
>>
>>   primary-dms100  Northern Telecom DMS-100 switch type for the U.S.
>>
>>   primary-dpnss   DPNSS switch type for Europe
>>
>>   primary-net5    NET5 switch type for UK, Europe, Asia and Australia
>>
>>   primary-ni      National ISDN Switch type for the U.S.
>>
>>   primary-ntt     NTT switch type for Japan
>>
>>   primary-qsig    QSIG switch type
>>
>>   primary-ts014   TS014 switch type for Australia (obsolete)
>>
>>
>>
>> Disclaimer: This e-mail communication and any attachments may  
>> contain confidential and privileged information and is for use by  
>> the designated addressee(s) named above only. If you are not the  
>> intended addressee, you are hereby notified that you have received  
>> this communication in error and that any use or reproduction of  
>> this email or its contents is strictly prohibited and may be  
>> unlawful. If you have received this communication in error, please  
>> notify us immediately by replying to this message and deleting it  
>> from your computer. Thank you.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From l.durso at gmail.com  Thu Oct 25 16:17:31 2007
From: l.durso at gmail.com (=?utf-8?B?TGVvbmFyZG8gRCdVcnNv?=)
Date: Thu, 25 Oct 2007 20:17:31 +0000
Subject: [cisco-voip] R: Re:  Callmanager cluster in Wan
In-Reply-To: <4720AB76.7070903@cisco.com>
References: <929274954-1193320798-cardhu_decombobulator_blackberry.rim.net-1398273527-@bxe006.bisx.produk.on.blackberry><4720AB76.7070903@cisco.com>
Message-ID: <1083021520-1193343385-cardhu_decombobulator_blackberry.rim.net-1036260840-@bxe006.bisx.produk.on.blackberry>

Thanks Wes

I will wait your next email about this question. 

Ciao
Leo
---
Leonardo D'Urso
l.durso at gmail.com

-----Original Message-----
From: Wes Sisk <wsisk at cisco.com>

Date: Thu, 25 Oct 2007 10:43:02 
To:l.durso at gmail.com
Cc:ciscovoip <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Callmanager cluster in Wan


I've seen many deployments that use this model.  Noteworthy items:
1.register a phone and gw in the US to the EU servers and test the 
timing.  With high RTT cm<->gw and cm<->phone users may notice a delay 
in audio cut through when answering calls.  Faststart for h323 and 
optimized mgcp help some, but delays can still become an issue.  I will 
post a followup message with details when I locate them.
2. dialpan on EU servers will become significantly more complex hosting 
PSTN patterns for US and EU
3. Consider SRST for the US site, possibly using multiple SRST routers 
to allow all phones to register in the event of any network outgate 
between US and EU.
4. if you deploy 3rd gen ipphones (7941/7961/7911/7970/7971, etc.) be 
aware of upgrade issues due to TFTP over high RTT.  Strongly consider 
upgrading to CM version and phone versions that support peer to peer 
firmware sharing. (cm6.0 srnd page 3-15)

More details of this are covered in the SRND:
http://www.cisco.com/go/srnd
Your scenario is covered under "multi-site wan with centralized call 
processing' but depends on information in other sectsions as well.


/Wes

Leonardo D'Urso wrote:
> Hi there
>
> A customer of mine have two site: one in EU, one in USA with two different clusters of callmanager. Actually there is a trunk that permits to call between. 
> I have users that want to have the same extension mobility profile on each side. 
> I have evaluated that with a latency of 120ms it is not possible to have 1 centralized ccm cluster infrastructure with 1pub and 2sub in EU and 1 sub in USA. But what happens if i have all ccm server in a single location EU and a single voice gateway with SRST in USA and the  ip phones that directly point to ccm cluster in EU?
>
> Regards
> Leo 
> ---
> Leonardo D'Urso
> l.durso at gmail.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From tednugent73 at gmail.com  Thu Oct 25 16:38:25 2007
From: tednugent73 at gmail.com (Ted Nugent)
Date: Thu, 25 Oct 2007 16:38:25 -0400
Subject: [cisco-voip] DMS 250?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0073C3DE2@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0073C3DE2@USNAEXCH.na.didata.local>
Message-ID: <22e6380b0710251338y2cb744d8vd8ac55b8129e481e@mail.gmail.com>

In my experience its more important to ask the provider what protocol the
switch is runing rather then the actual switchtype. DMS250 to my knowledge
can run multiple protocols, NI2, 5ESS, NI1 (nortels NI1 = Cisco DMS100), NI1
for the rest of the world is only a BRI protocol. If they give you the
option always go for Ni2...


On 10/25/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  What option should I set for a DMS-250?
>
>
>
>
>
> 1VG01(config)#isdn switch-type ?
>
>   primary-4ess    Lucent 4ESS switch type for the U.S.
>
>   primary-5ess    Lucent 5ESS switch type for the U.S.
>
>   primary-dms100  Northern Telecom DMS-100 switch type for the U.S.
>
>   primary-dpnss   DPNSS switch type for Europe
>
>   primary-net5    NET5 switch type for UK, Europe, Asia and Australia
>
>   primary-ni      National ISDN Switch type for the U.S.
>
>   primary-ntt     NTT switch type for Japan
>
>   primary-qsig    QSIG switch type
>
>   primary-ts014   TS014 switch type for Australia (obsolete)
>
>
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Dotun.Adebogun at metavante.com  Thu Oct 25 18:09:04 2007
From: Dotun.Adebogun at metavante.com (Dotun.Adebogun at metavante.com)
Date: Thu, 25 Oct 2007 17:09:04 -0500
Subject: [cisco-voip] Remote connection to CUE
Message-ID: <OF65CACAB7.70DA16F9-ON8625737F.007876A4-8625737F.0079AE6D@metavante.com>

I have 2 sites running CCME. 
Site1 has the CUE for voicemail. Site2 does not have CUE.
Can I setup the mailboxes for site2 users on site1, and hairpin accross
the WAN for voicemail services?

Thanks,

Dotun Adebogun
Metavante Corporation
Network Implementation Engineering
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From jason.aarons at us.didata.com  Thu Oct 25 18:47:19 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 25 Oct 2007 18:47:19 -0400
Subject: [cisco-voip] Nortel Option 51 Paging Trunk
Message-ID: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>

Customer has a Trunk port set for paging on a Nortel connected to a
no-name paging unit.

 

Is a "Nortel Paging Port" just a FXO?

 

I've connected up my VWIC-2FXO for Loop Start, I go offhook but never
get connected to paging unit.

 

Debug vpm signal

*Aug 17 05:02:16.115: htsp_timer_stop

*Aug 17 05:02:16.115: htsp_timer_stop2

*Aug 17 05:02:16.115: [2/1] set signal state = 0x4 timestamp = 0

*Aug 17 05:02:16.127: htsp_process_event: [2/1, FXOLS_ONHOOK,
E_HTSP_RELEASE_REQ]fxols_onhook_release

*Aug 17 05:02:16.219: htsp_process_event: [2/1, FXOLS_ONHOOK,
E_DSP_SIG_0100]

*Aug 17 05:02:17.139: htsp_timer_stop3 htsp_setup_req

*Aug 17 05:02:17.151: htsp_process_event: [2/1, FXOLS_ONHOOK,
E_HTSP_SETUP_REQ]fxols_onhook_setup

*Aug 17 05:02:17.151: [2/1] set signal state = 0xC timestamp = 0

*Aug 17 05:02:17.151: htsp_timer - 1300 msec

*Aug 17 05:02:17.303: htsp_process_event: [2/1, FXOLS_WAIT_DIAL_TONE,
E_DSP_SIG_1100]fxols_power_denial_proc

*Aug 17 05:02:17.307: htsp_timer2 - 1000 msec

*Aug 17 05:02:17.307: htsp_timer_stop

*Aug 17 05:02:18.307: htsp_process_event: [2/1, FXOLS_WAIT_DIAL_TONE,
E_HTSP_EVENT_TIMER2]fxols_timer2_proc

*Aug 17 05:02:18.307: htsp_timer_stop

*Aug 17 05:02:18.307: htsp_timer_stop2

*Aug 17 05:02:18.307: [2/1] set signal state = 0x4 timestamp = 0

*Aug 17 05:02:18.319: htsp_process_event: [2/1, FXOLS_ONHOOK,
E_HTSP_RELEASE_REQ]fxols_onhook_release

*Aug 17 05:02:18.411: htsp_process_event: [2/1, FXOLS_ONHOOK,
E_DSP_SIG_0100]




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From Roconnell at unislumin.com  Thu Oct 25 22:34:39 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Thu, 25 Oct 2007 22:34:39 -0400
Subject: [cisco-voip] CUBE or IP IP GW with MTP
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
Message-ID: <18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>

Hello all,

 

Quick question, on IOS routers you can use DSP resources as MTP. I
understand and implemented this many times in the past using SCCP
commands that point back to CallManager or CallManager Express. I'm just
wondering in the case of CUBE or IP-IP GW used as a SIP trunk to the
Service provider, and we are not using CallManager or CallManager
Express can I still use the DSP's for MTP's, Transcoding? In my case I'm
specifically looking for MTP's cause I need it for FAST Start H.323
calls.

 

According to the attached Document there is such thing as a "HARDWARE
based MTP" but the bullet stats that it uses SCCP

1.    MTPs ( and Transcoders ) can be controlled by Communications
Manager or the Cisco Unified Border Element (via SCCP)

So do they mean that I have to turn on "TELEPHONY-SERVICES" just to make
use of these MTP or am I missing something

 

Ryan

 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
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strictly prohibited and may be unlawful. If you have received this
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Attention:

Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.
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From mthompson729 at gmail.com  Thu Oct 25 23:53:31 2007
From: mthompson729 at gmail.com (Michael Thompson)
Date: Thu, 25 Oct 2007 23:53:31 -0400
Subject: [cisco-voip] Secondary DNs only have 7 ring choices?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>
Message-ID: <363f21ee0710252053n6da51c16gfcd38d87e08f26c1@mail.gmail.com>

Jason,
   an easy way to do it is to backup the DistinctiveRingList.xml (just
rename it) and then copy the RingList.xml to the DistinctiveRingList.xml.
you'll then have the same info in both the Default Ringer as well as the Per
Line Ringer.


On 10/24/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  Customer asked why their line 2, line 3, etc only have 7 ring choices in
> Settings > Ring Type > X when their default ring has 25 choices?
>
>
>
> I said "because it is what it is" (works on my 3 year old all the
> time)?..what's the real reason?
>
>
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
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From asobihoudai at yahoo.com  Fri Oct 26 00:20:06 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 25 Oct 2007 21:20:06 -0700 (PDT)
Subject: [cisco-voip] Remote connection to CUE
In-Reply-To: <OF65CACAB7.70DA16F9-ON8625737F.007876A4-8625737F.0079AE6D@metavante.com>
Message-ID: <854867.55681.qm@web57214.mail.re3.yahoo.com>

I have asked around searching for the answer to this
question and the consensus seems to be "NO."

--- Dotun.Adebogun at metavante.com wrote:

> I have 2 sites running CCME. 
> Site1 has the CUE for voicemail. Site2 does not have
> CUE.
> Can I setup the mailboxes for site2 users on site1,
> and hairpin accross
> the WAN for voicemail services?
> 
> Thanks,
> 
> Dotun Adebogun
> Metavante Corporation
> Network Implementation Engineering
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From jason.aarons at us.didata.com  Fri Oct 26 00:15:17 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 26 Oct 2007 00:15:17 -0400
Subject: [cisco-voip] Secondary DNs only have 7 ring choices?
In-Reply-To: <363f21ee0710252053n6da51c16gfcd38d87e08f26c1@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F00738440B@USNAEXCH.na.didata.local>
	<363f21ee0710252053n6da51c16gfcd38d87e08f26c1@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0073C4339@USNAEXCH.na.didata.local>

In Windows and CallManger 4.x this is pretty easy and I've done it
before, how do I do this in 5.1?

 

From: Michael Thompson [mailto:mthompson729 at gmail.com] 
Sent: Thursday, October 25, 2007 11:54 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Secondary DNs only have 7 ring choices?

 

Jason,

   an easy way to do it is to backup the DistinctiveRingList.xml (just
rename it) and then copy the RingList.xml to the
DistinctiveRingList.xml.  you'll then have the same info in both the
Default Ringer as well as the Per Line Ringer. 

 

On 10/24/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote: 

Customer asked why their line 2, line 3, etc only have 7 ring choices in
Settings > Ring Type > X when their default ring has 25 choices?

 

I said "because it is what it is" (works on my 3 year old all the
time).....what's the real reason?

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

 




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From tele at plexialab.it  Fri Oct 26 04:18:36 2007
From: tele at plexialab.it (tele)
Date: Fri, 26 Oct 2007 10:18:36 +0200
Subject: [cisco-voip] CUBE or IP IP GW with MTP
In-Reply-To: <18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
	<18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>
Message-ID: <1193386716.7226.125.camel@mda>

Hi,

Ryan yes you need turn on "telephony-services".
You can also use an external DSP farm for the MTP or transcoding.

http://www.cisco.com/en/US/products/ps6706/products_feature_guide09186a008076161a.html



:tele

On Thu, 2007-10-25 at 22:34 -0400, Ryan O'Connell wrote:
> Hello all,
> 
>  
> 
> Quick question, on IOS routers you can use DSP resources as MTP. I
> understand and implemented this many times in the past using SCCP
> commands that point back to CallManager or CallManager Express. I'm
> just wondering in the case of CUBE or IP-IP GW used as a SIP trunk to
> the Service provider, and we are not using CallManager or CallManager
> Express can I still use the DSP's for MTP's, Transcoding? In my case
> I'm specifically looking for MTP's cause I need it for FAST Start
> H.323 calls.
> 
>  
> 
> According to the attached Document there is such thing as a "HARDWARE
> based MTP" but the bullet stats that it uses SCCP
> 
> 1.    MTPs ( and Transcoders ) can be controlled by Communications
> Manager or the Cisco Unified Border Element (via SCCP)
> 
> So do they mean that I have to turn on "TELEPHONY-SERVICES" just to
> make use of these MTP or am I missing something
> 
>  
> 
> Ryan
> 
>  
> 
>  
> 
> 
> 
> ______________________________________________________________________
> Privileged/Confidential Information may be contained in this message.
> Disclosure to any person other than the named recipient is
> unauthorized.  If you are not the intended recipient, please delete
> all copies of this information and kindly notify the sender by reply
> email.  Opinions, conclusions and other information in this message
> that do not relate to the official business of UNIS LUMIN Inc. shall
> be understood as neither given nor endorsed by it.  UNIS LUMIN Inc.
> and any of its subsidiaries reserve the right to monitor all e-mail
> communications through its networks.  Thank you.
> 
> 
> ______________________________________________________________________
> 
> Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the intended
> addressee, you are hereby notified that you have received this
> communication in error and that any use or reproduction of this email
> or its contents is strictly prohibited and may be unlawful. If you
> have received this communication in error, please notify us
> immediately by replying to this message and deleting it from your
> computer. Thank you.
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From khalid_khursheed at hotmail.com  Fri Oct 26 04:30:40 2007
From: khalid_khursheed at hotmail.com (Syed Khalid Ali)
Date: Fri, 26 Oct 2007 13:30:40 +0500
Subject: [cisco-voip] Difference b/w Call Manager 4, 5 and 6
Message-ID: <BAY113-W25DF280BF6E66AADDDAD88E8960@phx.gbl>



hi,

i'm fairly new to voip world and currently preparing for cvoice exam. Can any one tell me the difference between CallManager 4,5 and 6.

eagrly waiting for the response

thanks

syed khalid ali

_________________________________________________________________
Discover the new Windows Vista
http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE
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From khalid_khursheed at hotmail.com  Fri Oct 26 04:56:24 2007
From: khalid_khursheed at hotmail.com (Syed Khalid Ali)
Date: Fri, 26 Oct 2007 13:56:24 +0500
Subject: [cisco-voip] FXO Port probelm with CCM 4.1 & 3845 gateway
Message-ID: <BAY113-W44C4688238F3D21006ADDE8960@phx.gbl>


hi

i have configure FXO port(s) with a 3845 gateway. setup ip phones with dual lines. line 2 serves as plar opx connection for FXO port(s). now when i dial to fxo port number, although the call is disconnected from the caller the ip phone is still ringing untill someone off-hook the phone.

also tried the supervisory dissconnet command and other stuff but all in vain. need advice

also how do i configure the ip phone so that whenever user presses the line-2 softkey it get the FXO port(s) line (not from PRI)

using call manager 4.1(3), ip phone 7940, 7941 and 7960.

thanks in advance

syed khalid ali
_________________________________________________________________
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx

From David.Sullivan at barnet.ac.uk  Fri Oct 26 04:44:44 2007
From: David.Sullivan at barnet.ac.uk (David Sullivan)
Date: Fri, 26 Oct 2007 09:44:44 +0100
Subject: [cisco-voip] OT -- Wireless to remote sites?
In-Reply-To: <1193342686.9977.29.camel@lt21223.campus.dmacc.edu>
Message-ID: <EDB47A2F10617E4095AE70CBBC723F611E2F78@oak.bc-ad.barnet.ac.uk>

cisco-voip-bounces at puck.nether.net wrote:
> On Thu, 2007-10-25 at 12:33 -0700, Scott Voll wrote:
>> is anyone using some kind of Wireless solution to a remote IP
>> Telephony site? 
>> 
>> I have a site that will be $1 million to build fiber to so I'm
>> looking for options.  I'm thinking something with a licensed band or
>> microwave.  anyone have experience with something like this?  if so,
>> what are you using?  uptime? fog or rain issues? latency? packet
>> loss? 
> 
We used to use a telco-grade licensed microwave system from Ceragon
providing a 100M connection over 3 1/2 miles before that site was
closed. It was expensive at the time but was a saving over our 2M
connection from the incumbent telco after 3 years (in addition to being
a lot faster). It was pretty much bombproof and only had one (hardware)
failure over the 5 years we used it through rain and shine.

Unlicensed 802.11b/g radios will be a lot cheaper and are an option
depending on the distance, we've used this for temporary installs over
short distances and they excel at this (ETSI EIRP power limits mean for
longer links you're generally pushing your luck in the UK), local
interference may or may not be an issue. Another good choice may be the
5GHz line of sight systems but I can't say I've used any of these.

We also used/use infra-red, these systems are unlicensed but free from
interference issues but can have issues with the weather, we tried them
over a long link (3 miles) but that was a complete waste of time but we
have a system that links a building 150m away across a road and car park
and it's completely solid and we've never had any problems over that
distance.

Your best option is to speak to a company who installs different types
of systems or several companies who each supply different systems and
ask them what they would recommend for your situation, a lot of these
systems depend on your local regulatory/licensing regime, weather,
hardware support of the system, how reliable do you want it to be, how
much are you willing to spend etc ...

David Sullivan
IT Services Systems Team
Barnet College
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From khalid_khursheed at hotmail.com  Fri Oct 26 06:40:19 2007
From: khalid_khursheed at hotmail.com (Syed Khalid Ali)
Date: Fri, 26 Oct 2007 15:40:19 +0500
Subject: [cisco-voip] OT -- Wireless to remote sites?
In-Reply-To: <f84a38d30710251233y4374391eobe39ab100fee48e7@mail.gmail.com>
References: <f84a38d30710251233y4374391eobe39ab100fee48e7@mail.gmail.com>
Message-ID: <BAY113-W30DE81E5F8DC7357C33293E8960@phx.gbl>


hi,

Yes, one of our clients is using wireless bridges (Nexus SmartBridges) to connect its branch offices. Connectivity is fine. They have 2-4 ip phones on branches that register with call manger 4.1 at HO site.
One of the site is also  on VSAT and voice is still fine.

Regards, 
Syed Khalid Ali


________________________________
Date: Thu, 25 Oct 2007 12:33:44 -0700
From: svoll.voip at gmail.com
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] OT -- Wireless to remote sites?

is anyone using some kind of Wireless solution to a remote IP Telephony site?

I have a site that will be $1 million to build fiber to so I'm looking for options.  I'm thinking something with a licensed band or microwave.  anyone have experience with something like this?  if so, what are you using?  uptime? fog or rain issues? latency? packet loss?

Thanks

Scott

PS.  you can unicast me off line.  Thanks



_________________________________________________________________
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx

From Richard.vanBerkel at telindus.nl  Fri Oct 26 06:57:24 2007
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Fri, 26 Oct 2007 12:57:24 +0200
Subject: [cisco-voip] CSA
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157ED03@tnl1mx1.telindus.intra>

Hi,

 

We have Unity 4.3 installed and have the following.

We have a backup off all databases running. However when CSA is running
the backup fails.

 

I know there was a Bug id (CSCsa95341
<http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?metho
d=fetchBugDetails&bugId=CSCsa95341>  ) for this with the previous
versions of Unity, but does anyone know if there is a possibility to add
a rule

of exception to the CSA?

 

Richard van Berkel

Senior Customer Service

Profile: MCSE-2003, Microsoft Exchange 2003, Cisco Certified Voip
Specialist

--------------------------------------------------------------------

Telindus B.V.

Savannahweg 19     

3542 AW Utrecht

Netherlands

--------------------------------------------------------------------

mailto: Richard.vanBerkel at telindus.nl
<mailto:mailto:%20Richard.vanBerkel at telindus.nl> 

tel:         +31 302 477 515 

fax:       +31 302 477 711 

mobile: +31 628 903 277

 

Telindus. Change things your way.

http://www.telindus.com/ <http://www.telindus.com/> 

	
	 	 
 


  

 

 

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From ErasmuE4 at telkom.co.za  Fri Oct 26 08:55:18 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Fri, 26 Oct 2007 14:55:18 +0200
Subject: [cisco-voip] automate stop and start of services
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01C09CBA@CNTRRA20-XCS01.telkom.co.za>


Hi 

 

Does anyone know of a way to automate the restarting of call manager 4
services? We have a problem with the de queuing of calls to our
attendant consoles in ccm 4-2-3 sr2b (cisco bug) and are waiting for a
software fix. In the meantime when the problem occurs we reset TCD and
cti services manually. We noticed that after resetting these services
that things acrry on working for a couple of days so we want to if
possible automate this process and maybe use the Microsoft Scheduler
to do this.

 

 

Any advice will be appreciated

 

 

 

Erik Erasmus



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From wsisk at cisco.com  Fri Oct 26 09:54:24 2007
From: wsisk at cisco.com (Wesley Sisk)
Date: Fri, 26 Oct 2007 09:54:24 -0400 (EDT)
Subject: [cisco-voip] CUBE or IP IP GW with MTP
In-Reply-To: <18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
	<18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>
Message-ID: <Pine.GSO.4.64.0710260952510.1665@clubhouse-8.cisco.com>

it depends on how you configure ip-ip gw.  it has flow-through and 
flow-around options that determine wheter the RTP stream flows through the 
DSP's on the router or if the router only rewrites the ip headers of the 
provided rtp packet.

/wes

From wsisk at cisco.com  Fri Oct 26 09:59:20 2007
From: wsisk at cisco.com (Wesley Sisk)
Date: Fri, 26 Oct 2007 09:59:20 -0400 (EDT)
Subject: [cisco-voip] automate stop and start of services
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F01C09CBA@CNTRRA20-XCS01.telkom.co.za>
References: <472E195E692EE74BAE4E7B2880A78C1F01C09CBA@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <Pine.GSO.4.64.0710260958100.1665@clubhouse-8.cisco.com>

write a .bat file:
*********
net stop "cisco callmanager"
net start "cisco callmanager"
*********
enable scheduler service and schedule this .bat to job when you desire. 
substitute the service name you need for 'cisco callmanager'.

/wes

On Fri, 26 Oct 2007, Erik Erasmus (E) wrote:

>
> Hi
>
>
>
> Does anyone know of a way to automate the restarting of call manager 4
> services? We have a problem with the de queuing of calls to our
> attendant consoles in ccm 4-2-3 sr2b (cisco bug) and are waiting for a
> software fix. In the meantime when the problem occurs we reset TCD and
> cti services manually. We noticed that after resetting these services
> that things acrry on working for a couple of days so we want to if
> possible automate this process and maybe use the Microsoft Scheduler
> to do this.
>
>
>
>
>
> Any advice will be appreciated
>
>
>
>
>
>
>
> Erik Erasmus
>
>
>
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> This e-mail is subject to the Telkom SA electronic communication legal notice, available at :
> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

From wsisk at cisco.com  Fri Oct 26 10:05:03 2007
From: wsisk at cisco.com (Wesley Sisk)
Date: Fri, 26 Oct 2007 10:05:03 -0400 (EDT)
Subject: [cisco-voip] CSA
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157ED03@tnl1mx1.telindus.intra>
References: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157ED03@tnl1mx1.telindus.intra>
Message-ID: <Pine.GSO.4.64.0710260959570.1665@clubhouse-8.cisco.com>

Hi Richard,

'Headless' CSA is just that - headless.  Rules cannot be modified.

defect sa95341 has headline:
SQL backup job agent owned by install account by default

This may change the job owner, but not clear that it's going to address 
CSA issue.  Most SQL backup tools can connect to SQL instances over the 
network and perform backup.  That may be a better option unless your 
backup software is on the 'supported' list for CSA. In which case you need 
to open a TAC case and get a new defect opened against CSA to correct 
rules for your backup software.. again, if your backup software is one of 
the supported ones with csa.

Another option is to purchase fully managed csa version. then you can 
import the headless rules and customize as you please.

/Wes

On Fri, 26 Oct 2007, Richard van Berkel wrote:

> Hi,
>
>
>
> We have Unity 4.3 installed and have the following.
>
> We have a backup off all databases running. However when CSA is running
> the backup fails.
>
>
>
> I know there was a Bug id (CSCsa95341
> <http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?metho
> d=fetchBugDetails&bugId=CSCsa95341>  ) for this with the previous
> versions of Unity, but does anyone know if there is a possibility to add
> a rule
>
> of exception to the CSA?
>
>
>
> Richard van Berkel
>
> Senior Customer Service
>
> Profile: MCSE-2003, Microsoft Exchange 2003, Cisco Certified Voip
> Specialist
>
> --------------------------------------------------------------------
>
> Telindus B.V.
>
> Savannahweg 19
>
> 3542 AW Utrecht
>
> Netherlands
>
> --------------------------------------------------------------------
>
> mailto: Richard.vanBerkel at telindus.nl
> <mailto:mailto:%20Richard.vanBerkel at telindus.nl>
>
> tel:         +31 302 477 515
>
> fax:       +31 302 477 711
>
> mobile: +31 628 903 277
>
>
>
> Telindus. Change things your way.
>
> http://www.telindus.com/ <http://www.telindus.com/>
>
>
>
>
>
>
>
>
>
>
>
>
>

From Nick.Kassel at ipitomi.com  Fri Oct 26 10:09:29 2007
From: Nick.Kassel at ipitomi.com (Nick Kassel)
Date: Fri, 26 Oct 2007 15:09:29 +0100
Subject: [cisco-voip] Arc Consoles losing connection
In-Reply-To: <4720BE57.1040303@cisco.com>
References: <475fda330710250839k66e43713r850e92d0243a8b11@mail.gmail.com>
	<4720BE57.1040303@cisco.com>
Message-ID: <970310A6A9EFBC4189C5B1C43EFDDB8BC7DECE@pitmx1.principal-it.com>

Wes

 

One of the phones attached to an Arc console has just gone down again,
when I checked the network statistics screen on the phone the elapsed
time was showing over 10 hours which was the time we would have rebooted
it yesterday. However the Debug Display still shows the Last=CM
Closed-TCP message which I now understand means it has tried to fail
over the back up call manager. 

 

Is there anywhere else I could check for information on why this phone
may be losing connection with the primary Call manager.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: 25 October 2007 17:04
To: Nick
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Arc Consoles losing connection

 

if the phone deregisters i believe ARC has to go down as CM will send
CTIDeviceClose to ARC.  
On the phone under settings->status->network statistics what is the
'Elapsed Time'?  This indicates the duration of time the phone has been
properly registered to its current CM. If the phone is unregistering
then ARC will most likely be affected.

/wes

Nick wrote: 

I have an issue with arc consoles losing connection this is happening to
a number of consoles in the same site when I check the phones the debug
display on the phone it says 

 

15:57:33 14: Name=SEP00146905158B Load= 7.0(3.0S) Last=CM-closed-TCP

each time it loses connection, we have tried re patching to different
switchports and different switches and still seem to be getting the same
issue, could this be an isue with the phone load as it is very old
firmware, all other consoles on the same arc server and other phones in
the same site don't appear to be getting the issue althouh it may not be
as noticeable as on an Arc console. 

 

Has anyone had this issue with Arc before or can make any suggestions on
what else to check

 



________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


______________________________________________________________________
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From asobihoudai at yahoo.com  Fri Oct 26 10:12:26 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 26 Oct 2007 07:12:26 -0700 (PDT)
Subject: [cisco-voip] Difference b/w Call Manager 4, 5 and 6
In-Reply-To: <BAY113-W25DF280BF6E66AADDDAD88E8960@phx.gbl>
Message-ID: <955696.38534.qm@web57202.mail.re3.yahoo.com>

http://www.cisco.com/en/US/products/sw/voicesw/ps556/

CCM4 = windows-based
CCM5/6 = linux OS-based

--- Syed Khalid Ali <khalid_khursheed at hotmail.com>
wrote:

> 
> 
> hi,
> 
> i'm fairly new to voip world and currently preparing
> for cvoice exam. Can any one tell me the difference
> between CallManager 4,5 and 6.
> 
> eagrly waiting for the response
> 
> thanks
> 
> syed khalid ali
> 
>
_________________________________________________________________
> Discover the new Windows Vista
>
http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE>
_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From Hesham.Moussa at vanco.co.uk  Fri Oct 26 10:19:06 2007
From: Hesham.Moussa at vanco.co.uk (Hesham Moussa)
Date: Fri, 26 Oct 2007 15:19:06 +0100
Subject: [cisco-voip] 7911 and overlays on a CME system
In-Reply-To: <876C8F9957792243B670CAEBDA19C6D101FEF6DF@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <876C8F9957792243B670CAEBDA19C6D1020FCDDD@LON-INT-MSGVS01.internal.vancocorp.com>

In case anyone else was wondering, initial testing shows it does and the
broadcast ring to all phones on the CME platform also seems to work too.

 

Will fully lab test next week.

 

H

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Hesham Moussa
Sent: 24 October 2007 14:54
To: cisco voip list
Subject: [cisco-voip] 7911 and overlays on a CME system

 

Hi all,

 

Does the 7911 support overlay buttons?

 

I am trying to setup a broadcast group using a very crude method of
multiple DNs with the same number and assigning these multiple DNs to
the 7911 but am worried that it doesn't support overlays at all.

 

Any pointers or confirmation as to whether yay or nay appreciated.

 

Cheers,

 

H



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From patrick.diener at gmail.com  Fri Oct 26 10:23:50 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Fri, 26 Oct 2007 16:23:50 +0200
Subject: [cisco-voip] Difference b/w Call Manager 4, 5 and 6
In-Reply-To: <BAY113-W25DF280BF6E66AADDDAD88E8960@phx.gbl>
References: <BAY113-W25DF280BF6E66AADDDAD88E8960@phx.gbl>
Message-ID: <c0deb5b80710260723x51b6247an15662f4da0292078@mail.gmail.com>

btw cvoice does not test anything callmanager related...

regards
patrick

On 10/26/07, Syed Khalid Ali <khalid_khursheed at hotmail.com> wrote:
>
>
> hi,
>
> i'm fairly new to voip world and currently preparing for cvoice exam. Can
> any one tell me the difference between CallManager 4,5 and 6.
>
> eagrly waiting for the response
>
> thanks
>
> syed khalid ali
>
> ________________________________
> Discover the new Windows Vista Learn more!
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From rkulagow at gmail.com  Fri Oct 26 10:24:25 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Fri, 26 Oct 2007 09:24:25 -0500
Subject: [cisco-voip] Intercom History
In-Reply-To: <185C6467-DC1D-469D-81E7-613466A7F453@cisco.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>	<470B8E1E.9874.0039.0@mtb.com>
	<470BD357.9040404@cisco.com>	<470BD1B9.9874.0039.0@mtb.com>
	<185C6467-DC1D-469D-81E7-613466A7F453@cisco.com>
Message-ID: <4721F899.4010603@gmail.com>

Ryan Ratliff wrote:
> No the intercom is only in 6.x however the phone load included with  
> SR5d added support for that feature.  The phone however isn't smart  
> enough to know that the CM it's registering with supports the feature  
> or not so it always shows the intercom history.

Ugh.  Cisco has managed to stuff a peer-to-peer firmware distribution 
system into the phone, but not a "capabilities" query during 
registration to CallManager that returns a boolean "display Intercom 
History bubble"?

This almost seems like a mis-feature to me.  I'm sure that someone, 
somewhere really, really loves this Intercom History thing, and I'll 
tell my account team that I don't. But does an organization with 1500 
7970 telephones really have any voice in these sorts of things?

From rkulagow at gmail.com  Fri Oct 26 10:28:38 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Fri, 26 Oct 2007 09:28:38 -0500
Subject: [cisco-voip] OT:  "Wesley"?
In-Reply-To: <Pine.GSO.4.64.0710260952510.1665@clubhouse-8.cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>	<18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>
	<Pine.GSO.4.64.0710260952510.1665@clubhouse-8.cisco.com>
Message-ID: <4721F996.7010905@gmail.com>

Wesley Sisk wrote:
...snip...

You need to tell the corporate people that no one is going to recognize 
"Wesley" and that it needs to go back to "Wes".

:)

From MillerS at DicksteinShapiro.COM  Fri Oct 26 10:34:09 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Fri, 26 Oct 2007 10:34:09 -0400
Subject: [cisco-voip] Intercom History
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
	<470B8E1E.9874.0039.0@mtb.com> <470BD357.9040404@cisco.com>
	<470BD1B9.9874.0039.0@mtb.com>
	<185C6467-DC1D-469D-81E7-613466A7F453@cisco.com>
	<4721F899.4010603@gmail.com>
Message-ID: <421CAA1A5300B84A87D200D70998578263DFAF@DCEX2.DSMO.COM>

 I missed something.  Is there a new (true) intercom feature available
in 6X?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Kulagowski
Sent: Friday, October 26, 2007 10:24 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Intercom History

Ryan Ratliff wrote:
> No the intercom is only in 6.x however the phone load included with  
> SR5d added support for that feature.  The phone however isn't smart  
> enough to know that the CM it's registering with supports the feature

> or not so it always shows the intercom history.

Ugh.  Cisco has managed to stuff a peer-to-peer firmware distribution 
system into the phone, but not a "capabilities" query during 
registration to CallManager that returns a boolean "display Intercom 
History bubble"?

This almost seems like a mis-feature to me.  I'm sure that someone, 
somewhere really, really loves this Intercom History thing, and I'll 
tell my account team that I don't. But does an organization with 1500 
7970 telephones really have any voice in these sorts of things?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


--------------------------------------------------------
This e-mail message and any attached files are confidential
and are intended solely for the use of the addressee(s) 
named above. This communication may contain material 
protected by attorney-client, work product, or other 
privileges. If you are not the intended recipient or person 
responsible for delivering this confidential communication 
to the intended recipient, you have received this 
communication in error, and any review, use, dissemination, 
forwarding, printing, copying, or other distribution of 
this e-mail message and any attached files is strictly 
prohibited. Dickstein Shapiro reserves the right to monitor 
any communication that is created, received, or sent on its 
network.  If you have received this confidential 
communication in error, please notify the sender 
immediately by reply e-mail message and permanently delete 
the original message. 

To reply to our email administrator directly, send an email 
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==============================================================================


From voicenoob at gmail.com  Fri Oct 26 10:36:48 2007
From: voicenoob at gmail.com (Voice Noob)
Date: Fri, 26 Oct 2007 09:36:48 -0500
Subject: [cisco-voip] Difference b/w Call Manager 4, 5 and 6
In-Reply-To: <c0deb5b80710260723x51b6247an15662f4da0292078@mail.gmail.com>
References: <BAY113-W25DF280BF6E66AADDDAD88E8960@phx.gbl>
	<c0deb5b80710260723x51b6247an15662f4da0292078@mail.gmail.com>
Message-ID: <535625f70710260736u5f81ae43kc23d311e1d859d70@mail.gmail.com>

You also shouldn't base your desision on what version to deploy strictly on
the operating system.



On 10/26/07, Patrick Diener <patrick.diener at gmail.com> wrote:
>
> btw cvoice does not test anything callmanager related...
>
> regards
> patrick
>
> On 10/26/07, Syed Khalid Ali <khalid_khursheed at hotmail.com> wrote:
> >
> >
> > hi,
> >
> > i'm fairly new to voip world and currently preparing for cvoice exam.
> Can
> > any one tell me the difference between CallManager 4,5 and 6.
> >
> > eagrly waiting for the response
> >
> > thanks
> >
> > syed khalid ali
> >
> > ________________________________
> > Discover the new Windows Vista Learn more!
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From voicenoob at gmail.com  Fri Oct 26 10:42:36 2007
From: voicenoob at gmail.com (Voice Noob)
Date: Fri, 26 Oct 2007 09:42:36 -0500
Subject: [cisco-voip] Intercom History
In-Reply-To: <421CAA1A5300B84A87D200D70998578263DFAF@DCEX2.DSMO.COM>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
	<470B8E1E.9874.0039.0@mtb.com> <470BD357.9040404@cisco.com>
	<470BD1B9.9874.0039.0@mtb.com>
	<185C6467-DC1D-469D-81E7-613466A7F453@cisco.com>
	<4721F899.4010603@gmail.com>
	<421CAA1A5300B84A87D200D70998578263DFAF@DCEX2.DSMO.COM>
Message-ID: <535625f70710260742t29e9d6bj8a6020b57b5aceb7@mail.gmail.com>

  Intercom yes paging no.

On 10/26/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
> I missed something.  Is there a new (true) intercom feature available
> in 6X?
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> MillerS at dicksteinshapiro.com
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
> Kulagowski
> Sent: Friday, October 26, 2007 10:24 AM
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Intercom History
>
> Ryan Ratliff wrote:
> > No the intercom is only in 6.x however the phone load included with
> > SR5d added support for that feature.  The phone however isn't smart
> > enough to know that the CM it's registering with supports the feature
>
> > or not so it always shows the intercom history.
>
> Ugh.  Cisco has managed to stuff a peer-to-peer firmware distribution
> system into the phone, but not a "capabilities" query during
> registration to CallManager that returns a boolean "display Intercom
> History bubble"?
>
> This almost seems like a mis-feature to me.  I'm sure that someone,
> somewhere really, really loves this Intercom History thing, and I'll
> tell my account team that I don't. But does an organization with 1500
> 7970 telephones really have any voice in these sorts of things?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential
> and are intended solely for the use of the addressee(s)
> named above. This communication may contain material
> protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
> responsible for delivering this confidential communication
> to the intended recipient, you have received this
> communication in error, and any review, use, dissemination,
> forwarding, printing, copying, or other distribution of
> this e-mail message and any attached files is strictly
> prohibited. Dickstein Shapiro reserves the right to monitor
> any communication that is created, received, or sent on its
> network.  If you have received this confidential
> communication in error, please notify the sender
> immediately by reply e-mail message and permanently delete
> the original message.
>
> To reply to our email administrator directly, send an email
> to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
>
>
> ==============================================================================
>
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>
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From wsisk at cisco.com  Fri Oct 26 11:56:24 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 26 Oct 2007 11:56:24 -0400 (EDT)
Subject: [cisco-voip] OT:  "Wesley"?
In-Reply-To: <4721F996.7010905@gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
	<18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>
	<Pine.GSO.4.64.0710260952510.1665@clubhouse-8.cisco.com>
	<4721F996.7010905@gmail.com>
Message-ID: <Pine.GSO.4.64.0710261155420.5522@clubhouse-6.cisco.com>

Should be fixed now.  Sorry for the confusion.
/^Wes$

On Fri, 26 Oct 2007, Robert Kulagowski wrote:

> Wesley Sisk wrote:
> ...snip...
>
> You need to tell the corporate people that no one is going to recognize 
> "Wesley" and that it needs to go back to "Wes".
>
> :)
>

From craig at staffin.org  Fri Oct 26 12:03:00 2007
From: craig at staffin.org (Craig Staffin)
Date: Fri, 26 Oct 2007 11:03:00 -0500
Subject: [cisco-voip] OT: "Wesley"?
In-Reply-To: <Pine.GSO.4.64.0710261155420.5522@clubhouse-6.cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
	<18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>
	<Pine.GSO.4.64.0710260952510.1665@clubhouse-8.cisco.com>
	<4721F996.7010905@gmail.com>
	<Pine.GSO.4.64.0710261155420.5522@clubhouse-6.cisco.com>
Message-ID: <9b76d88c0710260903r5522257cpfe02d5542b4985e2@mail.gmail.com>

Man I wish I had that kind of pull around Cisco

You change a form of documentation in less than 6 hours :)

Craig

On 10/26/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> Should be fixed now.  Sorry for the confusion.
> /^Wes$
>
> On Fri, 26 Oct 2007, Robert Kulagowski wrote:
>
> > Wesley Sisk wrote:
> > ...snip...
> >
> > You need to tell the corporate people that no one is going to recognize
> > "Wesley" and that it needs to go back to "Wes".
> >
> > :)
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003
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From wsisk at cisco.com  Fri Oct 26 12:51:18 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 26 Oct 2007 12:51:18 -0400 (EDT)
Subject: [cisco-voip] Intercom History
In-Reply-To: <421CAA1A5300B84A87D200D70998578263DFAF@DCEX2.DSMO.COM>
References: <253FAE7A06B11B49A3A29770B2B8BAFC03B482F1@CSEXCH01.charles-stanley.co.uk>
	<470B8E1E.9874.0039.0@mtb.com> <470BD357.9040404@cisco.com>
	<470BD1B9.9874.0039.0@mtb.com>
	<185C6467-DC1D-469D-81E7-613466A7F453@cisco.com>
	<4721F899.4010603@gmail.com>
	<421CAA1A5300B84A87D200D70998578263DFAF@DCEX2.DSMO.COM>
Message-ID: <Pine.GSO.4.64.0710261248110.5522@clubhouse-6.cisco.com>

yes, Intercom is availble in CM6.  CSCsk99117 is open to track 'Intercom 
History' as an issue for CM3.x, CM4.x, and CM5.x  We have 24 customers 
feedback that this is an 'undesirable' change.  Each additional unit of 
feedback counts.  Feedback can be additional TAC cases attached to the 
defect or direct feedback to the phone PM at CIPTUG next week.

/Wes

On Fri, 26 Oct 2007, Miller, Steve wrote:

> I missed something.  Is there a new (true) intercom feature available
> in 6X?
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> MillerS at dicksteinshapiro.com
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
> Kulagowski
> Sent: Friday, October 26, 2007 10:24 AM
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Intercom History
>
> Ryan Ratliff wrote:
>> No the intercom is only in 6.x however the phone load included with
>> SR5d added support for that feature.  The phone however isn't smart
>> enough to know that the CM it's registering with supports the feature
>
>> or not so it always shows the intercom history.
>
> Ugh.  Cisco has managed to stuff a peer-to-peer firmware distribution
> system into the phone, but not a "capabilities" query during
> registration to CallManager that returns a boolean "display Intercom
> History bubble"?
>
> This almost seems like a mis-feature to me.  I'm sure that someone,
> somewhere really, really loves this Intercom History thing, and I'll
> tell my account team that I don't. But does an organization with 1500
> 7970 telephones really have any voice in these sorts of things?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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>

From wsisk at cisco.com  Fri Oct 26 12:57:09 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 26 Oct 2007 12:57:09 -0400 (EDT)
Subject: [cisco-voip] Arc Consoles losing connection
In-Reply-To: <970310A6A9EFBC4189C5B1C43EFDDB8BC7DECE@pitmx1.principal-it.com>
References: <475fda330710250839k66e43713r850e92d0243a8b11@mail.gmail.com>
	<4720BE57.1040303@cisco.com>
	<970310A6A9EFBC4189C5B1C43EFDDB8BC7DECE@pitmx1.principal-it.com>
Message-ID: <Pine.GSO.4.64.0710261253040.5522@clubhouse-6.cisco.com>

Nick,

In that case it appears the phone stayed registered.  Next step are to 
check the processes in play between ARC and CM:

ARC--(tsp or jtapi)--CTIManager---CCM---Phone

I would get the CTIManager SDI and SDL logs from CM next and search for 
errors.  In CM3.x and 4.x CTI SDL errors are strings that begin with 
'0x8CC'.  The CTI SDL lines usually include the device name as well 
(SEP....).

If you're going to TAC, then get both the CTIManager CTI/SDL and the CM 
SDI/SDL from all nodes in the cluster around the time of the phone outage 
(+/- 10 mins).  Open a TAC case and send me the case number.

/Wes

On Fri, 26 Oct 2007, Nick Kassel wrote:

> Wes
>
>
>
> One of the phones attached to an Arc console has just gone down again,
> when I checked the network statistics screen on the phone the elapsed
> time was showing over 10 hours which was the time we would have rebooted
> it yesterday. However the Debug Display still shows the Last=CM
> Closed-TCP message which I now understand means it has tried to fail
> over the back up call manager.
>
>
>
> Is there anywhere else I could check for information on why this phone
> may be losing connection with the primary Call manager.
>
>
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
> Sent: 25 October 2007 17:04
> To: Nick
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Arc Consoles losing connection
>
>
>
> if the phone deregisters i believe ARC has to go down as CM will send
> CTIDeviceClose to ARC.
> On the phone under settings->status->network statistics what is the
> 'Elapsed Time'?  This indicates the duration of time the phone has been
> properly registered to its current CM. If the phone is unregistering
> then ARC will most likely be affected.
>
> /wes
>
> Nick wrote:
>
> I have an issue with arc consoles losing connection this is happening to
> a number of consoles in the same site when I check the phones the debug
> display on the phone it says
>
>
>
> 15:57:33 14: Name=SEP00146905158B Load= 7.0(3.0S) Last=CM-closed-TCP
>
> each time it loses connection, we have tried re patching to different
> switchports and different switches and still seem to be getting the same
> issue, could this be an isue with the phone load as it is very old
> firmware, all other consoles on the same arc server and other phones in
> the same site don't appear to be getting the issue althouh it may not be
> as noticeable as on an Arc console.
>
>
>
> Has anyone had this issue with Arc before or can make any suggestions on
> what else to check
>
>
>
>
>
> ________________________________
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From Roconnell at unislumin.com  Fri Oct 26 13:20:09 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Fri, 26 Oct 2007 13:20:09 -0400
Subject: [cisco-voip] CUBE or IP IP GW with MTP
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
	<18082B35CF0EBD4C8D265874508A615C073E5884@hermes.unislumin.com>
	<1193386716.7226.125.camel@mda>
Message-ID: <18082B35CF0EBD4C8D265874508A615C057658BC@hermes.unislumin.com>

Ok, 
 
Whats best practice
 
So If I have a Service Provider SIP trunk terminating lets say 100 DID's. From my IP to IP GW I'm pointing these DID's to various destinations. For example 40 DID's going to CallManager, 30 DID's going to a remote CallManager Express and the remaining 30 DID's pointing to something else like some of that new Microsoft Stuff (mediation server and OCS)
 
If I'm doing something like this do I setup all MTP / Transcoding on the IPtoIP GW and register it against Telephony Services or should I use each remote system to have their own resources to use?
Or
Do I pass all calls through one system ie (CallManger) and create a dialplan from there that brings all these systems together?
 
 
Hope this makes sense?
 
Ryan
 
<http://www.unislumin.com>  

________________________________

From: tele [mailto:tele at plexialab.it]
Sent: Fri 10/26/2007 2:18 AM
To: Ryan O'Connell
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CUBE or IP IP GW with MTP



Hi,

Ryan yes you need turn on "telephony-services".
You can also use an external DSP farm for the MTP or transcoding.

http://www.cisco.com/en/US/products/ps6706/products_feature_guide09186a008076161a.html



:tele

On Thu, 2007-10-25 at 22:34 -0400, Ryan O'Connell wrote:
> Hello all,
>
> 
>
> Quick question, on IOS routers you can use DSP resources as MTP. I
> understand and implemented this many times in the past using SCCP
> commands that point back to CallManager or CallManager Express. I'm
> just wondering in the case of CUBE or IP-IP GW used as a SIP trunk to
> the Service provider, and we are not using CallManager or CallManager
> Express can I still use the DSP's for MTP's, Transcoding? In my case
> I'm specifically looking for MTP's cause I need it for FAST Start
> H.323 calls.
>
> 
>
> According to the attached Document there is such thing as a "HARDWARE
> based MTP" but the bullet stats that it uses SCCP
>
> 1.    MTPs ( and Transcoders ) can be controlled by Communications
> Manager or the Cisco Unified Border Element (via SCCP)
>
> So do they mean that I have to turn on "TELEPHONY-SERVICES" just to
> make use of these MTP or am I missing something
>
> 
>
> Ryan
>
> 
>
> 
>
>
>
> ______________________________________________________________________
> Privileged/Confidential Information may be contained in this message.
> Disclosure to any person other than the named recipient is
> unauthorized.  If you are not the intended recipient, please delete
> all copies of this information and kindly notify the sender by reply
> email.  Opinions, conclusions and other information in this message
> that do not relate to the official business of UNIS LUMIN Inc. shall
> be understood as neither given nor endorsed by it.  UNIS LUMIN Inc.
> and any of its subsidiaries reserve the right to monitor all e-mail
> communications through its networks.  Thank you.
>
>
> ______________________________________________________________________
>
> Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the intended
> addressee, you are hereby notified that you have received this
> communication in error and that any use or reproduction of this email
> or its contents is strictly prohibited and may be unlawful. If you
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> _______________________________________________
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From jsteinberg at gmail.com  Fri Oct 26 13:28:13 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 26 Oct 2007 13:28:13 -0400
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0071FFC39@USNAEXCH.na.didata.local>
References: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com>
	<FDC137145729AD438CDCA0F9FC6F7236013FE273@mail.netechcorp.com>
	<f1b65f880710161505m5989eed5leec418d9ecd76b96@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0071FFC39@USNAEXCH.na.didata.local>
Message-ID: <f1b65f880710261028q35bca6c5l4ed73dbc7ecd62d4@mail.gmail.com>

I finally had some more time today to look back into this issue and
have been able to get this to work with 6.0.1.

-Create a Enterprise Root Microsoft Certificate Services on a W2K3
domain member server.
-Upload this root CA certificate to CUPS as a 'presence-trust' certificate.
-Add this root CA to your domain security group policy.  push group
policy to the clients.
-Create a CSR on my Exchange box via IIS.  Had this CSR signed by the
enterprise root CA.
-Upload the signed cert back to IIS on exchange. restart iis.
-Configure CUPS presence engine for Outlook.  The common name is the
name of the presence-trust certificate as viewed in the CUPS OS admin
certificate page.
-Rebooted presence.

It works - pretty cool too.


Justin

On 10/17/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> I checked with the BU when I had trouble getting it to work.  The
> response I received from their group is that is it not supported until
> 6.0(2).  It is not documented anywhere.  Have any of you been able to
> get this part to work?
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> Steinberg
> Sent: Tuesday, October 16, 2007 6:06 PM
> To: Jason Burton
> Cc: cisco voip list
> Subject: Re: [cisco-voip] CUPS6 integration to Exchange Calendaring
>
> Jason, Are you speaking about the SSL cert that I use for url such as:
>
> https://exchangeserver/Exchange/Jsteinberg/Calendar/
>
>
> Matt, where did you hear it wasn't working in CUPS 6.0.1 ?  It is
> mentioned in the release notes for 6.0.1.  Is there a bug filed
> against it?
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cups/6_0_1/rel_notes/cup60
> 1rn.html#wp478329
>
>
> On 10/16/07, Jason Burton <jburton at netechcorp.com> wrote:
> > I ran into a bug with it.  Are you using an extended validation SSL
> cert
> > (the one that changes IE7's address bar to green)?
> >
> > _____________________________
> > Jason Burton
> > Systems Engineer
> > CCIE Voice #17319
> > NETech Corporation
> > 317.660.3010 - Desk
> > 317.581.1919 - Fax
> > jburton at netechcorp.com
> >
> >
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> > Steinberg
> > Sent: Tuesday, October 16, 2007 12:17 PM
> > To: cisco voip list
> > Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
> >
> > Has anyone been successful with this?
> >
> > I am trying to get this to work but can't figure out what the 'Trust
> > Certificate Subject CN' field is for on the presence gateway page when
> > integrating to outlook.
> >
> > thanks.
> > justin
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>

From jason.aarons at us.didata.com  Fri Oct 26 14:40:09 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 26 Oct 2007 14:40:09 -0400
Subject: [cisco-voip] Phantom Phone?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0073C4D12@USNAEXCH.na.didata.local>

In CallManager 4.1 I would create a CTI Port that sends to a Unity
Mailbox for users that don't have a phone or desk but work remotely and
need a Voicemail box with dial-by-name.

 

In CallManager 5.1 due to licensing is this still the preferred method? 




-----------------------------------------
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From svoll.voip at gmail.com  Fri Oct 26 15:52:41 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 26 Oct 2007 12:52:41 -0700
Subject: [cisco-voip] Phantom Phone?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0073C4D12@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
	<C1FE15183DA37645BC0633BC604E44F0073C4D12@USNAEXCH.na.didata.local>
Message-ID: <f84a38d30710261252q58653435s535ca1f5d87fbbf@mail.gmail.com>

I don't believe there is any licensing issues with CTI ports so I think that
would work.

Scott


On 10/26/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  *In CallManager 4.1 I would create a CTI Port that sends to a Unity
> Mailbox for users that don't have a phone or desk but work remotely and need
> a Voicemail box with dial-by-name.*
>
> * *
>
> *In CallManager 5.1 due to licensing is this still the preferred method? *
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
> _______________________________________________
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From jonvoip at gmail.com  Fri Oct 26 16:51:11 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 26 Oct 2007 15:51:11 -0500
Subject: [cisco-voip] Phantom Phone?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0073C4D12@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0073C41EE@USNAEXCH.na.didata.local>
	<C1FE15183DA37645BC0633BC604E44F0073C4D12@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0710261351p62c51c8bgde8ab4a67c42f3d4@mail.gmail.com>

CTI ports are free, just create them and CFA them to vmail.



J

On 10/26/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> In CallManager 4.1 I would create a CTI Port that sends to a Unity Mailbox
> for users that don't have a phone or desk but work remotely and need a
> Voicemail box with dial-by-name.
>
>
>
> In CallManager 5.1 due to licensing is this still the preferred method?
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From rkulagow at gmail.com  Fri Oct 26 20:48:16 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Fri, 26 Oct 2007 19:48:16 -0500
Subject: [cisco-voip] So, what's a 7937G telephone?
Message-ID: <47228AD0.8090405@gmail.com>

Firmware for it is posted on CCO, but can't find marketing for it.  Some 
sort of conference phone I'm guessing?

From asobihoudai at yahoo.com  Fri Oct 26 20:50:24 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 26 Oct 2007 17:50:24 -0700 (PDT)
Subject: [cisco-voip] So, what's a 7937G telephone?
In-Reply-To: <47228AD0.8090405@gmail.com>
Message-ID: <14370.44766.qm@web57210.mail.re3.yahoo.com>

It's a very purdy more industrial-looking conference
phone...with the new blobby logo on it!

--- Robert Kulagowski <rkulagow at gmail.com> wrote:

> Firmware for it is posted on CCO, but can't find
> marketing for it.  Some 
> sort of conference phone I'm guessing?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From Timothy_Frazee at adp.com  Fri Oct 26 22:38:04 2007
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Fri, 26 Oct 2007 20:38:04 -0600
Subject: [cisco-voip] So, what's a 7937G telephone?
Message-ID: <a8df01c8183a$04e517fc$56cc7e8b@ds.ad.adp.com>

Still made by polycom?


-----Original Message-----
From: "Paul Choi" <asobihoudai at yahoo.com>
To: "Robert Kulagowski" <rkulagow at gmail.com>; "cisco voip list" <cisco-voip at puck.nether.net>
Sent: 10/26/07 6:52 PM
Subject: Re: [cisco-voip] So, what's a 7937G telephone?

It's a very purdy more industrial-looking conference
phone...with the new blobby logo on it!

--- Robert Kulagowski <rkulagow at gmail.com> wrote:

> Firmware for it is posted on CCO, but can't find
> marketing for it.  Some 
> sort of conference phone I'm guessing?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around 
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_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip


This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.

From asobihoudai at yahoo.com  Fri Oct 26 21:51:39 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 26 Oct 2007 18:51:39 -0700 (PDT)
Subject: [cisco-voip] So, what's a 7937G telephone?
In-Reply-To: <a8df01c8183a$04e517fc$56cc7e8b@ds.ad.adp.com>
Message-ID: <490664.80924.qm@web57202.mail.re3.yahoo.com>

Welp,

It's still shaped like a star so I'm going to assume
so.

Paul
--- "Frazee, Timothy" <Timothy_Frazee at adp.com> wrote:

> Still made by polycom?
> 
> 
> -----Original Message-----
> From: "Paul Choi" <asobihoudai at yahoo.com>
> To: "Robert Kulagowski" <rkulagow at gmail.com>; "cisco
> voip list" <cisco-voip at puck.nether.net>
> Sent: 10/26/07 6:52 PM
> Subject: Re: [cisco-voip] So, what's a 7937G
> telephone?
> 
> It's a very purdy more industrial-looking conference
> phone...with the new blobby logo on it!
> 
> --- Robert Kulagowski <rkulagow at gmail.com> wrote:
> 
> > Firmware for it is posted on CCO, but can't find
> > marketing for it.  Some 
> > sort of conference phone I'm guessing?
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> This message and any attachments are intended only
> for the use of the addressee and may contain
> information that is privileged and confidential. If
> the reader of the message is not the intended
> recipient or an authorized representative of the
> intended recipient, you are hereby notified that any
> dissemination of this communication is strictly
> prohibited. If you have received this communication
> in error, please notify us immediately by e-mail and
> delete the message and any attachments from your
> system.
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From jeff at ocjtech.us  Fri Oct 26 22:10:18 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Fri, 26 Oct 2007 21:10:18 -0500
Subject: [cisco-voip] So, what's a 7937G telephone?
In-Reply-To: <47228AD0.8090405@gmail.com>
References: <47228AD0.8090405@gmail.com>
Message-ID: <1193451019.3249.0.camel@lt21223.campus.dmacc.edu>

On Fri, 2007-10-26 at 19:48 -0500, Robert Kulagowski wrote:
> Firmware for it is posted on CCO, but can't find marketing for it.  Some 
> sort of conference phone I'm guessing?

http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900aecd806e021a.html

Jeff

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From jason.aarons at us.didata.com  Fri Oct 26 23:17:29 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 26 Oct 2007 23:17:29 -0400
Subject: [cisco-voip] Route Pattern > Route List reset
Message-ID: <C1FE15183DA37645BC0633BC604E44F0074023A2@USNAEXCH.na.didata.local>

I thought in 4.x a MGCP controlled gateway when reset via a Route
Pattern > Route List change would wait until there are no callers to
restart (media preservation).

 

I see clearly in 5.x it definitely drops the callers via reset, phones
show Temp Failure.

 

Did something change between 4.x and 5.x or is my memory wrong. I flip
between MGCP and H323 so much I forget what does what...

 

 




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From khalid_khursheed at hotmail.com  Sat Oct 27 01:53:29 2007
From: khalid_khursheed at hotmail.com (Syed Khalid Ali)
Date: Sat, 27 Oct 2007 10:53:29 +0500
Subject: [cisco-voip] Difference b/w Call Manager 4, 5 and 6
In-Reply-To: <535625f70710260736u5f81ae43kc23d311e1d859d70@mail.gmail.com>
References: <BAY113-W25DF280BF6E66AADDDAD88E8960@phx.gbl>
	<c0deb5b80710260723x51b6247an15662f4da0292078@mail.gmail.com> 
	<535625f70710260736u5f81ae43kc23d311e1d859d70@mail.gmail.com>
Message-ID: <BAY113-W350168FF722242652B52E7E8970@phx.gbl>



hi,

i know that i'm studying for cvoice exam, but i want to have a better understanding of different call manager version, just not on the basis of operating system. i meant to know the about feature and advantages and cons  of each version

regards,
syed khalid ali

________________________________
Date: Fri, 26 Oct 2007 09:36:48 -0500
From: voicenoob at gmail.com
To: patrick.diener at gmail.com
Subject: Re: [cisco-voip] Difference b/w Call Manager 4, 5 and 6
CC: khalid_khursheed at hotmail.com; cisco-voip at puck.nether.net

You also shouldn't base your desision on what version to deploy strictly on the operating system.



On 10/26/07, Patrick Diener <patrick.diener at gmail.com> wrote:
btw cvoice does not test anything callmanager related...

regards
patrick

On 10/26/07, Syed Khalid Ali < khalid_khursheed at hotmail.com> wrote:
>
>
> hi,
>
> i'm fairly new to voip world and currently preparing for cvoice exam. Can
> any one tell me the difference between CallManager 4,5 and 6.
>
> eagrly waiting for the response
>
> thanks
>
> syed khalid ali
>
> ________________________________
> Discover the new Windows Vista Learn more!
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From rkulagow at gmail.com  Sat Oct 27 10:38:09 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Sat, 27 Oct 2007 09:38:09 -0500
Subject: [cisco-voip] So, what's a 7937G telephone?
In-Reply-To: <1193451019.3249.0.camel@lt21223.campus.dmacc.edu>
References: <47228AD0.8090405@gmail.com>
	<1193451019.3249.0.camel@lt21223.campus.dmacc.edu>
Message-ID: <47234D51.5030006@gmail.com>

Jeffrey C. Ollie wrote:
> On Fri, 2007-10-26 at 19:48 -0500, Robert Kulagowski wrote:
>> Firmware for it is posted on CCO, but can't find marketing for it.  Some 
>> sort of conference phone I'm guessing?
> 
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900aecd806e021a.html

Thanks, guess I should have waited a day, because it sure wasn't there 
on Thursday when I searched for it.


From dwolgas1 at rochester.rr.com  Sat Oct 27 11:08:02 2007
From: dwolgas1 at rochester.rr.com (Dave Wolgast)
Date: Sat, 27 Oct 2007 11:08:02 -0400
Subject: [cisco-voip] So, what's a 7937G telephone?
In-Reply-To: <1193451019.3249.0.camel@lt21223.campus.dmacc.edu>
References: <47228AD0.8090405@gmail.com>
	<1193451019.3249.0.camel@lt21223.campus.dmacc.edu>
Message-ID: <e4491a9c0710270808x35d12d72q32638bbcdf3f46d0@mail.gmail.com>

Just curious...is there any Cisco IP Phone that can be tied into a room
system with embedded speakers and mics(controlled by, for example, a Polycom
Vortex unit)?

At one point, I know one could extend a 7970G, but that potentially
caused electrical issues with the phone, so the external jacks were removed.


On 10/26/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote:
>
> On Fri, 2007-10-26 at 19:48 -0500, Robert Kulagowski wrote:
> > Firmware for it is posted on CCO, but can't find marketing for it.  Some
> > sort of conference phone I'm guessing?
>
>
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900aecd806e021a.html
>
> Jeff
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
-- 
Dave Wolgast
Livonia, NY
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From jason.aarons at us.didata.com  Sat Oct 27 11:06:05 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sat, 27 Oct 2007 11:06:05 -0400
Subject: [cisco-voip] So, what's a 7937G telephone?
In-Reply-To: <47234D51.5030006@gmail.com>
References: <47228AD0.8090405@gmail.com><1193451019.3249.0.camel@lt21223.campus.dmacc.edu>
	<47234D51.5030006@gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0074023DA@USNAEXCH.na.didata.local>

Will it be on New Product Hold for 6 months or when is general shipment
scheduled to begin?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Kulagowski
Sent: Saturday, October 27, 2007 10:38 AM
To: cisco voip list
Subject: Re: [cisco-voip] So, what's a 7937G telephone?

Jeffrey C. Ollie wrote:
> On Fri, 2007-10-26 at 19:48 -0500, Robert Kulagowski wrote:
>> Firmware for it is posted on CCO, but can't find marketing for it.
Some 
>> sort of conference phone I'm guessing?
> 
>
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0
900aecd806e021a.html

Thanks, guess I should have waited a day, because it sure wasn't there 
on Thursday when I searched for it.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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From aelnagar at ACT-EG.COM  Sat Oct 27 11:16:43 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Sat, 27 Oct 2007 17:16:43 +0200
Subject: [cisco-voip] Multicast in a campus network
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED011DA699@mail.ACT-EG.COM>

Hello all;

I dont know if someone dealed before with IPTV system (but I am sure u did). The customer wants to deploy IPTV system (not cisco) and he told me that the system is using multicast for delivering TV stations to clients, I have read alot of documents regarding multiacst but I am not sure which configuration I should use to run this successfully.

The customer network consists of two backbone 6509 switch connected to it 3560 switches as edge switches, if anyone has ideas about this and how to configure these swithces (backbone and edge) to pass multicast traffic.

Also note that there will be clients in vlans other than the server vlan.

thanks

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From aelnagar at ACT-EG.COM  Sat Oct 27 11:48:58 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Sat, 27 Oct 2007 17:48:58 +0200
Subject: [cisco-voip] Multicast in a campus network
References: <88C8D019DA547C4BABECB9A806ED22ED011DA699@mail.ACT-EG.COM>
	<C1FE15183DA37645BC0633BC604E44F0074023DE@USNAEXCH.na.didata.local>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED011DA69A@mail.ACT-EG.COM>

Hello jason
 
thanks alot for your reply, I have read alot thorugh this and I know pim with its modes very well, but these all talks about multicast routing and routers configurations. I didnot pass by any document that explains these in a network where L3 switches is used for inter vlan routing. I read also about MVR but I dont know is it okay for my case or not.
________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Sat 27-Oct-07 5:37 PM
To: Ahmed Elnagar
Subject: RE: [cisco-voip] Multicast in a campus network



Research ip pim sparse-dense mode

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Saturday, October 27, 2007 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Multicast in a campus network

 

Hello all;

I dont know if someone dealed before with IPTV system (but I am sure u did). The customer wants to deploy IPTV system (not cisco) and he told me that the system is using multicast for delivering TV stations to clients, I have read alot of documents regarding multiacst but I am not sure which configuration I should use to run this successfully.

The customer network consists of two backbone 6509 switch connected to it 3560 switches as edge switches, if anyone has ideas about this and how to configure these swithces (backbone and edge) to pass multicast traffic.

Also note that there will be clients in vlans other than the server vlan.

thanks

________________________________

Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 

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From Ruben.Montes at eu.didata.com  Sat Oct 27 14:54:56 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Sat, 27 Oct 2007 20:54:56 +0200
Subject: [cisco-voip] Multicast in a campus network
References: <88C8D019DA547C4BABECB9A806ED22ED011DA699@mail.ACT-EG.COM><C1FE15183DA37645BC0633BC604E44F0074023DE@USNAEXCH.na.didata.local>
	<88C8D019DA547C4BABECB9A806ED22ED011DA69A@mail.ACT-EG.COM>
Message-ID: <FA233791F398B848B978464507739E8FDE3D26@EUBEBRUSVEX1.eu.didata.local>

Hello,

 

Inside a vlan and using L3 switches is exactly the same, you should enable pim and multicast routing: I would recommend using sparse mode instead dense mode. I would configure one 6509 as RP (and maybe another one as secondary for redundancy purposes) and all the leaf routers pointing to this router RP address (you can create a Loopback address). For the layer 2 switches (3560), it is enough with igmp snooping, which comes enabled by default...

 

HTH,

 

Ruben

 

________________________________

De: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] En nombre de Ahmed Elnagar
Enviado el: s?bado, 27 de octubre de 2007 17:49
Para: Jason Aarons (US)
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Multicast in a campus network

 

Hello jason

 

thanks alot for your reply, I have read alot thorugh this and I know pim with its modes very well, but these all talks about multicast routing and routers configurations. I didnot pass by any document that explains these in a network where L3 switches is used for inter vlan routing. I read also about MVR but I dont know is it okay for my case or not.

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Sat 27-Oct-07 5:37 PM
To: Ahmed Elnagar
Subject: RE: [cisco-voip] Multicast in a campus network

Research ip pim sparse-dense mode

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Saturday, October 27, 2007 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Multicast in a campus network

 

Hello all;

I dont know if someone dealed before with IPTV system (but I am sure u did). The customer wants to deploy IPTV system (not cisco) and he told me that the system is using multicast for delivering TV stations to clients, I have read alot of documents regarding multiacst but I am not sure which configuration I should use to run this successfully.

The customer network consists of two backbone 6509 switch connected to it 3560 switches as edge switches, if anyone has ideas about this and how to configure these swithces (backbone and edge) to pass multicast traffic.

Also note that there will be clients in vlans other than the server vlan.

thanks

________________________________

Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 

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From aelnagar at ACT-EG.COM  Sat Oct 27 15:04:22 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Sat, 27 Oct 2007 21:04:22 +0200
Subject: [cisco-voip] Multicast in a campus network
References: <88C8D019DA547C4BABECB9A806ED22ED011DA699@mail.ACT-EG.COM><C1FE15183DA37645BC0633BC604E44F0074023DE@USNAEXCH.na.didata.local>
	<88C8D019DA547C4BABECB9A806ED22ED011DA69A@mail.ACT-EG.COM>
	<FA233791F398B848B978464507739E8FDE3D26@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED011DA69B@mail.ACT-EG.COM>

Hello ruben;
 
Thank you first for ur reply, and yes that how I am starting to think about it but I am not sure that I will go with the sparse mode (I dont want to go into a lot of configuration) I will try dense mode first and if things are okay I will think of ur advice of using sparse.
 
thanks

 
________________________________

From: Ruben Montes (Europe) [mailto:Ruben.Montes at eu.didata.com]
Sent: Sat 27-Oct-07 8:54 PM
To: Ahmed Elnagar; Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Multicast in a campus network



Hello,

 

Inside a vlan and using L3 switches is exactly the same, you should enable pim and multicast routing: I would recommend using sparse mode instead dense mode. I would configure one 6509 as RP (and maybe another one as secondary for redundancy purposes) and all the leaf routers pointing to this router RP address (you can create a Loopback address). For the layer 2 switches (3560), it is enough with igmp snooping, which comes enabled by default...

 

HTH,

 

Ruben

 

________________________________

De: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] En nombre de Ahmed Elnagar
Enviado el: s?bado, 27 de octubre de 2007 17:49
Para: Jason Aarons (US)
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Multicast in a campus network

 

Hello jason

 

thanks alot for your reply, I have read alot thorugh this and I know pim with its modes very well, but these all talks about multicast routing and routers configurations. I didnot pass by any document that explains these in a network where L3 switches is used for inter vlan routing. I read also about MVR but I dont know is it okay for my case or not.

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Sat 27-Oct-07 5:37 PM
To: Ahmed Elnagar
Subject: RE: [cisco-voip] Multicast in a campus network

Research ip pim sparse-dense mode

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Saturday, October 27, 2007 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Multicast in a campus network

 

Hello all;

I dont know if someone dealed before with IPTV system (but I am sure u did). The customer wants to deploy IPTV system (not cisco) and he told me that the system is using multicast for delivering TV stations to clients, I have read alot of documents regarding multiacst but I am not sure which configuration I should use to run this successfully.

The customer network consists of two backbone 6509 switch connected to it 3560 switches as edge switches, if anyone has ideas about this and how to configure these swithces (backbone and edge) to pass multicast traffic.

Also note that there will be clients in vlans other than the server vlan.

thanks

________________________________

Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 

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From l.durso at gmail.com  Sat Oct 27 17:04:41 2007
From: l.durso at gmail.com (=?utf-8?B?TGVvbmFyZG8gRCdVcnNv?=)
Date: Sat, 27 Oct 2007 21:04:41 +0000
Subject: [cisco-voip] Intercluster extension mobility
Message-ID: <1321198905-1193519017-cardhu_decombobulator_blackberry.rim.net-1427276249-@bxe006.bisx.produk.on.blackberry>


Hi 

Is it possible for two callmanager cluster connected by an intercluster trunk to share the exmob service, so a user defined on the first cluster can register to its exmob service when he/she is under the other cluster?
I know there was a product but Cisco has done an acquisition and now this product is no more available. 

Kind Regards
Leo
---
Leonardo D'Urso
l.durso at gmail.com


From asobihoudai at yahoo.com  Sat Oct 27 17:07:18 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Sat, 27 Oct 2007 14:07:18 -0700 (PDT)
Subject: [cisco-voip] Difference b/w Call Manager 4, 5 and 6
In-Reply-To: <BAY113-W350168FF722242652B52E7E8970@phx.gbl>
Message-ID: <207819.23440.qm@web57208.mail.re3.yahoo.com>

Syed,

Please contact your local Cisco Learning Partner and
enroll in their CVOICE class. You will find that
CVOICE has very little CallManager material in it.



__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From wsisk at cisco.com  Sat Oct 27 23:13:17 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Sat, 27 Oct 2007 23:13:17 -0400
Subject: [cisco-voip] Intercluster extension mobility
In-Reply-To: <1321198905-1193519017-cardhu_decombobulator_blackberry.rim.net-1427276249-@bxe006.bisx.produk.on.blackberry>
References: <1321198905-1193519017-cardhu_decombobulator_blackberry.rim.net-1427276249-@bxe006.bisx.produk.on.blackberry>
Message-ID: <4723FE4D.4030805@cisco.com>

Not natively.  Litescape provides it by addon applications. /wes

Leonardo D'Urso wrote:
> Hi 
>
> Is it possible for two callmanager cluster connected by an intercluster trunk to share the exmob service, so a user defined on the first cluster can register to its exmob service when he/she is under the other cluster?
> I know there was a product but Cisco has done an acquisition and now this product is no more available. 
>
> Kind Regards
> Leo
> ---
> Leonardo D'Urso
> l.durso at gmail.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Sat Oct 27 23:18:02 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Sat, 27 Oct 2007 23:18:02 -0400
Subject: [cisco-voip] R: Re:  Callmanager cluster in Wan
In-Reply-To: <1083021520-1193343385-cardhu_decombobulator_blackberry.rim.net-1036260840-@bxe006.bisx.produk.on.blackberry>
References: <929274954-1193320798-cardhu_decombobulator_blackberry.rim.net-1398273527-@bxe006.bisx.produk.on.blackberry><4720AB76.7070903@cisco.com>
	<1083021520-1193343385-cardhu_decombobulator_blackberry.rim.net-1036260840-@bxe006.bisx.produk.on.blackberry>
Message-ID: <4723FF6A.8030409@cisco.com>

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From lelio at uoguelph.ca  Sun Oct 28 01:18:37 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun, 28 Oct 2007 01:18:37 -0400
Subject: [cisco-voip] OT (sort of): cisco telepresence vs air force VR system
Message-ID: <000a01c81921$fecc5bc0$6601a8c0@pdp11>

http://gizmodo.com/gadgets/military-sim-almost-brings-halo-3-to-reality/military-vr-simulator-is-closest-thing-ever-to-real+life-halo-3-315435.php

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From nikola at worldnet.att.net  Sun Oct 28 20:20:42 2007
From: nikola at worldnet.att.net (Nikola Stojsin)
Date: Sun, 28 Oct 2007 20:20:42 -0400
Subject: [cisco-voip] H323 dial-peers question
Message-ID: <000f01c819c1$8af4a310$ca1ea8c0@T42p>

I have a question - I have just set up a set of H323 dial-peers with
CallManager and Unity and DID extensions. The configuration below works, but
it looks ugly to me. If someone can take a look at it and let me know  what
should be done to it. Thanks!

 

(Note: 4900 is the Unity AA pilot; CM extensions are the same as the last
four digits of the DIDs. Yes, telco sends ten digits; please do not ask me
why. 

 

voice translation-rule 10

 rule 1 /......4300/ /4900/

 rule 2 /......4301/ /4301/

 rule 3 /......4302/ /4302/

 rule 4 /......4303/ /4303/

 rule 5 /......4304/ /4304/

 rule 6 /......4305/ /4305/

 rule 7 /......4306/ /4306/

 rule 8 /......4307/ /4307/

 rule 9 /......4308/ /4308/

 rule 10 /.*/ /4900/

!

voice translation-profile To_Unity

 translate called 10

!

dial-peer voice 9 pots

 description To_PSTN 

 destination-pattern 0T

 direct-inward-dial

 port 0/1/0:23

!

dial-peer voice 4900 voip

 description PSTN_To_Unity_Translating 

 translation-profile outgoing To_Unity

 destination-pattern ......4...

 voice-class codec 1

 voice-class h323 1

 session target ipv4:192.168.57.15

 dtmf-relay h245-alphanumeric

 ip qos dscp cs3 signaling

 no vad   

 

Again, thanks in advance!

Nikola

 

 

------------------------------
Nikola Stojsin
Nikola at att.net
(917) 558-1423

(917) 591-9382 (fax)
------------------------------

 

 

 

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From Roconnell at unislumin.com  Sun Oct 28 22:20:35 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Sun, 28 Oct 2007 22:20:35 -0400
Subject: [cisco-voip] H323 dial-peers question
References: <000f01c819c1$8af4a310$ca1ea8c0@T42p>
Message-ID: <18082B35CF0EBD4C8D265874508A615C057658D3@hermes.unislumin.com>

Many ways of accomplishing this. One way of doing it is let CallManager do the digit stripping. You can use the "significant digits" on the H.323 page of Callmanager to only accept 4 digits instead of 10. Then you can setup one incoming, (or multiple depending if you have a Pub and a Sub in your environment) dialpeer that sends all 10 digits to Callmanager.
If you need to do any translations outside of this you can do it all in callmanager with partitions and translation patterns
 
Example

 

dial-peer voice 1 voip

 description Incoming from PSTN

 preference 1

 incoming called-number .

 session target ipv4:<<ccmsub IP>>

 dtmf-relay h245-alphanumeric

 no vad   

 

dial-peer voice 2 voip

 description Incoming from PSTN

 preference 2

 incoming called-number .

 session target ipv4:<<ccmpub IP>>

 dtmf-relay h245-alphanumeric

 no vad   

 

 




________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Nikola Stojsin
Sent: Sun 10/28/2007 6:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] H323 dial-peers question



I have a question - I have just set up a set of H323 dial-peers with CallManager and Unity and DID extensions. The configuration below works, but it looks ugly to me. If someone can take a look at it and let me know  what should be done to it. Thanks!

 

(Note: 4900 is the Unity AA pilot; CM extensions are the same as the last four digits of the DIDs. Yes, telco sends ten digits; please do not ask me why. 

 

voice translation-rule 10

 rule 1 /......4300/ /4900/

 rule 2 /......4301/ /4301/

 rule 3 /......4302/ /4302/

 rule 4 /......4303/ /4303/

 rule 5 /......4304/ /4304/

 rule 6 /......4305/ /4305/

 rule 7 /......4306/ /4306/

 rule 8 /......4307/ /4307/

 rule 9 /......4308/ /4308/

 rule 10 /.*/ /4900/

!

voice translation-profile To_Unity

 translate called 10

!

dial-peer voice 9 pots

 description To_PSTN 

 destination-pattern 0T

 direct-inward-dial

 port 0/1/0:23

!

dial-peer voice 4900 voip

 description PSTN_To_Unity_Translating 

 translation-profile outgoing To_Unity

 destination-pattern ......4...

 voice-class codec 1

 voice-class h323 1

 session target ipv4:192.168.57.15

 dtmf-relay h245-alphanumeric

 ip qos dscp cs3 signaling

 no vad   

 

Again, thanks in advance!

Nikola

 

 

------------------------------
Nikola Stojsin
Nikola at att.net
(917) 558-1423

(917) 591-9382 (fax)
------------------------------

 

 

 


Attention:

Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.
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From ErasmuE4 at telkom.co.za  Mon Oct 29 01:35:32 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 29 Oct 2007 07:35:32 +0200
Subject: [cisco-voip] automate stop and start of services
In-Reply-To: <Pine.GSO.4.64.0710260958100.1665@clubhouse-8.cisco.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01C09CEB@CNTRRA20-XCS01.telkom.co.za>

Thanks wes

-----Original Message-----
From: Wesley Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, October 26, 2007 3:59 PM
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] automate stop and start of services

write a .bat file:
*********
net stop "cisco callmanager"
net start "cisco callmanager"
*********
enable scheduler service and schedule this .bat to job when you
desire. 
substitute the service name you need for 'cisco callmanager'.

/wes

On Fri, 26 Oct 2007, Erik Erasmus (E) wrote:

>
> Hi
>
>
>
> Does anyone know of a way to automate the restarting of call manager
4
> services? We have a problem with the de queuing of calls to our
> attendant consoles in ccm 4-2-3 sr2b (cisco bug) and are waiting for
a
> software fix. In the meantime when the problem occurs we reset TCD
and
> cti services manually. We noticed that after resetting these
services
> that things acrry on working for a couple of days so we want to if
> possible automate this process and maybe use the Microsoft Scheduler
> to do this.
>
>
>
>
>
> Any advice will be appreciated
>
>
>
>
>
>
>
> Erik Erasmus
>
>
>
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> This e-mail is subject to the Telkom SA electronic communication
legal notice, available at :
> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

From cisco-voip at ibh.net  Mon Oct 29 08:15:39 2007
From: cisco-voip at ibh.net (Andre Beck)
Date: Mon, 29 Oct 2007 13:15:39 +0100
Subject: [cisco-voip] DST change delay
Message-ID: <20071029121539.GD9131@ibh.de>

Hi,

last weekend our local time changed from CEST back to CET. Now DST
issues are a seemingly neverending story, but it generally worked, at
least time was correct on monday morning. I was informed, however, that
the phones displayed the wrong time for the complete duration of sunday,
they just went correct on the day change to monday.

CCM is 5.1.3.1000-12.

Anyone with similar problems? Fixes?

TIA,
Andre.
-- 
    .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From MillerS at DicksteinShapiro.COM  Mon Oct 29 08:24:57 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 29 Oct 2007 08:24:57 -0400
Subject: [cisco-voip] DST change delay
References: <20071029121539.GD9131@ibh.de>
Message-ID: <421CAA1A5300B84A87D200D70998578263DFDA@DCEX2.DSMO.COM>

All of our phones had the old time as of yesterday (SUN) so I reset all
of the phones, 20 at time so that they would show the correct time.

Our 7936 and 7912 phones STILL show the incorrect time and I assume they
will all week.  Luckily, all of those models are in conference rooms and
all of the 7961s that people use at their desk are now displaying the
correct time.  If one of the 7961s is unplugged or loses connectivity
with the Call Manager, they again revert to the old time and need to be
reset.

I am looking for solutions so that I don't have to do this again in the
Spring. 


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andre Beck
Sent: Monday, October 29, 2007 8:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] DST change delay

Hi,

last weekend our local time changed from CEST back to CET. Now DST
issues are a seemingly neverending story, but it generally worked, at
least time was correct on monday morning. I was informed, however, that
the phones displayed the wrong time for the complete duration of sunday,
they just went correct on the day change to monday.

CCM is 5.1.3.1000-12.

Anyone with similar problems? Fixes?

TIA,
Andre.
-- 
    .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From cisco-voip at ibh.net  Mon Oct 29 08:43:18 2007
From: cisco-voip at ibh.net (Andre Beck)
Date: Mon, 29 Oct 2007 13:43:18 +0100
Subject: [cisco-voip] H323 dial-peers question
In-Reply-To: <000f01c819c1$8af4a310$ca1ea8c0@T42p>
References: <000f01c819c1$8af4a310$ca1ea8c0@T42p>
Message-ID: <20071029124318.GE9131@ibh.de>

Hi Nikola,

On Sun, Oct 28, 2007 at 08:20:42PM -0400, Nikola Stojsin wrote:
> 
> voice translation-rule 10
>  rule 1 /......4300/ /4900/
>  rule 2 /......4301/ /4301/
>  rule 3 /......4302/ /4302/
>  rule 4 /......4303/ /4303/
>  rule 5 /......4304/ /4304/
>  rule 6 /......4305/ /4305/
>  rule 7 /......4306/ /4306/
>  rule 8 /......4307/ /4307/
>  rule 9 /......4308/ /4308/
>  rule 10 /.*/ /4900/

Independent of everything else, you could merge rules 2..9 into a single
one, using regular subexpression backreference (ed-style):

 rule 2 /......\(430[1-8]\)/ /\1/

With your catch-all rule 10, you can also leave out rule 1. So as a result,
you end up with just

 rule 2 /......\(430[1-8]\)/ /\1/
 rule 10 /.*/ /4900/

doing the exact same thing. As there are only 15 rules available,
compacting them as much as possible is a relevant art, just hindered
by the regex model used here not implementing | slices.

BTW, you should think about anchoring your regexps. It may not seem
necessary now, but it makes things clearer and less error prone over
time, when the environment changes unexpectedly.

BTW^2, called party number translations are highly sensitive to
overlap receiving situations. Seems you don't have to deal with one,
but if you had to, be aware that /.*/ on the LHS might terminate an
overlap receiving in progress, routing every call to 4900 instantly.
In such situations, I'm using a construct like

  rule 15 /^123456\(..*\)$/ /\1/

to force the overlap receiving to continue. Don't ask me why exactly
it works (even with extensions longer than two digits), but it does,
while \(.*\) does not. I assume it's because .* can match nothing, so
123456.* will match 123456 before any overlap receiving is invoked,
while 123456..* will require at least another digit, which allows overlap
receiving to start, and after that IOS somehow manages to complete it
correctly (on T302 expiry).

HTH,
Andre.
-- 
    .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From d4rkf1ber at gmail.com  Mon Oct 29 08:51:59 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Mon, 29 Oct 2007 07:51:59 -0500
Subject: [cisco-voip] Ughh DST Issues!
Message-ID: <a1b07ea60710290551g2a3768c7v955af853b8171bd4@mail.gmail.com>

CallManagers show the correct time and time zone, but the phones all
show the incorrect time now.  They are one hour behind, so basically
they changed over this weekend based on the old schedule and not the
new schedule which should be next weekend.

I have tried resetting the phones but that didn't do anything.

Since the CallManagers are all correct I don't know what else to do or
try at this point.

Any thoughts?   I have checked the date and time group in CallManager
it is right, the CallManagers themselves show the correct date and
time.  But the phones are off an hour.

From wsisk at cisco.com  Mon Oct 29 09:09:38 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 29 Oct 2007 08:09:38 -0500
Subject: [cisco-voip] Ughh DST Issues!
In-Reply-To: <a1b07ea60710290551g2a3768c7v955af853b8171bd4@mail.gmail.com>
References: <a1b07ea60710290551g2a3768c7v955af853b8171bd4@mail.gmail.com>
Message-ID: <4725DB92.6040308@cisco.com>

Hi Dark,

No CM version, no phone model, no phone load version. Not much we can 
say other than read the FN:
http://cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml

/Wes

Dark Fiber wrote:
> CallManagers show the correct time and time zone, but the phones all
> show the incorrect time now.  They are one hour behind, so basically
> they changed over this weekend based on the old schedule and not the
> new schedule which should be next weekend.
>
> I have tried resetting the phones but that didn't do anything.
>
> Since the CallManagers are all correct I don't know what else to do or
> try at this point.
>
> Any thoughts?   I have checked the date and time group in CallManager
> it is right, the CallManagers themselves show the correct date and
> time.  But the phones are off an hour.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Mon Oct 29 09:10:52 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 29 Oct 2007 08:10:52 -0500
Subject: [cisco-voip] DST change delay
In-Reply-To: <20071029121539.GD9131@ibh.de>
References: <20071029121539.GD9131@ibh.de>
Message-ID: <4725DBDC.7030807@cisco.com>

need to know what phone model and load you are using. Have you updated 
per the FN?
http://cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml

/Wes

Andre Beck wrote:
> Hi,
>
> last weekend our local time changed from CEST back to CET. Now DST
> issues are a seemingly neverending story, but it generally worked, at
> least time was correct on monday morning. I was informed, however, that
> the phones displayed the wrong time for the complete duration of sunday,
> they just went correct on the day change to monday.
>
> CCM is 5.1.3.1000-12.
>
> Anyone with similar problems? Fixes?
>
> TIA,
> Andre.
>   

From wsisk at cisco.com  Mon Oct 29 09:12:06 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 29 Oct 2007 08:12:06 -0500
Subject: [cisco-voip] DST change delay
In-Reply-To: <421CAA1A5300B84A87D200D70998578263DFDA@DCEX2.DSMO.COM>
References: <20071029121539.GD9131@ibh.de>
	<421CAA1A5300B84A87D200D70998578263DFDA@DCEX2.DSMO.COM>
Message-ID: <4725DC26.2050707@cisco.com>

Hi Steve,

Getting closer.  We would still need to know the CM version and load 
(version) for each phone involved. Did you follow the FN?
http://cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml

/Wes

Miller, Steve wrote:
> All of our phones had the old time as of yesterday (SUN) so I reset all
> of the phones, 20 at time so that they would show the correct time.
>
> Our 7936 and 7912 phones STILL show the incorrect time and I assume they
> will all week.  Luckily, all of those models are in conference rooms and
> all of the 7961s that people use at their desk are now displaying the
> correct time.  If one of the 7961s is unplugged or loses connectivity
> with the Call Manager, they again revert to the old time and need to be
> reset.
>
> I am looking for solutions so that I don't have to do this again in the
> Spring. 
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> MillerS at dicksteinshapiro.com 
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andre Beck
> Sent: Monday, October 29, 2007 8:16 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] DST change delay
>
> Hi,
>
> last weekend our local time changed from CEST back to CET. Now DST
> issues are a seemingly neverending story, but it generally worked, at
> least time was correct on monday morning. I was informed, however, that
> the phones displayed the wrong time for the complete duration of sunday,
> they just went correct on the day change to monday.
>
> CCM is 5.1.3.1000-12.
>
> Anyone with similar problems? Fixes?
>
> TIA,
> Andre.
>   

From d4rkf1ber at gmail.com  Mon Oct 29 09:33:33 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Mon, 29 Oct 2007 08:33:33 -0500
Subject: [cisco-voip] Ughh DST Issues!
In-Reply-To: <4725DB92.6040308@cisco.com>
References: <a1b07ea60710290551g2a3768c7v955af853b8171bd4@mail.gmail.com>
	<4725DB92.6040308@cisco.com>
Message-ID: <a1b07ea60710290633l6207c55exafb5a183acb6cf50@mail.gmail.com>

Awfully witty and insightful there.  Happy monday morning to you to. :-P

As per the FN:

http://cisco.com/en/US/products/hw/phones/ps379/products_field_notice09186a00807d3d9e.shtml

"Cisco Unified IP Phone models 7910G, 7910G+SW, 7940G and 7960G (SCCP)
 Time information comes from Cisco Unified CallManager / CallManager
Express. Update CallManager or CallManager Express to correct the new
DST dates."


Well like I said the patches were applied to the CallManagers and they
are obviously working as they showed the correct time and everything.
And based on the FN 7940s and 7960s get there time from CallManagers
obviously.  So it doesn't indicate you need to do anything on the
phones right?

CallManager 4.1(3)sr4d

All phones are 7960s running P00308000400 which as I understand is an SCCP load.

So CallManagers are patched and working fine and show correct time.
Phones are 7960s with an SCCP load, and based on the FN they should
just get time from CallManagers.


Any thoughts as to what I am missing or overlooking here?





On 10/29/07, Wes Sisk <wsisk at cisco.com> wrote:
> Hi Dark,
>
> No CM version, no phone model, no phone load version. Not much we can
> say other than read the FN:
> http://cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml
>
> /Wes
>
> Dark Fiber wrote:
> > CallManagers show the correct time and time zone, but the phones all
> > show the incorrect time now.  They are one hour behind, so basically
> > they changed over this weekend based on the old schedule and not the
> > new schedule which should be next weekend.
> >
> > I have tried resetting the phones but that didn't do anything.
> >
> > Since the CallManagers are all correct I don't know what else to do or
> > try at this point.
> >
> > Any thoughts?   I have checked the date and time group in CallManager
> > it is right, the CallManagers themselves show the correct date and
> > time.  But the phones are off an hour.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From cisco-voip at ibh.net  Mon Oct 29 10:07:11 2007
From: cisco-voip at ibh.net (Andre Beck)
Date: Mon, 29 Oct 2007 15:07:11 +0100
Subject: [cisco-voip] DST change delay
In-Reply-To: <4725DBDC.7030807@cisco.com>
References: <20071029121539.GD9131@ibh.de> <4725DBDC.7030807@cisco.com>
Message-ID: <20071029140710.GF9131@ibh.de>

Hi Wes,

On Mon, Oct 29, 2007 at 08:10:52AM -0500, Wes Sisk wrote:
> need to know what phone model and load you are using.

Models: Various. AFAIK the reports came from 7960 users, but this doesn't
mean others weren't affected. It was just sunday and the effect was only
observed by some and by chance.

Loads in use here are:

7912	CP7912080002SCCP060817A
7940/60	P00308000700
7970	SCCP70.8-3-2S

There is another report that the time fixed itself after the phone had
a call. I remember that there have been issues in the past with phone's
clocks drifting away several minutes when the phone was idle, which
would correct itself by simply going offhook for a moment. Maybe that
is the same thing, phones only syncing time every now and then, thus
not noticing such a major change as a DST jump.

> Have you updated per the FN?

Updated what?

> http://cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml

I don't think this applies to me. CET DST rules have been stable for
a decade now (the last change I remember was in the Win95 era), even
the antique RHEL 3 underneath CCM5 has the correct timezone data for
CET I'm sure. I don't expect any DST issues after such a long time for
the solutions to mature - US and especially NZ changed their rules
somewhat suddenly which is another issue of course.

BTW, I also remember similar reports from CCME users in the spring, which
somehow corrected themselves before I could investigate. It's 7940/60s
there as well. Maybe it's all the same problem with these phones not
syncing their time when idle?
 
> Andre Beck wrote:
> >
> >last weekend our local time changed from CEST back to CET. Now DST
> >issues are a seemingly neverending story, but it generally worked, at
> >least time was correct on monday morning. I was informed, however, that
> >the phones displayed the wrong time for the complete duration of sunday,
> >they just went correct on the day change to monday.
> >
> >CCM is 5.1.3.1000-12.

TIA,
Andre.
-- 
    .sig making fun of Santa Claus Operation currently unavailable

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From tim.reimers at asheville.k12.nc.us  Mon Oct 29 10:13:45 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon, 29 Oct 2007 10:13:45 -0400
Subject: [cisco-voip] H.323 ISDN question..
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1AFC4@webmail.asheville.k12.nc.us>

 

Here's a possibly OT question, but perhaps not OT.

 

We run CCM 4.01 and have a 2651 terminating two ISDN PRIs in H.323 mode
(not MGCP)

The 2651 runs c2600-ipvoice-mz.123-8.T3

 

Here's the question-

 

We have some Sony PCS-11 and a Tandberg video conferencing unit.

We normally use video conferencing sites that support IP connectivity,
but I'm now hearing of sites that staff want to conference with which
are ISDN only.

 

Is it possible to somehow register the Sony/Tandberg units as H.323
clients on the phone system and have them use the ISDN PRI channels for
making video-calls ??

 

 

It would seem like perhaps with the CCM and the ISDN we already own that
we possibly already have a lot of the pieces necessary to allow our
video conferencing units to use ISDN when needed, via some sort of
connectivity through the CCM/2651 H.323 gateway??

 

 

Or can I/should I think in terms of them using the 2651 as an MCU
directly and not involve the CCM at all??

 

Or am I talking about two completely different usages of the ISDN PRI
and I can't use the PRI for both CCM voice calls AND for video
conferencing?

 

Not being familiar with how videoconferencing over ISDN works for sure,
I'm not certain if ISDN is ISDN, or whether there's some different setup
going on for VOIP versus video calls..

 

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From erickbee at gmail.com  Mon Oct 29 10:38:14 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Mon, 29 Oct 2007 09:38:14 -0500
Subject: [cisco-voip] 8.2.1S phoneload and DST
Message-ID: <f4445faf0710290738h559c7a8hf6384eb58ee709b@mail.gmail.com>

Does the newer 8.3.3 phone load fix the issue where the phone needs to
be reset manually for the new time to take effect? That is what the FN
states still. Had a bunch of 3rd gen phones be a hour off this morning
until they were manually reset.

From keli at carocomp.ro  Mon Oct 29 10:42:13 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Mon, 29 Oct 2007 16:42:13 +0200
Subject: [cisco-voip] H.323 ISDN question..
In-Reply-To: <050944EE48FF384F9C663F7048C47DA702E1AFC4@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA702E1AFC4@webmail.asheville.k12.nc.us>
Message-ID: <4725F145.2050602@carocomp.ro>

I'm not quite sure about Tandberg, but I know that Sony units can call 
either point-to-point H.323, or they need a Gatekeeper.

With a GK, on the other hand, you might be able to integrate the whole 
thing into the exisitng phone network, especially, if you use your 
gateways with H.323 as well.

The details, however, are unfortunately beyond my knowledge :)

regards,
Zoltan

Tim Reimers wrote:
>
> Here?s a possibly OT question, but perhaps not OT.
>
> We run CCM 4.01 and have a 2651 terminating two ISDN PRIs in H.323 
> mode (not MGCP)
>
> The 2651 runs c2600-ipvoice-mz.123-8.T3
>
> Here?s the question-
>
> We have some Sony PCS-11 and a Tandberg video conferencing unit.
>
> We normally use video conferencing sites that support IP connectivity, 
> but I?m now hearing of sites that staff want to conference with which 
> are ISDN only.
>
> Is it possible to somehow register the Sony/Tandberg units as H.323 
> clients on the phone system and have them use the ISDN PRI channels 
> for making video-calls ??
>
> It would seem like perhaps with the CCM and the ISDN we already own 
> that we possibly already have a lot of the pieces necessary to allow 
> our video conferencing units to use ISDN when needed, via some sort of 
> connectivity through the CCM/2651 H.323 gateway??
>
> Or can I/should I think in terms of them using the 2651 as an MCU 
> directly and not involve the CCM at all??
>
> Or am I talking about two completely different usages of the ISDN PRI 
> and I can?t use the PRI for both CCM voice calls AND for video 
> conferencing?
>
> Not being familiar with how videoconferencing over ISDN works for 
> sure, I?m not certain if ISDN is ISDN, or whether there?s some 
> different setup going on for VOIP versus video calls..
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From Andrius-conf at elsis.lt  Mon Oct 29 11:01:35 2007
From: Andrius-conf at elsis.lt (Andrius Kislas)
Date: Mon, 29 Oct 2007 17:01:35 +0200
Subject: [cisco-voip] H.323 ISDN question..
In-Reply-To: <050944EE48FF384F9C663F7048C47DA702E1AFC4@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA702E1AFC4@webmail.asheville.k12.nc.us>
Message-ID: <4725F5CF.6090208@elsis.lt>

Correct me if I am wrong, but I think you need dedicated h323-to-h320
(like 3527 MCU) gateway if you want to connect H323 conferencing devices
to ISDN conferencing devices (that work using h320).

Andrius


Tim Reimers wrote:
>  
> 
> Here?s a possibly OT question, but perhaps not OT.
> 
>  
> 
> We run CCM 4.01 and have a 2651 terminating two ISDN PRIs in H.323 mode
> (not MGCP)
> 
> The 2651 runs c2600-ipvoice-mz.123-8.T3
> 
>  
> 
> Here?s the question-
> 
>  
> 
> We have some Sony PCS-11 and a Tandberg video conferencing unit.
> 
> We normally use video conferencing sites that support IP connectivity,
> but I?m now hearing of sites that staff want to conference with which
> are ISDN only.
> 
>  
> 
> Is it possible to somehow register the Sony/Tandberg units as H.323
> clients on the phone system and have them use the ISDN PRI channels for
> making video-calls ??
> 
>  
> 
>  
> 
> It would seem like perhaps with the CCM and the ISDN we already own that
>  we possibly already have a lot of the pieces necessary to allow our
> video conferencing units to use ISDN when needed, via some sort of
> connectivity through the CCM/2651 H.323 gateway??
> 
>  
> 
>  
> 
> Or can I/should I think in terms of them using the 2651 as an MCU
> directly and not involve the CCM at all??
> 
>  
> 
> Or am I talking about two completely different usages of the ISDN PRI
> and I can?t use the PRI for both CCM voice calls AND for video conferencing?
> 
>  
> 
> Not being familiar with how videoconferencing over ISDN works for sure,
> I?m not certain if ISDN is ISDN, or whether there?s some different setup
> going on for VOIP versus video calls..
> 
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From rkulagow at gmail.com  Mon Oct 29 11:15:09 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Mon, 29 Oct 2007 10:15:09 -0500
Subject: [cisco-voip] 8.2.1S phoneload and DST
In-Reply-To: <f4445faf0710290738h559c7a8hf6384eb58ee709b@mail.gmail.com>
References: <f4445faf0710290738h559c7a8hf6384eb58ee709b@mail.gmail.com>
Message-ID: <4725F8FD.2080408@gmail.com>

Erick Bergquist wrote:
> Does the newer 8.3.3 phone load fix the issue where the phone needs to
> be reset manually for the new time to take effect? That is what the FN
> states still. Had a bunch of 3rd gen phones be a hour off this morning
> until they were manually reset.

According to the release notes, this was fixed in 8.2.2 and higher when 
they included the new Java:

"
Include Sun Java patch to address 2007 Daylight Savings Time (DST) changes

http://www.cisco.com/pcgi-bin/Support/Bugtool/onebug.pl?bugid=CSCse47164
"

We deployed 8.2.2SR4 this weekend to fix that issue.  We didn't go to 
8.3.x because of the "Intercom History" default.  (Opened a TAC case 
btw, so hopefully the counter against bug CSCsk99117 is increasing fast 
enough that Something Is Done about it.)



From rkulagow at gmail.com  Mon Oct 29 11:26:25 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Mon, 29 Oct 2007 10:26:25 -0500
Subject: [cisco-voip] 8.2.1S phoneload and DST
In-Reply-To: <f4445faf0710290738h559c7a8hf6384eb58ee709b@mail.gmail.com>
References: <f4445faf0710290738h559c7a8hf6384eb58ee709b@mail.gmail.com>
Message-ID: <4725FBA1.3070104@gmail.com>

[Sent on behalf of Katherine, who's having posting issues]

All this DST talk is making me nervous for next weekend.  All my phones
have the correct time this morning.  I have 7941s and 7961s and a few
7911s.  I have Call Manager 4.1(3)sr3c and the load on the phones is
SCCP41.8-2-1ES5S for the 7941s and 61s and for the 7911s its
SCCP11.8-2-1ES5S.  Am I going to be alright next Sunday?  Field Notes
confuse me, sorry.  We did all the updates in the Spring before the time
change.  Do we need to do something else?

Thanks, Kathy

Katherine O. Dupree
Randolph College
Main 243
Telecommunications Support/Infrastructure Coordinator
kdupree at randolphcollege.edu
(434) 947 8371

From jvw at medicineforthedefense.com  Mon Oct 29 11:33:22 2007
From: jvw at medicineforthedefense.com (Weigand, John V.)
Date: Mon, 29 Oct 2007 11:33:22 -0400
Subject: [cisco-voip] FXS/FXO - Fax Lines?
Message-ID: <1E196438D2BA76448E5A077C8C99329D065B340C@lmiexmb01.corp.medicineforthedefense.com>

Serious Medicine for the Defense R 

This may seem like a stupid question, but I'm having trouble grasping
the difference between FXS and FXO ports. Currently, we have all our
POTS fax lines going out through FXS ports on our Callmanager gateways,
which is the way our VAR set it up. Can we setup those FXO ports as POTS
lines and attach fax machines to those, or will that not work?

 

Thanks in advance.


John V. Weigand
Help Desk Support

Litigation Management, Inc.
300 Allen-Bradley Drive
Suite 200
Mayfield Heights, OH 44124





Tel: 
Fax: 440-484-2020
Cell: 
email: jvw at medicineforthedefense.com 




PRIVILEGE AND CONFIDENTIALITY NOTICE 

The information in this electronic mail is intended for the named recipients only. It may contain privileged and confidential material and may be protected under law by the Health Insurance Portability and Accountability Act. Any use of this information by anyone other than the intended receiver is prohibited. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, copying, or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately by replying to this electronic e-mail or by calling (800) 778-5424. Please delete it from your computer. Thank you.
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From tim.reimers at asheville.k12.nc.us  Mon Oct 29 11:38:11 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon, 29 Oct 2007 11:38:11 -0400
Subject: [cisco-voip] H.323 ISDN question..
In-Reply-To: <4725F145.2050602@carocomp.ro>
References: <050944EE48FF384F9C663F7048C47DA702E1AFC4@webmail.asheville.k12.nc.us>
	<4725F145.2050602@carocomp.ro>
Message-ID: <050944EE48FF384F9C663F7048C47DA707ED2960@webmail.asheville.k12.nc.us>

That's kinda my thought--- that perhaps all I need to do is configure
the 2651 to be a gatekeeper as well as perhaps be able to act as a
gateway.

I have extra DIDs I could assign to the videoconference units.

What would be _really_ cool would be to have them have VOIP extensions
and be able to use Softphone/Ekiga clients with them, etc...

-----Original Message-----
From: Kelemen Zoltan [mailto:keli at carocomp.ro] 
Sent: Monday, October 29, 2007 10:42 AM
To: Tim Reimers
Cc: Cisco Voip Mailing list
Subject: Re: [cisco-voip] H.323 ISDN question..

I'm not quite sure about Tandberg, but I know that Sony units can call 
either point-to-point H.323, or they need a Gatekeeper.

With a GK, on the other hand, you might be able to integrate the whole 
thing into the exisitng phone network, especially, if you use your 
gateways with H.323 as well.

The details, however, are unfortunately beyond my knowledge :)

regards,
Zoltan

Tim Reimers wrote:
>
> Here's a possibly OT question, but perhaps not OT.
>
> We run CCM 4.01 and have a 2651 terminating two ISDN PRIs in H.323 
> mode (not MGCP)
>
> The 2651 runs c2600-ipvoice-mz.123-8.T3
>
> Here's the question-
>
> We have some Sony PCS-11 and a Tandberg video conferencing unit.
>
> We normally use video conferencing sites that support IP connectivity,

> but I'm now hearing of sites that staff want to conference with which 
> are ISDN only.
>
> Is it possible to somehow register the Sony/Tandberg units as H.323 
> clients on the phone system and have them use the ISDN PRI channels 
> for making video-calls ??
>
> It would seem like perhaps with the CCM and the ISDN we already own 
> that we possibly already have a lot of the pieces necessary to allow 
> our video conferencing units to use ISDN when needed, via some sort of

> connectivity through the CCM/2651 H.323 gateway??
>
> Or can I/should I think in terms of them using the 2651 as an MCU 
> directly and not involve the CCM at all??
>
> Or am I talking about two completely different usages of the ISDN PRI 
> and I can't use the PRI for both CCM voice calls AND for video 
> conferencing?
>
> Not being familiar with how videoconferencing over ISDN works for 
> sure, I'm not certain if ISDN is ISDN, or whether there's some 
> different setup going on for VOIP versus video calls..
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From matt at saskin.net  Mon Oct 29 11:43:23 2007
From: matt at saskin.net (Matthew Saskin)
Date: Mon, 29 Oct 2007 11:43:23 -0400
Subject: [cisco-voip] FXS/FXO - Fax Lines?
In-Reply-To: <1E196438D2BA76448E5A077C8C99329D065B340C@lmiexmb01.corp.medicineforthedefense.com>
References: <1E196438D2BA76448E5A077C8C99329D065B340C@lmiexmb01.corp.medicineforthedefense.com>
Message-ID: <4725FF9B.4060608@saskin.net>

At the simplest:
FXS provides battery current and ring voltage
FXO receives battery current and ring voltage.

They are not interchangeable - FXO's will be used when terminating an 
analog log provided by a PABX, from a telco, etc.

FXS ports are necessary when you need to generate dial-tone/ring, i.e.; 
for analog phones, fax machines, etc.

-matt

Weigand, John V. wrote:
> 	
> 
> Litigation Management
> 
> 	
> 
> Serious Medicine for the Defense ? <http://www.medicineforthedefense.com/>
> 
> 	
> 	
> 
> This may seem like a stupid question, but I?m having trouble grasping 
> the difference between FXS and FXO ports. Currently, we have all our 
> POTS fax lines going out through FXS ports on our Callmanager gateways, 
> which is the way our VAR set it up. Can we setup those FXO ports as POTS 
> lines and attach fax machines to those, or will that not work?
> 
>  
> 
> Thanks in advance.
> 
> 
> 	*John V. Weigand*
> Help Desk Support
> 
> Litigation Management, Inc.
> 300 Allen-Bradley Drive
> Suite 200
> Mayfield Heights, OH 44124
> 
> 	
> 
> 
> 
> 
> Tel:
> Fax: 440-484-2020
> Cell:
> email: <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com> 
> <mailto:jvw at medicineforthedefense.com>jvw at medicineforthedefense.com 
> <mailto:jvw at medicineforthedefense.com>
> 
> 
> PRIVILEGE AND CONFIDENTIALITY NOTICE
> 
> The information in this electronic mail is intended for the named 
> recipients only. It may contain privileged and confidential material and 
> may be protected under law by the Health Insurance Portability and 
> Accountability Act. Any use of this information by anyone other than the 
> intended receiver is prohibited. If the reader of this message is not 
> the intended recipient, you are hereby notified that any dissemination, 
> distribution, copying, or other use of this message or its attachments 
> is strictly prohibited. If you have received this message in error, 
> please notify the sender immediately by replying to this electronic 
> e-mail or by calling (800) 778-5424. Please delete it from your 
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From tim.reimers at asheville.k12.nc.us  Mon Oct 29 11:46:30 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon, 29 Oct 2007 11:46:30 -0400
Subject: [cisco-voip] FXS/FXO - Fax Lines?
In-Reply-To: <1E196438D2BA76448E5A077C8C99329D065B340C@lmiexmb01.corp.medicineforthedefense.com>
References: <1E196438D2BA76448E5A077C8C99329D065B340C@lmiexmb01.corp.medicineforthedefense.com>
Message-ID: <050944EE48FF384F9C663F7048C47DA707ED2961@webmail.asheville.k12.nc.us>

Here's a link that's fairly good with pictures even (which I find handy
to really "get" things sometimes..)

 

http://www.nch.com.au/kb/10050.html

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Weigand, John
V.
Sent: Monday, October 29, 2007 11:33 AM
To: cisco voip list
Subject: [cisco-voip] FXS/FXO - Fax Lines?

 

 

 Litigation
Management<http://www.medicineforthedefense.com/main/gifs/LMI-side-pms69
0wBlack.jpg> 

Serious Medicine for the Defense (r)
<http://www.medicineforthedefense.com/>  

 

 

This may seem like a stupid question, but I'm having trouble grasping
the difference between FXS and FXO ports. Currently, we have all our
POTS fax lines going out through FXS ports on our Callmanager gateways,
which is the way our VAR set it up. Can we setup those FXO ports as POTS
lines and attach fax machines to those, or will that not work?

 

Thanks in advance.

 

 

John V. Weigand
Help Desk Support

Litigation Management, Inc.
300 Allen-Bradley Drive 
Suite 200
Mayfield Heights, OH 44124




Tel: 
Fax: 440-484-2020
Cell: 
email: <mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com>
<mailto:jvw at medicineforthedefense.com> jvw at medicineforthedefense.com 

 


PRIVILEGE AND CONFIDENTIALITY NOTICE 

The information in this electronic mail is intended for the named
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From rkulagow at gmail.com  Mon Oct 29 11:57:05 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Mon, 29 Oct 2007 10:57:05 -0500
Subject: [cisco-voip] H.323 ISDN question..
In-Reply-To: <4725F5CF.6090208@elsis.lt>
References: <050944EE48FF384F9C663F7048C47DA702E1AFC4@webmail.asheville.k12.nc.us>
	<4725F5CF.6090208@elsis.lt>
Message-ID: <472602D1.7090702@gmail.com>

Andrius Kislas wrote:
> Correct me if I am wrong, but I think you need dedicated h323-to-h320
> (like 3527 MCU) gateway if you want to connect H323 conferencing devices
> to ISDN conferencing devices (that work using h320).

Here's my findings so far on this:

http://www.gossamer-threads.com/lists/cisco/voip/64916?search_string=128K;#64916

From vmemaillist at gmail.com  Mon Oct 29 12:04:32 2007
From: vmemaillist at gmail.com (VM)
Date: Mon, 29 Oct 2007 12:04:32 -0400
Subject: [cisco-voip] FXS/FXO - Fax Lines?
In-Reply-To: <4725FF9B.4060608@saskin.net>
References: <1E196438D2BA76448E5A077C8C99329D065B340C@lmiexmb01.corp.medicineforthedefense.com>
	<4725FF9B.4060608@saskin.net>
Message-ID: <A799D831-5467-47F1-B008-D4FA961749ED@gmail.com>

POTS = Plain Old Telephone Service
FXO = Foreign Exchange Office (i.e. TELCO Switch)
FXS = Foreign Exchange Station (i.e. analog telephone)

For some reason, Cisco identifies these hardware components by the  
devices that are intended to be connected to it such that FXS devices  
(analog phones, modems, faxes, etc.) connect to the FXS cards and the  
FXO cards connect to analog lines being provided with line current  
and voltage.  However, the FXO card is actually a FXS device (acts  
line a station expecting line current and voltage) and the FXS card  
is actually a FXO device (acts like a Telco switch providing line  
current and voltage).

Connecting FXS devices to Cisco FXO cards will not work.

On Oct 29, 2007, at 11:43 AM, Matthew Saskin wrote:

> At the simplest:
> FXS provides battery current and ring voltage
> FXO receives battery current and ring voltage.
>
> They are not interchangeable - FXO's will be used when terminating an
> analog log provided by a PABX, from a telco, etc.
>
> FXS ports are necessary when you need to generate dial-tone/ring,  
> i.e.;
> for analog phones, fax machines, etc.
>
> -matt
>
> Weigand, John V. wrote:
>> 	
>>
>> Litigation Management
>>
>> 	
>>
>> Serious Medicine for the Defense ? <http:// 
>> www.medicineforthedefense.com/>
>>
>> 	
>> 	
>>
>> This may seem like a stupid question, but I?m having trouble grasping
>> the difference between FXS and FXO ports. Currently, we have all our
>> POTS fax lines going out through FXS ports on our Callmanager  
>> gateways,
>> which is the way our VAR set it up. Can we setup those FXO ports  
>> as POTS
>> lines and attach fax machines to those, or will that not work?
>>
>>
>>
>> Thanks in advance.
>>
>>
>> 	*John V. Weigand*
>> Help Desk Support
>>
>> Litigation Management, Inc.
>> 300 Allen-Bradley Drive
>> Suite 200
>> Mayfield Heights, OH 44124
>>
>> 	
>>
>>
>>
>>
>> Tel:
>> Fax: 440-484-2020
>> Cell:
>> email: <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>
>> <mailto:jvw at medicineforthedefense.com>jvw at medicineforthedefense.com
>> <mailto:jvw at medicineforthedefense.com>
>>
>>
>> PRIVILEGE AND CONFIDENTIALITY NOTICE
>>
>> The information in this electronic mail is intended for the named
>> recipients only. It may contain privileged and confidential  
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>> may be protected under law by the Health Insurance Portability and
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From KDupree at randolphcollege.edu  Mon Oct 29 10:46:42 2007
From: KDupree at randolphcollege.edu (Kathy Dupree)
Date: Mon, 29 Oct 2007 10:46:42 -0400
Subject: [cisco-voip] DST
Message-ID: <6E497ADB607656479C24E6D7BF6B505A09B941E5@exchange.randolphcollege.edu>

All this DST talk is making me nervous for next weekend.  All my phones
have the correct time this morning.  I have 7941s and 7961s and a few
7911s.  I have Call Manager 4.1(3)sr3c and the load on the phones is
SCCP41.8-2-1ES5S for the 7941s and 61s and for the 7911s its
SCCP11.8-2-1ES5S.  Am I going to be alright next Sunday?  Field Notes
confuse me, sorry.  We did all the updates in the Spring before the time
change.  Do we need to do something else?

 

Thanks, Kathy

 

Katherine O. Dupree

Randolph College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at randolphcollege.edu <mailto:kdupree at rmwc.edu> 

(434) 947 8371

 

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From lakpriya at gmail.com  Mon Oct 29 12:19:55 2007
From: lakpriya at gmail.com (Ravindra Lakpriya)
Date: Mon, 29 Oct 2007 21:49:55 +0530
Subject: [cisco-voip] hello guys need some help on LDAP intergration
Message-ID: <4535c2ca0710290919m5431728eic078efa6dd7cfbbe@mail.gmail.com>

hello
Guys i have download the SDK 3.3 and i have configure ldapsearch.asp file.
then i have add ip phone service directing that URL. as a result of that i
got the service in my phone. and when i'm browsing that service all i can
see is a cording. can you help me out. thanking you in advance,

Ravindra
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From rkulagow at gmail.com  Mon Oct 29 12:37:21 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Mon, 29 Oct 2007 11:37:21 -0500
Subject: [cisco-voip] Help for 7970 "Product Specific Information" is
	incomplete on 4.1.3SR5d
Message-ID: <47260C41.8050106@gmail.com>

The "Log Server" part of it says
..."the format is" and then nothing.

What's the format?

Also, here's what I'm trying to accomplish with all this AUX port 
questions.  I have a site where the users say they have "phantom" 
phones.  Phones autodial from the directory, mysteriously won't answer, etc.

If I look at the CM logs, I'll see the call coming in from the phone, 
but that doesn't tell me why the _phone_ thinks that it needed to make a 
call, such as went off hook, user press {foo}, or whatever.

And it's more than one device, which makes me think it's not faulty hook 
switches.

From Erik.Stillman at Shearman.com  Mon Oct 29 13:01:53 2007
From: Erik.Stillman at Shearman.com (Erik Stillman)
Date: Mon, 29 Oct 2007 13:01:53 -0400
Subject: [cisco-voip] PRI issue - down on inbound calls now
In-Reply-To: <NYEXM44Fu2gJ1aipOrQ0001d2e6@NYEXMB2.shearman.com>
Message-ID: <643A3A543DB5224499BC3D73473AEC2792226B@NYEXMB2.shearman.com>

 
Has anyone seen this in the q.931 debugs?

 Facility i = 0x91A11502020


It keeps showing up with every inbound failure?

-Erik


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net@SHSDOMAIN On Behalf Of
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Sent: Monday, October 29, 2007 11:47 AM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 56, Issue 172

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From SCASPER at mtb.com  Mon Oct 29 13:22:36 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Mon, 29 Oct 2007 13:22:36 -0400
Subject: [cisco-voip]  CCMPPID
In-Reply-To: <20071029124318.GE9131@ibh.de>
References: <000f01c819c1$8af4a310$ca1ea8c0@T42p>
	<20071029124318.GE9131@ibh.de>
Message-ID: <4725DE97.9874.0039.0@mtb.com>


 Any info out there on how the Peer Firmware Sharing feature works? I have seen the readme file and a couple of brief descriptions in phone admin docs but I was wondering if there was anything else with some more detail. 

Thanks! 
Steve

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From jason.aarons at us.didata.com  Mon Oct 29 14:07:41 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 29 Oct 2007 14:07:41 -0400
Subject: [cisco-voip] Called Party Number is blank....
Message-ID: <C1FE15183DA37645BC0633BC604E44F007402D68@USNAEXCH.na.didata.local>


Fax 954-851-1946
Inbound H323 PSTN 2811 172.16.10.10

Outbound H323 Castelle 3825 172.16.315

Fax Press doesn't get that call was for 1946 due to Called Party Number
is blank.....but how did IOS know to route 1946 to fax press port
properly?


089849: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: Applying typeplan
for sw-type 0x3 is 0x2 0x1, Calling num 9043383245
089850: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: Applying typeplan
for sw-type 0x3 is 0x4 0x1, Called num 
089851: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: TX -> SETUP pd =
8  callref = 0x0087 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transfer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98381 
                Exclusive, Channel 1 
        Calling Party Number i = 0x2183, '9043383245' 
                Plan:ISDN, Type:National 
        Called Party Number i = 0xC1 
                Plan:ISDN, Type:Subscriber(local)
089852: Oct 29 14:03:42.852 EDT: ISDN Se0/3/0:23 Q931: RX <- CALL_PROC
pd = 8  callref = 0x8087 
        Channel ID i = 0xA98381


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From craig at staffin.org  Mon Oct 29 14:33:33 2007
From: craig at staffin.org (Craig Staffin)
Date: Mon, 29 Oct 2007 13:33:33 -0500
Subject: [cisco-voip] Called Party Number is blank....
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F007402D68@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F007402D68@USNAEXCH.na.didata.local>
Message-ID: <9b76d88c0710291133w61fb5d65r789ce00fbf48296c@mail.gmail.com>

Are you useing H323 dial-peers?

Craig

On 10/29/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
> Fax 954-851-1946
> Inbound H323 PSTN 2811 172.16.10.10
>
> Outbound H323 Castelle 3825 172.16.315
>
> Fax Press doesn't get that call was for 1946 due to Called Party Number
> is blank.....but how did IOS know to route 1946 to fax press port
> properly?
>
>
> 089849: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: Applying typeplan
> for sw-type 0x3 is 0x2 0x1, Calling num 9043383245
> 089850: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: Applying typeplan
> for sw-type 0x3 is 0x4 0x1, Called num
> 089851: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: TX -> SETUP pd =
> 8  callref = 0x0087
>         Bearer Capability i = 0x8090A2
>                 Standard = CCITT
>                 Transfer Capability = Speech
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA98381
>                 Exclusive, Channel 1
>         Calling Party Number i = 0x2183, '9043383245'
>                 Plan:ISDN, Type:National
>         Called Party Number i = 0xC1
>                 Plan:ISDN, Type:Subscriber(local)
> 089852: Oct 29 14:03:42.852 EDT: ISDN Se0/3/0:23 Q931: RX <- CALL_PROC
> pd = 8  callref = 0x8087
>         Channel ID i = 0xA98381
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
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-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003
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From jason.aarons at us.didata.com  Mon Oct 29 14:38:36 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 29 Oct 2007 14:38:36 -0400
Subject: [cisco-voip] Called Party Number is blank....
In-Reply-To: <9b76d88c0710291133w61fb5d65r789ce00fbf48296c@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F007402D68@USNAEXCH.na.didata.local>
	<9b76d88c0710291133w61fb5d65r789ce00fbf48296c@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F007402E50@USNAEXCH.na.didata.local>

H323 yes, I think I found it, direct-inward-dial was missing -jason

 

From: cmstaffin at gmail.com [mailto:cmstaffin at gmail.com] On Behalf Of
Craig Staffin
Sent: Monday, October 29, 2007 2:34 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Called Party Number is blank....

 

Are you useing H323 dial-peers?

Craig

On 10/29/07, Jason Aarons (US) <jason.aarons at us.didata.com > wrote:


Fax 954-851-1946
Inbound H323 PSTN 2811 172.16.10.10

Outbound H323 Castelle 3825 172.16.315

Fax Press doesn't get that call was for 1946 due to Called Party Number
is blank.....but how did IOS know to route 1946 to fax press port
properly? 


089849: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: Applying typeplan
for sw-type 0x3 is 0x2 0x1, Calling num 9043383245
089850: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: Applying typeplan
for sw-type 0x3 is 0x4 0x1, Called num 
089851: Oct 29 14:03:42.840 EDT: ISDN Se0/3/0:23 Q931: TX -> SETUP pd =
8  callref = 0x0087
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech 
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98381
                Exclusive, Channel 1
        Calling Party Number i = 0x2183, '9043383245' 
                Plan:ISDN, Type:National
        Called Party Number i = 0xC1
                Plan:ISDN, Type:Subscriber(local)
089852: Oct 29 14:03:42.852 EDT: ISDN Se0/3/0:23 Q931: RX <- CALL_PROC
pd = 8  callref = 0x8087 
        Channel ID i = 0xA98381


-----------------------------------------
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This e-mail communication and any attachments may contain
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unlawful.  If you have received this communication in error, please
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https://puck.nether.net/mailman/listinfo/cisco-voip 




-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003 




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From jgrace at adcapnet.com  Mon Oct 29 16:00:28 2007
From: jgrace at adcapnet.com (James Grace)
Date: Mon, 29 Oct 2007 16:00:28 -0400
Subject: [cisco-voip] cm 6.0 personal directory issues
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E8F0@mail3.adcapnetsvc.local>

?
I'm trying to setup personal directory in cm 6.0  and when I try to log into my ipphones with my user / password  I get an PD logon unsuccessful error.  im able to log into ccmuser and add the entries there.  but unable to log in on the phone to view entries
 
James Grace 
<http://www.adcapnet.com/>  
Disclaimer For ADCAP Network Systems

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From jason.aarons at us.didata.com  Mon Oct 29 16:23:23 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 29 Oct 2007 16:23:23 -0400
Subject: [cisco-voip] cm 6.0 personal directory issues
In-Reply-To: <49AFE548BC7F7E49A1A4A2F28AF1914901E8F0@mail3.adcapnetsvc.local>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E8F0@mail3.adcapnetsvc.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00745D326@USNAEXCH.na.didata.local>

Remember password is the PIN, not the "password" on the phone services.
Try resetting your User PIN in CCMAdmin.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Monday, October 29, 2007 4:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] cm 6.0 personal directory issues

 

?

I'm trying to setup personal directory in cm 6.0  and when I try to log
into my ipphones with my user / password  I get an PD logon unsuccessful
error.  im able to log into ccmuser and add the entries there.  but
unable to log in on the phone to view entries

 

James Grace 

 

Disclaimer For ADCAP Network Systems
 
"The information transmitted is intended only for the person or entity
to which it is addressed and may
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This original email was sent to the internet for delivery at 29 Oct 2007
15:59:19 -0400



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From bcarter at sentinel.com  Mon Oct 29 21:47:51 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 29 Oct 2007 20:47:51 -0500
Subject: [cisco-voip] MGCP Gateway and FAX to SCCP ATA
Message-ID: <6A0744639B86DD459433108DBF101D26019FE9DC@MAIL01.sentinel.com>

What is the recommended configuration for a MGCP gateway and an ATA to
support FAX ?  CM 5.1 and the latest SCCP version for the ATA.

From kenjohnson at letu.edu  Mon Oct 29 22:01:17 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Mon, 29 Oct 2007 21:01:17 -0500
Subject: [cisco-voip] So, what's a 7937G telephone?
In-Reply-To: <e4491a9c0710270808x35d12d72q32638bbcdf3f46d0@mail.gmail.com>
References: <47228AD0.8090405@gmail.com><1193451019.3249.0.camel@lt21223.campus.dmacc.edu>
	<e4491a9c0710270808x35d12d72q32638bbcdf3f46d0@mail.gmail.com>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EF88DAB6@rachael-reborn.letnet.net>

The 7970 jacks weren't removed - just covered under a sticker. (At least
on the 7970G - unsure about the later 797X models as I don't yet own
any). If you take off the sticker (that's really the hardest part :-),
plug the phone into the wall (it won't activate the menu options for
external jacks if you use POE) and set the mic and speaker options for
external in preferences it works fine. We do it in a number of places
both for AV-tied in conference phones, remote presentations, auditorium
tie-ins. Etc.
 
Ken Johnson
   Mgr. Network Services,
   Information Technology
   LeTourneau University
_________________________________
 E-mail: kenjohnson at letu.edu
 Helpdesk: (903) 233-3500
 Phone: (903) 233-3520
 Web: http://www.letu.edu/infotech/ <http://www.letu.edu/infotech/> 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dave Wolgast
Sent: Saturday, October 27, 2007 10:08 AM
To: Cisco VOIP Newsletter - puck.nether.net
Subject: Re: [cisco-voip] So, what's a 7937G telephone?


Just curious...is there any Cisco IP Phone that can be tied into a room
system with embedded speakers and mics(controlled by, for example, a
Polycom Vortex unit)?
 
At one point, I know one could extend a 7970G, but that potentially
caused electrical issues with the phone, so the external jacks were
removed.

 
On 10/26/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote: 

	On Fri, 2007-10-26 at 19:48 -0500, Robert Kulagowski wrote:
	> Firmware for it is posted on CCO, but can't find marketing for
it.  Some 
	> sort of conference phone I'm guessing?
	
	
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0
900aecd806e021a.html 
	
	Jeff
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	
	
	


-- 
Dave Wolgast
Livonia, NY 
________________________________


Spam <http://antispam.letu.edu/canit/b.php?i=8556976&m=d51937435c59&c=s>

Not spam
<http://antispam.letu.edu/canit/b.php?i=8556976&m=d51937435c59&c=n> 
Forget previous vote
<http://antispam.letu.edu/canit/b.php?i=8556976&m=d51937435c59&c=f> 

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From svoll.voip at gmail.com  Mon Oct 29 22:34:00 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 29 Oct 2007 19:34:00 -0700
Subject: [cisco-voip] MGCP Gateway and FAX to SCCP ATA
In-Reply-To: <6A0744639B86DD459433108DBF101D26019FE9DC@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26019FE9DC@MAIL01.sentinel.com>
Message-ID: <f84a38d30710291934r49f521bbsd1406981738825e8@mail.gmail.com>

Fax pass through

on the fax turn off ECM and Super G3

Scott

On Oct 29, 2007 6:47 PM, Carter, Bill <bcarter at sentinel.com> wrote:
> What is the recommended configuration for a MGCP gateway and an ATA to
> support FAX ?  CM 5.1 and the latest SCCP version for the ATA.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From svoll.voip at gmail.com  Mon Oct 29 22:36:58 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 29 Oct 2007 19:36:58 -0700
Subject: [cisco-voip] DST
In-Reply-To: <6E497ADB607656479C24E6D7BF6B505A09B941E5@exchange.randolphcollege.edu>
References: <6E497ADB607656479C24E6D7BF6B505A09B941E5@exchange.randolphcollege.edu>
Message-ID: <f84a38d30710291936m7afdd3d6y4f00c2013f127d6@mail.gmail.com>

if your good right now you should be good next weekend.  everyone was
upgrading to fix yesterday.

I'm down at CIPTUG and the hotel we are staying at didn't work.  the
7:am wake up happened at 8am.

Scott

On Oct 29, 2007 7:46 AM, Kathy Dupree <KDupree at randolphcollege.edu> wrote:
>
>
>
> All this DST talk is making me nervous for next weekend.  All my phones have
> the correct time this morning.  I have 7941s and 7961s and a few 7911s.  I
> have Call Manager 4.1(3)sr3c and the load on the phones is SCCP41.8-2-1ES5S
> for the 7941s and 61s and for the 7911s its SCCP11.8-2-1ES5S.  Am I going to
> be alright next Sunday?  Field Notes confuse me, sorry.  We did all the
> updates in the Spring before the time change.  Do we need to do something
> else?
>
>
>
> Thanks, Kathy
>
>
>
> Katherine O. Dupree
>
> Randolph College
>
> Main 243
>
> Telecommunications Support/Infrastructure Coordinator
>
> kdupree at randolphcollege.edu
>
> (434) 947 8371
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jgrace at adcapnet.com  Mon Oct 29 22:39:37 2007
From: jgrace at adcapnet.com (James Grace)
Date: Mon, 29 Oct 2007 22:39:37 -0400
Subject: [cisco-voip] cm 6.0 personal address book
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E8F4@mail3.adcapnetsvc.local>

?
is there a way to block a user from modifying their personal address book.  I would like to add there entries myself and give them only view rights to it.  
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
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This original email was sent to the internet for delivery at 29 Oct 2007 22:42:00 -0400
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From jason.aarons at us.didata.com  Mon Oct 29 22:42:14 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 29 Oct 2007 22:42:14 -0400
Subject: [cisco-voip] Castelle Fax Press turn off ECM and Super G3
In-Reply-To: <f84a38d30710291934r49f521bbsd1406981738825e8@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26019FE9DC@MAIL01.sentinel.com>
	<f84a38d30710291934r49f521bbsd1406981738825e8@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00745D640@USNAEXCH.na.didata.local>

I had a customer on the phone with Castelle Fax Press technical support
(they don't seem very knowledgeable of their equipment I noticed) and
had an issue escalated to Castelle development who told me they don't
have an option to turn off ECM or set the speed to G3.

Anyone else setup one of their boxes via T1 PRI back to ISR router?  I
find it hard to believe they can't turn of ECM or disable Super G3...

Also Castelle is looking at adding T.38 support down the road, seemed to
me they are way behind the times.

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From daniel.argueta at comcast.net  Tue Oct 30 11:27:01 2007
From: daniel.argueta at comcast.net (Daniel Argueta)
Date: Tue, 30 Oct 2007 08:27:01 -0700
Subject: [cisco-voip] CM4.1 to 5.x Upgrade (DMA Questions)
Message-ID: <BF9C2A278ED44A9AAF459C0D468F9FBE@main>

Does anyone have any detailed information about what the DMA actually does during the validation process? It looks like i'm being able to take a backup of all the information, but the validation is what fails when i'm trying to run this utility......it errors out with some Database errors......i'm guessing it's creating a temp database and using the schema it would use in 5.x and inserting the information into it as a 'test'? Do any of you have any information on the actual process this takes? I've been messing with this for about 2 weeks now and can't seem to successfully complete the DMA process.....except for this time i'm stuck and don't know how to fix it......

Thanks,


Daniel Argueta
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From WOCAMPO at broward.org  Tue Oct 30 10:26:05 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Tue, 30 Oct 2007 10:26:05 -0400
Subject: [cisco-voip] Cisco Unity
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA63A@UVEXVS01.bc.broward.cty>

Hi Folks,

 

Is there a direct path from Up-grading Cisco Unity from 4.0(5) to 5.0

 

Regards,

 

Walter Ocampo

System Network Analyst

Office of Information Technology

540 S.E 3rd. Ave, Suite 300

Fort Lauderdale, FL 33301

(954) 357 8041 Cell (954) 632 8228

wocampo at broward.org <mailto:wocampo at broward.org> 

 

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From wsisk at cisco.com  Tue Oct 30 11:12:53 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 30 Oct 2007 10:12:53 -0500
Subject: [cisco-voip] CM4.1 to 5.x Upgrade (DMA Questions)
In-Reply-To: <BF9C2A278ED44A9AAF459C0D468F9FBE@main>
References: <BF9C2A278ED44A9AAF459C0D468F9FBE@main>
Message-ID: <472749F5.9030802@cisco.com>

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From daniel.argueta at comcast.net  Tue Oct 30 12:30:45 2007
From: daniel.argueta at comcast.net (Daniel Argueta)
Date: Tue, 30 Oct 2007 09:30:45 -0700
Subject: [cisco-voip] CM4.1 to 5.x Upgrade (DMA Questions)
References: <BF9C2A278ED44A9AAF459C0D468F9FBE@main>
	<472749F5.9030802@cisco.com>
Message-ID: <C20C32F32B4D4256933DD5EC55DDBA83@main>

Thanks,

I actually saw the erros in the log and believe to have corrected them now.......i am giving the DMA another run....(keeping fingers crossed). Thanks for the response.

-Daniel
  ----- Original Message ----- 
  From: Wes Sisk 
  To: Daniel Argueta 
  Cc: cisco-voip at puck.nether.net 
  Sent: Tuesday, October 30, 2007 8:12 AM
  Subject: Re: [cisco-voip] CM4.1 to 5.x Upgrade (DMA Questions)


  the errors are most likely errors in your database.  in the DMA output it lists log files you need to retrive.  collect those, zip them up, and attach to your TAC case.

  what dma does:
  1. if cm ver < 5.0 or < 4.2, then export windows database
  2. import win database to ccm_w database in windows install of informix with fkconstraints.  error on any constraint failures
  3. migrate ccm_w informix database to ccm database in informix in new cm6.x schema.

  Errors can occur at both steps 2 and 3.  

  /Wes

  Daniel Argueta wrote: 
    Does anyone have any detailed information about what the DMA actually does during the validation process? It looks like i'm being able to take a backup of all the information, but the validation is what fails when i'm trying to run this utility......it errors out with some Database errors......i'm guessing it's creating a temp database and using the schema it would use in 5.x and inserting the information into it as a 'test'? Do any of you have any information on the actual process this takes? I've been messing with this for about 2 weeks now and can't seem to successfully complete the DMA process.....except for this time i'm stuck and don't know how to fix it......

    Thanks,


    Daniel Argueta
----------------------------------------------------------------------------
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From rleetun at co.boulder.co.us  Tue Oct 30 11:46:50 2007
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue, 30 Oct 2007 09:46:50 -0600
Subject: [cisco-voip] CallManager 4.x (vs.) 6.x
Message-ID: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us>

Hello all,

We are having a discussion about why is a Windows platform better than a
Linux platform when it comes to CallManager, or is Linux better than
Windows?  In your opinions, is it better to go from 4.x to 6.x or wait
for 7.x to roll out next summer?  Or is it better to stay on the 4.x
train?  How many of you have switched to the Linux platform from the
Windows platform?

Thanks in advance.

Rob
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From jason.aarons at us.didata.com  Tue Oct 30 11:50:35 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 30 Oct 2007 11:50:35 -0400
Subject: [cisco-voip] CallManager 4.x (vs.) 6.x
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E917@USNAEXCH.na.didata.local>

In Windows you have root access, which I really miss. Watching tac try to type a long SQL select statement from a command line is pretty funny as it usually takes several tries to get it right. I'd rather have log files in \var\logs then in RTMT. Trying to edit text files like RingList.xml takes more steps.
 
I was more concerned that Cisco kept shipping Windows 2000/SQL 2000 so long after Microsoft stopped supporting them and put it on extended maintenance.  Cisco is going to have a lot of old Windows 2000 boxes for years to come.  It was liking buying a new PC in 2007and getting Windows 95.
 
Nothing wrong with 5.x/6.x, they have some great features but I do wish Cisco could release the same version numbers in Linix\Windows 2003 at the same time.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Leetun, Rob
Sent: Tue 10/30/2007 11:46 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager 4.x (vs.) 6.x



Hello all,

We are having a discussion about why is a Windows platform better than a Linux platform when it comes to CallManager, or is Linux better than Windows?  In your opinions, is it better to go from 4.x to 6.x or wait for 7.x to roll out next summer?  Or is it better to stay on the 4.x train?  How many of you have switched to the Linux platform from the Windows platform?

Thanks in advance.

Rob




-----------------------------------------
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designated addressee(s) named above only.  If you are not the
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From Richard.vanBerkel at telindus.nl  Tue Oct 30 12:06:19 2007
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Tue, 30 Oct 2007 17:06:19 +0100
Subject: [cisco-voip] Unity does not connect.
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157ED98@tnl1mx1.telindus.intra>

Rudi,

 

 

We had Unity working.

We could select the voicemail pilot. 

We left  a message at the subscriber voicemail box and the subscriber got the MWI-ON message.

After listening to the message we deleted the message. However the MWI stayed on. 

 

We forced the MWI-OFF by dialing the MWI-OFF number (after changing some calling search spaces).

 

Strange thing we have now is that we are able to dial the voicemail pilot, however it keeps ringing and doesn't get connected to the voicemail.

 

I?m able to dial the voicemail ports (after changing some CSS). But I don't see them entering the port monitor tool.

 

Below I have the output of the dbwalker utility.

Any reason why my voicemail does not let the wav file play??

 

================================================================

Running dbWalker version:4.0.39 

0:0:Search for '(error)' or '(warning)' to find logged errors and warnings. 

Starting walk at:10/30/2007 16:04:31 PM

Local Unity Version=4.2(1.0)

Local server name= SPFSVUNITY1

Backend is Exchange 2000 or 2003

================================================================ 

Running dbWalker based on user input.  User=SPFBEHEER\UnityAdmin


Storage Info


    Unity is logging diagnostic and data files to: D:\CommServer\Logs\

        That drive has:

             41948.93 Megs total

             39343.71 Megs free

             93.79% available free space

 

    Unity is temporarily storing voice messages for UMR to: D:\CommServer\UnityMTA\

        That drive has:

             41948.93 Megs total

             39343.71 Megs free

             93.79% available free space

 


*** Starting location object walk ***


    Checking cross box transfer and login tables


*** Starting DTMFAccessID Table walk ***


     26 entries checked.

    Checking for duplication ParentObjectID entries in DTMFAccessID table


*** Starting General System Checks ***


    Primary server name found='SPFSVUNITY1'

    Checking LastModifiedTime column for all entries in the Configuration table


*** Starting subscriber walk ***


*** Starting call handler walk ***


*** Starting subscriber template walk ***


*** Starting directory handler walk ***


*** Starting interview handler walk ***


*** Starting Distribution List walk ***


*** Starting routing rules walk ***


Rule Name=Attempt Forward to Greeting

Rule Name=Attempt Sign-In

Rule Name=Default Call Handler

        Set to go to call handler with alias=openinggreetingch


*** Checking For Orphaned Table Entries ***


*** Check For Orphaned WAV Files ***


1605:(error) stranded wav file name=D:\CommServer\Stream Files\ESubscriberVoiceName.wav

   You can fix this by selecting the 'Removed orphaned WAV files automatically' option and re running dbWalker

1605:(error) stranded wav file name=D:\CommServer\Stream Files\UnaddressedFaxesVoiceName.wav

   You can fix this by selecting the 'Removed orphaned WAV files automatically' option and re running dbWalker


*** Check for orphaned GlobalSubscriber table entries ***


 


Database walk complete at:10/30/2007 16:04:39 PM


Errors Found:  2

Fixed Count:  0

Warning Count:  0

 

 

 

Richard van Berkel

 

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From bcarter at sentinel.com  Tue Oct 30 12:06:36 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Tue, 30 Oct 2007 11:06:36 -0500
Subject: [cisco-voip] Urgent: Output attenuation - What is louder
Message-ID: <6A0744639B86DD459433108DBF101D26019FEA23@MAIL01.sentinel.com>

Phone--PSTN--GW--IP Phone

In the conversation between the IP Phone and PSTN Phone, I want the PSTN
Phone received volume louder.  I am getting conflicting information on
output attenuation.

Which setting would increase the volume:

Output attenuation 5
Or
Output attenuation -5

?
 
 

From daniel.argueta at comcast.net  Tue Oct 30 13:03:25 2007
From: daniel.argueta at comcast.net (Daniel Argueta)
Date: Tue, 30 Oct 2007 10:03:25 -0700
Subject: [cisco-voip] Unity and new exchange server
Message-ID: <2B0AC0D2865946CFAC9118F2CE5435DD@main>

Hello All,


Do any of you have any information on what configuration changes need to take place if we are moving to a new Exchange server? We are currently running Unity 4.0(5) and Exchange 2003. Our old mail server ran out of space and we are in the process of moving everyone and everything to a much more powerful server with more storage. The name of the server is going to be different....

Thanks,

Daniel Argueta
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From jason.aarons at us.didata.com  Tue Oct 30 12:11:40 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 30 Oct 2007 12:11:40 -0400
Subject: [cisco-voip] X Missed Calls and Personal Address Book (CCM 5.1.2)
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E918@USNAEXCH.na.didata.local>

CallManger 5.1.2

Q1 With shared lines the missed calls represents all lines. We can't remove "X missed calls" but it is a Global Setting effecting all phones, how are other companies handling this.
 
Q2 On the phone under Directories > Personal Address Book (we setup in CCMUser) I have to login with UserId and UserPIN, is there a inactivity for timeout? if I reset a phone in ccmadin I assume they have to log in again?

 



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From Matt.Slaga at us.didata.com  Tue Oct 30 12:29:49 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 30 Oct 2007 12:29:49 -0400
Subject: [cisco-voip] Unity and new exchange server
In-Reply-To: <2B0AC0D2865946CFAC9118F2CE5435DD@main>
References: <2B0AC0D2865946CFAC9118F2CE5435DD@main>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00745DCA9@USNAEXCH.na.didata.local>

Daniel-

 

Are you replacing the Exchange server or adding a new one?

 

Once you have the new Exchange server up, run the Permissions Wizard
against the new message store.  Then run the Message Store Configuration
Wizard (Control Panel -> Add Remove Programs -> Message Store
Configuration Wizard) and point to the new server.  This will move the
Unity_XXX accounts.  

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel Argueta
Sent: Tuesday, October 30, 2007 1:03 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity and new exchange server

 

Hello All,

 

 

Do any of you have any information on what configuration changes need to
take place if we are moving to a new Exchange server? We are currently
running Unity 4.0(5) and Exchange 2003. Our old mail server ran out of
space and we are in the process of moving everyone and everything to a
much more powerful server with more storage. The name of the server is
going to be different....

 

Thanks,

 

Daniel Argueta




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From smgustafson at gmail.com  Tue Oct 30 12:38:43 2007
From: smgustafson at gmail.com (Steve G)
Date: Tue, 30 Oct 2007 09:38:43 -0700
Subject: [cisco-voip] Urgent: Output attenuation - What is louder
In-Reply-To: <6A0744639B86DD459433108DBF101D26019FEA23@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26019FEA23@MAIL01.sentinel.com>
Message-ID: <8a1f34b50710300938x329f2b2ex4875de66ebe78e47@mail.gmail.com>

I would say "output attenuation -5" because that would be doing less
attenuation.  Remember to 'shut' 'no shut' the voice port for this change to
take effect.

On 10/30/07, Carter, Bill <bcarter at sentinel.com> wrote:
>
> Phone--PSTN--GW--IP Phone
>
> In the conversation between the IP Phone and PSTN Phone, I want the PSTN
> Phone received volume louder.  I am getting conflicting information on
> output attenuation.
>
> Which setting would increase the volume:
>
> Output attenuation 5
> Or
> Output attenuation -5
>
> ?
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From VLoschiavo at data-corporation.com  Tue Oct 30 12:42:41 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Tue, 30 Oct 2007 12:42:41 -0400
Subject: [cisco-voip] Urgent: Output attenuation - What is louder
In-Reply-To: <8a1f34b50710300938x329f2b2ex4875de66ebe78e47@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26019FEA23@MAIL01.sentinel.com>
	<8a1f34b50710300938x329f2b2ex4875de66ebe78e47@mail.gmail.com>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF5713703066@MAIL02.hq.data-corporation.com>

you are correct.  -5 would be louder in this case.
 
However, while you are on a call...you can change these values and
immediately hear the results.
 
____________________________________________________________
Vincent Loschiavo
Cisco Practice Director
Datacorp
18503 Pines Blvd
Suite 213
Pembroke Pines, FL, 33029

"Cisco Advanced Unified Communications Partner"
Cell Phone:       786-282-1164
Office Phone:    954-671-5669
Office Main:      305-594-6933
____________________________________________________________ 
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Steve G
Sent: Tuesday, October 30, 2007 12:39 PM
To: Carter, Bill
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Urgent: Output attenuation - What is louder


I would say "output attenuation -5" because that would be doing less
attenuation.  Remember to 'shut' 'no shut' the voice port for this
change to take effect.


On 10/30/07, Carter, Bill <bcarter at sentinel.com> wrote: 

	Phone--PSTN--GW--IP Phone
	
	In the conversation between the IP Phone and PSTN Phone, I want
the PSTN
	Phone received volume louder.  I am getting conflicting
information on
	output attenuation.
	
	Which setting would increase the volume:
	
	Output attenuation 5
	Or
	Output attenuation -5
	
	?
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip 
	


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From daniel.argueta at comcast.net  Tue Oct 30 13:52:48 2007
From: daniel.argueta at comcast.net (Daniel Argueta)
Date: Tue, 30 Oct 2007 10:52:48 -0700
Subject: [cisco-voip] Unity and new exchange server
References: <2B0AC0D2865946CFAC9118F2CE5435DD@main>
	<C1FE15183DA37645BC0633BC604E44F00745DCA9@USNAEXCH.na.didata.local>
Message-ID: <52CEB0321CEB4A4DBCBBB15FB4F9D7A6@main>

Thanks for all of the responses!!!

We are adding a new one and slowly migrating everyone over...eventually getting rid of the old one......so far everyone has been getting emails, but no message light notification for those on the new one.

-Daniel
  ----- Original Message ----- 
  From: Matt Slaga (US) 
  To: Daniel Argueta ; cisco-voip at puck.nether.net 
  Sent: Tuesday, October 30, 2007 9:29 AM
  Subject: RE: [cisco-voip] Unity and new exchange server


  Daniel-

   

  Are you replacing the Exchange server or adding a new one?

   

  Once you have the new Exchange server up, run the Permissions Wizard against the new message store.  Then run the Message Store Configuration Wizard (Control Panel -> Add Remove Programs -> Message Store Configuration Wizard) and point to the new server.  This will move the Unity_XXX accounts.  

   

   

   

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel Argueta
  Sent: Tuesday, October 30, 2007 1:03 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Unity and new exchange server

   

  Hello All,

   

   

  Do any of you have any information on what configuration changes need to take place if we are moving to a new Exchange server? We are currently running Unity 4.0(5) and Exchange 2003. Our old mail server ran out of space and we are in the process of moving everyone and everything to a much more powerful server with more storage. The name of the server is going to be different....

   

  Thanks,

   

  Daniel Argueta



------------------------------------------------------------------------------


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From Matt.Slaga at us.didata.com  Tue Oct 30 13:00:17 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 30 Oct 2007 13:00:17 -0400
Subject: [cisco-voip] Unity and new exchange server
In-Reply-To: <52CEB0321CEB4A4DBCBBB15FB4F9D7A6@main>
References: <2B0AC0D2865946CFAC9118F2CE5435DD@main><C1FE15183DA37645BC0633BC604E44F00745DCA9@USNAEXCH.na.didata.local>
	<52CEB0321CEB4A4DBCBBB15FB4F9D7A6@main>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00745DD14@USNAEXCH.na.didata.local>

Sounds like permissions wizard needs to be run against the new exchange
box.  Unity probably does not have rights to look at the inbox for new
messages.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel Argueta
Sent: Tuesday, October 30, 2007 1:53 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity and new exchange server

 

Thanks for all of the responses!!!

 

We are adding a new one and slowly migrating everyone over...eventually
getting rid of the old one......so far everyone has been getting emails,
but no message light notification for those on the new one.

 

-Daniel

	----- Original Message ----- 

	From: Matt Slaga (US) <mailto:Matt.Slaga at us.didata.com>  

	To: Daniel Argueta <mailto:daniel.argueta at comcast.net>  ;
cisco-voip at puck.nether.net 

	Sent: Tuesday, October 30, 2007 9:29 AM

	Subject: RE: [cisco-voip] Unity and new exchange server

	 

	Daniel-

	 

	Are you replacing the Exchange server or adding a new one?

	 

	Once you have the new Exchange server up, run the Permissions
Wizard against the new message store.  Then run the Message Store
Configuration Wizard (Control Panel -> Add Remove Programs -> Message
Store Configuration Wizard) and point to the new server.  This will move
the Unity_XXX accounts.  

	 

	 

	 

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel Argueta
	Sent: Tuesday, October 30, 2007 1:03 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Unity and new exchange server

	 

	Hello All,

	 

	 

	Do any of you have any information on what configuration changes
need to take place if we are moving to a new Exchange server? We are
currently running Unity 4.0(5) and Exchange 2003. Our old mail server
ran out of space and we are in the process of moving everyone and
everything to a much more powerful server with more storage. The name of
the server is going to be different....

	 

	Thanks,

	 

	Daniel Argueta

________________________________

	Disclaimer: This e-mail communication and any attachments may
contain confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the intended
addressee, you are hereby notified that you have received this
communication in error and that any use or reproduction of this email or
its contents is strictly prohibited and may be unlawful. If you have
received this communication in error, please notify us immediately by
replying to this message and deleting it from your computer. Thank you. 




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From rkulagow at gmail.com  Tue Oct 30 13:04:56 2007
From: rkulagow at gmail.com (Robert Kulagowski)
Date: Tue, 30 Oct 2007 12:04:56 -0500
Subject: [cisco-voip] Urgent: Output attenuation - What is louder
In-Reply-To: <CB4BC8C24EF6E149AED9FDB9B9AF5713703066@MAIL02.hq.data-corporation.com>
References: <6A0744639B86DD459433108DBF101D26019FEA23@MAIL01.sentinel.com>	<8a1f34b50710300938x329f2b2ex4875de66ebe78e47@mail.gmail.com>
	<CB4BC8C24EF6E149AED9FDB9B9AF5713703066@MAIL02.hq.data-corporation.com>
Message-ID: <47276438.7090808@gmail.com>

If you want incoming voice from PSTN to sound louder, then you'd use 
input gain -3 on the voice port

If you want your outgoing audio to the PSTN to sound louder, then you'd 
use output attenuation -6.

But, if your signal to the PSTN is too "hot", then you'll run into 
problems.  We had an issue with DTMF digits to a conference calling 
system not registering correctly until we removed the output attenuation.

From daniel.argueta at comcast.net  Tue Oct 30 14:14:42 2007
From: daniel.argueta at comcast.net (Daniel Argueta)
Date: Tue, 30 Oct 2007 11:14:42 -0700
Subject: [cisco-voip] CM4.1 to 5.x Upgrade (DMA Questions)
References: <BF9C2A278ED44A9AAF459C0D468F9FBE@main>
	<472749F5.9030802@cisco.com>
Message-ID: <9CD7B04283FC4B798C3A0FD6146FC994@main>

Nope.....darn thing failed again....this is the only error message in the whole DMAErrors.log:


********************************************************************************

The error number is [-691]
This record was not processed because another record it needs, 
(see reference name listed below), is missing.  This likely may be due to a previous error 
already reported in the log above, or the result of Directory Services Export issues 
reported separately, or due to missing Product, Locale, or other add-on definition CSV 
files that should exist on the source system but do not.
Suspected Field Is [DMABack_Admin.fk_enduserdirgroupmap_fkenduser] 
TABLE NAME IS [enduserdirgroupmap ] 

fkdirgroup                '47b0de74-db3a-4be9-aa92-47a7921e0f26'

fkenduser                 'b3104b58-5f47-4a2d-bf98-cadeccebbf2d'

pkid                      '31abc01e-d680-4aa1-bbd4-4401b715ddb3'
********************************************************************************

Here is what I find in the installdbw1.log in reference to the error:


10/30/2007 09:25:02.592 installdb|   DBUtil::CopyTable insert into enduserdirgroupmap (fkdirgroup,fkenduser,pkid) values ('47b0de74-db3a-4be9-aa92-47a7921e0f26','b3104b58-5f47-4a2d-bf98-cadeccebbf2d','31abc01e-d680-4aa1-bbd4-4401b715ddb3')|
10/30/2007 09:25:02.592 installdb|   DBUtil::CopyTable ErrorCode=-691, SQLState=HY000|
10/30/2007 09:25:02.592 installdb|*ERROR* Error executing "insert into enduserdirgroupmap (fkdirgroup,fkenduser,pkid) values ('47b0de74-db3a-4be9-aa92-47a7921e0f26','b3104b58-5f47-4a2d-bf98-cadeccebbf2d','31abc01e-d680-4aa1-bbd4-4401b715ddb3'): [Informix][Informix ODBC Driver][Informix]Unspecified System Error =  -691. sqlerrm(DMABack_Admin.fk_enduserdirgroupmap_fkenduser)|
10/30/2007 09:25:02.592 installdb|<--DBUtil::CopyTable |
10/30/2007 09:25:02.592 installdb|   DataConverter_4to5::CopyTables update statistics medium for table enduserdirgroupmap|
10/30/2007 09:25:02.608 installdb|-->SysUtil::DMACanceled |


I found this page on the informix error 391: 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00807f3307.shtml

but unfortunately the solution does apply to us....Any ideas will be greatly appreciated...

-Daniel


  ----- Original Message ----- 
  From: Wes Sisk 
  To: Daniel Argueta 
  Cc: cisco-voip at puck.nether.net 
  Sent: Tuesday, October 30, 2007 8:12 AM
  Subject: Re: [cisco-voip] CM4.1 to 5.x Upgrade (DMA Questions)


  the errors are most likely errors in your database.  in the DMA output it lists log files you need to retrive.  collect those, zip them up, and attach to your TAC case.

  what dma does:
  1. if cm ver < 5.0 or < 4.2, then export windows database
  2. import win database to ccm_w database in windows install of informix with fkconstraints.  error on any constraint failures
  3. migrate ccm_w informix database to ccm database in informix in new cm6.x schema.

  Errors can occur at both steps 2 and 3.  

  /Wes

  Daniel Argueta wrote: 
    Does anyone have any detailed information about what the DMA actually does during the validation process? It looks like i'm being able to take a backup of all the information, but the validation is what fails when i'm trying to run this utility......it errors out with some Database errors......i'm guessing it's creating a temp database and using the schema it would use in 5.x and inserting the information into it as a 'test'? Do any of you have any information on the actual process this takes? I've been messing with this for about 2 weeks now and can't seem to successfully complete the DMA process.....except for this time i'm stuck and don't know how to fix it......

    Thanks,


    Daniel Argueta
----------------------------------------------------------------------------
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From ealeatherman at gmail.com  Tue Oct 30 13:40:16 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 30 Oct 2007 13:40:16 -0400
Subject: [cisco-voip] Ughh DST Issues!
In-Reply-To: <a1b07ea60710290633l6207c55exafb5a183acb6cf50@mail.gmail.com>
References: <a1b07ea60710290551g2a3768c7v955af853b8171bd4@mail.gmail.com>
	<4725DB92.6040308@cisco.com>
	<a1b07ea60710290633l6207c55exafb5a183acb6cf50@mail.gmail.com>
Message-ID: <94a1afde0710301040r349366c0gc8986e2ca85702e8@mail.gmail.com>

FYI

We ran into a few phones running 8.2.1s code still (7961's and 7911's), for
whatever reason they did not get reset when we upgraded firmware. Their
clocks set back this weekend. There is a asterisk on them in the field
notice that explains why, see below. Most of our phones of those models are
running 8.2.2 and were not affected. At least in our case this was a known
and fixed bug, we just weren't 100% upgraded.

Ed

****IMPORTANT NOTE***: For 7906G, 7911G, 7941G/GE, 7961G/GE, 7970G, 7971GE,
in addition to the above phone loads, there is one manual workaround that
will be needed in order to get the phones to display the correct time.*

There is a Daylight Savings Time issue which will be fixed in an upcoming
phone release, having the following workaround:

- Reset the IP phone from the CCMAdmin for the phone to get the correct
time.

Please be aware that each time IP Phones lose power, the System
Administrator will need to reset the IP Phones from CCMAdmin to get the
correct time.


On 10/29/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
>
> Awfully witty and insightful there.  Happy monday morning to you to. :-P
>
> As per the FN:
>
>
> http://cisco.com/en/US/products/hw/phones/ps379/products_field_notice09186a00807d3d9e.shtml
>
> "Cisco Unified IP Phone models 7910G, 7910G+SW, 7940G and 7960G (SCCP)
> Time information comes from Cisco Unified CallManager / CallManager
> Express. Update CallManager or CallManager Express to correct the new
> DST dates."
>
>
> Well like I said the patches were applied to the CallManagers and they
> are obviously working as they showed the correct time and everything.
> And based on the FN 7940s and 7960s get there time from CallManagers
> obviously.  So it doesn't indicate you need to do anything on the
> phones right?
>
> CallManager 4.1(3)sr4d
>
> All phones are 7960s running P00308000400 which as I understand is an SCCP
> load.
>
> So CallManagers are patched and working fine and show correct time.
> Phones are 7960s with an SCCP load, and based on the FN they should
> just get time from CallManagers.
>
>
> Any thoughts as to what I am missing or overlooking here?
>
>
>
>
>
> On 10/29/07, Wes Sisk <wsisk at cisco.com> wrote:
> > Hi Dark,
> >
> > No CM version, no phone model, no phone load version. Not much we can
> > say other than read the FN:
> >
> http://cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a00807ca437.shtml
> >
> > /Wes
> >
> > Dark Fiber wrote:
> > > CallManagers show the correct time and time zone, but the phones all
> > > show the incorrect time now.  They are one hour behind, so basically
> > > they changed over this weekend based on the old schedule and not the
> > > new schedule which should be next weekend.
> > >
> > > I have tried resetting the phones but that didn't do anything.
> > >
> > > Since the CallManagers are all correct I don't know what else to do or
> > > try at this point.
> > >
> > > Any thoughts?   I have checked the date and time group in CallManager
> > > it is right, the CallManagers themselves show the correct date and
> > > time.  But the phones are off an hour.
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From voicenoob at gmail.com  Tue Oct 30 14:28:33 2007
From: voicenoob at gmail.com (Voice Noob)
Date: Tue, 30 Oct 2007 13:28:33 -0500
Subject: [cisco-voip] CallManager 4.x (vs.) 6.x
In-Reply-To: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us>
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us>
Message-ID: <535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com>

I think the windows version is better because it does not lock you out of
the system like the Linux version. you have more control.

On 10/30/07, Leetun, Rob <rleetun at co.boulder.co.us> wrote:
>
>  Hello all,
>
> We are having a discussion about why is a Windows platform better than a
> Linux platform when it comes to CallManager, or is Linux better than
> Windows?  In your opinions, is it better to go from 4.x to 6.x or wait for
> 7.x to roll out next summer?  Or is it better to stay on the 4.x train?
> How many of you have switched to the Linux platform from the Windows
> platform?
>
> Thanks in advance.
>
> Rob
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From VLoschiavo at data-corporation.com  Tue Oct 30 15:08:13 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Tue, 30 Oct 2007 15:08:13 -0400
Subject: [cisco-voip] CallManager 4.x (vs.) 6.x
In-Reply-To: <535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com>
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us>
	<535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voice Noob
Sent: Tuesday, October 30, 2007 2:29 PM
To: Leetun, Rob
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager 4.x (vs.) 6.x


I think the windows version is better because it does not lock you out
of the system like the Linux version. you have more control.


On 10/30/07, Leetun, Rob <rleetun at co.boulder.co.us> wrote: 

	Hello all,

	We are having a discussion about why is a Windows platform
better than a Linux platform when it comes to CallManager , or is Linux
better than Windows?  In your opinions, is it better to go from 4.x to
6.x or wait for 7.x to roll out next summer?  Or is it better to stay on
the 4.x train?  How many of you have switched to the Linux platform from
the Windows platform?

	Thanks in advance.

	Rob


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip 
	

 
 
 
All, 

I've found both the windows and linux versions to be quite stable.
 
I haven't been locked out of the linux platform...so I can't speak to
that, but I do like the method of patching on linux.  Seems much cleaner
to me.  (Plus the ability to roll back from a patch).
 
(I'm also an avid *nix user) :)
 
-V
 
____________________________________________________________
Vincent Loschiavo
Cisco Practice Director
Datacorp
18503 Pines Blvd
Suite 213
Pembroke Pines, FL, 33029

"Cisco Advanced Unified Communications Partner"
Cell Phone:       786-282-1164
Office Phone:    954-671-5669
Office Main:      305-594-6933
____________________________________________________________ 
 

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From ducotedt at stfran.com  Tue Oct 30 13:32:46 2007
From: ducotedt at stfran.com (Darren Ducote)
Date: Tue, 30 Oct 2007 12:32:46 -0500
Subject: [cisco-voip] IPCC express 4.5 icd.aef script manipulation
Message-ID: <A46FC8D5BB84EB4D9B0CE8CBDCDCD06517464BBA@xch07.stfran.net>

Hello everyone,

We have installed IPCCx 4.5 and use CCM5.

We have a service desk and would like to use the icd.aef script with a few modifications. I would like the play a welcome message and then dump everyone into a queue to wait for available agents. During this time, I would like to play a message that will allow the user to press "1" to leave a message.

At this time it would be a call forward to our voicemail pilot number.

Anyone doing anything like this? I have made some modification with moderate success using the "get digits" string. Thanks!

________________________________
This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission.


***St. Francis Medical Center and affiliates encrypt email containing Patient Health Information.  You will receive a notice with instructions on how to open any encrypted email you receive.  To learn more, go to this web site: http://userawareness.zixcorp.com/sites/index.php?b=141&type=1
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From wsisk at cisco.com  Tue Oct 30 18:10:48 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 30 Oct 2007 17:10:48 -0500
Subject: [cisco-voip] CM4.1 to 5.x Upgrade (DMA Questions)
In-Reply-To: <9CD7B04283FC4B798C3A0FD6146FC994@main>
References: <BF9C2A278ED44A9AAF459C0D468F9FBE@main>	<472749F5.9030802@cisco.com>
	<9CD7B04283FC4B798C3A0FD6146FC994@main>
Message-ID: <4727ABE8.1020400@cisco.com>

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From martind at cae.com  Tue Oct 30 18:17:59 2007
From: martind at cae.com (Daniel Martin)
Date: Tue, 30 Oct 2007 18:17:59 -0400
Subject: [cisco-voip] Upgrade to V6.0 hanging
Message-ID: <09881513E25DD549B26BF7F38ECAF8DA56E541@caemex01.caecorp.cae.com>

Hi,

 

I'm having a problem upgrading my prod publisher from v5.1.1 to
v6.0(1a). I was able to do it without any problems on my lab cluster.

 

I'm using the V5.x to V6.0(1a) Upgrade DVD which I used to do the lab. 

 

When I start the upgrade, it downloads from DVD and after a while it
just hangs. Sometimes it's a 1000Mb, 800MB and even around 650MB.

 

I can see the CPU hit 100% and stay there. That's when it hangs.

 

The server is a 7825H1 with 2GB of RAM, 72GB hard drive and a 3.04GHz
CPU which meets the hardware requirement. 

 

Anybody have this problem before?

 

Thanks,

 

Daniel.

 

Daniel Martin

Groupe Reseau / Networks Group

CAE Inc.

St-Laurent, Quebec Canada

H4T 1G6

Tel:  1-514-341-2000 x4878

Fax: 1-514-340-5583

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From wsisk at cisco.com  Tue Oct 30 18:24:10 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 30 Oct 2007 17:24:10 -0500
Subject: [cisco-voip] Upgrade to V6.0 hanging
In-Reply-To: <09881513E25DD549B26BF7F38ECAF8DA56E541@caemex01.caecorp.cae.com>
References: <09881513E25DD549B26BF7F38ECAF8DA56E541@caemex01.caecorp.cae.com>
Message-ID: <4727AF0A.6000303@cisco.com>

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From martind at cae.com  Tue Oct 30 18:28:25 2007
From: martind at cae.com (Daniel Martin)
Date: Tue, 30 Oct 2007 18:28:25 -0400
Subject: [cisco-voip] Upgrade to V6.0 hanging
In-Reply-To: <09881513E25DD549B26BF7F38ECAF8DA56E541@caemex01.caecorp.cae.com>
References: <09881513E25DD549B26BF7F38ECAF8DA56E541@caemex01.caecorp.cae.com>
Message-ID: <09881513E25DD549B26BF7F38ECAF8DA56E543@caemex01.caecorp.cae.com>

RTMT reports that RisDC is causing Core Dumps each time I try the
upgrade.

 

TAC had told me to upgrade to fix the core dump issue but the upgrade is
causing dumps also. I seem to be going in circles here.

 

I'll have to go back to TAC I guess.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel Martin
Sent: Tuesday, October 30, 2007 6:18 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Upgrade to V6.0 hanging

 

Hi,

 

I'm having a problem upgrading my prod publisher from v5.1.1 to
v6.0(1a). I was able to do it without any problems on my lab cluster.

 

I'm using the V5.x to V6.0(1a) Upgrade DVD which I used to do the lab. 

 

When I start the upgrade, it downloads from DVD and after a while it
just hangs. Sometimes it's a 1000Mb, 800MB and even around 650MB.

 

I can see the CPU hit 100% and stay there. That's when it hangs.

 

The server is a 7825H1 with 2GB of RAM, 72GB hard drive and a 3.04GHz
CPU which meets the hardware requirement. 

 

Anybody have this problem before?

 

Thanks,

 

Daniel.

 

Daniel Martin

Groupe Reseau / Networks Group

CAE Inc.

St-Laurent, Quebec Canada

H4T 1G6

Tel:  1-514-341-2000 x4878

Fax: 1-514-340-5583

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From drogers at la-networks.com  Tue Oct 30 18:40:46 2007
From: drogers at la-networks.com (Dan Rogers)
Date: Tue, 30 Oct 2007 15:40:46 -0700
Subject: [cisco-voip] cisco-voip Digest, Vol 56, Issue 182
In-Reply-To: <mailman.3916.1193783355.13771.cisco-voip@puck.nether.net>
References: <mailman.3916.1193783355.13771.cisco-voip@puck.nether.net>
Message-ID: <25BF73C165551F469D3F5F1FBA7C506311BC7D@lanet-ex01.ad.la-networks.com>

Daniel,

My first post so please forgive any errors in format.

Looking at your hardware platform, it is consistent with a bug that I
recall reading about where upgrades will cause a hang. I can look up the
bug if you like, but if you wish to take a peek there is a download
under Callmanager Utilies (or some such) at CCO. It specifically cites
your hardware platform as being one that can be affected by periodic
hangs upon upgrade. There is a patch that can be applied to rectify the
issue. Please let me know if I need to be more specific with where to
look for the patch.

Dan Rogers
LA Networks Services
(O) 310.943.4118
(F) 310.943.4101
drogers at la-networks.com
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
cisco-voip-request at puck.nether.net
Sent: Tuesday, October 30, 2007 3:29 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 56, Issue 182

Send cisco-voip mailing list submissions to
	cisco-voip at puck.nether.net

To subscribe or unsubscribe via the World Wide Web, visit
	https://puck.nether.net/mailman/listinfo/cisco-voip
or, via email, send a message with subject or body 'help' to
	cisco-voip-request at puck.nether.net

You can reach the person managing the list at
	cisco-voip-owner at puck.nether.net

When replying, please edit your Subject line so it is more specific
than "Re: Contents of cisco-voip digest..."


Today's Topics:

   1. Re: CM4.1 to 5.x Upgrade (DMA Questions) (Wes Sisk)
   2. Upgrade to V6.0 hanging (Daniel Martin)
   3. Re: Upgrade to V6.0 hanging (Wes Sisk)
   4. Re: Upgrade to V6.0 hanging (Daniel Martin)


----------------------------------------------------------------------

Message: 1
Date: Tue, 30 Oct 2007 17:10:48 -0500
From: Wes Sisk <wsisk at cisco.com>
Subject: Re: [cisco-voip] CM4.1 to 5.x Upgrade (DMA Questions)
To: Daniel Argueta <daniel.argueta at comcast.net>
Cc: , cisco-voip at puck.nether.net
Message-ID: <4727ABE8.1020400 at cisco.com>
Content-Type: text/plain; charset="us-ascii"

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------------------------------

Message: 2
Date: Tue, 30 Oct 2007 18:17:59 -0400
From: "Daniel Martin" <martind at cae.com>
Subject: [cisco-voip] Upgrade to V6.0 hanging
To: <cisco-voip at puck.nether.net>
Message-ID:
	
<09881513E25DD549B26BF7F38ECAF8DA56E541 at caemex01.caecorp.cae.com>
Content-Type: text/plain; charset="us-ascii"

Hi,

 

I'm having a problem upgrading my prod publisher from v5.1.1 to
v6.0(1a). I was able to do it without any problems on my lab cluster.

 

I'm using the V5.x to V6.0(1a) Upgrade DVD which I used to do the lab. 

 

When I start the upgrade, it downloads from DVD and after a while it
just hangs. Sometimes it's a 1000Mb, 800MB and even around 650MB.

 

I can see the CPU hit 100% and stay there. That's when it hangs.

 

The server is a 7825H1 with 2GB of RAM, 72GB hard drive and a 3.04GHz
CPU which meets the hardware requirement. 

 

Anybody have this problem before?

 

Thanks,

 

Daniel.

 

Daniel Martin

Groupe Reseau / Networks Group

CAE Inc.

St-Laurent, Quebec Canada

H4T 1G6

Tel:  1-514-341-2000 x4878

Fax: 1-514-340-5583

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------------------------------

Message: 3
Date: Tue, 30 Oct 2007 17:24:10 -0500
From: Wes Sisk <wsisk at cisco.com>
Subject: Re: [cisco-voip] Upgrade to V6.0 hanging
To: Daniel Martin <martind at cae.com>
Cc: cisco-voip at puck.nether.net
Message-ID: <4727AF0A.6000303 at cisco.com>
Content-Type: text/plain; charset="us-ascii"

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------------------------------

Message: 4
Date: Tue, 30 Oct 2007 18:28:25 -0400
From: "Daniel Martin" <martind at cae.com>
Subject: Re: [cisco-voip] Upgrade to V6.0 hanging
To: "Daniel Martin" <martind at cae.com>, <cisco-voip at puck.nether.net>
Message-ID:
	
<09881513E25DD549B26BF7F38ECAF8DA56E543 at caemex01.caecorp.cae.com>
Content-Type: text/plain; charset="us-ascii"

RTMT reports that RisDC is causing Core Dumps each time I try the
upgrade.

 

TAC had told me to upgrade to fix the core dump issue but the upgrade is
causing dumps also. I seem to be going in circles here.

 

I'll have to go back to TAC I guess.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel Martin
Sent: Tuesday, October 30, 2007 6:18 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Upgrade to V6.0 hanging

 

Hi,

 

I'm having a problem upgrading my prod publisher from v5.1.1 to
v6.0(1a). I was able to do it without any problems on my lab cluster.

 

I'm using the V5.x to V6.0(1a) Upgrade DVD which I used to do the lab. 

 

When I start the upgrade, it downloads from DVD and after a while it
just hangs. Sometimes it's a 1000Mb, 800MB and even around 650MB.

 

I can see the CPU hit 100% and stay there. That's when it hangs.

 

The server is a 7825H1 with 2GB of RAM, 72GB hard drive and a 3.04GHz
CPU which meets the hardware requirement. 

 

Anybody have this problem before?

 

Thanks,

 

Daniel.

 

Daniel Martin

Groupe Reseau / Networks Group

CAE Inc.

St-Laurent, Quebec Canada

H4T 1G6

Tel:  1-514-341-2000 x4878

Fax: 1-514-340-5583

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------------------------------

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

End of cisco-voip Digest, Vol 56, Issue 182
*******************************************

From dustin.s.fowler at gmail.com  Tue Oct 30 20:19:11 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Tue, 30 Oct 2007 20:19:11 -0400
Subject: [cisco-voip] IPCC express 4.5 icd.aef script manipulation
In-Reply-To: <A46FC8D5BB84EB4D9B0CE8CBDCDCD06517464BBA@xch07.stfran.net>
References: <A46FC8D5BB84EB4D9B0CE8CBDCDCD06517464BBA@xch07.stfran.net>
Message-ID: <4727ca02.157f400a.7d36.735a@mx.google.com>

I think this might be what you are looking for.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Darren Ducote
Sent: Tuesday, October 30, 2007 1:33 PM
To: 'cisco-voip at puck.nether.net'
Subject: [cisco-voip] IPCC express 4.5 icd.aef script manipulation

 

Hello everyone,

 

We have installed IPCCx 4.5 and use CCM5.

 

We have a service desk and would like to use the icd.aef script with a few
modifications. I would like the play a welcome message and then dump
everyone into a queue to wait for available agents. During this time, I
would like to play a message that will allow the user to press "1" to leave
a message.

 

At this time it would be a call forward to our voicemail pilot number.

 

Anyone doing anything like this? I have made some modification with moderate
success using the "get digits" string. Thanks!

 

  _____  

This message contains confidential information and is intended only for the
individual named. If you are not the named addressee you should not
disseminate, distribute or copy this e-mail. Please notify the sender
immediately by e-mail if you have received this e-mail by mistake and delete
this e-mail from your system. E-mail transmission cannot be guaranteed to be
secure or error-free as information could be intercepted, corrupted, lost,
destroyed, arrive late or incomplete, or contain viruses. The sender
therefore does not accept liability for any errors or omissions in the
contents of this message, which arise as a result of e-mail transmission.


***St. Francis Medical Center and affiliates encrypt email containing
Patient Health Information. You will receive a notice with instructions on
how to open any encrypted email you receive. To learn more about this click
<http://userawareness.zixcorp.com/sites/index.php?b=141&type=1> Here 

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From martind at cae.com  Tue Oct 30 20:38:51 2007
From: martind at cae.com (Daniel Martin)
Date: Tue, 30 Oct 2007 20:38:51 -0400
Subject: [cisco-voip] Upgrade to V6.0 hanging
In-Reply-To: <4727AF0A.6000303@cisco.com>
References: <09881513E25DD549B26BF7F38ECAF8DA56E541@caemex01.caecorp.cae.com>
	<4727AF0A.6000303@cisco.com>
Message-ID: <09881513E25DD549B26BF7F38ECAF8DA56E545@caemex01.caecorp.cae.com>

I guess it was.

 

FTP worked like a charm. Thanks Wes.

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Tuesday, October 30, 2007 6:24 PM
To: Daniel Martin
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Upgrade to V6.0 hanging

 

Can you the upgrade file on SFTP server rather than disk?  This sounds
like disk read problems. /wes

Daniel Martin wrote: 

Hi,

 

I'm having a problem upgrading my prod publisher from v5.1.1 to
v6.0(1a). I was able to do it without any problems on my lab cluster.

 

I'm using the V5.x to V6.0(1a) Upgrade DVD which I used to do the lab. 

 

When I start the upgrade, it downloads from DVD and after a while it
just hangs. Sometimes it's a 1000Mb, 800MB and even around 650MB.

 

I can see the CPU hit 100% and stay there. That's when it hangs.

 

The server is a 7825H1 with 2GB of RAM, 72GB hard drive and a 3.04GHz
CPU which meets the hardware requirement. 

 

Anybody have this problem before?

 

Thanks,

 

Daniel.

 

Daniel Martin

Groupe Reseau / Networks Group

CAE Inc.

St-Laurent, Quebec Canada

H4T 1G6

Tel:  1-514-341-2000 x4878

Fax: 1-514-340-5583

 


________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From svoll.voip at gmail.com  Wed Oct 31 00:09:57 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 30 Oct 2007 21:09:57 -0700
Subject: [cisco-voip] Castelle Fax Press turn off ECM and Super G3
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00745D640@USNAEXCH.na.didata.local>
References: <6A0744639B86DD459433108DBF101D26019FE9DC@MAIL01.sentinel.com>
	<f84a38d30710291934r49f521bbsd1406981738825e8@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F00745D640@USNAEXCH.na.didata.local>
Message-ID: <f84a38d30710302109q5bf52f59p9e4fa2c7b3cdb6cb@mail.gmail.com>

is this a Fax server or a fax machine?

if it's a fax server it might be a fax-relay vs fax-passthrough issue
or VAD on your dial peer.

if fax machine do they even support ECM and Super G3.  I've never
heard of Castelle.

Scott

On Oct 29, 2007 7:42 PM, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> I had a customer on the phone with Castelle Fax Press technical support
> (they don't seem very knowledgeable of their equipment I noticed) and
> had an issue escalated to Castelle development who told me they don't
> have an option to turn off ECM or set the speed to G3.
>
> Anyone else setup one of their boxes via T1 PRI back to ISR router?  I
> find it hard to believe they can't turn of ECM or disable Super G3...
>
> Also Castelle is looking at adding T.38 support down the road, seemed to
> me they are way behind the times.
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
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> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From svoll.voip at gmail.com  Wed Oct 31 00:15:28 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 30 Oct 2007 21:15:28 -0700
Subject: [cisco-voip] Cisco Unity
In-Reply-To: <F4DD4283A3EF194B84D774C60FB7F786024DA63A@UVEXVS01.bc.broward.cty>
References: <F4DD4283A3EF194B84D774C60FB7F786024DA63A@UVEXVS01.bc.broward.cty>
Message-ID: <f84a38d30710302115h5320decfxddaa7c83e9e2c0db@mail.gmail.com>

yep.  read the upgrade doc's.  looks like the same process of
upgrading from 4.03 to 4.05.  Just make sure you get the upgrade
license before you start.

Scott

On Oct 30, 2007 7:26 AM, Ocampo, Walter <WOCAMPO at broward.org> wrote:
>
>
>
> Hi Folks,
>
>
>
> Is there a direct path from Up-grading Cisco Unity from 4.0(5) to 5.0
>
>
>
> Regards,
>
>
>
> Walter Ocampo
>
> System Network Analyst
>
> Office of Information Technology
>
> 540 S.E 3rd. Ave, Suite 300
>
> Fort Lauderdale, FL 33301
>
> (954) 357 8041 Cell (954) 632 8228
>
> wocampo at broward.org
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From svoll.voip at gmail.com  Wed Oct 31 00:17:54 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 30 Oct 2007 21:17:54 -0700
Subject: [cisco-voip] Unity and new exchange server
In-Reply-To: <52CEB0321CEB4A4DBCBBB15FB4F9D7A6@main>
References: <2B0AC0D2865946CFAC9118F2CE5435DD@main>
	<C1FE15183DA37645BC0633BC604E44F00745DCA9@USNAEXCH.na.didata.local>
	<52CEB0321CEB4A4DBCBBB15FB4F9D7A6@main>
Message-ID: <f84a38d30710302117r77717d47o71cd908a388b6710@mail.gmail.com>

just for clarification.... it's not exchange 2007 right?  if so, you
will need to upgrade to Unity 5.0

Scott

On Oct 30, 2007 10:52 AM, Daniel Argueta <daniel.argueta at comcast.net> wrote:
>
> Thanks for all of the responses!!!
>
> We are adding a new one and slowly migrating everyone over...eventually
> getting rid of the old one......so far everyone has been getting emails, but
> no message light notification for those on the new one.
>
> -Daniel
>
>
> ----- Original Message -----
> From: Matt Slaga (US)
> To: Daniel Argueta ; cisco-voip at puck.nether.net
> Sent: Tuesday, October 30, 2007 9:29 AM
> Subject: RE: [cisco-voip] Unity and new exchange server
>
>
>
> Daniel-
>
>
>
> Are you replacing the Exchange server or adding a new one?
>
>
>
> Once you have the new Exchange server up, run the Permissions Wizard against
> the new message store.  Then run the Message Store Configuration Wizard
> (Control Panel -> Add Remove Programs -> Message Store Configuration Wizard)
> and point to the new server.  This will move the Unity_XXX accounts.
>
>
>
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel Argueta
> Sent: Tuesday, October 30, 2007 1:03 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Unity and new exchange server
>
>
>
>
> Hello All,
>
>
>
>
>
>
>
>
> Do any of you have any information on what configuration changes need to
> take place if we are moving to a new Exchange server? We are currently
> running Unity 4.0(5) and Exchange 2003. Our old mail server ran out of space
> and we are in the process of moving everyone and everything to a much more
> powerful server with more storage. The name of the server is going to be
> different....
>
>
>
>
>
> Thanks,
>
>
>
>
>
> Daniel Argueta
>
> ________________________________
>
>
>
>
> Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From svoll.voip at gmail.com  Wed Oct 31 00:31:36 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 30 Oct 2007 21:31:36 -0700
Subject: [cisco-voip] CallManager 4.x (vs.) 6.x
In-Reply-To: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us>
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us>
Message-ID: <f84a38d30710302131q84e3e08m5f934daeb3e44d96@mail.gmail.com>

Both have good and bad.

linux version is nice with the dual partitions for upgrading and backing out.

Windows has a little easier access to the DB.  As far as being locked
out....... I think some of it is just a readjustement to new ways of
getting to the same data.

personally I run CM 4.1.3 but just did a install of 5.1.2 for one of
my School districts.  Since I'm so late in the game right now I'm
waiting for 6.1 in January to start shipping and will most likely
upgrade to 6.1 just so I don't have to turn around and do a second
major upgrade.

With all that said...... at Open Mic tomorrow I'm going to ask to make
SFTP optional rather then  mandatory.  there is no reason why a server
on the same network needs to be SFTP for a stupid backup.  If a Hack
wanted my backup file he can have it ;-)

just my 2 cents.

Scott

PS.  I love that there is no java problems with 5.x

On Oct 30, 2007 8:46 AM, Leetun, Rob <rleetun at co.boulder.co.us> wrote:
>
>
> Hello all,
>
> We are having a discussion about why is a Windows platform better than a
> Linux platform when it comes to CallManager, or is Linux better than
> Windows?  In your opinions, is it better to go from 4.x to 6.x or wait for
> 7.x to roll out next summer?  Or is it better to stay on the 4.x train?  How
> many of you have switched to the Linux platform from the Windows platform?
>
> Thanks in advance.
>
> Rob
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From svoll.voip at gmail.com  Wed Oct 31 00:34:09 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 30 Oct 2007 21:34:09 -0700
Subject: [cisco-voip] Unity does not connect.
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157ED98@tnl1mx1.telindus.intra>
References: <B42A9A0E2A14B943A3B7EA2FA7A0D8860157ED98@tnl1mx1.telindus.intra>
Message-ID: <f84a38d30710302134pe5495e1y4f98b5fe286e0e34@mail.gmail.com>

Can you use DNA and see where the number is going?

TSP issue?

Scott

On Oct 30, 2007 9:06 AM, Richard van Berkel
<Richard.vanBerkel at telindus.nl> wrote:
>
>
>
> Rudi,
>
>
>
>
>
> We had Unity working.
>
> We could select the voicemail pilot.
>
> We left  a message at the subscriber voicemail box and the subscriber got
> the MWI-ON message.
>
> After listening to the message we deleted the message. However the MWI
> stayed on.
>
>
>
> We forced the MWI-OFF by dialing the MWI-OFF number (after changing some
> calling search spaces).
>
>
>
> Strange thing we have now is that we are able to dial the voicemail pilot,
> however it keeps ringing and doesn't get connected to the voicemail.
>
>
>
> I?m able to dial the voicemail ports (after changing some CSS). But I don't
> see them entering the port monitor tool.
>
>
>
> Below I have the output of the dbwalker utility.
>
> Any reason why my voicemail does not let the wav file play??
>
>
>
> ================================================================
>
> Running dbWalker version:4.0.39
>
> 0:0:Search for '(error)' or '(warning)' to find logged errors and warnings.
>
> Starting walk at:10/30/2007 16:04:31 PM
>
> Local Unity Version=4.2(1.0)
>
> Local server name= SPFSVUNITY1
>
> Backend is Exchange 2000 or 2003
>
> ================================================================
>
> Running dbWalker based on user input.  User=SPFBEHEER\UnityAdmin
> Storage Info
>
>     Unity is logging diagnostic and data files to: D:\CommServer\Logs\
>
>         That drive has:
>
>              41948.93 Megs total
>
>              39343.71 Megs free
>
>              93.79% available free space
>
>
>
>     Unity is temporarily storing voice messages for UMR to:
> D:\CommServer\UnityMTA\
>
>         That drive has:
>
>              41948.93 Megs total
>
>              39343.71 Megs free
>
>              93.79% available free space
>
>
> *** Starting location object walk ***
>
>     Checking cross box transfer and login tables
> *** Starting DTMFAccessID Table walk ***
>
>      26 entries checked.
>
>     Checking for duplication ParentObjectID entries in DTMFAccessID table
> *** Starting General System Checks ***
>
>     Primary server name found='SPFSVUNITY1'
>
>     Checking LastModifiedTime column for all entries in the Configuration
> table
> *** Starting subscriber walk ***
> *** Starting call handler walk ***
> *** Starting subscriber template walk ***
> *** Starting directory handler walk ***
> *** Starting interview handler walk ***
> *** Starting Distribution List walk ***
> *** Starting routing rules walk ***
>
> Rule Name=Attempt Forward to Greeting
>
> Rule Name=Attempt Sign-In
>
> Rule Name=Default Call Handler
>
>         Set to go to call handler with alias=openinggreetingch
> *** Checking For Orphaned Table Entries ***
> *** Check For Orphaned WAV Files ***
>
> 1605:(error) stranded wav file name=D:\CommServer\Stream
> Files\ESubscriberVoiceName.wav
>
>    You can fix this by selecting the 'Removed orphaned WAV files
> automatically' option and re running dbWalker
>
> 1605:(error) stranded wav file name=D:\CommServer\Stream
> Files\UnaddressedFaxesVoiceName.wav
>
>    You can fix this by selecting the 'Removed orphaned WAV files
> automatically' option and re running dbWalker
> *** Check for orphaned GlobalSubscriber table entries ***
>
>
> Database walk complete at:10/30/2007 16:04:39 PM
>
> Errors Found:  2
>
> Fixed Count:  0
>
> Warning Count:  0
>
>
>
>
>
>
>
> Richard van Berkel
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From khalid_khursheed at hotmail.com  Wed Oct 31 03:22:10 2007
From: khalid_khursheed at hotmail.com (Syed Khalid Ali)
Date: Wed, 31 Oct 2007 12:22:10 +0500
Subject: [cisco-voip] Asktrik or Open SIP
Message-ID: <BAY113-W1543B217F59118332BCAB2E8930@phx.gbl>


hi,

is Asterisk or OpenSER SIP is compatible with Cisco SIP User Agents (Gateways)?

regards

syed khalid ali


_________________________________________________________________
Connect to the next generation of MSN Messenger?
http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline

From keli at carocomp.ro  Wed Oct 31 03:21:09 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Wed, 31 Oct 2007 09:21:09 +0200
Subject: [cisco-voip] Castelle Fax Press turn off ECM and Super G3
In-Reply-To: <f84a38d30710302109q5bf52f59p9e4fa2c7b3cdb6cb@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26019FE9DC@MAIL01.sentinel.com>	<f84a38d30710291934r49f521bbsd1406981738825e8@mail.gmail.com>	<C1FE15183DA37645BC0633BC604E44F00745D640@USNAEXCH.na.didata.local>
	<f84a38d30710302109q5bf52f59p9e4fa2c7b3cdb6cb@mail.gmail.com>
Message-ID: <47282CE5.6020101@carocomp.ro>

A quick googling tells me it's a fax server, and (if it wasn't obvious 
from that alone :P) it's max speed is 14,4kbps. So it's definitely not 
Super G3 (v.34bis, up to 33.6 kbps). It's not even G3 (v.34, 28.8kbps)

Also, ECM may or may not be supported at all. It's probably not, though.

I'd say, for now, the safest bet anyway is to run with passthrough fax, 
don't try to relay it.

regards,
  Zoltan

Scott Voll wrote:
> is this a Fax server or a fax machine?
>
> if it's a fax server it might be a fax-relay vs fax-passthrough issue
> or VAD on your dial peer.
>
> if fax machine do they even support ECM and Super G3.  I've never
> heard of Castelle.
>
> Scott
>
> On Oct 29, 2007 7:42 PM, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>   
>> I had a customer on the phone with Castelle Fax Press technical support
>> (they don't seem very knowledgeable of their equipment I noticed) and
>> had an issue escalated to Castelle development who told me they don't
>> have an option to turn off ECM or set the speed to G3.
>>
>> Anyone else setup one of their boxes via T1 PRI back to ISR router?  I
>> find it hard to believe they can't turn of ECM or disable Super G3...
>>
>> Also Castelle is looking at adding T.38 support down the road, seemed to
>> me they are way behind the times.
>>
>> -----------------------------------------
>> Disclaimer:
>>
>> This e-mail communication and any attachments may contain
>> confidential and privileged information and is for use by the
>> designated addressee(s) named above only.  If you are not the
>> intended addressee, you are hereby notified that you have received
>> this communication in error and that any use or reproduction of
>> this email or its contents is strictly prohibited and may be
>> unlawful.  If you have received this communication in error, please
>> notify us immediately by replying to this message and deleting it
>> from your computer. Thank you.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>     
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From aman.chugh at gmail.com  Wed Oct 31 03:56:19 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Wed, 31 Oct 2007 13:26:19 +0530
Subject: [cisco-voip] Asktrik or Open SIP
In-Reply-To: <BAY113-W1543B217F59118332BCAB2E8930@phx.gbl>
References: <BAY113-W1543B217F59118332BCAB2E8930@phx.gbl>
Message-ID: <d4c697ad0710310056i435aa2c5jd5f4bc1d81ac72f6@mail.gmail.com>

Yes, Cisco Voip Gateways work with Asterisk.

Do a little googling and you should find configuration examples of the same.

Aman


On 10/31/07, Syed Khalid Ali <khalid_khursheed at hotmail.com> wrote:
>
>
> hi,
>
> is Asterisk or OpenSER SIP is compatible with Cisco SIP User Agents
> (Gateways)?
>
> regards
>
> syed khalid ali
>
>
> _________________________________________________________________
> Connect to the next generation of MSN Messenger
>
> http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From ranjan.ganesh at gavsin.com  Wed Oct 31 06:29:25 2007
From: ranjan.ganesh at gavsin.com (Ranjan Babu .G)
Date: Wed, 31 Oct 2007 15:59:25 +0530
Subject: [cisco-voip] CCM 6 IP address change
References: <mailman.3916.1193783355.13771.cisco-voip@puck.nether.net>
Message-ID: <6A808E3BD461C94DACA49D21E46E30BC02289C89@mail.gavsin.com>


Hi, 

I'm having a v6.0(1)on my lab in vmware  image.

1.The image having assigned ip .now I want to change the IP address .Any
impact on ip change.

2. How to access the CCMadminpage from CLI of  Linux box.

Thanks and Regards,
Ranjan 


From e.dumbledore at googlemail.com  Wed Oct 31 06:38:23 2007
From: e.dumbledore at googlemail.com (Stift)
Date: Wed, 31 Oct 2007 10:38:23 +0000
Subject: [cisco-voip] Unified CM Telephony Subsystem is Out of Service
Message-ID: <fg9lvg$te2$1@ger.gmane.org>

Hi,

we try to running an CUCCX 5 Installation. Every service is working, 
only the important CM Telephony Subsystem is out of service.

Hi,

we've set up IPCC5 connected to CCM5.1. Everything worked finde, but 
then we found out that we had used a wrong license file and had to 
rebuild the IPCC from scratch (re-installed OS and IPCC).
We deleted the CTI-Ports, the CTI-Routepoints, the "CTI 1"-user and the 
RmCm-User from the CM before the rebuilt.

After the initial setup, it seems to be allright (users are created), 
but the system goes to "partial service", the Telephony subsystem is out 
of service.

The eventlog shows an error message saying that the connection to the 
CTI gets lost.

Any ideas?


From lemon at lemon.za.net  Wed Oct 31 07:14:41 2007
From: lemon at lemon.za.net (Lemon)
Date: Wed, 31 Oct 2007 11:14:41 +0000
Subject: [cisco-voip] 5400 isdn erros
Message-ID: <94e031d30710310414x72246b23ha77fbd00bcf6aa6c@mail.gmail.com>

Hi,

I have the following:

Cisco IOS Software, 5400 Software (C5400-IS-M), Version 12.4(13a), RELEASE SOFTW
ARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2007 by Cisco Systems, Inc.
Compiled Wed 07-Mar-07 01:01 by prod_rel_team

ROM: System Bootstrap, Version 12.3(12r)PI6b, RELEASE SOFTWARE (fc1)

MULCC_VG uptime is 3 hours, 3 minutes
System returned to ROM by reload at 08:07:32 GMT Wed Oct 31 2007
System restarted at 08:08:12 GMT Wed Oct 31 2007
System image file is "flash:c5400-is-mz.124-13a.bin"

Cisco AS5400XM (BCM) processor (revision 0x21) with 393215K/131072K bytes of mem
ory.


and a debug gives me the following:

Oct 31 10:57:08.573: ISDN Se7/0:15 Q931: RX <- CONNECT pd = 8  callref = 0x80B6
        Date/Time i = 0x070A1F0A3900
Oct 31 10:57:08.577: ISDN Se7/0:15 Q931: TX -> CONNECT_ACK pd = 8  callref = 0x0
0B6
Oct 31 10:57:14.085: ISDN Se7/1:15 Q931: RX <- SETUP pd = 8  callref = 0x1430
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98392
                Exclusive, Channel 18
        Progress Ind i = 0x8583 - Origination address is non-ISDN
        Calling Party Number i = 0x2181, '12345678'
                Plan:ISDN, Type:National
        Called Party Number i = 0x81, '1234'
                Plan:ISDN, Type:Unknown
Oct 31 10:57:14.089: ISDN Se7/1:15 **ERROR**: call_incoming: Received a call id
0x7B with a bad bearercap from 14611904 on b channel 18
Oct 31 10:57:14.089: ISDN Se7/1:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x
9430
        Cause i = 0x80D8 - Incompatible destination

Any ideas what could be casuing this

Thnx

From jonas at consign.se  Wed Oct 31 07:26:54 2007
From: jonas at consign.se (Jonas Jonsson)
Date: Wed, 31 Oct 2007 12:26:54 +0100
Subject: [cisco-voip] 5400 isdn erros
In-Reply-To: <94e031d30710310414x72246b23ha77fbd00bcf6aa6c@mail.gmail.com>
References: <94e031d30710310414x72246b23ha77fbd00bcf6aa6c@mail.gmail.com>
Message-ID: <459E635B-139E-4005-8AEF-242DDDF8AE1C@consign.se>

Hi,

Check CCO ref: http://www.cisco.com/warp/public/788/signalling/h323-isdn-callfailure.html

Looks like the same issue.

Bearer Capability i = 0x8090A3
CCO: 0x8090A3 Speech


  		/// Br, Jonas

On 31 okt 2007, at 12.14, Lemon wrote:

> Hi,
>
> I have the following:
>
> Cisco IOS Software, 5400 Software (C5400-IS-M), Version 12.4(13a),  
> RELEASE SOFTW
> ARE (fc1)
> Technical Support: http://www.cisco.com/techsupport
> Copyright (c) 1986-2007 by Cisco Systems, Inc.
> Compiled Wed 07-Mar-07 01:01 by prod_rel_team
>
> ROM: System Bootstrap, Version 12.3(12r)PI6b, RELEASE SOFTWARE (fc1)
>
> MULCC_VG uptime is 3 hours, 3 minutes
> System returned to ROM by reload at 08:07:32 GMT Wed Oct 31 2007
> System restarted at 08:08:12 GMT Wed Oct 31 2007
> System image file is "flash:c5400-is-mz.124-13a.bin"
>
> Cisco AS5400XM (BCM) processor (revision 0x21) with 393215K/131072K  
> bytes of mem
> ory.
>
>
> and a debug gives me the following:
>
> Oct 31 10:57:08.573: ISDN Se7/0:15 Q931: RX <- CONNECT pd = 8   
> callref = 0x80B6
>        Date/Time i = 0x070A1F0A3900
> Oct 31 10:57:08.577: ISDN Se7/0:15 Q931: TX -> CONNECT_ACK pd = 8   
> callref = 0x0
> 0B6
> Oct 31 10:57:14.085: ISDN Se7/1:15 Q931: RX <- SETUP pd = 8  callref  
> = 0x1430
>        Sending Complete
>        Bearer Capability i = 0x8090A3
>                Standard = CCITT
>                Transfer Capability = Speech
>                Transfer Mode = Circuit
>                Transfer Rate = 64 kbit/s
>        Channel ID i = 0xA98392
>                Exclusive, Channel 18
>        Progress Ind i = 0x8583 - Origination address is non-ISDN
>        Calling Party Number i = 0x2181, '12345678'
>                Plan:ISDN, Type:National
>        Called Party Number i = 0x81, '1234'
>                Plan:ISDN, Type:Unknown
> Oct 31 10:57:14.089: ISDN Se7/1:15 **ERROR**: call_incoming:  
> Received a call id
> 0x7B with a bad bearercap from 14611904 on b channel 18
> Oct 31 10:57:14.089: ISDN Se7/1:15 Q931: TX -> RELEASE_COMP pd = 8   
> callref = 0x
> 9430
>        Cause i = 0x80D8 - Incompatible destination
>
> Any ideas what could be casuing this
>
> Thnx
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From BGI at Helios-it.com  Wed Oct 31 07:20:14 2007
From: BGI at Helios-it.com (Benny Giebens)
Date: Wed, 31 Oct 2007 12:20:14 +0100
Subject: [cisco-voip] XML app to register a phone into a line group...??
Message-ID: <1BFA23C9ABCB94458CF57BD0A6BCCBB9028B6F3A@idefix.Helios-it.com>

Hi, 
i'm looking for an application that can be run on a phone thru xml. 
Here is the problem. 
Incoming calls are handled by hunt Pilot/Hunt List/Line groups 
Now we need some kind of handling of the line groups, because the
members change a lot in those line groups. 
Is there some existing xml app, that allows you to register your number
under a line group, so that once registered you are also receiving the
phone calls for that hunt pilot. 
Ideal would be if this is done on the phone (we all use extension
mobility so have all xml capable phones). 
An other solution could be a web page accessible for normal users to
register them self via the webpage (off course not via the admin pages).

Anybody knows if such a solution exisist or can be made? 
Rgds 
Benny Giebens

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From aelnagar at ACT-EG.COM  Wed Oct 31 08:53:33 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Wed, 31 Oct 2007 14:53:33 +0200
Subject: [cisco-voip] Multicast in a campus network
References: <88C8D019DA547C4BABECB9A806ED22ED011DA699@mail.ACT-EG.COM><C1FE15183DA37645BC0633BC604E44F0074023DE@USNAEXCH.na.didata.local>
	<88C8D019DA547C4BABECB9A806ED22ED011DA69A@mail.ACT-EG.COM>
	<FA233791F398B848B978464507739E8FDE3D26@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED011DA6A9@mail.ACT-EG.COM>

thanks for all replies.
 
simply I enabled multicast routing on the backbones, enabled pim on the int vlans that will be a part of the multicast domain.
and just make sure that igmp is enabled on the edge switches
thanks alot

________________________________

From: Ruben Montes (Europe) [mailto:Ruben.Montes at eu.didata.com]
Sent: Sat 27-Oct-07 8:54 PM
To: Ahmed Elnagar; Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Multicast in a campus network



Hello,

 

Inside a vlan and using L3 switches is exactly the same, you should enable pim and multicast routing: I would recommend using sparse mode instead dense mode. I would configure one 6509 as RP (and maybe another one as secondary for redundancy purposes) and all the leaf routers pointing to this router RP address (you can create a Loopback address). For the layer 2 switches (3560), it is enough with igmp snooping, which comes enabled by default...

 

HTH,

 

Ruben

 

________________________________

De: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] En nombre de Ahmed Elnagar
Enviado el: s?bado, 27 de octubre de 2007 17:49
Para: Jason Aarons (US)
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Multicast in a campus network

 

Hello jason

 

thanks alot for your reply, I have read alot thorugh this and I know pim with its modes very well, but these all talks about multicast routing and routers configurations. I didnot pass by any document that explains these in a network where L3 switches is used for inter vlan routing. I read also about MVR but I dont know is it okay for my case or not.

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Sat 27-Oct-07 5:37 PM
To: Ahmed Elnagar
Subject: RE: [cisco-voip] Multicast in a campus network

Research ip pim sparse-dense mode

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Saturday, October 27, 2007 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Multicast in a campus network

 

Hello all;

I dont know if someone dealed before with IPTV system (but I am sure u did). The customer wants to deploy IPTV system (not cisco) and he told me that the system is using multicast for delivering TV stations to clients, I have read alot of documents regarding multiacst but I am not sure which configuration I should use to run this successfully.

The customer network consists of two backbone 6509 switch connected to it 3560 switches as edge switches, if anyone has ideas about this and how to configure these swithces (backbone and edge) to pass multicast traffic.

Also note that there will be clients in vlans other than the server vlan.

thanks

________________________________

Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 

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From KWhitson at cmbtec.com  Wed Oct 31 08:50:41 2007
From: KWhitson at cmbtec.com (Karl Whitson)
Date: Wed, 31 Oct 2007 08:50:41 -0400
Subject: [cisco-voip] Directories URL button
Message-ID: <8A4D67E299AA254CAB2B5985EEDD86583EDF1F@cmb-bm-sbs.CMB-BM.Local>

With Call Manager 5.1 I was wondering if you can assign one of the buttons on the phone to be a one button touch to get to the Personal Fast Dials screen under Directories?  I know that you can assign an IP Phone Service URL to a button, but could I also do the same thing to get tot eh Personal Fast Dials section under the Directories key?  
 
Is it possible to assign the Personal Fast Dials to an IP phone service that points to the section under the Directories key?
 
Karl Whitson
CMB Technologies
DID:  508.422.8413
Cell:  603.930.4189
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From ErasmuE4 at telkom.co.za  Wed Oct 31 09:31:01 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 31 Oct 2007 15:31:01 +0200
Subject: [cisco-voip] XML app to register a phone into a line group...??
In-Reply-To: <1BFA23C9ABCB94458CF57BD0A6BCCBB9028B6F3A@idefix.Helios-it.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01C09E1E@CNTRRA20-XCS01.telkom.co.za>


In call manager express one can dynamically log in and out of a
huntgroup when using a dynamcic huntgroups with wildcards - like:

 

*, *, *  etc would allow three people to login as needed - I use it
for the attendants 

 

Maybe cal manager supports something similar

 

 

Erik Erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Benny Giebens
Sent: Wednesday, October 31, 2007 1:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] XML app to register a phone into a line
group...??

 

Hi, 
i'm looking for an application that can be run on a phone thru xml. 
Here is the problem. 
Incoming calls are handled by hunt Pilot/Hunt List/Line groups 
Now we need some kind of handling of the line groups, because the
members change a lot in those line groups. 
Is there some existing xml app, that allows you to register your
number under a line group, so that once registered you are also
receiving the phone calls for that hunt pilot. 
Ideal would be if this is done on the phone (we all use extension
mobility so have all xml capable phones). 
An other solution could be a web page accessible for normal users to
register them self via the webpage (off course not via the admin
pages). 
Anybody knows if such a solution exisist or can be made? 
Rgds 
Benny Giebens



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail is subject to the Telkom SA electronic communication legal notice, available at :
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From voicenoob at gmail.com  Wed Oct 31 09:39:46 2007
From: voicenoob at gmail.com (Voice Noob)
Date: Wed, 31 Oct 2007 08:39:46 -0500
Subject: [cisco-voip] CallManager 4.x (vs.) 6.x
In-Reply-To: <CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us>
	<535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com>
	<CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>
Message-ID: <535625f70710310639x2ba2ace7k9a6d3081e663210c@mail.gmail.com>

Well you can't get access to the OS like you can with the Windows version. I
would call that being locked out.


On 10/30/07, Vince Loschiavo <VLoschiavo at data-corporation.com> wrote:
>
>  ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Voice Noob
> *Sent:* Tuesday, October 30, 2007 2:29 PM
> *To:* Leetun, Rob
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] CallManager 4.x (vs.) 6.x
>
>
>  I think the windows version is better because it does not lock you out of
> the system like the Linux version. you have more control.
>
> On 10/30/07, Leetun, Rob <rleetun at co.boulder.co.us> wrote:
> >
> >  Hello all,
> >
> > We are having a discussion about why is a Windows platform better than a
> > Linux platform when it comes to CallManager , or is Linux better than
> > Windows?  In your opinions, is it better to go from 4.x to 6.x or wait
> > for 7.x to roll out next summer?  Or is it better to stay on the 4.xtrain?  How many of you have switched to the Linux
> > platform from the Windows platform?
> >
> > Thanks in advance.
> >
> > Rob
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> All,
>
> I've found both the windows and linux versions to be quite stable.
>
> I haven't been locked out of the linux platform...so I can't speak to
> that, but I do like the method of patching on linux.  Seems much cleaner to
> me.  (Plus the ability to roll back from a patch).
>
> (I'm also an avid *nix user) :)
>
> -V
>
> ____________________________________________________________
> Vincent Loschiavo
> Cisco Practice Director
> Datacorp
> 18503 Pines Blvd
> Suite 213
> Pembroke Pines, FL, 33029
>
> "Cisco Advanced Unified Communications Partner"
> Cell Phone:       786-282-1164
> Office Phone:    954-671-5669
> Office Main:      305-594-6933
> ____________________________________________________________
>
>
>
>
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From rsingleton at morsco.com  Wed Oct 31 11:20:31 2007
From: rsingleton at morsco.com (Robert)
Date: Wed, 31 Oct 2007 10:20:31 -0500
Subject: [cisco-voip] Phone registration status
Message-ID: <1193844031.7955.15.camel@robert.morsco.com>

Greetings, All!

I have need to be able to see if a given telephone is registered. I
presume there is a way to check in the database but have not found
anything that seems obvious. Is there an SQL field that shows
registration status and optionally, the IP address of the device?

Thanks!

Robert


From Matt.Slaga at us.didata.com  Wed Oct 31 11:24:02 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 31 Oct 2007 11:24:02 -0400
Subject: [cisco-voip] Phone registration status
In-Reply-To: <1193844031.7955.15.camel@robert.morsco.com>
References: <1193844031.7955.15.camel@robert.morsco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F007499DE8@USNAEXCH.na.didata.local>

Registration status is not stored in the database, but in RIS.  The only
way to get this information is to either get it from AXL or from the
phone's web service.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Sent: Wednesday, October 31, 2007 11:21 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Phone registration status

Greetings, All!

I have need to be able to see if a given telephone is registered. I
presume there is a way to check in the database but have not found
anything that seems obvious. Is there an SQL field that shows
registration status and optionally, the IP address of the device?

Thanks!

Robert

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From jason.aarons at us.didata.com  Wed Oct 31 11:23:46 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 31 Oct 2007 11:23:46 -0400
Subject: [cisco-voip] Unity Connection 2.0 Listen to VM want to forward by
	extension not dial-by-name
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us><535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com><CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>
	<535625f70710310639x2ba2ace7k9a6d3081e663210c@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E925@USNAEXCH.na.didata.local>

After listening to a message in Unity Connection 2.0 I wish to enter a extension to forward a message to, instead the system requires I dial-by-name the extension I want to forward the message to.
 
Can't I just enter a extension? Service setting somewhere?

	 




-----------------------------------------
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From wsisk at cisco.com  Wed Oct 31 11:26:24 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 31 Oct 2007 10:26:24 -0500
Subject: [cisco-voip] Phone registration status
In-Reply-To: <1193844031.7955.15.camel@robert.morsco.com>
References: <1193844031.7955.15.camel@robert.morsco.com>
Message-ID: <47289EA0.6000809@cisco.com>

Hi Robert,

Device registration status is not stored in the SQL database.  It is far 
too 'realtime' to log in permanent storage.

That said there are a few ways to query device registration status:
1. mmfspy from the command prompt of the CM server
2. SNMP looking at the device table in the CCM mib.  It gives the same 
information as mmfspy
3. http://<server>/ccmadmin/reports/DeviceListX.asp gives the same 
information as mmfspy

DeviceListX.asp is only available in 4.x.  It is removed in CM5.x and 
replaced with the AXL Serviceability kit.

/Wes

Robert wrote:
> Greetings, All!
>
> I have need to be able to see if a given telephone is registered. I
> presume there is a way to check in the database but have not found
> anything that seems obvious. Is there an SQL field that shows
> registration status and optionally, the IP address of the device?
>
> Thanks!
>
> Robert
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From lelio at uoguelph.ca  Wed Oct 31 11:26:27 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 31 Oct 2007 11:26:27 -0400
Subject: [cisco-voip] Phone registration status
References: <1193844031.7955.15.camel@robert.morsco.com>
	<C1FE15183DA37645BC0633BC604E44F007499DE8@USNAEXCH.na.didata.local>
Message-ID: <01da01c81bd2$67a50810$56196883@cfs.uoguelph.ca>

you can also see the registration status from the callmanager web page.

but you probably already knew that.

----- Original Message ----- 
From: "Matt Slaga (US)" <Matt.Slaga at us.didata.com>
To: "Robert" <rsingleton at morsco.com>; <cisco-voip at puck.nether.net>
Sent: Wednesday, October 31, 2007 11:24 AM
Subject: Re: [cisco-voip] Phone registration status


> Registration status is not stored in the database, but in RIS.  The only
> way to get this information is to either get it from AXL or from the
> phone's web service.
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
> Sent: Wednesday, October 31, 2007 11:21 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Phone registration status
> 
> Greetings, All!
> 
> I have need to be able to see if a given telephone is registered. I
> presume there is a way to check in the database but have not found
> anything that seems obvious. Is there an SQL field that shows
> registration status and optionally, the IP address of the device?
> 
> Thanks!
> 
> Robert
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> -----------------------------------------
> Disclaimer:
> 
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From Brian.VanBenschoten at inacom.com  Wed Oct 31 11:50:22 2007
From: Brian.VanBenschoten at inacom.com (Brian Van Benschoten)
Date: Wed, 31 Oct 2007 10:50:22 -0500
Subject: [cisco-voip] Wes- Can you forward this suggestion on to the Call
	Manger BU or whoever manages the ordering tool - CUCM-BE can
	only select 10 packs of DLUs
Message-ID: <153D0FAADC2C90459D202D16D24A94AE0A70C5B9@exch1.inacom-msn.com>

Cisco Team:

When ordering CUCM-BE you can only select from 10-packs of DLUs.  We
usually end up with 15-30 PAK codes to individually register on CCO.
This takes time away from engineering that can be spent working with the
client.

Can the ordering tool be  modified to allow a range of DLU quantities ?

 

Can you please forward on to the correct BU manager ?

Is there a external mailer for the CM business unit ?

Thanks

 

<screen capture below>

 

This is what I get to pick from to configure/order it properly - 10
packs...

 

 

 

 

________________________________________

Brian Van Benschoten 

Practice Leader - Unified Communications

Consulting Engineer - CCIE # 5421

Inacom Information Systems

3001 West Beltline Highway

Madison, WI  53713  USA

608-661-7780 (IP Phone)

608-661-7701 (FAX)

608-661-7777 (Customer Service)

brian.vanbenschoten at inacom.com

http://www.inacom.com <http://www.inacom.com/> 

 

 

 

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From jsteinberg at gmail.com  Wed Oct 31 12:33:11 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 31 Oct 2007 12:33:11 -0400
Subject: [cisco-voip] XML app to register a phone into a line group...??
In-Reply-To: <1BFA23C9ABCB94458CF57BD0A6BCCBB9028B6F3A@idefix.Helios-it.com>
References: <1BFA23C9ABCB94458CF57BD0A6BCCBB9028B6F3A@idefix.Helios-it.com>
Message-ID: <f1b65f880710310933x1044eb24s980106036d44e3fb@mail.gmail.com>

hunt group 'hlog' softkey in CM4.2 or CM6.

On 10/31/07, Benny Giebens <BGI at helios-it.com> wrote:
>
>
>
>
> Hi,
>  i'm looking for an application that can be run on a phone thru xml.
>  Here is the problem.
>  Incoming calls are handled by hunt Pilot/Hunt List/Line groups
>  Now we need some kind of handling of the line groups, because the members
> change a lot in those line groups.
>  Is there some existing xml app, that allows you to register your number
> under a line group, so that once registered you are also receiving the phone
> calls for that hunt pilot.
>  Ideal would be if this is done on the phone (we all use extension mobility
> so have all xml capable phones).
>  An other solution could be a web page accessible for normal users to
> register them self via the webpage (off course not via the admin pages).
>  Anybody knows if such a solution exisist or can be made?
>  Rgds
>  Benny Giebens
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From abate at dei.unipd.it  Wed Oct 31 11:39:01 2007
From: abate at dei.unipd.it (abate at dei.unipd.it)
Date: Wed, 31 Oct 2007 16:39:01 +0100
Subject: [cisco-voip] unsubscribe from mailing list please
Message-ID: <20071031163901.0427fvfxw8gcgc88@mail.dei.unipd.it>



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.


From BGI at Helios-it.com  Wed Oct 31 12:43:06 2007
From: BGI at Helios-it.com (Benny Giebens)
Date: Wed, 31 Oct 2007 17:43:06 +0100
Subject: [cisco-voip] XML app to register a phone into a line group...??
In-Reply-To: <f1b65f880710310933x1044eb24s980106036d44e3fb@mail.gmail.com>
References: <1BFA23C9ABCB94458CF57BD0A6BCCBB9028B6F3A@idefix.Helios-it.com>
	<f1b65f880710310933x1044eb24s980106036d44e3fb@mail.gmail.com>
Message-ID: <1BFA23C9ABCB94458CF57BD0A6BCCBB9028B711A@idefix.Helios-it.com>

Ok,
I see .... but is there somewhere info on how this key acts?
In what hunt list/group does the phone logsin, etc....

Rgds
Benny


-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: woensdag 31 oktober 2007 17:33
To: Benny Giebens
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] XML app to register a phone into a line
group...??

hunt group 'hlog' softkey in CM4.2 or CM6.

On 10/31/07, Benny Giebens <BGI at helios-it.com> wrote:
>
>
>
>
> Hi,
>  i'm looking for an application that can be run on a phone thru xml.
>  Here is the problem.
>  Incoming calls are handled by hunt Pilot/Hunt List/Line groups
>  Now we need some kind of handling of the line groups, because the
members
> change a lot in those line groups.
>  Is there some existing xml app, that allows you to register your
number
> under a line group, so that once registered you are also receiving the
phone
> calls for that hunt pilot.
>  Ideal would be if this is done on the phone (we all use extension
mobility
> so have all xml capable phones).
>  An other solution could be a web page accessible for normal users to
> register them self via the webpage (off course not via the admin
pages).
>  Anybody knows if such a solution exisist or can be made?
>  Rgds
>  Benny Giebens
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From axe7961 at hotmail.com  Wed Oct 31 12:50:23 2007
From: axe7961 at hotmail.com (Axel Wilzinski)
Date: Wed, 31 Oct 2007 12:50:23 -0400
Subject: [cisco-voip] How to set default PRI destination to Operator using
	MGCP
Message-ID: <BLU109-W58E0046EA380F2A37306D6BA930@phx.gbl>

How can I set all unassigned DN's that come in on a PRI to a default 4 digit extension?  I'm going crazy trying to figure this out.  Basically, I want all unassigned numbers in our DID range to automatically go to our operator.
 
We are using CCM 4.2.3 with MGCP gateways.
 
Thanks in advance.
Axel 
_________________________________________________________________
Climb to the top of the charts!? Play Star Shuffle:? the word scramble challenge with star power.
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From rsingleton at morsco.com  Wed Oct 31 12:59:11 2007
From: rsingleton at morsco.com (Robert)
Date: Wed, 31 Oct 2007 11:59:11 -0500
Subject: [cisco-voip] Phone registration status
In-Reply-To: <01da01c81bd2$67a50810$56196883@cfs.uoguelph.ca>
References: <1193844031.7955.15.camel@robert.morsco.com>
	<C1FE15183DA37645BC0633BC604E44F007499DE8@USNAEXCH.na.didata.local>
	<01da01c81bd2$67a50810$56196883@cfs.uoguelph.ca>
Message-ID: <1193849951.7955.37.camel@robert.morsco.com>

On Wed, 2007-10-31 at 11:26 -0400, Lelio Fulgenzi wrote:
> you can also see the registration status from the callmanager web page.
> but you probably already knew that.

hehehhee Yeah... I need a non-interactive way

On Wed, 2007-10-31 at 10:26 -0500, Wes Sisk wrote:

> That said there are a few ways to query device registration status:
> 1. mmfspy from the command prompt of the CM server
> 2. SNMP looking at the device table in the CCM mib.  It gives the
> same information as mmfspy
> 3. http://<server>/ccmadmin/reports/DeviceListX.asp gives the same 
> information as mmfspy

mmfspy - spiffy tool, very quick, but interactive. :/

SNMP - will have to investigate this, but at first glance, this may be a
good bet for what I'm doing here...

DeviceListX.asp - also spiffy and I can wget the info, though it does
not come back quickly since it gives me the status of not just one
device but all of them. :) Takes about 7.5 seconds for 1144 lines.

I have been moving all my command line tools to a web server so that
more than just the two admins can benefit from them. In the process, I
have combined several of them and added some functionality. Every time I
add something cool, it makes me want more and registration status and/or
the phone's IP is the next bit I'd like to see on the telephone info
page :)

Thanks, especially to Wes, for all the help!

Robert




From lemon at lemon.za.net  Wed Oct 31 13:28:14 2007
From: lemon at lemon.za.net (Lemon)
Date: Wed, 31 Oct 2007 17:28:14 +0000
Subject: [cisco-voip] How to set default PRI destination to Operator
	using MGCP
In-Reply-To: <BLU109-W58E0046EA380F2A37306D6BA930@phx.gbl>
References: <BLU109-W58E0046EA380F2A37306D6BA930@phx.gbl>
Message-ID: <94e031d30710311028l5df1b66do9d53b329fc73b1c7@mail.gmail.com>

On 10/31/07, Axel Wilzinski <axe7961 at hotmail.com> wrote:
> How can I set all unassigned DN's that come in on a PRI to a default 4 digit
> extension?  I'm going crazy trying to figure this out.  Basically, I want
> all unassigned numbers in our DID range to automatically go to our operator.

create a translation pattern. lets say your extension range is 1000 -
1499, then create a translation pattern like 1XXX and have it ring at
the operator.

From lelio at uoguelph.ca  Wed Oct 31 13:43:58 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 31 Oct 2007 13:43:58 -0400
Subject: [cisco-voip] Phone registration status
References: <1193844031.7955.15.camel@robert.morsco.com><C1FE15183DA37645BC0633BC604E44F007499DE8@USNAEXCH.na.didata.local><01da01c81bd2$67a50810$56196883@cfs.uoguelph.ca>
	<1193849951.7955.37.camel@robert.morsco.com>
Message-ID: <02b101c81be5$9de991d0$56196883@cfs.uoguelph.ca>

OK.

You can also set up alarms using CallManager alarms and a syslog host and 
some processing scrips.

----- Original Message ----- 
From: "Robert" <rsingleton at morsco.com>
To: <cisco-voip at puck.nether.net>
Sent: Wednesday, October 31, 2007 12:59 PM
Subject: Re: [cisco-voip] Phone registration status


> On Wed, 2007-10-31 at 11:26 -0400, Lelio Fulgenzi wrote:
>> you can also see the registration status from the callmanager web page.
>> but you probably already knew that.
>
> hehehhee Yeah... I need a non-interactive way
>
> On Wed, 2007-10-31 at 10:26 -0500, Wes Sisk wrote:
>
>> That said there are a few ways to query device registration status:
>> 1. mmfspy from the command prompt of the CM server
>> 2. SNMP looking at the device table in the CCM mib.  It gives the
>> same information as mmfspy
>> 3. http://<server>/ccmadmin/reports/DeviceListX.asp gives the same
>> information as mmfspy
>
> mmfspy - spiffy tool, very quick, but interactive. :/
>
> SNMP - will have to investigate this, but at first glance, this may be a
> good bet for what I'm doing here...
>
> DeviceListX.asp - also spiffy and I can wget the info, though it does
> not come back quickly since it gives me the status of not just one
> device but all of them. :) Takes about 7.5 seconds for 1144 lines.
>
> I have been moving all my command line tools to a web server so that
> more than just the two admins can benefit from them. In the process, I
> have combined several of them and added some functionality. Every time I
> add something cool, it makes me want more and registration status and/or
> the phone's IP is the next bit I'd like to see on the telephone info
> page :)
>
> Thanks, especially to Wes, for all the help!
>
> Robert
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From ducotedt at stfran.com  Wed Oct 31 10:01:26 2007
From: ducotedt at stfran.com (Darren Ducote)
Date: Wed, 31 Oct 2007 09:01:26 -0500
Subject: [cisco-voip] IPCC express 4.5 icd.aef script manipulation
In-Reply-To: <4727ca02.157f400a.7d36.735a@mx.google.com>
Message-ID: <A46FC8D5BB84EB4D9B0CE8CBDCDCD06517464BBF@xch07.stfran.net>

Thanks everyone for the help.

By the time this email went out, I have the script completed. We went from box to complete installation in 3 days without ever seeing the product.

We ended up using the "menu" option. If anyone would like a copy just let me know!

This site is great by the way.

________________________________
From: Dustin S Fowler [mailto:dustin.s.fowler at gmail.com]
Sent: Tuesday, October 30, 2007 7:19 PM
To: Darren Ducote; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC express 4.5 icd.aef script manipulation

I think this might be what you are looking for.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Darren Ducote
Sent: Tuesday, October 30, 2007 1:33 PM
To: 'cisco-voip at puck.nether.net'
Subject: [cisco-voip] IPCC express 4.5 icd.aef script manipulation

Hello everyone,

We have installed IPCCx 4.5 and use CCM5.

We have a service desk and would like to use the icd.aef script with a few modifications. I would like the play a welcome message and then dump everyone into a queue to wait for available agents. During this time, I would like to play a message that will allow the user to press "1" to leave a message.

At this time it would be a call forward to our voicemail pilot number.

Anyone doing anything like this? I have made some modification with moderate success using the "get digits" string. Thanks!

________________________________
This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission.

***St. Francis Medical Center and affiliates encrypt email containing Patient Health Information. You will receive a notice with instructions on how to open any encrypted email you receive. To learn more about this click Here<http://userawareness.zixcorp.com/sites/index.php?b=141&type=1>

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This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission.


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From david.peck1 at us.army.mil  Wed Oct 31 14:07:20 2007
From: david.peck1 at us.army.mil (Peck, David Mr CIV USA AMC)
Date: Wed, 31 Oct 2007 14:07:20 -0400
Subject: [cisco-voip]  OS 2000.4.4 breaks the SCSI array?
Message-ID: <E10CBD3DEEB2AA43AE13915F630F76BFD87DBD@wate010be000001.nae.ds.army.mil>

Did you ever get this resolved? I am having the same issue with one of
my Call managers.

Thanks
Dave

From jgrace at adcapnet.com  Wed Oct 31 14:51:47 2007
From: jgrace at adcapnet.com (James Grace)
Date: Wed, 31 Oct 2007 14:51:47 -0400
Subject: [cisco-voip] trying to access pca unity 2 error
Message-ID: <49AFE548BC7F7E49A1A4A2F28AF1914901E90C@mail3.adcapnetsvc.local>

?
when i try to loging to unity con 2.0 pca, i get 
 
type Status report

message /ciscopca/home.do

description The requested resource (/ciscopca/home.do) is not available.

Have anybody ran into this

 

James 
 
<http://www.adcapnet.com/>  
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
recipient is prohibited. If you received this message in error, please contact the sender and delete the
material from all computers."

This original email was sent to the internet for delivery at 31 Oct 2007 14:52:39 -0400
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From rsingleton at morsco.com  Wed Oct 31 15:28:10 2007
From: rsingleton at morsco.com (Robert)
Date: Wed, 31 Oct 2007 14:28:10 -0500
Subject: [cisco-voip] Phone registration status
In-Reply-To: <02b101c81be5$9de991d0$56196883@cfs.uoguelph.ca>
References: <1193844031.7955.15.camel@robert.morsco.com>
	<C1FE15183DA37645BC0633BC604E44F007499DE8@USNAEXCH.na.didata.local>
	<01da01c81bd2$67a50810$56196883@cfs.uoguelph.ca>
	<1193849951.7955.37.camel@robert.morsco.com>
	<02b101c81be5$9de991d0$56196883@cfs.uoguelph.ca>
Message-ID: <1193858890.7955.56.camel@robert.morsco.com>

On Wed, 2007-10-31 at 13:43 -0400, Lelio Fulgenzi wrote:
> OK.
> 
> You can also set up alarms using CallManager alarms and a syslog host and 
> some processing scrips.

That is on my todo list as well, to get notification when a phone goes
unregistered. Could be handy for a lot of things.

The immediate need however is for my web app to be able look up the
current status of a given phone when I pull up the info on that DN.

Thanks again!

Robert


From jonvoip at gmail.com  Wed Oct 31 16:45:07 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 31 Oct 2007 15:45:07 -0500
Subject: [cisco-voip] Error message on an NFAS gateway...
Message-ID: <5d093f9a0710311345t75ab4f64p212f0753f55a10b8@mail.gmail.com>

It is a 2811...

Inbound works fine, outbound gets this message:


Oct 31 20:43:20.352: %CALL_CONTROL-6-NUMBERING_TYPE_NO_MATCH: The
called number numbering type did not matchThe event for the callid
1000 is being discarded.

Here is dial-peer 1000

dial-peer voice 1000 pots
 numbering-type national
 preference 1
 destination-pattern 1[2-9]..[2-9]......
 port 0/0/1:23
 forward-digits 10


Anyone know what the problem is?



Jonathan

From jsteinberg at gmail.com  Wed Oct 31 16:50:50 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 31 Oct 2007 16:50:50 -0400
Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
In-Reply-To: <f1b65f880710261028q35bca6c5l4ed73dbc7ecd62d4@mail.gmail.com>
References: <f1b65f880710160917p769cb591u8d02af6e50d381e6@mail.gmail.com>
	<FDC137145729AD438CDCA0F9FC6F7236013FE273@mail.netechcorp.com>
	<f1b65f880710161505m5989eed5leec418d9ecd76b96@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0071FFC39@USNAEXCH.na.didata.local>
	<f1b65f880710261028q35bca6c5l4ed73dbc7ecd62d4@mail.gmail.com>
Message-ID: <f1b65f880710311350q6b970e43kcc1d525176442e8@mail.gmail.com>

Anybody able to get CUPS exchange calendaring to work when exchange
users are located on more than one exchange server?

I can only configure a single exchange server in CUPS.  Calendaring
data for users on this server works.  However, I haven't had any luck
with calendaring data integrating to CUPS when the users are located
on another exchange server.

thx

justin

On 10/26/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
> I finally had some more time today to look back into this issue and
> have been able to get this to work with 6.0.1.
>
> -Create a Enterprise Root Microsoft Certificate Services on a W2K3
> domain member server.
> -Upload this root CA certificate to CUPS as a 'presence-trust' certificate.
> -Add this root CA to your domain security group policy.  push group
> policy to the clients.
> -Create a CSR on my Exchange box via IIS.  Had this CSR signed by the
> enterprise root CA.
> -Upload the signed cert back to IIS on exchange. restart iis.
> -Configure CUPS presence engine for Outlook.  The common name is the
> name of the presence-trust certificate as viewed in the CUPS OS admin
> certificate page.
> -Rebooted presence.
>
> It works - pretty cool too.
>
>
> Justin
>
> On 10/17/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> > I checked with the BU when I had trouble getting it to work.  The
> > response I received from their group is that is it not supported until
> > 6.0(2).  It is not documented anywhere.  Have any of you been able to
> > get this part to work?
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> > Steinberg
> > Sent: Tuesday, October 16, 2007 6:06 PM
> > To: Jason Burton
> > Cc: cisco voip list
> > Subject: Re: [cisco-voip] CUPS6 integration to Exchange Calendaring
> >
> > Jason, Are you speaking about the SSL cert that I use for url such as:
> >
> > https://exchangeserver/Exchange/Jsteinberg/Calendar/
> >
> >
> > Matt, where did you hear it wasn't working in CUPS 6.0.1 ?  It is
> > mentioned in the release notes for 6.0.1.  Is there a bug filed
> > against it?
> >
> > http://www.cisco.com/en/US/docs/voice_ip_comm/cups/6_0_1/rel_notes/cup60
> > 1rn.html#wp478329
> >
> >
> > On 10/16/07, Jason Burton <jburton at netechcorp.com> wrote:
> > > I ran into a bug with it.  Are you using an extended validation SSL
> > cert
> > > (the one that changes IE7's address bar to green)?
> > >
> > > _____________________________
> > > Jason Burton
> > > Systems Engineer
> > > CCIE Voice #17319
> > > NETech Corporation
> > > 317.660.3010 - Desk
> > > 317.581.1919 - Fax
> > > jburton at netechcorp.com
> > >
> > >
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> > > Steinberg
> > > Sent: Tuesday, October 16, 2007 12:17 PM
> > > To: cisco voip list
> > > Subject: [cisco-voip] CUPS6 integration to Exchange Calendaring
> > >
> > > Has anyone been successful with this?
> > >
> > > I am trying to get this to work but can't figure out what the 'Trust
> > > Certificate Subject CN' field is for on the presence gateway page when
> > > integrating to outlook.
> > >
> > > thanks.
> > > justin
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the
> > designated addressee(s) named above only.  If you are not the
> > intended addressee, you are hereby notified that you have received
> > this communication in error and that any use or reproduction of
> > this email or its contents is strictly prohibited and may be
> > unlawful.  If you have received this communication in error, please
> > notify us immediately by replying to this message and deleting it
> > from your computer. Thank you.
> >
>

From jsteinberg at gmail.com  Wed Oct 31 16:53:10 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 31 Oct 2007 16:53:10 -0400
Subject: [cisco-voip] Error message on an NFAS gateway...
In-Reply-To: <5d093f9a0710311345t75ab4f64p212f0753f55a10b8@mail.gmail.com>
References: <5d093f9a0710311345t75ab4f64p212f0753f55a10b8@mail.gmail.com>
Message-ID: <f1b65f880710311353u68d402cdr346cf874cac22c50@mail.gmail.com>

why do you have 'numbering-type' under the POTS dial-peer?



On 10/31/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> It is a 2811...
>
> Inbound works fine, outbound gets this message:
>
>
> Oct 31 20:43:20.352: %CALL_CONTROL-6-NUMBERING_TYPE_NO_MATCH: The
> called number numbering type did not matchThe event for the callid
> 1000 is being discarded.
>
> Here is dial-peer 1000
>
> dial-peer voice 1000 pots
>  numbering-type national
>  preference 1
>  destination-pattern 1[2-9]..[2-9]......
>  port 0/0/1:23
>  forward-digits 10
>
>
> Anyone know what the problem is?
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From pat-cv at wcyv.com  Wed Oct 31 19:49:44 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Wed, 31 Oct 2007 19:49:44 -0400
Subject: [cisco-voip] Unity Connection 2.0 Listen to VM want to forward
 by	extension not dial-by-name
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E925@USNAEXCH.na.didata.local>
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us><535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com><CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>	<535625f70710310639x2ba2ace7k9a6d3081e663210c@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F013E925@USNAEXCH.na.didata.local>
Message-ID: <47291498.6000202@wcyv.com>

Try the 'Send Message Settings' page for your user account.

-------- Original Message  --------
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Connection 2.0 Listen to VM want to 
forward by	extension not dial-by-name
Date: 10/31/2007 11:23 AM

> After listening to a message in Unity Connection 2.0 I wish to enter a 
> extension to forward a message to, instead the system requires I 
> dial-by-name the extension I want to forward the message to.
>  
> Can't I just enter a extension? Service setting somewhere?
> 
>      
> 
> ------------------------------------------------------------------------
> 
> * Disclaimer: This e-mail communication and any attachments may contain 
> confidential and privileged information and is for use by the designated 
> addressee(s) named above only. If you are not the intended addressee, 
> you are hereby notified that you have received this communication in 
> error and that any use or reproduction of this email or its contents is 
> strictly prohibited and may be unlawful. If you have received this 
> communication in error, please notify us immediately by replying to this 
> message and deleting it from your computer. Thank you. *
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From pat-cv at wcyv.com  Wed Oct 31 19:55:41 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Wed, 31 Oct 2007 19:55:41 -0400
Subject: [cisco-voip] trying to access pca unity 2 error
In-Reply-To: <49AFE548BC7F7E49A1A4A2F28AF1914901E90C@mail3.adcapnetsvc.local>
References: <49AFE548BC7F7E49A1A4A2F28AF1914901E90C@mail3.adcapnetsvc.local>
Message-ID: <472915FD.5010609@wcyv.com>

I've seen similar when the services are still starting. Check the UC 
serviceability page and verify everything is up.

-------- Original Message  --------
From: "James Grace" <jgrace at adcapnet.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] trying to access pca unity 2 error
Date: 10/31/2007 2:51 PM

> ?
> when i try to loging to unity con 2.0 pca, i get
>  
> 
> *type* Status report
> 
> *message* _/ciscopca/home.do_
> 
> *description* _The requested resource (/ciscopca/home.do) is not available._
> 
> _Have anybody ran into this_
> 
> __ 
> 
> James
>  
> <http://www.adcapnet.com/> 
> 
> Disclaimer For ADCAP Network Systems
> 
> "The information transmitted is intended only for the person or entity to which it is addressed and may
> contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
> or taking of any action in reliance upon this information by persons or entities other than the intended
> recipient is prohibited. If you received this message in error, please contact the sender and delete the
> material from all computers."
> 
> This original email was sent to the internet for delivery at 31 Oct 2007 14:52:39 -0400
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From greencucumber2007 at yahoo.com  Wed Oct 31 19:56:00 2007
From: greencucumber2007 at yahoo.com (Cucumber Green)
Date: Wed, 31 Oct 2007 16:56:00 -0700 (PDT)
Subject: [cisco-voip] Phone registration log
Message-ID: <995995.9541.qm@web44802.mail.sp1.yahoo.com>

Is there any log where I can go and see at what time certain phones got registered with callmanager? (using CUCM 4.2(3) )
 __________________________________________________
Do You Yahoo!?
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From jason.aarons at us.didata.com  Wed Oct 31 21:48:17 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 31 Oct 2007 21:48:17 -0400
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)
In-Reply-To: <47291498.6000202@wcyv.com>
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us><535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com><CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>	<535625f70710310639x2ba2ace7k9a6d3081e663210c@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F013E925@USNAEXCH.na.didata.local>
	<47291498.6000202@wcyv.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0074D8863@USNAEXCH.na.didata.local>

Below was a comment from a manager moving from Nortel Option 61c to
Cisco CallManager 5.1.2, I'm don't think the "live keypad" is a feature
in callmanager, if you enter a extension of another user you must press
the dial button when done.


"One major loss from a user standpoint has been that the keypad is not
live. In the Nortel system you could enter another phone extension 5846
and it would just dial without any additional effort, therefore calling
my extension. Each user has one extra step for every call they make by
pressing Dial softkey in Cisco. I am surprised that Cisco can not
program the same defaults." -customer



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From Andrew.Riley at getronics.com  Wed Oct 31 22:03:45 2007
From: Andrew.Riley at getronics.com (Riley, Andrew)
Date: Thu, 1 Nov 2007 13:03:45 +1100
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us><535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com><CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>	<535625f70710310639x2ba2ace7k9a6d3081e663210c@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F013E925@USNAEXCH.na.didata.local>
	<47291498.6000202@wcyv.com>
	<C1FE15183DA37645BC0633BC604E44F0074D8863@USNAEXCH.na.didata.local>
Message-ID: <A49371C6FF101B46BD3E3A8C89985F023DCAAE@excausy200.australia.unity>

lift the handset (like traditional phones) dial the number and call goes through.
 
Where is the extra step?

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Thu 1/11/2007 12:48 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)



Below was a comment from a manager moving from Nortel Option 61c to
Cisco CallManager 5.1.2, I'm don't think the "live keypad" is a feature
in callmanager, if you enter a extension of another user you must press
the dial button when done.


"One major loss from a user standpoint has been that the keypad is not
live. In the Nortel system you could enter another phone extension 5846
and it would just dial without any additional effort, therefore calling
my extension. Each user has one extra step for every call they make by
pressing Dial softkey in Cisco. I am surprised that Cisco can not
program the same defaults." -customer



-----------------------------------------
Disclaimer:

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From wsisk at cisco.com  Wed Oct 31 22:20:44 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 31 Oct 2007 21:20:44 -0500
Subject: [cisco-voip] Phone registration log
In-Reply-To: <995995.9541.qm@web44802.mail.sp1.yahoo.com>
References: <995995.9541.qm@web44802.mail.sp1.yahoo.com>
Message-ID: <472937FC.4010106@cisco.com>

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From jason.aarons at us.didata.com  Wed Oct 31 22:22:38 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 31 Oct 2007 22:22:38 -0400
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)
In-Reply-To: <A49371C6FF101B46BD3E3A8C89985F023DCAAE@excausy200.australia.unity>
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us><535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com><CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>	<535625f70710310639x2ba2ace7k9a6d3081e663210c@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F013E925@USNAEXCH.na.didata.local>
	<47291498.6000202@wcyv.com>
	<C1FE15183DA37645BC0633BC604E44F0074D8863@USNAEXCH.na.didata.local>
	<A49371C6FF101B46BD3E3A8C89985F023DCAAE@excausy200.australia.unity>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0074D8881@USNAEXCH.na.didata.local>

He wants specifically to enter an extension number on the keypad without pressing the dial softkey and have it call the extension. The Nortel automatically knows a extension has been answer, place the phone in speakerphone mode and connects the call. I'm sure he will get used to pressing the dial softkey in time.

Your extra step is lifting the handset <grin>.


-----Original Message-----
From: Riley, Andrew [mailto:Andrew.Riley at getronics.com] 
Sent: Wednesday, October 31, 2007 10:04 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Nortel vs Cisco (live keypad feature)

lift the handset (like traditional phones) dial the number and call goes through.
 
Where is the extra step?

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Thu 1/11/2007 12:48 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)



Below was a comment from a manager moving from Nortel Option 61c to
Cisco CallManager 5.1.2, I'm don't think the "live keypad" is a feature
in callmanager, if you enter a extension of another user you must press
the dial button when done.


"One major loss from a user standpoint has been that the keypad is not
live. In the Nortel system you could enter another phone extension 5846
and it would just dial without any additional effort, therefore calling
my extension. Each user has one extra step for every call they make by
pressing Dial softkey in Cisco. I am surprised that Cisco can not
program the same defaults." -customer



-----------------------------------------
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_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




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From wsisk at cisco.com  Wed Oct 31 22:28:13 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 31 Oct 2007 21:28:13 -0500
Subject: [cisco-voip] Phone registration status
In-Reply-To: <1193849951.7955.37.camel@robert.morsco.com>
References: <1193844031.7955.15.camel@robert.morsco.com>	<C1FE15183DA37645BC0633BC604E44F007499DE8@USNAEXCH.na.didata.local>	<01da01c81bd2$67a50810$56196883@cfs.uoguelph.ca>
	<1193849951.7955.37.camel@robert.morsco.com>
Message-ID: <472939BD.9050805@cisco.com>

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From jason.aarons at us.didata.com  Wed Oct 31 22:44:45 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 31 Oct 2007 22:44:45 -0400
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)
In-Reply-To: <1495a01c81c30$4fbb9661$3c84a8c0@europe.unity>
References: <1495a01c81c30$4fbb9661$3c84a8c0@europe.unity>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0074D888A@USNAEXCH.na.didata.local>

Correct, Somehow Nortel knows you've entered a matching extension when you press 5656 and automatically does speakerphone and dials. He only pressed "5656" and got speakerphone and dialing.

-----Original Message-----
From: Riley, Andrew [mailto:Andrew.Riley at getronics.com] 
Sent: Wednesday, October 31, 2007 10:39 PM
To: Jason Aarons (US)
Subject: RE: [cisco-voip] Nortel vs Cisco (live keypad feature)

Understood. So the nortel feature behaves as if the phone was always off hook

-----Original Message-----
From: "Jason Aarons (US)" <jason.aarons at us.didata.com>
To: "Riley, Andrew" <Andrew.Riley at getronics.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Sent: 1/11/07 13:23
Subject: RE: [cisco-voip] Nortel vs Cisco (live keypad feature)

He wants specifically to enter an extension number on the keypad without pressing the dial softkey and have it call the extension. The Nortel automatically knows a extension has been answer, place the phone in speakerphone mode and connects the call. I'm sure he will get used to pressing the dial softkey in time.

Your extra step is lifting the handset <grin>.


-----Original Message-----
From: Riley, Andrew [mailto:Andrew.Riley at getronics.com] 
Sent: Wednesday, October 31, 2007 10:04 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Nortel vs Cisco (live keypad feature)

lift the handset (like traditional phones) dial the number and call goes through.
 
Where is the extra step?

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Thu 1/11/2007 12:48 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)



Below was a comment from a manager moving from Nortel Option 61c to
Cisco CallManager 5.1.2, I'm don't think the "live keypad" is a feature
in callmanager, if you enter a extension of another user you must press
the dial button when done.


"One major loss from a user standpoint has been that the keypad is not
live. In the Nortel system you could enter another phone extension 5846
and it would just dial without any additional effort, therefore calling
my extension. Each user has one extra step for every call they make by
pressing Dial softkey in Cisco. I am surprised that Cisco can not
program the same defaults." -customer



-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
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notify us immediately by replying to this message and deleting it
from your computer. Thank you.
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




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From jsteinberg at gmail.com  Wed Oct 31 23:59:59 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 31 Oct 2007 23:59:59 -0400
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0074D888A@USNAEXCH.na.didata.local>
References: <1495a01c81c30$4fbb9661$3c84a8c0@europe.unity>
	<C1FE15183DA37645BC0633BC604E44F0074D888A@USNAEXCH.na.didata.local>
Message-ID: <f1b65f880710312059g6fd821d5i54a5247bc356709a@mail.gmail.com>

you did a good install for them if this is what they are complaining about :)

On 10/31/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> Correct, Somehow Nortel knows you've entered a matching extension when you press 5656 and automatically does speakerphone and dials. He only pressed "5656" and got speakerphone and dialing.
>
> -----Original Message-----
> From: Riley, Andrew [mailto:Andrew.Riley at getronics.com]
> Sent: Wednesday, October 31, 2007 10:39 PM
> To: Jason Aarons (US)
> Subject: RE: [cisco-voip] Nortel vs Cisco (live keypad feature)
>
> Understood. So the nortel feature behaves as if the phone was always off hook
>
> -----Original Message-----
> From: "Jason Aarons (US)" <jason.aarons at us.didata.com>
> To: "Riley, Andrew" <Andrew.Riley at getronics.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
> Sent: 1/11/07 13:23
> Subject: RE: [cisco-voip] Nortel vs Cisco (live keypad feature)
>
> He wants specifically to enter an extension number on the keypad without pressing the dial softkey and have it call the extension. The Nortel automatically knows a extension has been answer, place the phone in speakerphone mode and connects the call. I'm sure he will get used to pressing the dial softkey in time.
>
> Your extra step is lifting the handset <grin>.
>
>
> -----Original Message-----
> From: Riley, Andrew [mailto:Andrew.Riley at getronics.com]
> Sent: Wednesday, October 31, 2007 10:04 PM
> To: Jason Aarons (US); cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Nortel vs Cisco (live keypad feature)
>
> lift the handset (like traditional phones) dial the number and call goes through.
>
> Where is the extra step?
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
> Sent: Thu 1/11/2007 12:48 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)
>
>
>
> Below was a comment from a manager moving from Nortel Option 61c to
> Cisco CallManager 5.1.2, I'm don't think the "live keypad" is a feature
> in callmanager, if you enter a extension of another user you must press
> the dial button when done.
>
>
> "One major loss from a user standpoint has been that the keypad is not
> live. In the Nortel system you could enter another phone extension 5846
> and it would just dial without any additional effort, therefore calling
> my extension. Each user has one extra step for every call they make by
> pressing Dial softkey in Cisco. I am surprised that Cisco can not
> program the same defaults." -customer
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
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> designated addressee(s) named above only.  If you are not the
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> notify us immediately by replying to this message and deleting it
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>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
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>

From lelio at uoguelph.ca  Wed Oct 31 23:03:56 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 31 Oct 2007 23:03:56 -0400
Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)
References: <963CE94787005149A52C17F33A028E120A3209FD@EVS1B.boco.co.boulder.co.us><535625f70710301128h48f4586ak63d2a737fba8e565@mail.gmail.com><CB4BC8C24EF6E149AED9FDB9B9AF57137030B8@MAIL02.hq.data-corporation.com>	<535625f70710310639x2ba2ace7k9a6d3081e663210c@mail.gmail.com><C1FE15183DA37645BC0633BC604E44F013E925@USNAEXCH.na.didata.local><47291498.6000202@wcyv.com><C1FE15183DA37645BC0633BC604E44F0074D8863@USNAEXCH.na.didata.local><A49371C6FF101B46BD3E3A8C89985F023DCAAE@excausy200.australia.unity>
	<C1FE15183DA37645BC0633BC604E44F0074D8881@USNAEXCH.na.didata.local>
Message-ID: <002d01c81c33$d8819560$6501a8c0@digibox>

I believe Cisco calls this "on-hook" dialing. I believe it used to work with 
7970s, but with the introduction of Abbreviated Dialing (and the AbbrDial 
softkey and no ability to disable it - why on earth would you want to do 
that?) you have to press the dial softkey.

I can see/appreciate the live keypad feature. I'd like it.

Not as much as being able to forward non-prime lines, but nonetheless.
----- Original Message ----- 
From: "Jason Aarons (US)" <jason.aarons at us.didata.com>
To: "Riley, Andrew" <Andrew.Riley at getronics.com>; 
<cisco-voip at puck.nether.net>
Sent: Wednesday, October 31, 2007 10:22 PM
Subject: Re: [cisco-voip] Nortel vs Cisco (live keypad feature)


> He wants specifically to enter an extension number on the keypad without 
> pressing the dial softkey and have it call the extension. The Nortel 
> automatically knows a extension has been answer, place the phone in 
> speakerphone mode and connects the call. I'm sure he will get used to 
> pressing the dial softkey in time.
>
> Your extra step is lifting the handset <grin>.
>
>
> -----Original Message-----
> From: Riley, Andrew [mailto:Andrew.Riley at getronics.com]
> Sent: Wednesday, October 31, 2007 10:04 PM
> To: Jason Aarons (US); cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Nortel vs Cisco (live keypad feature)
>
> lift the handset (like traditional phones) dial the number and call goes 
> through.
>
> Where is the extra step?
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
> Sent: Thu 1/11/2007 12:48 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Nortel vs Cisco (live keypad feature)
>
>
>
> Below was a comment from a manager moving from Nortel Option 61c to
> Cisco CallManager 5.1.2, I'm don't think the "live keypad" is a feature
> in callmanager, if you enter a extension of another user you must press
> the dial button when done.
>
>
> "One major loss from a user standpoint has been that the keypad is not
> live. In the Nortel system you could enter another phone extension 5846
> and it would just dial without any additional effort, therefore calling
> my extension. Each user has one extra step for every call they make by
> pressing Dial softkey in Cisco. I am surprised that Cisco can not
> program the same defaults." -customer
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> -----------------------------------------
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>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
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> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
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> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


