From JeanPaul.Vega at telemark-peru.com  Wed Aug  1 04:02:55 2007
From: JeanPaul.Vega at telemark-peru.com (JeanPaul Vega)
Date: Wed, 1 Aug 2007 03:02:55 -0500
Subject: [cisco-voip] virtual access down
Message-ID: <82627BE34A80064098D4ADFE72AE0F561811EFB205@limasrv2.lima.local>


Hi:

I have some questions. I hope you can help me.
In my work, we have 2 routers connected to spain. We use ip telephony, and some times we have problem with the calls.
When I see the log of the routers, I found problems in the interface virtual access(see the logs)

*Jul 31 01:12:25.319: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is down: holding time expired
*Jul 31 01:12:56.807: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is up: new adjacency
*Jul 31 01:12:56.963: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is down: peer restarted
*Jul 31 01:13:01.651: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is up: new adjacency
*Jul 31 12:36:42.448: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is down: holding time expired
*Jul 31 12:36:54.564: %LINK-3-UPDOWN: Interface Virtual-Access2, changed state to down
*Jul 31 12:36:54.716: %LINK-3-UPDOWN: Interface Virtual-Access3, changed state to up
*Jul 31 12:36:55.564: %LINEPROTO-5-UPDOWN: Line protocol on Interface Virtual-Access2, changed state to down
*Jul 31 12:36:55.716: %LINEPROTO-5-UPDOWN: Line protocol on Interface Virtual-Access3, changed state to up
*Jul 31 12:36:57.232: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access3) is up: new adjacency
*Jul 31 12:51:00.340: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access3) is down: holding time expired
*Jul 31 12:51:42.504: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access3) is up: new adjacency


Ih the the problem was physical, I don`t saw any log reference interface serial down or protocol in interface serial.

What's the mean of "line protocol on virtual access down"?

Regards.

Jean Paul.



Jean Paul Vega L.
Departamento de Tecnolog?a

TELEMARK SPAIN SL- Sucursal Per?
AV. JOSE PARDO NRO. 805 MIRAFLORES
T +511-2114747 EXT 4741


________________________________
AVISO DE CONFIDENCIALIDAD:
Este correo electr?nico contiene informaci?n estrictamente confidencial no susceptible de ser distribuida. Si usted no es el destinatario de este mensaje, por favor no publicarlo, no copiarlo o tomar cualquier otro tipo de acci?n sobre esta transmisi?n. Si recibi? este mensaje por error, por favor proceda a eliminarlo y notificar al remitente

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From Brian.Henry at apptis.com  Wed Aug  1 05:27:19 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Wed, 1 Aug 2007 05:27:19 -0400
Subject: [cisco-voip] virtual access down
References: <82627BE34A80064098D4ADFE72AE0F561811EFB205@limasrv2.lima.local>
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6FE@CH-EXCH-02.intelltech.apptis.inc>

?
It seems that you have Virtual Templates configured for your interfaces for possibly a ATM circuit for ppp multipoint (QoS) features.
 
It also looks like you are having problems with the link because of the EIGRP statements in the logs.
 
Look at the main interface in question for errors "sh controll x/x"  If this is a consistent then contact your ISP to get them to look at the line, if its random its still provider but its just taking random hits of the circuit being up.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of JeanPaul Vega
Sent: Wed 8/1/2007 4:02 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] virtual access down



 

Hi:

 

I have some questions. I hope you can help me.

In my work, we have 2 routers connected to spain. We use ip telephony, and some times we have problem with the calls.

When I see the log of the routers, I found problems in the interface virtual access(see the logs)

 

*Jul 31 01:12:25.319: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is down: holding time expired

*Jul 31 01:12:56.807: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is up: new adjacency

*Jul 31 01:12:56.963: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is down: peer restarted

*Jul 31 01:13:01.651: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is up: new adjacency

*Jul 31 12:36:42.448: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access2) is down: holding time expired

*Jul 31 12:36:54.564: %LINK-3-UPDOWN: Interface Virtual-Access2, changed state to down

*Jul 31 12:36:54.716: %LINK-3-UPDOWN: Interface Virtual-Access3, changed state to up

*Jul 31 12:36:55.564: %LINEPROTO-5-UPDOWN: Line protocol on Interface Virtual-Access2, changed state to down

*Jul 31 12:36:55.716: %LINEPROTO-5-UPDOWN: Line protocol on Interface Virtual-Access3, changed state to up

*Jul 31 12:36:57.232: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access3) is up: new adjacency

*Jul 31 12:51:00.340: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access3) is down: holding time expired

*Jul 31 12:51:42.504: %DUAL-5-NBRCHANGE: IP-EIGRP(0) 113: Neighbor 192.168.112.1 (Virtual-Access3) is up: new adjacency

 

 

Ih the the problem was physical, I don`t saw any log reference interface serial down or protocol in interface serial. 

 

What?EUR(tm)s the mean of ?EURoeline protocol on virtual access down?EUR??

 

Regards.

 

Jean Paul.

 

 

 

Jean Paul Vega L.

Departamento de Tecnolog???a

 

TELEMARK SPAIN SL- Sucursal Per???

AV. JOSE PARDO NRO. 805 MIRAFLORES

T +511-2114747 EXT 4741

 


________________________________

AVISO DE CONFIDENCIALIDAD:
Este correo electr???nico contiene informaci???n estrictamente confidencial no susceptible de ser distribuida. Si usted no es el destinatario de este mensaje, por favor no publicarlo, no copiarlo o tomar cualquier otro tipo de acci???n sobre esta transmisi???n. Si recibi??? este mensaje por error, por favor proceda a eliminarlo y notificar al remitente


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From Andrew.Riley at getronics.com  Wed Aug  1 07:54:50 2007
From: Andrew.Riley at getronics.com (Riley, Andrew)
Date: Wed, 1 Aug 2007 21:54:50 +1000
Subject: [cisco-voip] No Call Proceeding received for CM to Avaya PBX via
	TDM/H323 trunk
Message-ID: <E4B3AF9D5AA27041A2C8A5B9BC5518D90CC2FF30@excausy100.australia.unity>

Hi there, 
 
Has anyone had success/experience with connecting an Avaya PBX to a
Cisco Call Manager via a TDM and H323 link. See topology below. I have
seen all the Cisco doc's about doing H323 direct to CCM but this is not
feasible in this particular environment.
 
IP phone------Cisco CM
----mgcp-----3845-----q931------2811-----h323--------Avaya
PBX------Phone
 
|
 
Analog Phone
 
Calls from the analog phone are successful end to the Avaya
Calls from Avaya to the IP Phone are successful
Calls from the IP Phone to the Avaya Fail.
 
Attached are the traces from the 2811 for a failed call. We are seeing
the h225 setup go out to the Avaya, but the Avaya doesn't respond with a
call proceeding. The Avaya signal's its phone so it rings but as the
2811 doesn't receive CALLPROC h225 times out and the IP Phone gets slow
busy tone. The strange thing is that the analog phone calling the same
pattern work no problem. 
 
My suspicion is that the Avaya is dropping the call as it doesn't like
some information in the IE bits of the h225 setup for a call from the IP
Phone but as there is less info in the analog call setup it proceeds.
The Avaya folk have not been able to provide reason for not sending the
CALLPROC back.
 
Any input is much appreciated. 

Andrew
 
 

Andrew Riley
Technical Consultant - Communication Services
CCIE # 18285 Voice 

Getronics Australia Pty Limited
Getronics combines the service capabilities of the original Dutch
company with those of Wang Global, acquired in 1999, and of the Olivetti
systems and services division. We are ranked second worldwide in network
and desktop outsourcing and fourth in network consulting and integration
(Source: IDC 2002-2003).

2 Minna Close
Belrose NSW 2085
Australia
Tel:       +61 2 9847 7309
Fax:      +61 2 9847 7378
Mobile:  +61 402 894 793
Email:    andrew.riley at getronics.com

Communication Services | Security Services | Application Services |
Workspace Management Services | Technology Transformation Services

www.getronics.com.au <http://www.getronics.com.au/>  

 

Please note that whilst we take all care, neither Getronics nor the
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From SCASPER at mtb.com  Wed Aug  1 08:08:33 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Wed, 01 Aug 2007 08:08:33 -0400
Subject: [cisco-voip] Duplicate CDR globalCallID_callID
Message-ID: <46B03F82.9874.0039.0@mtb.com>

Do CDR records involved in Conference calls or other modifications use the same globalCallID_callID as the originating call? I am seeing duplicates on the same server and Call Manager has not been restarted in a while. Any other thoughts?
 
 
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 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.voip at verizon.net  Wed Aug  1 08:45:23 2007
From: cisco.voip at verizon.net (cisco.voip)
Date: Wed, 01 Aug 2007 08:45:23 -0400
Subject: [cisco-voip] Cisco srtp CM 4.2.3
References: <mailman.9003.1185969334.1529.cisco-voip@puck.nether.net>
Message-ID: <000701c7d439$d4b31900$661b1b0a@cs.myharris.net>

Has anyone implemented  SRTP on Call Manager 4.2.3?
How much of a hassle is it, and which server is recommended for the
certifcates  other than using the call manager.

Can it be implemented on a phoen by phone basis?

Thanks,



From jonvoip at gmail.com  Wed Aug  1 09:03:56 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 1 Aug 2007 08:03:56 -0500
Subject: [cisco-voip] 0 (Operator Ext) in SRST
In-Reply-To: <003b01c7d399$8c9c2990$47196883@cfs.uoguelph.ca>
References: <003b01c7d399$8c9c2990$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0708010603q2462e9a5l168bd23bf33fe438@mail.gmail.com>

I am a big fan of a num-exp for 0 to the registered extension.


Jonathan

On 7/31/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>
> Just wondering if there is a special way to handle phones dialing 0 in SRST
> mode? I was contemplating entering a numexp command with 0 and the
> destination I want to use but thought I'd ask first.
>
> --------------------------------------------------------------------------------
> 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
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From CarlosOrtiz at bayviewfinancial.com  Wed Aug  1 08:56:51 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 1 Aug 2007 08:56:51 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
Message-ID: <OFF256D627.A725EE1C-ON8525732A.0046D683-8525732A.00471FF1@bftg.com>

When viewing the DestIPAddr for a call in the CDR database of the 
calldetailrecord table a numeric number is shown.  Anyone know how to 
convert this to IP address format?

Carlos
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From lelio at uoguelph.ca  Wed Aug  1 09:10:53 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 1 Aug 2007 09:10:53 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
References: <OFF256D627.A725EE1C-ON8525732A.0046D683-8525732A.00471FF1@bftg.com>
Message-ID: <005701c7d43d$63f654d0$47196883@cfs.uoguelph.ca>

I see dotted decimal form a.b.c.d

what do you see?

--------------------------------------------------------------------------------
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
  ----- Original Message ----- 
  From: CarlosOrtiz at bayviewfinancial.com 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, August 01, 2007 8:56 AM
  Subject: [cisco-voip] DestIPAddr field in CDR record



  When viewing the DestIPAddr for a call in the CDR database of the calldetailrecord table a numeric number is shown.  Anyone know how to convert this to IP address format? 

  Carlos


------------------------------------------------------------------------------


  _______________________________________________
  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  Wed Aug  1 09:16:06 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 1 Aug 2007 09:16:06 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
In-Reply-To: <005701c7d43d$63f654d0$47196883@cfs.uoguelph.ca>
Message-ID: <OF7D665CD6.37CDC837-ON8525732A.0048AB8B-8525732A.0048E2D2@bftg.com>

I see a 9 digit numeric number.   Are you looking in the CDR database, 
calldetailrecord table?  I started searching the archives and I did find 
this email which you posted a reply to !




lelio at uoguelph 

Sep 27, 2006, 5:00 PM 

Post #2 of 4 (258 views) 
Permalink 


Re: CDR IP Address Conversion Remove Highlighting [In reply to] 


found this using google... 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a0080382d82.html#wp1003527 



----- Original Message ----- 
From: STEVEN CASPER 
To: cisco-voip[at]puck.nether.net 
Sent: Wednesday, September 27, 2006 7:55 PM 
Subject: [cisco-voip] CDR IP Address Conversion 



Does anyone have a tool or formula they could share that can be used to 
convert the signed decimal value in the CDR database to an IP address? 

Thanks! 
Steve 







"Lelio Fulgenzi" <lelio at uoguelph.ca> 
Sent by: cisco-voip-bounces at puck.nether.net
08/01/2007 09:10 AM

To
<cisco-voip at puck.nether.net>, <CarlosOrtiz at bayviewfinancial.com>
cc

Subject
Re: [cisco-voip] DestIPAddr field in CDR record






I see dotted decimal form a.b.c.d
 
what do you see?
 
--------------------------------------------------------------------------------
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
----- Original Message ----- 
From: CarlosOrtiz at bayviewfinancial.com 
To: cisco-voip at puck.nether.net 
Sent: Wednesday, August 01, 2007 8:56 AM
Subject: [cisco-voip] DestIPAddr field in CDR record


When viewing the DestIPAddr for a call in the CDR database of the 
calldetailrecord table a numeric number is shown.  Anyone know how to 
convert this to IP address format? 

Carlos 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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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 lelio at uoguelph.ca  Wed Aug  1 09:23:26 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 1 Aug 2007 09:23:26 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
References: <OF7D665CD6.37CDC837-ON8525732A.0048AB8B-8525732A.0048E2D2@bftg.com>
Message-ID: <008601c7d43f$24d49c10$47196883@cfs.uoguelph.ca>

sorry, i was using the tool. it probably did the conversion for me.

--------------------------------------------------------------------------------
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
  ----- Original Message ----- 
  From: CarlosOrtiz at bayviewfinancial.com 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Wednesday, August 01, 2007 9:16 AM
  Subject: Re: [cisco-voip] DestIPAddr field in CDR record



  I see a 9 digit numeric number.   Are you looking in the CDR database, calldetailrecord table?  I started searching the archives and I did find this email which you posted a reply to ! 

                    lelio at uoguelph 

                    Sep 27, 2006, 5:00 PM 

                    Post #2 of 4 (258 views) 
                    Permalink  
                          Re: CDR IP Address Conversion Remove Highlighting [In reply to]  

------------------------------------------------------
                         


                    found this using google... 

                    http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a0080382d82.html#wp1003527 


                    ----- Original Message ----- 
                    From: STEVEN CASPER 
                    To: cisco-voip[at]puck.nether.net 
                    Sent: Wednesday, September 27, 2006 7:55 PM 
                    Subject: [cisco-voip] CDR IP Address Conversion 



                    Does anyone have a tool or formula they could share that can be used to 
                    convert the signed decimal value in the CDR database to an IP address? 

                    Thanks! 
                    Steve  

             

       







        "Lelio Fulgenzi" <lelio at uoguelph.ca> 
        Sent by: cisco-voip-bounces at puck.nether.net 
        08/01/2007 09:10 AM 
       To <cisco-voip at puck.nether.net>, <CarlosOrtiz at bayviewfinancial.com>  
              cc  
              Subject Re: [cisco-voip] DestIPAddr field in CDR record 

              

       



  I see dotted decimal form a.b.c.d 
    
  what do you see? 
    
  --------------------------------------------------------------------------------
  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 
  ----- Original Message ----- 
  From: CarlosOrtiz at bayviewfinancial.com 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, August 01, 2007 8:56 AM 
  Subject: [cisco-voip] DestIPAddr field in CDR record 


  When viewing the DestIPAddr for a call in the CDR database of the calldetailrecord table a numeric number is shown.  Anyone know how to convert this to IP address format? 

  Carlos 


------------------------------------------------------------------------------

  _______________________________________________
  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 CarlosOrtiz at bayviewfinancial.com  Wed Aug  1 09:27:52 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 1 Aug 2007 09:27:52 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
In-Reply-To: <008601c7d43f$24d49c10$47196883@cfs.uoguelph.ca>
Message-ID: <OF67B3F7AA.0446CDAC-ON8525732A.0049F01D-8525732A.0049F6C3@bftg.com>

Which tool?





"Lelio Fulgenzi" <lelio at uoguelph.ca> 
Sent by: cisco-voip-bounces at puck.nether.net
08/01/2007 09:23 AM

To
<CarlosOrtiz at bayviewfinancial.com>
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] DestIPAddr field in CDR record






sorry, i was using the tool. it probably did the conversion for me.
 
--------------------------------------------------------------------------------
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
----- Original Message ----- 
From: CarlosOrtiz at bayviewfinancial.com 
To: Lelio Fulgenzi 
Cc: cisco-voip at puck.nether.net 
Sent: Wednesday, August 01, 2007 9:16 AM
Subject: Re: [cisco-voip] DestIPAddr field in CDR record


I see a 9 digit numeric number.   Are you looking in the CDR database, 
calldetailrecord table?  I started searching the archives and I did find 
this email which you posted a reply to ! 




lelio at uoguelph 

Sep 27, 2006, 5:00 PM 

Post #2 of 4 (258 views) 
Permalink 


Re: CDR IP Address Conversion Remove Highlighting [In reply to] 



found this using google... 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a0080382d82.html#wp1003527 



----- Original Message ----- 
From: STEVEN CASPER 
To: cisco-voip[at]puck.nether.net 
Sent: Wednesday, September 27, 2006 7:55 PM 
Subject: [cisco-voip] CDR IP Address Conversion 



Does anyone have a tool or formula they could share that can be used to 
convert the signed decimal value in the CDR database to an IP address? 

Thanks! 
Steve 







"Lelio Fulgenzi" <lelio at uoguelph.ca> 
Sent by: cisco-voip-bounces at puck.nether.net 
08/01/2007 09:10 AM 


To
<cisco-voip at puck.nether.net>, <CarlosOrtiz at bayviewfinancial.com> 
cc

Subject
Re: [cisco-voip] DestIPAddr field in CDR record








I see dotted decimal form a.b.c.d 
  
what do you see? 
  
--------------------------------------------------------------------------------
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 
----- Original Message ----- 
From: CarlosOrtiz at bayviewfinancial.com 
To: cisco-voip at puck.nether.net 
Sent: Wednesday, August 01, 2007 8:56 AM 
Subject: [cisco-voip] DestIPAddr field in CDR record 


When viewing the DestIPAddr for a call in the CDR database of the 
calldetailrecord table a numeric number is shown.  Anyone know how to 
convert this to IP address format? 

Carlos 

_______________________________________________
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 lelio at uoguelph.ca  Wed Aug  1 09:29:36 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 1 Aug 2007 09:29:36 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
References: <OF67B3F7AA.0446CDAC-ON8525732A.0049F01D-8525732A.0049F6C3@bftg.com>
Message-ID: <009601c7d440$012cd7e0$47196883@cfs.uoguelph.ca>

The CAR/ART tool , whatever you call it.

--------------------------------------------------------------------------------
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
  ----- Original Message ----- 
  From: CarlosOrtiz at bayviewfinancial.com 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net ; cisco-voip-bounces at puck.nether.net 
  Sent: Wednesday, August 01, 2007 9:27 AM
  Subject: Re: [cisco-voip] DestIPAddr field in CDR record



  Which tool? 




        "Lelio Fulgenzi" <lelio at uoguelph.ca> 
        Sent by: cisco-voip-bounces at puck.nether.net 
        08/01/2007 09:23 AM 
       To <CarlosOrtiz at bayviewfinancial.com>  
              cc cisco-voip at puck.nether.net  
              Subject Re: [cisco-voip] DestIPAddr field in CDR record 

              

       



  sorry, i was using the tool. it probably did the conversion for me. 
    
  --------------------------------------------------------------------------------
  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 
  ----- Original Message ----- 
  From: CarlosOrtiz at bayviewfinancial.com 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Wednesday, August 01, 2007 9:16 AM 
  Subject: Re: [cisco-voip] DestIPAddr field in CDR record 


  I see a 9 digit numeric number.   Are you looking in the CDR database, calldetailrecord table?  I started searching the archives and I did find this email which you posted a reply to ! 

                    lelio at uoguelph 

                    Sep 27, 2006, 5:00 PM 

                    Post #2 of 4 (258 views) 
                    Permalink  
                          Re: CDR IP Address Conversion Remove Highlighting [In reply to]  

------------------------------------------------------
                         



                    found this using google... 

                    http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a0080382d82.html#wp1003527 


                    ----- Original Message ----- 
                    From: STEVEN CASPER 
                    To: cisco-voip[at]puck.nether.net 
                    Sent: Wednesday, September 27, 2006 7:55 PM 
                    Subject: [cisco-voip] CDR IP Address Conversion 



                    Does anyone have a tool or formula they could share that can be used to 
                    convert the signed decimal value in the CDR database to an IP address? 

                    Thanks! 
                    Steve  

             

       







        "Lelio Fulgenzi" <lelio at uoguelph.ca> 
        Sent by: cisco-voip-bounces at puck.nether.net 
        08/01/2007 09:10 AM 
       
              To <cisco-voip at puck.nether.net>, <CarlosOrtiz at bayviewfinancial.com>  
              cc  
              Subject Re: [cisco-voip] DestIPAddr field in CDR record 


              

       




  I see dotted decimal form a.b.c.d 
   
  what do you see? 
   
  --------------------------------------------------------------------------------
  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 
  ----- Original Message ----- 
  From: CarlosOrtiz at bayviewfinancial.com 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, August 01, 2007 8:56 AM 
  Subject: [cisco-voip] DestIPAddr field in CDR record 


  When viewing the DestIPAddr for a call in the CDR database of the calldetailrecord table a numeric number is shown.  Anyone know how to convert this to IP address format? 

  Carlos 


------------------------------------------------------------------------------

  _______________________________________________
  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 matt at saskin.net  Wed Aug  1 09:41:50 2007
From: matt at saskin.net (Matthew Saskin)
Date: Wed, 01 Aug 2007 09:41:50 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
In-Reply-To: <009601c7d440$012cd7e0$47196883@cfs.uoguelph.ca>
References: <OF67B3F7AA.0446CDAC-ON8525732A.0049F01D-8525732A.0049F6C3@bftg.com>
	<009601c7d440$012cd7e0$47196883@cfs.uoguelph.ca>
Message-ID: <46B08D9E.8090101@saskin.net>

IP addresses are stored in the cdr table in a decimal format.  Here is 
an (ugly) SQL server function I wrote a while back that does the 
conversion to dotted decimal format....once again, it's ugly and only 
relatively efficient, but it works :)

-matt

------------------------------------------------------------
------------------------------------------------------------
CREATE  function dbo.decimalToIP(@ip INT)
RETURNS VARCHAR(15)
AS
BEGIN

DECLARE @binip BINARY(4)
SET @binip = CONVERT(BINARY(4), at ip)

DECLARE @charip varchar(8)
SET @charip = CONVERT(varchar(8), at binip)

DECLARE @1 int, @2 int, @3 int, @4 int

SET @4 = ASCII(SUBSTRING(@charip,1,1))
SET @3 = ASCII(SUBSTRING(@charip,2,1))
SET @2 = ASCII(SUBSTRING(@charip,3,1))
SET @1 = ASCII(SUBSTRING(@charip,4,1))

DECLARE @output varchar(15)
SET @output = CONVERT(varchar, at 1) + '.' + CONVERT(varchar, at 2) + '.' + 
CONVERT(varchar, at 3) + '.' + CONVERT(varchar, at 4)
RETURN @output

END
------------------------------------------------------------
------------------------------------------------------------




Lelio Fulgenzi wrote:
> The CAR/ART tool , whatever you call it.
>  
> --------------------------------------------------------------------------------
> 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
> 
>     ----- Original Message -----
>     *From:* CarlosOrtiz at bayviewfinancial.com
>     <mailto:CarlosOrtiz at bayviewfinancial.com>
>     *To:* Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
>     *Cc:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     ; cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
>     *Sent:* Wednesday, August 01, 2007 9:27 AM
>     *Subject:* Re: [cisco-voip] DestIPAddr field in CDR record
> 
> 
>     Which tool?
> 
> 
> 
> 
>     *"Lelio Fulgenzi" <lelio at uoguelph.ca <mailto:lelio at uoguelph.ca>>*
>     Sent by: cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
> 
>     08/01/2007 09:23 AM
> 
>     	
>     To
>     	<CarlosOrtiz at bayviewfinancial.com
>     <mailto:CarlosOrtiz at bayviewfinancial.com>>
>     cc
>     	cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     Subject
>     	Re: [cisco-voip] DestIPAddr field in CDR record
> 
> 
>     	
> 
> 
> 
> 
> 
>     sorry, i was using the tool. it probably did the conversion for me.
>      
>     --------------------------------------------------------------------------------
>     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
>     ----- Original Message -----
>     *From:* _CarlosOrtiz at bayviewfinancial.com_
>     <mailto:CarlosOrtiz at bayviewfinancial.com>
>     *To:* _Lelio Fulgenzi_ <mailto:lelio at uoguelph.ca>
>     *Cc:* _cisco-voip at puck.nether.net_ <mailto:cisco-voip at puck.nether.net>
>     *Sent:* Wednesday, August 01, 2007 9:16 AM
>     *Subject:* Re: [cisco-voip] DestIPAddr field in CDR record
> 
> 
>     I see a 9 digit numeric number.   Are you looking in the CDR
>     database, calldetailrecord table?  I started searching the archives
>     and I did find this email which you posted a reply to !
> 
> 
>     *_lelio at uoguelph_*
>     <http://www.gossamer-threads.com/lists/engine?user=2532;list=cisco>
> 
>     Sep 27, 2006, 5:00 PM
> 
>     Post #2 of 4 (258 views) _
>     __Permalink_
>     <http://www.gossamer-threads.com/lists/cisco/voip/52547#52547> 	
>     *Re: CDR IP Address Conversion* _Remove Highlighting_
>     <http://www.gossamer-threads.com/lists/cisco/voip/52547?nohighlight=1#52547>
>     [_In reply to_
>     <http://www.gossamer-threads.com/lists/cisco/voip/52546?search_string=CDR;#52546>]
> 
>     ------------------------------------------------------------------------
> 
> 
> 
> 
>     found this using google... _
>     __
>     __http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a0080382d82.html#wp1003527_
> 
> 
> 
>     ----- Original Message -----
>     From: STEVEN CASPER
>     To: cisco-voip[at]puck.nether.net
>     Sent: Wednesday, September 27, 2006 7:55 PM
>     Subject: [cisco-voip] *CDR* IP Address Conversion
> 
> 
> 
>     Does anyone have a tool or formula they could share that can be used to
>     convert the signed decimal value in the *CDR* database to an IP
>     address?
> 
>     Thanks!
>     Steve
> 
> 
> 
> 
> 
> 
> 
> 
>     *"Lelio Fulgenzi" <lelio at uoguelph.ca>*
>     Sent by: cisco-voip-bounces at puck.nether.net
> 
>     08/01/2007 09:10 AM
> 
>     	
>     To
>     	<cisco-voip at puck.nether.net>, <CarlosOrtiz at bayviewfinancial.com>
>     cc
>     	
>     Subject
>     	Re: [cisco-voip] DestIPAddr field in CDR record
> 
> 
> 
>     	
> 
> 
> 
> 
> 
> 
>     I see dotted decimal form a.b.c.d
>      
>     what do you see?
>      
>     --------------------------------------------------------------------------------
>     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
>     ----- Original Message ----- *
>     From:* _CarlosOrtiz at bayviewfinancial.com_
>     <mailto:CarlosOrtiz at bayviewfinancial.com> *
>     To:* _cisco-voip at puck.nether.net_ <mailto:cisco-voip at puck.nether.net> *
>     Sent:* Wednesday, August 01, 2007 8:56 AM *
>     Subject:* [cisco-voip] DestIPAddr field in CDR record
> 
> 
>     When viewing the DestIPAddr for a call in the CDR database of the
>     calldetailrecord table a numeric number is shown.  Anyone know how
>     to convert this to IP address format?
> 
>     Carlos
> 
>     ------------------------------------------------------------------------
> 
>     _______________________________________________
>     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________________________________________________
>     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 aizhar at ccbootcamp.com  Wed Aug  1 10:07:04 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Wed, 1 Aug 2007 07:07:04 -0700
Subject: [cisco-voip] 0 (Operator Ext) in SRST
References: <5d093f9a0708010603q2462e9a5l168bd23bf33fe438@mail.gmail.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C366356110A@nlimail.nlidomain>

I'd prefer the alias command, under the call-manager-fallback, since it only works while in srst mode.
 
-------------------------------------------
thanks,
Avner Izhar
CCIE#15999 (Voice), CCVP.
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

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
Sent: Wed 8/1/2007 6:03 AM
To: Lelio Fulgenzi
Cc: cisco voip list
Subject: Re: [cisco-voip] 0 (Operator Ext) in SRST




I am a big fan of a num-exp for 0 to the registered extension.


Jonathan

On 7/31/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>
> Just wondering if there is a special way to handle phones dialing 0 in SRST
> mode? I was contemplating entering a numexp command with 0 and the
> destination I want to use but thought I'd ask first.
>
> --------------------------------------------------------------------------------
> 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
> _______________________________________________
> 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  Wed Aug  1 10:12:38 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 01 Aug 2007 10:12:38 -0400
Subject: [cisco-voip] Duplicate CDR globalCallID_callID
In-Reply-To: <46B03F82.9874.0039.0@mtb.com>
References: <46B03F82.9874.0039.0@mtb.com>
Message-ID: <46B094D6.1000601@cisco.com>

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From lelio at uoguelph.ca  Wed Aug  1 10:14:56 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 1 Aug 2007 10:14:56 -0400
Subject: [cisco-voip] 0 (Operator Ext) in SRST
References: <5d093f9a0708010603q2462e9a5l168bd23bf33fe438@mail.gmail.com>
	<E76ACA8C13EF1847BC69538B522C366356110A@nlimail.nlidomain>
Message-ID: <001201c7d446$56d027f0$47196883@cfs.uoguelph.ca>

Re: [cisco-voip] 0 (Operator Ext) in SRST [bcc][faked-from]Thanks guys.

This brings to light another question....as far as I can tell, SRST is triggered when something attempts to register to the router.

For example, we are not using our voice gateway as our data router, so it is totally possible, although unlikely that our phones may loose WAN connectivity but the router (and MGCP gateways) do not. In this instance, I believe the phones will register to the router, activating SRST, but the MGCP gateways will continue to be registered with HQ subscribers.

While not ideal, at least the phones can still communicate to each other. Come to think of it, this might be one reason to have the analog ports registered as H323 so they will always be able to send 911 calls out.

--------------------------------------------------------------------------------
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
  ----- Original Message ----- 
  From: Avner Izhar 
  To: Jonathan Charles ; Lelio Fulgenzi 
  Cc: cisco voip list 
  Sent: Wednesday, August 01, 2007 10:07 AM
  Subject: RE: [cisco-voip] 0 (Operator Ext) in SRST


  I'd prefer the alias command, under the call-manager-fallback, since it only works while in srst mode.

  -------------------------------------------
  thanks,
  Avner Izhar
  CCIE#15999 (Voice), CCVP.
  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

------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
  Sent: Wed 8/1/2007 6:03 AM
  To: Lelio Fulgenzi
  Cc: cisco voip list
  Subject: Re: [cisco-voip] 0 (Operator Ext) in SRST


  I am a big fan of a num-exp for 0 to the registered extension.


  Jonathan

  On 7/31/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  >
  >
  > Just wondering if there is a special way to handle phones dialing 0 in SRST
  > mode? I was contemplating entering a numexp command with 0 and the
  > destination I want to use but thought I'd ask first.
  >
  > --------------------------------------------------------------------------------
  > 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
  > _______________________________________________
  > 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  Wed Aug  1 10:16:20 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 01 Aug 2007 10:16:20 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
In-Reply-To: <OFF256D627.A725EE1C-ON8525732A.0046D683-8525732A.00471FF1@bftg.com>
References: <OFF256D627.A725EE1C-ON8525732A.0046D683-8525732A.00471FF1@bftg.com>
Message-ID: <46B095B4.9010006@cisco.com>

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From bob at smalltime.com  Wed Aug  1 10:32:33 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 01 Aug 2007 09:32:33 -0500
Subject: [cisco-voip] Added memory to a server,
	now it thinks I've got a 6 Ghz CPU?
Message-ID: <46B09981.1000406@smalltime.com>

7825-I-3.0-IPC1/CC1 (aka, IBM x306 8836-C1X); put in two 1GB sticks of 
Crucial Memory, 
(http://www.crucial.com/store/mpartspecs.aspx?mtbpoid=32F534BEA5CA7304)

# Module Size: 2GB kit (1GBx2)
# Package: 184-pin DIMM
# Feature: DDR PC3200
# Specs: DDR PC3200 ? CL=3 ? Unbuffered ? ECC ? DDR400 ? 2.6V ? 128Meg x 72

However, when I boot, the initial POST says that I've got a 6Ghz CPU!

Going into the BIOS config then tells me that I've got a 1.4 Ghz CPU and 
a 100Mhz bus clock, so things seem to be way off.  System still runs, 
but I have no idea if this is cosmetic or something actually wrong with 
the box.

BIOS is KEE140AUS, which I believe is the most current.

A "regular" 7825 tells me that it has a 3Ghz CPU, system clock of 200Mhz.

Any ideas?


From pfabian at telesphere.com  Wed Aug  1 10:35:18 2007
From: pfabian at telesphere.com (Pete Fabian)
Date: Wed, 1 Aug 2007 07:35:18 -0700
Subject: [cisco-voip] DestIPAddr field in CDR record
In-Reply-To: <46B095B4.9010006@cisco.com>
Message-ID: <008501c7d449$2ef815a0$4bfea8c0@telesphereltd.com>

To convert the signed decimal value to an IP address

 

Step 1 Convert the database display (-1139627840) to a hex value.

The hex value equals 0xBC12A8C0.

 

Step 2 Reverse the hex bytes, as shown below:

CO A8 12 BC

 

Step 3 Convert the bytes from hex to decimal, as shown below:

192 168 18 188

 

Step 4 The IP address displays in the following format:

192.168.18.188 

 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Wednesday, August 01, 2007 7:16 AM
To: CarlosOrtiz at bayviewfinancial.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] DestIPAddr field in CDR record

 

documentation on how the number is stored:
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_progr
amming_usage_guide09186a00806ed410.html#wp29039


i think this has octets out of order, but it's easy enuf to mod to your
needs:

#!/usr/bin/perl

my $raw = -1139627840;
my @addr = unpack('C4', pack('i', $raw));

my $ip = join('.', @addr);
print $ip, "\n";

/Wes

CarlosOrtiz at bayviewfinancial.com wrote: 


When viewing the DestIPAddr for a call in the CDR database of the
calldetailrecord table a numeric number is shown.  Anyone know how to
convert this to IP address format? 

Carlos 

 





  _____  



 
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
  
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From jonvoip at gmail.com  Wed Aug  1 10:45:14 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 1 Aug 2007 09:45:14 -0500
Subject: [cisco-voip] 0 (Operator Ext) in SRST
In-Reply-To: <E76ACA8C13EF1847BC69538B522C366356110A@nlimail.nlidomain>
References: <5d093f9a0708010603q2462e9a5l168bd23bf33fe438@mail.gmail.com>
	<E76ACA8C13EF1847BC69538B522C366356110A@nlimail.nlidomain>
Message-ID: <5d093f9a0708010745g81cbb3fl647d996ff403faee@mail.gmail.com>

Yeah, right... forgot about that...


J

On 8/1/07, Avner Izhar <aizhar at ccbootcamp.com> wrote:
>
>
>
> I'd prefer the alias command, under the call-manager-fallback, since it only
> works while in srst mode.
>
>  ------------------------------------------- thanks, Avner
> Izhar CCIE#15999 (Voice), CCVP. 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
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net on behalf of
> Jonathan Charles
> Sent: Wed 8/1/2007 6:03 AM
> To: Lelio Fulgenzi
> Cc: cisco voip list
> Subject: Re: [cisco-voip] 0 (Operator Ext) in SRST
>
>
>
>
> I am a big fan of a num-exp for 0 to the registered extension.
>
>
> Jonathan
>
> On 7/31/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >
> >
> > Just wondering if there is a special way to handle phones dialing 0 in
> SRST
> > mode? I was contemplating entering a numexp command with 0 and the
> > destination I want to use but thought I'd ask first.
> >
> >
> --------------------------------------------------------------------------------
> > 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
> > _______________________________________________
> > 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  Wed Aug  1 10:46:24 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 1 Aug 2007 09:46:24 -0500
Subject: [cisco-voip] Unity 5 Schema Extensions for AD documentation
Message-ID: <5d093f9a0708010746w32421554q32cd887b98f53fc0@mail.gmail.com>

I have a customer that requires documentation on what fields are being
added to AD for Unity 5, I have one for Unity 4.X, but we will be
installing Unity 5 in a month.

Does anyone know where one can find this info?



Jonathan

From jonvoip at gmail.com  Wed Aug  1 11:11:05 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 1 Aug 2007 10:11:05 -0500
Subject: [cisco-voip] DestIPAddr field in CDR record
In-Reply-To: <OFF256D627.A725EE1C-ON8525732A.0046D683-8525732A.00471FF1@bftg.com>
References: <OFF256D627.A725EE1C-ON8525732A.0046D683-8525732A.00471FF1@bftg.com>
Message-ID: <5d093f9a0708010811j37b96a27xf7475966a936914b@mail.gmail.com>

Plug that number into Windows Calculator.

Click on the HEX button to convert it to hex

Break it into two character chunks.

Convert each chunk back into decimal.

Once you have four, reverse them and that is the IP address.


Basically, what you are looking at is a decimal representation of the
IP address in binary. You need to convert it back to the four
decimals.



Jonathan

On 8/1/07, CarlosOrtiz at bayviewfinancial.com
<CarlosOrtiz at bayviewfinancial.com> wrote:
>
> When viewing the DestIPAddr for a call in the CDR database of the
> calldetailrecord table a numeric number is shown.  Anyone know how to
> convert this to IP address format?
>
> Carlos
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From toddnh65 at gmail.com  Wed Aug  1 11:23:59 2007
From: toddnh65 at gmail.com (Todd Franklin)
Date: Wed, 1 Aug 2007 10:23:59 -0500
Subject: [cisco-voip] No Rings, Voicemail
Message-ID: <6101bda50708010823r3688f269t7de6967c73f6f78a@mail.gmail.com>

I am setting up a "greeting only" line for our company so when people call
in, they can just hear some automated information.
How do I make it so that line won't ring, just fall right into voicemail?

And I want to set up my secretary with me as her 2nd line, but it shouldn't
audibly ring at her phone, but she should see I am on the
phone and who with (caller ID).  How do I tell her 2nd line to not ring (or
can I just crank the volume all the way down)?? these are 7940s.

Thanks.
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From lelio at uoguelph.ca  Wed Aug  1 11:36:39 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 1 Aug 2007 11:36:39 -0400
Subject: [cisco-voip] No Rings, Voicemail
References: <6101bda50708010823r3688f269t7de6967c73f6f78a@mail.gmail.com>
Message-ID: <003301c7d451$c0e91a10$47196883@cfs.uoguelph.ca>


under the transfer settings "send directly to greeting"

change the ring settings on her second line to whatever you want in the line configuration page. 

Ring Setting (Phone Idle)  Use System Default, Disable, Flash Only, Ring Once, Ring  
Ring Setting (Phone Active)**  Use System Default, Disable, Flash Only, Ring Once, Ring, Beep Only   

System Defaults are configured in CallManager Service Parameters I believe.



  ----- Original Message ----- 
  From: Todd Franklin 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, August 01, 2007 11:23 AM
  Subject: [cisco-voip] No Rings, Voicemail


  I am setting up a "greeting only" line for our company so when people call in, they can just hear some automated information.
  How do I make it so that line won't ring, just fall right into voicemail?

  And I want to set up my secretary with me as her 2nd line, but it shouldn't audibly ring at her phone, but she should see I am on the
  phone and who with (caller ID).  How do I tell her 2nd line to not ring (or can I just crank the volume all the way down)?? these are 7940s. 

  Thanks.



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From bob at smalltime.com  Wed Aug  1 11:42:39 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 01 Aug 2007 10:42:39 -0500
Subject: [cisco-voip] Added memory to a server,
 now it thinks I've got a 6 Ghz CPU?
In-Reply-To: <46B09981.1000406@smalltime.com>
References: <46B09981.1000406@smalltime.com>
Message-ID: <46B0A9EF.2000105@smalltime.com>

> BIOS is KEE140AUS, which I believe is the most current.
> 
> A "regular" 7825 tells me that it has a 3Ghz CPU, system clock of 200Mhz.

Put the same memory into a different system and the CPU stays at 3Ghz 
and reflects the new memory.

Put the original 512MB sticks back into the machine that's reporting a 
6Ghz CPU, and it's still saying that it's a 6Ghz CPU in POST, but the 
IBM BIOS config screen says that it's a 1.4Ghz CPU with a 100Mhz bus.

Sounds like a busted system?

From svoll.voip at gmail.com  Wed Aug  1 11:50:40 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 1 Aug 2007 08:50:40 -0700
Subject: [cisco-voip] VGW not registered in CCM but is on GW
Message-ID: <f84a38d30708010850u7a879f08i8716b926e6a6b22a@mail.gmail.com>

I have a 3845 running 12.4(15)T1 via MGCP to the CM 5.1.2.

on the VGW a show ccm shows registered but on the CM is shows the correct IP
address but unregistered.

I have the service MGCPapp command on my dial-peers  and all the normal
ccm-manager commands along with MGCP commands.

is it not registered because I don't have anything plugged into the PRI
ports?

TIA

Scott
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From lelio at uoguelph.ca  Wed Aug  1 11:54:43 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 1 Aug 2007 11:54:43 -0400
Subject: [cisco-voip] VGW not registered in CCM but is on GW
References: <f84a38d30708010850u7a879f08i8716b926e6a6b22a@mail.gmail.com>
Message-ID: <00a701c7d454$4725cef0$47196883@cfs.uoguelph.ca>

I believe all layers have to be up before it will show registered on CallManager. But it will show connected to CallManager subscribers in the show ccm on the router as long as data connectivity from the router to the subscribers is available.

That is what I experienced.

You can try a cross-over cable between two local ports to simulate SP connectivity or a loopback might work as well.

http://www.ciscohq.com/FAQs/Cisco-FAQs/Building-T1-Looback-and-Crossover-Cable.html
--------------------------------------------------------------------------------
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
  ----- Original Message ----- 
  From: Scott Voll 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, August 01, 2007 11:50 AM
  Subject: [cisco-voip] VGW not registered in CCM but is on GW


  I have a 3845 running 12.4(15)T1 via MGCP to the CM 5.1.2.

  on the VGW a show ccm shows registered but on the CM is shows the correct IP address but unregistered.

  I have the service MGCPapp command on my dial-peers  and all the normal ccm-manager commands along with MGCP commands.

  is it not registered because I don't have anything plugged into the PRI ports?

  TIA

  Scott


------------------------------------------------------------------------------


  _______________________________________________
  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  Wed Aug  1 11:52:44 2007
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed, 1 Aug 2007 11:52:44 -0400
Subject: [cisco-voip] VGW not registered in CCM but is on GW
References: <f84a38d30708010850u7a879f08i8716b926e6a6b22a@mail.gmail.com>
Message-ID: <FDC137145729AD438CDCA0F9FC6F7236D8E44E@mail.netechcorp.com>

That is correct, it won't show registered until your PRI comes up.
 
Jason Burton  CCIE - Voice #17319
Netech Corporation
jburton at netechcorp.com
Ph:(317)581-1900 x416

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Scott Voll
Sent: Wed 8/1/2007 11:50 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VGW not registered in CCM but is on GW


I have a 3845 running 12.4(15)T1 via MGCP to the CM 5.1.2.
 
on the VGW a show ccm shows registered but on the CM is shows the correct IP address but unregistered.
 
I have the service MGCPapp command on my dial-peers  and all the normal ccm-manager commands along with MGCP commands.
 
is it not registered because I don't have anything plugged into the PRI ports?
 
TIA
 
Scott

From AFrutin at apcapital.com  Wed Aug  1 12:44:48 2007
From: AFrutin at apcapital.com (Frutin, Amy)
Date: Wed, 1 Aug 2007 12:44:48 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
Message-ID: <AB0A35F2486F81468971AA68558BE71906B61479@smieln01winm01.micoanet.com>

There's some info in the 'CallManager Best Practices' book on using an
Excel formula to convert the 32-bit signed integer value in CDR to
dotted-decimal format:

--------------------

The following steps show you how to convert an IP address to
human-readable format. After you have exported a CDR into an Excel file,
open the spreadsheet, add a row at the top, and follow these steps:

Step 1. Install or verify that the optional Analysis ToolPak package has
been installed (check for the Analysis ToolPak under Tools > Add-Ins).
Without the optional ToolPak installed, the formula does not operate.
You can install the Analysis ToolPak from the Microsoft Office Suite
installations CDs or check the Microsoft Office\Office\ Library\Analysis
directory.

Step 2. Insert the following long formula into the cell to display the
value as an IP address. Replace all four instances of H3 with the field
in the spreadsheet that you want to convert to a human-readable form:

=CONCATENATE(HEX2DEC(MID((DEC2HEX(ABS(H3),8)),7,2)),".",HEX2DEC(MID((DEC
2HEX(ABS(H3),8)),5,2)),".",HEX2DEC(MID((DEC2HEX(ABS(H3),8)),3,2)),".",HE
X2DEC(MID((DEC2HEX(ABS(H3),8)),1,2)))

The IP address is displayed in dotted-decimal format.

--------------------

 

I found that it wasn't calculating correctly in some cases, depending on
the length of the integer, so I ended up using this even longer ugly
formula (I wanted to convert all the IP addresses in Column H of my
exported data, so I inserted a column to the right of that, and copied
the formula all the way down):

 

=CONCATENATE(HEX2DEC(RIGHT(DEC2HEX(H2,8),2)),".",HEX2DEC(MID(DEC2HEX(H2,
8),LEN(DEC2HEX(H2,8))-3,2)),".",HEX2DEC(MID(DEC2HEX(H2,8),LEN(DEC2HEX(H2
,8))-5,2)),".",HEX2DEC(MID(DEC2HEX(H2,8),LEN(DEC2HEX(H2,8))-7,2)))


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From wsisk at cisco.com  Wed Aug  1 13:14:31 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 01 Aug 2007 13:14:31 -0400
Subject: [cisco-voip] VGW not registered in CCM but is on GW
In-Reply-To: <f84a38d30708010850u7a879f08i8716b926e6a6b22a@mail.gmail.com>
References: <f84a38d30708010850u7a879f08i8716b926e6a6b22a@mail.gmail.com>
Message-ID: <46B0BF77.6040001@cisco.com>

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From SCASPER at mtb.com  Wed Aug  1 13:22:52 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Wed, 01 Aug 2007 13:22:52 -0400
Subject: [cisco-voip] Duplicate CDR globalCallID_callID
In-Reply-To: <46B094D6.1000601@cisco.com>
References: <46B03F82.9874.0039.0@mtb.com> <46B094D6.1000601@cisco.com>
Message-ID: <46B0892E.9874.0039.0@mtb.com>

I did but I may be blind. I see no reference to this particular field.
 
Steve

>>> "Wes Sisk" <wsisk at cisco.com> 8/1/2007 10:12 AM >>>
did you check the conference section of the CDR definition doc?
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_programming_usage_guide09186a00806ed410.html#wp61027 

/Wes

STEVEN CASPER wrote: 


Do CDR records involved in Conference calls or other modifications use the same globalCallID_callID as the originating call? I am seeing duplicates on the same server and Call Manager has not been restarted in a while. Any other thoughts?
 
 
Thanks,

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026 
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From svoll.voip at gmail.com  Wed Aug  1 13:30:55 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 1 Aug 2007 10:30:55 -0700
Subject: [cisco-voip] CM 5.1.3?
Message-ID: <f84a38d30708011030h57beeb31j6214b02b27687026@mail.gmail.com>

I thought I heard that CM 5.1.3 was coming out this week.  is it not?

Scott
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From wsisk at cisco.com  Wed Aug  1 13:37:20 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 01 Aug 2007 13:37:20 -0400
Subject: [cisco-voip] CM 5.1.3?
In-Reply-To: <f84a38d30708011030h57beeb31j6214b02b27687026@mail.gmail.com>
References: <f84a38d30708011030h57beeb31j6214b02b27687026@mail.gmail.com>
Message-ID: <46B0C4D0.3030208@cisco.com>

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From LuisYrigoyen at bayviewfinancial.com  Wed Aug  1 14:08:29 2007
From: LuisYrigoyen at bayviewfinancial.com (LuisYrigoyen at bayviewfinancial.com)
Date: Wed, 1 Aug 2007 14:08:29 -0400
Subject: [cisco-voip] DestIPAddr field in CDR record
In-Reply-To: <46B08D9E.8090101@saskin.net>
Message-ID: <OFCD0C6D6F.D633E966-ON8525732A.006365C2-8525732A.0063A7B3@bftg.com>

Using Matthew's excellent query and the SQL built-in dateadd function, you 
can run queries with results for human eyes.

Install Matthew's function and then run somthing like this:

Select 
        dbo.decimaltoip(origipaddr)as origipaddr, 
        dbo.decimaltoip(destipaddr)as destipaddr, 
        callingpartynumber, 
        originalcalledpartynumber, 
        finalcalledpartynumber, 
        dateadd(ss,datetimeconnect,'19700101'), 
        dateadd(ss,datetimedisconnect,'19700101'), 
        origdevicename, 
        destdevicename

From
        calldetailrecord

where
        dateadd(ss,datetimeconnect,'19700101') > '2007-08-01 16:20:51.000'




Luis Yrigoyen

Network Operations
Bayview Financial, L.P.
4425 Ponce de Leon Blvd.
Coral Gables, FL 33146

T 305.817.5083
F 305.646.4744
E luisyrigoyen at bayviewfinancial.com




Matthew Saskin <matt at saskin.net> 
Sent by: cisco-voip-bounces at puck.nether.net
08/01/2007 09:41 AM

To
Lelio Fulgenzi <lelio at uoguelph.ca>
cc
CarlosOrtiz at bayviewfinancial.com, cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] DestIPAddr field in CDR record






IP addresses are stored in the cdr table in a decimal format.  Here is 
an (ugly) SQL server function I wrote a while back that does the 
conversion to dotted decimal format....once again, it's ugly and only 
relatively efficient, but it works :)

-matt

------------------------------------------------------------
------------------------------------------------------------
CREATE  function dbo.decimalToIP(@ip INT)
RETURNS VARCHAR(15)
AS
BEGIN

DECLARE @binip BINARY(4)
SET @binip = CONVERT(BINARY(4), at ip)

DECLARE @charip varchar(8)
SET @charip = CONVERT(varchar(8), at binip)

DECLARE @1 int, @2 int, @3 int, @4 int

SET @4 = ASCII(SUBSTRING(@charip,1,1))
SET @3 = ASCII(SUBSTRING(@charip,2,1))
SET @2 = ASCII(SUBSTRING(@charip,3,1))
SET @1 = ASCII(SUBSTRING(@charip,4,1))

DECLARE @output varchar(15)
SET @output = CONVERT(varchar, at 1) + '.' + CONVERT(varchar, at 2) + '.' + 
CONVERT(varchar, at 3) + '.' + CONVERT(varchar, at 4)
RETURN @output

END
------------------------------------------------------------
------------------------------------------------------------




Lelio Fulgenzi wrote:
> The CAR/ART tool , whatever you call it.
> 
> 
--------------------------------------------------------------------------------
> 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
> 
>     ----- Original Message -----
>     *From:* CarlosOrtiz at bayviewfinancial.com
>     <mailto:CarlosOrtiz at bayviewfinancial.com>
>     *To:* Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
>     *Cc:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     ; cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
>     *Sent:* Wednesday, August 01, 2007 9:27 AM
>     *Subject:* Re: [cisco-voip] DestIPAddr field in CDR record
> 
> 
>     Which tool?
> 
> 
> 
> 
>     *"Lelio Fulgenzi" <lelio at uoguelph.ca <mailto:lelio at uoguelph.ca>>*
>     Sent by: cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
> 
>     08/01/2007 09:23 AM
> 
> 
>     To
>                <CarlosOrtiz at bayviewfinancial.com
>     <mailto:CarlosOrtiz at bayviewfinancial.com>>
>     cc
>                cisco-voip at puck.nether.net <
mailto:cisco-voip at puck.nether.net>
>     Subject
>                Re: [cisco-voip] DestIPAddr field in CDR record
> 
> 
> 
> 
> 
> 
> 
> 
>     sorry, i was using the tool. it probably did the conversion for me.
> 
> 
--------------------------------------------------------------------------------
>     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
>     ----- Original Message -----
>     *From:* _CarlosOrtiz at bayviewfinancial.com_
>     <mailto:CarlosOrtiz at bayviewfinancial.com>
>     *To:* _Lelio Fulgenzi_ <mailto:lelio at uoguelph.ca>
>     *Cc:* _cisco-voip at puck.nether.net_ <
mailto:cisco-voip at puck.nether.net>
>     *Sent:* Wednesday, August 01, 2007 9:16 AM
>     *Subject:* Re: [cisco-voip] DestIPAddr field in CDR record
> 
> 
>     I see a 9 digit numeric number.   Are you looking in the CDR
>     database, calldetailrecord table?  I started searching the archives
>     and I did find this email which you posted a reply to !
> 
> 
>     *_lelio at uoguelph_*
>     <http://www.gossamer-threads.com/lists/engine?user=2532;list=cisco>
> 
>     Sep 27, 2006, 5:00 PM
> 
>     Post #2 of 4 (258 views) _
>     __Permalink_
>     <http://www.gossamer-threads.com/lists/cisco/voip/52547#52547>  
>     *Re: CDR IP Address Conversion* _Remove Highlighting_
>     <
http://www.gossamer-threads.com/lists/cisco/voip/52547?nohighlight=1#52547
>
>     [_In reply to_
>     <
http://www.gossamer-threads.com/lists/cisco/voip/52546?search_string=CDR;#52546
>]
> 
> ------------------------------------------------------------------------
> 
> 
> 
> 
>     found this using google... _
>     __
> 
__http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a0080382d82.html#wp1003527_
> 
> 
> 
>     ----- Original Message -----
>     From: STEVEN CASPER
>     To: cisco-voip[at]puck.nether.net
>     Sent: Wednesday, September 27, 2006 7:55 PM
>     Subject: [cisco-voip] *CDR* IP Address Conversion
> 
> 
> 
>     Does anyone have a tool or formula they could share that can be used 
to
>     convert the signed decimal value in the *CDR* database to an IP
>     address?
> 
>     Thanks!
>     Steve
> 
> 
> 
> 
> 
> 
> 
> 
>     *"Lelio Fulgenzi" <lelio at uoguelph.ca>*
>     Sent by: cisco-voip-bounces at puck.nether.net
> 
>     08/01/2007 09:10 AM
> 
> 
>     To
>                <cisco-voip at puck.nether.net>, 
<CarlosOrtiz at bayviewfinancial.com>
>     cc
> 
>     Subject
>                Re: [cisco-voip] DestIPAddr field in CDR record
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>     I see dotted decimal form a.b.c.d
> 
>     what do you see?
> 
> 
--------------------------------------------------------------------------------
>     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
>     ----- Original Message ----- *
>     From:* _CarlosOrtiz at bayviewfinancial.com_
>     <mailto:CarlosOrtiz at bayviewfinancial.com> *
>     To:* _cisco-voip at puck.nether.net_ <mailto:cisco-voip at puck.nether.net
> *
>     Sent:* Wednesday, August 01, 2007 8:56 AM *
>     Subject:* [cisco-voip] DestIPAddr field in CDR record
> 
> 
>     When viewing the DestIPAddr for a call in the CDR database of the
>     calldetailrecord table a numeric number is shown.  Anyone know how
>     to convert this to IP address format?
> 
>     Carlos
> 
> ------------------------------------------------------------------------
> 
>     _______________________________________________
>     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________________________________________________
>     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 PRuttman at foley.com  Wed Aug  1 18:30:26 2007
From: PRuttman at foley.com (Ruttman, Pete G.)
Date: Wed, 1 Aug 2007 17:30:26 -0500
Subject: [cisco-voip] IOS conferencing question
Message-ID: <5DB17A1560B7EE4186747DF43A6D4C6C6EC48B@WIMKEEX07.foleylaw.com>

We have 3845 routers with NM-HDV2 modules at each site to support local
PRIs and conferencing.  We have PVDM-64 installed on the HDV2 modules.
I am setting up conferencing on the routers and I want to eliminate
unnecessary codec from the list of codecs supported.  We use 7960 and
790x phones.  Callmanager is configured for calls over the WAN to be
g.729 and we use g.711 internally to each office.   A user at any site
can envoke local conference resource to conferenc PSTN calls,  local
phones and phones located across the WAN.

I can leave the following config but it wastes DSP resources:

dspfarm profile 1 conference
 description office Conference 1
 codec g711ulaw
 codec g729r8
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729br8
 associate application SCCP
 maximum sessions 4


What can I remove from the codec list?

Thanks,

Pete Ruttman

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 jsteinberg at gmail.com  Wed Aug  1 21:14:53 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 1 Aug 2007 21:14:53 -0400
Subject: [cisco-voip] No Call Proceeding received for CM to Avaya PBX
	via TDM/H323 trunk
In-Reply-To: <E4B3AF9D5AA27041A2C8A5B9BC5518D90CC2FF30@excausy100.australia.unity>
References: <E4B3AF9D5AA27041A2C8A5B9BC5518D90CC2FF30@excausy100.australia.unity>
Message-ID: <f1b65f880708011814v335936ebr27082fa951f186e9@mail.gmail.com>

Try adding  'no isdn outgoing display-ie' under the serial interface on both
the 3845 & the 2811.

Justin

On 8/1/07, Riley, Andrew <Andrew.Riley at getronics.com> wrote:
>
>  Hi there,
>
> Has anyone had success/experience with connecting an Avaya PBX to a Cisco
> Call Manager via a TDM and H323 link. See topology below. I have seen all
> the Cisco doc's about doing H323 direct to CCM but this is not feasible in
> this particular environment.
>
> IP phone------Cisco CM
> ----mgcp-----3845-----q931------2811-----h323--------Avaya PBX------Phone
>
>                                                                             |
>                                                                        Analog
> Phone
>
> Calls from the analog phone are successful end to the Avaya
> Calls from Avaya to the IP Phone are successful
> Calls from the IP Phone to the Avaya Fail.
>
> Attached are the traces from the 2811 for a failed call. We are seeing the
> h225 setup go out to the Avaya, but the Avaya doesn't respond with a call
> proceeding. The Avaya signal's its phone so it rings but as the 2811 doesn't
> receive CALLPROC h225 times out and the IP Phone gets slow busy tone. The
> strange thing is that the analog phone calling the same pattern work no
> problem.
>
> My suspicion is that the Avaya is dropping the call as it doesn't like
> some information in the IE bits of the h225 setup for a call from the IP
> Phone but as there is less info in the analog call setup it proceeds.  The
> Avaya folk have not been able to provide reason for not sending the CALLPROC
> back.
>
> Any input is much appreciated.
>
> Andrew
>
>
>
> *Andrew Riley
> Technical Consultant - Communication Services
> CCIE # 18285 Voice*
>
> *Getronics Australia Pty Limited*
> Getronics combines the service capabilities of the original Dutch company
> with those of Wang Global, acquired in 1999, and of the Olivetti systems and
> services division. We are ranked second worldwide in network and desktop
> outsourcing and fourth in network consulting and integration (Source: IDC
> 2002-2003).
>
> 2 Minna Close
> Belrose NSW 2085
> Australia
> Tel:       +61 2 9847 7309
> Fax:      +61 2 9847 7378
> Mobile:  +61 402 894 793
> Email:    *andrew.riley at getronics.com*
>
> Communication Services | Security Services | Application Services |
> Workspace Management Services | Technology Transformation Services
>
> *www.getronics.com.au*
>
> Please note that whilst we take all care, neither Getronics nor the sender
> accepts any responsibility for viruses and it is your responsibility to scan
> for viruses. The contents are intended only for use by the addressee and may
> contain confidential and/or privileged material and any use by other than
> the intended recipient is prohibited. If you received this in error, please
> inform the sender and/or addressee immediately and delete the material.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From ErasmuE4 at telkom.co.za  Thu Aug  2 08:11:27 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Thu, 2 Aug 2007 14:11:27 +0200
Subject: [cisco-voip] voicemail on a huntlist
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F0182E6FB@CNTRRA20-XCS01.telkom.co.za>


I have a customer with a lot of hunt groups on call manager 4-2. They
complaint that for some hunt lists they have complains where customers
don't get to a person. I think these hunt groups are under staffed and
now want to suggest that they try using a general voicemail mailbox as
a last member. They have unity connection.

 

It is not clear to me how to implement this. I have done several
normal phones with Unity connection and voicemail works fine. But how
does one get voicemail on a hunt. Does it sit on the hunt pilot?.

 

The hunt pilot is DN 4455. Do I now create mailbox with extension 4455
and then how do I get the hunt to revert to voicemail after it has
exhausted the hunt group members. Basically in ccm - what do I do. I
also want to know how would multiple people then have access to this
one common mailbox. Does it need to be configured as a special mailbox
on connection and how would they dial the mailbox to clean it out.
When you dial the voicemail pilot normally your extension takes you to
your voicemail. Also - how would message waiting indication then works
or would one not use it and people simply have to randomly dial in a
check for messages.

 

Any help appreciated

 

 

 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
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From gokhanciscottl at yahoo.com  Thu Aug  2 09:12:34 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Thu, 2 Aug 2007 06:12:34 -0700 (PDT)
Subject: [cisco-voip] lab design with CCM - IPCC - Unity
Message-ID: <810131.88559.qm@web39604.mail.mud.yahoo.com>

helo

I wanna design a lab for studying
Will it be enough if i buy 2 MCS servers, (one for CCM and one for Unity.)   
For IPCC do i have to buy another MCS or it is already embedded software into CCM?

And what else do  you suggest me to lab design.
thansk a  lot



for CCM            MCS7815I2-K9-CMA2 hw/sw CallMgr 5.1 7815-I2 Appliance, 0 seats

For Unity          MCS-7815-I2-ECS1 MCS-7815-I2; Tower; Unity and Unity Bridge; 2GB; Win2K


       
____________________________________________________________________________________
Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase.
http://farechase.yahoo.com/
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From ErasmuE4 at telkom.co.za  Thu Aug  2 09:26:43 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Thu, 2 Aug 2007 15:26:43 +0200
Subject: [cisco-voip] lab design with CCM - IPCC - Unity
In-Reply-To: <810131.88559.qm@web39604.mail.mud.yahoo.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F0182E71C@CNTRRA20-XCS01.telkom.co.za>


In the recent ccm versions cisco gives a 5 seat ipcc away fro free
co-resident on the call manager purchases - the latest is it is
enhanced and not standard or premium so you should be able to do a lot
with it co-resident on call manager feature wise - but not what
premium can do. But it is separate software and you sen yor pak code
that comes with the call manager software to cisco to get the ipcc
license file to activate it. If you really have a big deployment or
loads of seats (not a lab) then you will need separate hardware.

 

For ccm and unity the two servers should do. You can also consider
running ccm in vmware so that you can save snapshots etc or make
multiple people use one lab but you need a powerful machine and lots
of memory and disk space/  In this group I have ssen tips on how to
use vmware with ver 5 ccm etc.

 

 

 

erik

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of gokhan senol
Sent: Thursday, August 02, 2007 3:13 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] lab design with CCM - IPCC - Unity

 

helo

 

I wanna design a lab for studying

Will it be enough if i buy 2 MCS servers, (one for CCM and one for
Unity.)   

For IPCC do i have to buy another MCS or it is already embedded
software into CCM?

 

And what else do  you suggest me to lab design.

thansk a  lot

 

 

 

for CCM            MCS7815I2-K9-CMA2 hw/sw CallMgr 5.1 7815-I2
Appliance, 0 seats

For Unity          MCS-7815-I2-ECS1 MCS-7815-I2; Tower; Unity and
Unity Bridge; 2GB; Win2K


 

 

________________________________

Pinpoint customers
<http://us.rd.yahoo.com/evt=48250/*http:/searchmarketing.yahoo.com/arp
/sponsoredsearch_v9.php?o=US2226&cmp=Yahoo&ctv=AprNI&s=Y&s2=EM&b=50>
who are looking for what you sell. 



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From CarlosOrtiz at bayviewfinancial.com  Thu Aug  2 09:27:07 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 2 Aug 2007 09:27:07 -0400
Subject: [cisco-voip] Multicast MOH Design
Message-ID: <OFA707C93B.CE1A050D-ON8525732B.00494817-8525732B.0049E535@bftg.com>

Does anyone have a setup where they have a multicast enabled MPLS network 
that is set up to do MOH to the remote sites? 

I am thinking of setting up multiple multicast enabled audio sources and 
streaming them to the remote sites.    Each remote site may need 2-4 
different streams because different business groups want different MOH. 
This would eliminate local streaming from flash because when in SRST the 
local flash MOH is used and would be incorrectly played for all business 
groups.

Any thoughts or known gotchas would be appreciated.

Carlos
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From ErasmuE4 at telkom.co.za  Thu Aug  2 09:41:38 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Thu, 2 Aug 2007 15:41:38 +0200
Subject: [cisco-voip] Multicast MOH Design
In-Reply-To: <OFA707C93B.CE1A050D-ON8525732B.00494817-8525732B.0049E535@bftg.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F0182E726@CNTRRA20-XCS01.telkom.co.za>


I have finally sorted multicast MOH using the branch flashes with
multicast to the pstn gateway etc etc without using the "require mtp
tick box on the ccm gateway config"  but cannot help you with MOH
across MPLS as the networks we have as MPLS option here normally don't
have multicasting enabled and because bandwidth is so costly it is not
viable.

 

However - if you need advice to get it done streaming at branches and
having multicast MOH everywhere - let me know

 

 

erik

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Thursday, August 02, 2007 3:27 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Multicast MOH Design

 


Does anyone have a setup where they have a multicast enabled MPLS
network that is set up to do MOH to the remote sites?   

I am thinking of setting up multiple multicast enabled audio sources
and streaming them to the remote sites.    Each remote site may need
2-4 different streams because different business groups want different
MOH.  This would eliminate local streaming from flash because when in
SRST the local flash MOH is used and would be incorrectly played for
all business groups. 

Any thoughts or known gotchas would be appreciated. 

Carlos



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From CarlosOrtiz at bayviewfinancial.com  Thu Aug  2 09:57:14 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 2 Aug 2007 09:57:14 -0400
Subject: [cisco-voip] Multicast MOH Design
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F0182E726@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <OF545A5BF5.788883A8-ON8525732B.004C4DE5-8525732B.004CA716@bftg.com>

I currently have one office that I setup with  2 different MOH streams. 
One of them is being streamed from the local flash (multicast spoof) and 
the other is being unicast to the other group members (only about 5 
people).  I now have gotten a request for 3 MOH streams and it is a much 
bigger office.  That is why I am asking about streaming all sources from a 
centralized location.  I assume this will work but just wanted to see if 
anyone had any pointers.

Carlos



"Erik Erasmus (E)" <ErasmuE4 at telkom.co.za> 
08/02/2007 09:41 AM

To
<CarlosOrtiz at bayviewfinancial.com>, <cisco-voip at puck.nether.net>
cc

Subject
RE: [cisco-voip] Multicast MOH Design






I have finally sorted multicast MOH using the branch flashes with 
multicast to the pstn gateway etc etc without using the ?require mtp tick 
box on the ccm gateway config?  but cannot help you with MOH across MPLS 
as the networks we have as MPLS option here normally don?t have 
multicasting enabled and because bandwidth is so costly it is not viable.
 
However ? if you need advice to get it done streaming at branches and 
having multicast MOH everywhere ? let me know
 
 
erik
 

From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Thursday, August 02, 2007 3:27 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Multicast MOH Design
 

Does anyone have a setup where they have a multicast enabled MPLS network 
that is set up to do MOH to the remote sites?   

I am thinking of setting up multiple multicast enabled audio sources and 
streaming them to the remote sites.    Each remote site may need 2-4 
different streams because different business groups want different MOH. 
This would eliminate local streaming from flash because when in SRST the 
local flash MOH is used and would be incorrectly played for all business 
groups. 

Any thoughts or known gotchas would be appreciated. 

Carlos

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From wsisk at cisco.com  Thu Aug  2 11:17:01 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 02 Aug 2007 11:17:01 -0400
Subject: [cisco-voip] voicemail on a huntlist
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F0182E6FB@CNTRRA20-XCS01.telkom.co.za>
References: <472E195E692EE74BAE4E7B2880A78C1F0182E6FB@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <46B1F56D.50908@cisco.com>

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From Matt.Slaga at us.didata.com  Thu Aug  2 11:47:19 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 2 Aug 2007 11:47:19 -0400
Subject: [cisco-voip] Multicast MOH Design
In-Reply-To: <OF545A5BF5.788883A8-ON8525732B.004C4DE5-8525732B.004CA716@bftg.com>
References: <472E195E692EE74BAE4E7B2880A78C1F0182E726@CNTRRA20-XCS01.telkom.co.za>
	<OF545A5BF5.788883A8-ON8525732B.004C4DE5-8525732B.004CA716@bftg.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006541817@USNAEXCH.na.didata.local>

You will also need to ensure that your MPLS provider allows for multicast on the wan links.  Most like to charge extra for that ?feature?.

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of CarlosOrtiz at bayviewfinancial.com
Sent: Thursday, August 02, 2007 9:57 AM
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Multicast MOH Design

 


I currently have one office that I setup with  2 different MOH streams.  One of them is being streamed from the local flash (multicast spoof) and the other is being unicast to the other group members (only about 5 people).  I now have gotten a request for 3 MOH streams and it is a much bigger office.  That is why I am asking about streaming all sources from a centralized location.  I assume this will work but just wanted to see if anyone had any pointers. 

Carlos 



"Erik Erasmus (E)" <ErasmuE4 at telkom.co.za> 

08/02/2007 09:41 AM 

To

<CarlosOrtiz at bayviewfinancial.com>, <cisco-voip at puck.nether.net> 

cc

	
Subject

RE: [cisco-voip] Multicast MOH Design

 

		




I have finally sorted multicast MOH using the branch flashes with multicast to the pstn gateway etc etc without using the ?require mtp tick box on the ccm gateway config?  but cannot help you with MOH across MPLS as the networks we have as MPLS option here normally don?t have multicasting enabled and because bandwidth is so costly it is not viable. 
  
However ? if you need advice to get it done streaming at branches and having multicast MOH everywhere ? let me know 
  
  
erik 
  

 

________________________________


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of CarlosOrtiz at bayviewfinancial.com
Sent: Thursday, August 02, 2007 3:27 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Multicast MOH Design 
  

Does anyone have a setup where they have a multicast enabled MPLS network that is set up to do MOH to the remote sites?   

I am thinking of setting up multiple multicast enabled audio sources and streaming them to the remote sites.    Each remote site may need 2-4 different streams because different business groups want different MOH.  This would eliminate local streaming from flash because when in SRST the local flash MOH is used and would be incorrectly played for all business groups. 

Any thoughts or known gotchas would be appreciated. 

Carlos 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

 



-----------------------------------------
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
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From matt at saskin.net  Thu Aug  2 13:04:41 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 02 Aug 2007 13:04:41 -0400
Subject: [cisco-voip] Multicast MOH Design
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006541817@USNAEXCH.na.didata.local>
References: <472E195E692EE74BAE4E7B2880A78C1F0182E726@CNTRRA20-XCS01.telkom.co.za>	<OF545A5BF5.788883A8-ON8525732B.004C4DE5-8525732B.004CA716@bftg.com>
	<C1FE15183DA37645BC0633BC604E44F006541817@USNAEXCH.na.didata.local>
Message-ID: <46B20EA9.5080402@saskin.net>

Poor mans fix - build GRE tunnels across the MPLS links and do whatever 
the heck you want, at an additional administrative cost of course...

-matt

Matt Slaga (US) wrote:
> You will also need to ensure that your MPLS provider allows for 
> multicast on the wan links.  Most like to charge extra for that ?feature?.
> 
>  
> 
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of 
> *CarlosOrtiz at bayviewfinancial.com
> *Sent:* Thursday, August 02, 2007 9:57 AM
> *To:* Erik Erasmus (E)
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Multicast MOH Design
> 
>  
> 
> 
> I currently have one office that I setup with  2 different MOH streams. 
>  One of them is being streamed from the local flash (multicast spoof) 
> and the other is being unicast to the other group members (only about 5 
> people).  I now have gotten a request for 3 MOH streams and it is a much 
> bigger office.  That is why I am asking about streaming all sources from 
> a centralized location.  I assume this will work but just wanted to see 
> if anyone had any pointers.
> 
> Carlos
> 
> *"Erik Erasmus (E)" <ErasmuE4 at telkom.co.za>*
> 
> 08/02/2007 09:41 AM
> 
> 	
> 
> To
> 
> 	
> 
> <CarlosOrtiz at bayviewfinancial.com>, <cisco-voip at puck.nether.net>
> 
> cc
> 
> 	
> 
> Subject
> 
> 	
> 
> RE: [cisco-voip] Multicast MOH Design
> 
>  
> 
> 	
> 
> 
> 
> 
> I have finally sorted multicast MOH using the branch flashes with 
> multicast to the pstn gateway etc etc without using the ?require mtp 
> tick box on the ccm gateway config?  but cannot help you with MOH across 
> MPLS as the networks we have as MPLS option here normally don?t have 
> multicasting enabled and because bandwidth is so costly it is not viable.
>  
> However ? if you need advice to get it done streaming at branches and 
> having multicast MOH everywhere ? let me know
>  
>  
> erik
>  
> 
>  
> 
> ------------------------------------------------------------------------
> 
> 
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of 
> *CarlosOrtiz at bayviewfinancial.com*
> Sent:* Thursday, August 02, 2007 3:27 PM*
> To:* cisco-voip at puck.nether.net*
> Subject:* [cisco-voip] Multicast MOH Design
>  
> 
> Does anyone have a setup where they have a multicast enabled MPLS 
> network that is set up to do MOH to the remote sites?  
> 
> I am thinking of setting up multiple multicast enabled audio sources and 
> streaming them to the remote sites.    Each remote site may need 2-4 
> different streams because different business groups want different MOH. 
>  This would eliminate local streaming from flash because when in SRST 
> the local flash MOH is used and would be incorrectly played for all 
> business groups.
> 
> Any thoughts or known gotchas would be appreciated.
> 
> Carlos
> 
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> This e-mail and its contents are subject to the Telkom SA Limited
> e-mail legal notice available at
> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> 
>  
> 
> ------------------------------------------------------------------------
> 
> * 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  Thu Aug  2 13:53:50 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 2 Aug 2007 10:53:50 -0700
Subject: [cisco-voip] FXS with DID = unregistered
Message-ID: <f84a38d30708021053gb41721eue02b45d464d43408@mail.gmail.com>

newbee question:

I've never used a FXS on a VGW before.  Always used ATA's.

so I have a EVM-HD and I setup the first FXS port. it registers just fine.
I put a analog phone on it and it works but has no DN so I can't call it.

CM 5.1.2 I place a DN on the FXS port and it goes unregistered and will not
register.  Am I doing something wrong?

Through me a bone.

Many thanks

Scott
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From d4rkf1ber at gmail.com  Thu Aug  2 14:37:48 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Thu, 2 Aug 2007 13:37:48 -0500
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
Message-ID: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>

In the last couple of weeks I have been dealing with e911/911 issues
internally at work.  In trying to resolve and provide the best possible
solution for 911 calling to all employees throughout the region I ended
having to pull in our legal and regulatory folks to address some of these
issues and what they felt we needed to provide at a minimum to our
employees, so that we make sure we are completely covered.

In doing so I begin to layout the network and phone system for them
explaining what we have and where we are going and such.  Upon seeing this
the regulatory folk begin telling me that this is wrong / illegal.
Basically saying that the way we have deployed phones and Call Manager and
such is wrong and needs to be corrected.  I was seriously taken aback by
this, never did I question our phone deployment if you will, I mean heck
this meeting was about 911 you know.

The first thing they tell me is that DID's are for internal calling / usage,
and not meant for external or incoming calling?  I was like blown away, this
is completely OPPOSITE of what I have always thought and known.  DID's are
direct inward dialing, you can 100 DID's if you will on a PRI from your
service provider and assign them to individuals internally so that users can
have a direct inward number from the outside.  Heck even wikipedia "which I
know is not the end all source of all knowledge" but fairly reliable states
exactly what I have always known DID's as.

Next they begin telling me I have to get circuits and DID's / numbers that
correspond to each of my physical locations!

Basically, right now we have various small locations spread out in different
cities.  Say as an example Houston, Dallas and San Antonio.  Main office say
is Dallas, my call managers and gateways are there in Dallas.  I get PRI's
to terminate there and I have a large block of DID's all for Dallas.

All ip phones are setup with four digit extension, and are tied to a DID
from Dallas even though the phone and user may be in Houston or something.
User in Houston places a call it goes out of Dallas, and incoming calls
obviously go into Dallas then over our fiber to Houston.  Blah blah, nothing
new there I know alot of places that do the same thing.

Anyway, so they tell me I can't do this.  That legally I have to get
circuits in each market, and provide those users numbers in that market.

I just don't buy this.  I mean the past three employees I have been at and
managed the call managers they all had similar setups and I am not the one
who set them up so I know I was not the only one who believed it was
perfectly acceptable to do things this way.

I would love some facts to use to show that it is perfectly acceptable to do
this.  I can't find anything from a legal perspective to substantiate what
they said or what I believe.

I even pointed out VoIP providers like Vonage and stuff, and said if it was
illegal to do then don't you think these companies that base their business
on just that sort of thing would exist?  That's one of the main selling
points you can get a number from any market in the US pretty much no matter
where you live.  Of course their answer was they are regulated differently.
And who knows maybe so.

Any thoughts or arguments out there on this?  Would love to be able to point
to some law or case or something regarding this rather then just thoughts
and examples since they would have more weight for me to prove my point.
But I would love to hear anything at this point.
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From mh at markholloway.com  Thu Aug  2 14:59:17 2007
From: mh at markholloway.com (Mark Holloway)
Date: Thu, 2 Aug 2007 11:59:17 -0700
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
In-Reply-To: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
Message-ID: <000901c7d537$3a754530$af5fcf90$@com>

Hey man, had to skim through this one quick (busy day at work) but let me
see if I'm getting this right.  You are assigning DID's from Dallas to users
in Houston and San Antonio?  Yikes! How are you handling 911 for Houston/San
Antonio users?  You should have an FXO on Houston's router and San Antonio's
router with a 911 dial peer that displays a Houston or San Antonio TN to
their local PSAP.  You are in major FCC violation for not providing local
911 access.  If Houston dials 911 and goes to a Dallas PSAP, that person is
screwed.  That's a major law suit and a major FCC violation.  The reason
Vonage can do it is they have the ability to give you a DID anywhere, but
route 911 to your local PSAP.  You definitely need some FXO cards and 911
dial peers in each of your remote locations.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dark Fiber
Sent: Thursday, August 02, 2007 11:38 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Need some facts regarding DID's, Enterprise VoIP and
LATAs

 

In the last couple of weeks I have been dealing with e911/911 issues
internally at work.  In trying to resolve and provide the best possible
solution for 911 calling to all employees throughout the region I ended
having to pull in our legal and regulatory folks to address some of these
issues and what they felt we needed to provide at a minimum to our
employees, so that we make sure we are completely covered. 

 

In doing so I begin to layout the network and phone system for them
explaining what we have and where we are going and such.  Upon seeing this
the regulatory folk begin telling me that this is wrong / illegal.
Basically saying that the way we have deployed phones and Call Manager and
such is wrong and needs to be corrected.  I was seriously taken aback by
this, never did I question our phone deployment if you will, I mean heck
this meeting was about 911 you know. 

 

The first thing they tell me is that DID's are for internal calling / usage,
and not meant for external or incoming calling?  I was like blown away, this
is completely OPPOSITE of what I have always thought and known.  DID's are
direct inward dialing, you can 100 DID's if you will on a PRI from your
service provider and assign them to individuals internally so that users can
have a direct inward number from the outside.  Heck even wikipedia "which I
know is not the end all source of all knowledge" but fairly reliable states
exactly what I have always known DID's as. 

 

Next they begin telling me I have to get circuits and DID's / numbers that
correspond to each of my physical locations!

 

Basically, right now we have various small locations spread out in different
cities.  Say as an example Houston, Dallas and San Antonio.  Main office say
is Dallas, my call managers and gateways are there in Dallas.  I get PRI's
to terminate there and I have a large block of DID's all for Dallas. 

 

All ip phones are setup with four digit extension, and are tied to a DID
from Dallas even though the phone and user may be in Houston or something.
User in Houston places a call it goes out of Dallas, and incoming calls
obviously go into Dallas then over our fiber to Houston.  Blah blah, nothing
new there I know alot of places that do the same thing. 

 

Anyway, so they tell me I can't do this.  That legally I have to get
circuits in each market, and provide those users numbers in that market.  

 

I just don't buy this.  I mean the past three employees I have been at and
managed the call managers they all had similar setups and I am not the one
who set them up so I know I was not the only one who believed it was
perfectly acceptable to do things this way. 

 

I would love some facts to use to show that it is perfectly acceptable to do
this.  I can't find anything from a legal perspective to substantiate what
they said or what I believe.

 

I even pointed out VoIP providers like Vonage and stuff, and said if it was
illegal to do then don't you think these companies that base their business
on just that sort of thing would exist?  That's one of the main selling
points you can get a number from any market in the US pretty much no matter
where you live.  Of course their answer was they are regulated differently.
And who knows maybe so. 

 

Any thoughts or arguments out there on this?  Would love to be able to point
to some law or case or something regarding this rather then just thoughts
and examples since they would have more weight for me to prove my point.
But I would love to hear anything at this point. 

 

 

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From bob at smalltime.com  Thu Aug  2 15:05:45 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Thu, 02 Aug 2007 14:05:45 -0500
Subject: [cisco-voip] Need some facts regarding DID's,
 Enterprise VoIP and LATAs
In-Reply-To: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
Message-ID: <46B22B09.90708@smalltime.com>

If you're going to provide for 911 at each location, then you're going 
to need to get _something_ at that site.  It can be as simple as a 
standard analog line that you connect to a FXO port on a voice gateway, 
with appropriate routing in CM to send a 911 call out from that location.

PSAPs (Public Safety Answering Points) aren't necessarily linked to each 
other, so if someone in Houston is having a heart attack, and all calls 
go out from Dallas, there's nothing much that the PSAP in Dallas can do 
about it.

You don't need to renumber, or give users in your other locations 
numbers "DIDs" in those locations.  Just make sure that a call in 
Houston goes out through Houston, and a call in Dallas goes out through 
Dallas, etc.  Which pretty much means that you're going to need a 
"circuit" at those locations.

One caveat:  you may be able to get someone like one of the E911 VoIP 
service providers involved (www.911enable.com is one); you'd setup 
something like a SIP trunk and route e911 calls to them.  You'd tell 
them which DIDs are located in which city, and allow them to handle the 
911 call and send it to the correct PSAP.  (In which case 911 calls 
wouldn't traverse your PRI - they'd go out over your ethernet to the 
provider)


From matt at saskin.net  Thu Aug  2 15:09:56 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 02 Aug 2007 15:09:56 -0400
Subject: [cisco-voip] Need some facts regarding DID's,
 Enterprise VoIP and LATAs
In-Reply-To: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
Message-ID: <46B22C04.5010005@saskin.net>

The other responses answered the 911 portions correctly.

WRT the other issue, there is no legal problem with what you're doing - 
it's similar to tail-end hop-off.  It can become questionable overseas 
but I'm pretty certain that it's completely allowed within the United 
States.  In fact, the only country I know it's illegal in that I do work 
with is India.

-matt

Dark Fiber wrote:
> In the last couple of weeks I have been dealing with e911/911 issues 
> internally at work.  In trying to resolve and provide the best possible 
> solution for 911 calling to all employees throughout the region I ended 
> having to pull in our legal and regulatory folks to address some of 
> these issues and what they felt we needed to provide at a minimum to our 
> employees, so that we make sure we are completely covered.
>  
> In doing so I begin to layout the network and phone system for them 
> explaining what we have and where we are going and such.  Upon seeing 
> this the regulatory folk begin telling me that this is wrong / illegal.  
> Basically saying that the way we have deployed phones and Call 
> Manager and such is wrong and needs to be corrected.  I was seriously 
> taken aback by this, never did I question our phone deployment if you 
> will, I mean heck this meeting was about 911 you know.
>  
> The first thing they tell me is that DID's are for internal calling / 
> usage, and not meant for external or incoming calling?  I was like blown 
> away, this is completely OPPOSITE of what I have always thought and 
> known.  DID's are direct inward dialing, you can 100 DID's if you will 
> on a PRI from your service provider and assign them to individuals 
> internally so that users can have a direct inward number from the 
> outside.  Heck even wikipedia "which I know is not the end all source of 
> all knowledge" but fairly reliable states exactly what I have always 
> known DID's as.
>  
> Next they begin telling me I have to get circuits and DID's / numbers 
> that correspond to each of my physical locations!
>  
> Basically, right now we have various small locations spread out in 
> different cities.  Say as an example Houston, Dallas and San Antonio.  
> Main office say is Dallas, my call managers and gateways are there in 
> Dallas.  I get PRI's to terminate there and I have a large block of 
> DID's all for Dallas.
>  
> All ip phones are setup with four digit extension, and are tied to a DID 
> from Dallas even though the phone and user may be in Houston or 
> something.  User in Houston places a call it goes out of Dallas, and 
> incoming calls obviously go into Dallas then over our fiber to Houston.  
> Blah blah, nothing new there I know alot of places that do the same thing.
>  
> Anyway, so they tell me I can't do this.  That legally I have to get 
> circuits in each market, and provide those users numbers in that market. 
>  
> I just don't buy this.  I mean the past three employees I have been at 
> and managed the call managers they all had similar setups and I am not 
> the one who set them up so I know I was not the only one who believed it 
> was perfectly acceptable to do things this way.
>  
> I would love some facts to use to show that it is perfectly acceptable 
> to do this.  I can't find anything from a legal perspective to 
> substantiate what they said or what I believe.
>  
> I even pointed out VoIP providers like Vonage and stuff, and said if it 
> was illegal to do then don't you think these companies that base their 
> business on just that sort of thing would exist?  That's one of the main 
> selling points you can get a number from any market in the US pretty 
> much no matter where you live.  Of course their answer was they are 
> regulated differently.  And who knows maybe so.
>  
> Any thoughts or arguments out there on this?  Would love to be able to 
> point to some law or case or something regarding this rather then just 
> thoughts and examples since they would have more weight for me to prove 
> my point.  But I would love to hear anything at this point.
>  
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From BillS at psu.edu  Thu Aug  2 15:12:30 2007
From: BillS at psu.edu (Bill Simon)
Date: Thu, 2 Aug 2007 15:12:30 -0400
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
In-Reply-To: <46B22C04.5010005@saskin.net>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
	<46B22C04.5010005@saskin.net>
Message-ID: <01E9BE53DDA8974BA1A4E4C6DB8BA15B011CEFDD@tnshestia.tnsadmin.tns.psu.edu>

Can you confirm this?

We were told that, unless licensed as a "telco," (CLEC) we may not do
any toll-bypass by sending traffic over VoIP to a location with a local
gateway.
  

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matthew Saskin
Sent: Thursday, August 02, 2007 3:10 PM
To: Dark Fiber
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Need some facts regarding DID's, Enterprise
VoIP and LATAs

The other responses answered the 911 portions correctly.

WRT the other issue, there is no legal problem with what you're doing - 
it's similar to tail-end hop-off.  It can become questionable overseas 
but I'm pretty certain that it's completely allowed within the United 
States.  In fact, the only country I know it's illegal in that I do work

with is India.

-matt

Dark Fiber wrote:
> In the last couple of weeks I have been dealing with e911/911 issues 
> internally at work.  In trying to resolve and provide the best
possible 
> solution for 911 calling to all employees throughout the region I
ended 
> having to pull in our legal and regulatory folks to address some of 
> these issues and what they felt we needed to provide at a minimum to
our 
> employees, so that we make sure we are completely covered.
>  
> In doing so I begin to layout the network and phone system for them 
> explaining what we have and where we are going and such.  Upon seeing 
> this the regulatory folk begin telling me that this is wrong /
illegal.  
> Basically saying that the way we have deployed phones and Call 
> Manager and such is wrong and needs to be corrected.  I was seriously 
> taken aback by this, never did I question our phone deployment if you 
> will, I mean heck this meeting was about 911 you know.
>  
> The first thing they tell me is that DID's are for internal calling / 
> usage, and not meant for external or incoming calling?  I was like
blown 
> away, this is completely OPPOSITE of what I have always thought and 
> known.  DID's are direct inward dialing, you can 100 DID's if you will

> on a PRI from your service provider and assign them to individuals 
> internally so that users can have a direct inward number from the 
> outside.  Heck even wikipedia "which I know is not the end all source
of 
> all knowledge" but fairly reliable states exactly what I have always 
> known DID's as.
>  
> Next they begin telling me I have to get circuits and DID's / numbers 
> that correspond to each of my physical locations!
>  
> Basically, right now we have various small locations spread out in 
> different cities.  Say as an example Houston, Dallas and San Antonio.

> Main office say is Dallas, my call managers and gateways are there in 
> Dallas.  I get PRI's to terminate there and I have a large block of 
> DID's all for Dallas.
>  
> All ip phones are setup with four digit extension, and are tied to a
DID 
> from Dallas even though the phone and user may be in Houston or 
> something.  User in Houston places a call it goes out of Dallas, and 
> incoming calls obviously go into Dallas then over our fiber to
Houston.  
> Blah blah, nothing new there I know alot of places that do the same
thing.
>  
> Anyway, so they tell me I can't do this.  That legally I have to get 
> circuits in each market, and provide those users numbers in that
market. 
>  
> I just don't buy this.  I mean the past three employees I have been at

> and managed the call managers they all had similar setups and I am not

> the one who set them up so I know I was not the only one who believed
it 
> was perfectly acceptable to do things this way.
>  
> I would love some facts to use to show that it is perfectly acceptable

> to do this.  I can't find anything from a legal perspective to 
> substantiate what they said or what I believe.
>  
> I even pointed out VoIP providers like Vonage and stuff, and said if
it 
> was illegal to do then don't you think these companies that base their

> business on just that sort of thing would exist?  That's one of the
main 
> selling points you can get a number from any market in the US pretty 
> much no matter where you live.  Of course their answer was they are 
> regulated differently.  And who knows maybe so.
>  
> Any thoughts or arguments out there on this?  Would love to be able to

> point to some law or case or something regarding this rather then just

> thoughts and examples since they would have more weight for me to
prove 
> my point.  But I would love to hear anything at this point.
>  
>  
> 
> 
>
------------------------------------------------------------------------
> 
> _______________________________________________
> 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 e.dumbledore at googlemail.com  Thu Aug  2 15:19:37 2007
From: e.dumbledore at googlemail.com (Stift)
Date: Thu, 02 Aug 2007 21:19:37 +0200
Subject: [cisco-voip] Presence on CME
Message-ID: <f8taob$qjp$1@sea.gmane.org>


I have configured presence service on a CME 4.2 (12.4.11.XW2)

The BLF-speeddial works fine, but the presence call-list seems not 
working. I tested it with a 7970, but in no directory or call list 
(missed calls, placed calls and so on) i can't see the status of the 
monitored line.

Is it a bug?


From matt at saskin.net  Thu Aug  2 15:24:02 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 02 Aug 2007 15:24:02 -0400
Subject: [cisco-voip] Need some facts regarding DID's,
 Enterprise VoIP and LATAs
In-Reply-To: <01E9BE53DDA8974BA1A4E4C6DB8BA15B011CEFDD@tnshestia.tnsadmin.tns.psu.edu>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
	<46B22C04.5010005@saskin.net>
	<01E9BE53DDA8974BA1A4E4C6DB8BA15B011CEFDD@tnshestia.tnsadmin.tns.psu.edu>
Message-ID: <46B22F52.9040705@saskin.net>

I'll dig around.  I know ignorance doesn't make it legal, but everyone 
has seen literally hundreds of organizations that do this, including the 
documentation on Cisco's site outlining exactly how tail-end hop-off can 
be configured.

-matt

Bill Simon wrote:
> Can you confirm this?
> 
> We were told that, unless licensed as a "telco," (CLEC) we may not do
> any toll-bypass by sending traffic over VoIP to a location with a local
> gateway.
>   
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matthew Saskin
> Sent: Thursday, August 02, 2007 3:10 PM
> To: Dark Fiber
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Need some facts regarding DID's, Enterprise
> VoIP and LATAs
> 
> The other responses answered the 911 portions correctly.
> 
> WRT the other issue, there is no legal problem with what you're doing - 
> it's similar to tail-end hop-off.  It can become questionable overseas 
> but I'm pretty certain that it's completely allowed within the United 
> States.  In fact, the only country I know it's illegal in that I do work
> 
> with is India.
> 
> -matt
> 
> Dark Fiber wrote:
>> In the last couple of weeks I have been dealing with e911/911 issues 
>> internally at work.  In trying to resolve and provide the best
> possible 
>> solution for 911 calling to all employees throughout the region I
> ended 
>> having to pull in our legal and regulatory folks to address some of 
>> these issues and what they felt we needed to provide at a minimum to
> our 
>> employees, so that we make sure we are completely covered.
>>  
>> In doing so I begin to layout the network and phone system for them 
>> explaining what we have and where we are going and such.  Upon seeing 
>> this the regulatory folk begin telling me that this is wrong /
> illegal.  
>> Basically saying that the way we have deployed phones and Call 
>> Manager and such is wrong and needs to be corrected.  I was seriously 
>> taken aback by this, never did I question our phone deployment if you 
>> will, I mean heck this meeting was about 911 you know.
>>  
>> The first thing they tell me is that DID's are for internal calling / 
>> usage, and not meant for external or incoming calling?  I was like
> blown 
>> away, this is completely OPPOSITE of what I have always thought and 
>> known.  DID's are direct inward dialing, you can 100 DID's if you will
> 
>> on a PRI from your service provider and assign them to individuals 
>> internally so that users can have a direct inward number from the 
>> outside.  Heck even wikipedia "which I know is not the end all source
> of 
>> all knowledge" but fairly reliable states exactly what I have always 
>> known DID's as.
>>  
>> Next they begin telling me I have to get circuits and DID's / numbers 
>> that correspond to each of my physical locations!
>>  
>> Basically, right now we have various small locations spread out in 
>> different cities.  Say as an example Houston, Dallas and San Antonio.
> 
>> Main office say is Dallas, my call managers and gateways are there in 
>> Dallas.  I get PRI's to terminate there and I have a large block of 
>> DID's all for Dallas.
>>  
>> All ip phones are setup with four digit extension, and are tied to a
> DID 
>> from Dallas even though the phone and user may be in Houston or 
>> something.  User in Houston places a call it goes out of Dallas, and 
>> incoming calls obviously go into Dallas then over our fiber to
> Houston.  
>> Blah blah, nothing new there I know alot of places that do the same
> thing.
>>  
>> Anyway, so they tell me I can't do this.  That legally I have to get 
>> circuits in each market, and provide those users numbers in that
> market. 
>>  
>> I just don't buy this.  I mean the past three employees I have been at
> 
>> and managed the call managers they all had similar setups and I am not
> 
>> the one who set them up so I know I was not the only one who believed
> it 
>> was perfectly acceptable to do things this way.
>>  
>> I would love some facts to use to show that it is perfectly acceptable
> 
>> to do this.  I can't find anything from a legal perspective to 
>> substantiate what they said or what I believe.
>>  
>> I even pointed out VoIP providers like Vonage and stuff, and said if
> it 
>> was illegal to do then don't you think these companies that base their
> 
>> business on just that sort of thing would exist?  That's one of the
> main 
>> selling points you can get a number from any market in the US pretty 
>> much no matter where you live.  Of course their answer was they are 
>> regulated differently.  And who knows maybe so.
>>  
>> Any thoughts or arguments out there on this?  Would love to be able to
> 
>> point to some law or case or something regarding this rather then just
> 
>> thoughts and examples since they would have more weight for me to
> prove 
>> my point.  But I would love to hear anything at this point.
>>  
>>  
>>
>>
>>
> ------------------------------------------------------------------------
>> _______________________________________________
>> 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  Thu Aug  2 15:26:14 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 2 Aug 2007 12:26:14 -0700 (PDT)
Subject: [cisco-voip] FXS with DID = unregistered
In-Reply-To: <f84a38d30708021053gb41721eue02b45d464d43408@mail.gmail.com>
Message-ID: <396321.15686.qm@web57208.mail.re3.yahoo.com>

First of all, what kind of VG is this? VG248? VG224?

VG248 ports can be assigned DNs on the VG itself as
well as under CCM.


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

> I've never used a FXS on a VGW before.  Always used
> ATA's.
> 
> so I have a EVM-HD and I setup the first FXS port.
> it registers just fine.
> I put a analog phone on it and it works but has no
> DN so I can't call it.
> 
> CM 5.1.2 I place a DN on the FXS port and it goes
> unregistered and will not
> register.  Am I doing something wrong?
> 
> Through me a bone.
> 
> Many thanks
> 



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

From MillerS at DicksteinShapiro.COM  Thu Aug  2 15:30:48 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Thu, 2 Aug 2007 15:30:48 -0400
Subject: [cisco-voip] Question About the Message Store Wizard
Message-ID: <B9B5B75DF9145F44A21111DD92CC490007C92806@DCEX2.DSMO.COM>

We are migrating our Exchange mail stores to a SAN this weekend and they
will be in a different place from where they currently reside.  We have
been told by some experts that we need to run the permissions wizard and
told by others that we only need to run the Unity Message Store Wizard
in order to fix the mailboxes after the move.
 
We are on Unity 4.2.1 and have a redundant failover server.  Our
Exchange Servers are not clustered.  Does anyone have any experience
running the Message Store Wizard?  I haven't heard much mention of it.
 
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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From svoll.voip at gmail.com  Thu Aug  2 15:59:01 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 2 Aug 2007 12:59:01 -0700
Subject: [cisco-voip] FXS with DID = unregistered
In-Reply-To: <396321.15686.qm@web57208.mail.re3.yahoo.com>
References: <f84a38d30708021053gb41721eue02b45d464d43408@mail.gmail.com>
	<396321.15686.qm@web57208.mail.re3.yahoo.com>
Message-ID: <f84a38d30708021259lce91ea1pdc19e94b03c19995@mail.gmail.com>

3845 with EVM-HD running 12.4 code.

I'm trying to do it via CCM.

Scott


On 8/2/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> First of all, what kind of VG is this? VG248? VG224?
>
> VG248 ports can be assigned DNs on the VG itself as
> well as under CCM.
>
>
> --- Scott Voll <svoll.voip at gmail.com> wrote:
>
> > I've never used a FXS on a VGW before.  Always used
> > ATA's.
> >
> > so I have a EVM-HD and I setup the first FXS port.
> > it registers just fine.
> > I put a analog phone on it and it works but has no
> > DN so I can't call it.
> >
> > CM 5.1.2 I place a DN on the FXS port and it goes
> > unregistered and will not
> > register.  Am I doing something wrong?
> >
> > Through me a bone.
> >
> > Many thanks
> >
>
>
>
>
>
> ____________________________________________________________________________________
> Need a vacation? Get great deals
> to amazing places on Yahoo! Travel.
> http://travel.yahoo.com/
>
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From asobihoudai at yahoo.com  Thu Aug  2 16:05:18 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 2 Aug 2007 13:05:18 -0700 (PDT)
Subject: [cisco-voip] FXS with DID = unregistered
In-Reply-To: <f84a38d30708021259lce91ea1pdc19e94b03c19995@mail.gmail.com>
Message-ID: <281469.7942.qm@web57207.mail.re3.yahoo.com>

Call control protocol ->

H.323? MGCP? SIP? Must be MGCP since you're trying to
control the port via CCM.

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

> 3845 with EVM-HD running 12.4 code.
> 
> I'm trying to do it via CCM.
> 
> Scott
> 
> 
> On 8/2/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> >
> > First of all, what kind of VG is this? VG248?
> VG224?
> >
> > VG248 ports can be assigned DNs on the VG itself
> as
> > well as under CCM.
> >
> >
> > --- Scott Voll <svoll.voip at gmail.com> wrote:
> >
> > > I've never used a FXS on a VGW before.  Always
> used
> > > ATA's.
> > >
> > > so I have a EVM-HD and I setup the first FXS
> port.
> > > it registers just fine.
> > > I put a analog phone on it and it works but has
> no
> > > DN so I can't call it.
> > >
> > > CM 5.1.2 I place a DN on the FXS port and it
> goes
> > > unregistered and will not
> > > register.  Am I doing something wrong?
> > >
> > > Through me a bone.
> > >
> > > Many thanks
> > >
> >
> >
> >
> >
> >
> >
>
____________________________________________________________________________________
> > Need a vacation? Get great deals
> > to amazing places on Yahoo! Travel.
> > http://travel.yahoo.com/
> >
> 



       
____________________________________________________________________________________
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 asobihoudai at yahoo.com  Thu Aug  2 15:30:14 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 2 Aug 2007 12:30:14 -0700 (PDT)
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
In-Reply-To: <46B22B09.90708@smalltime.com>
Message-ID: <14687.16406.qm@web57203.mail.re3.yahoo.com>

What you could do is, if management would be kosher
with this, have several dedicated 911 red phones with
straight POTS lines attached to them bypassing any 911
router business whatsoever. This is what one of our
customers did for several of their offices and they
seem to work well with the solution.

--- Robert Kulagowski <bob at smalltime.com> wrote:

> If you're going to provide for 911 at each location,
> then you're going 
> to need to get _something_ at that site.  It can be
> as simple as a 
> standard analog line that you connect to a FXO port
> on a voice gateway, 
> with appropriate routing in CM to send a 911 call
> out from that location.
> 
> PSAPs (Public Safety Answering Points) aren't
> necessarily linked to each 
> other, so if someone in Houston is having a heart
> attack, and all calls 
> go out from Dallas, there's nothing much that the
> PSAP in Dallas can do 
> about it.
> 
> You don't need to renumber, or give users in your
> other locations 
> numbers "DIDs" in those locations.  Just make sure
> that a call in 
> Houston goes out through Houston, and a call in
> Dallas goes out through 
> Dallas, etc.  Which pretty much means that you're
> going to need a 
> "circuit" at those locations.
> 
> One caveat:  you may be able to get someone like one
> of the E911 VoIP 
> service providers involved (www.911enable.com is
> one); you'd setup 
> something like a SIP trunk and route e911 calls to
> them.  You'd tell 
> them which DIDs are located in which city, and allow
> them to handle the 
> 911 call and send it to the correct PSAP.  (In which
> case 911 calls 
> wouldn't traverse your PRI - they'd go out over your
> ethernet to the 
> provider)
> 
> _______________________________________________
> 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 wsisk at cisco.com  Thu Aug  2 17:19:51 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 02 Aug 2007 17:19:51 -0400
Subject: [cisco-voip] FXS with DID = unregistered
In-Reply-To: <f84a38d30708021259lce91ea1pdc19e94b03c19995@mail.gmail.com>
References: <f84a38d30708021053gb41721eue02b45d464d43408@mail.gmail.com>	<396321.15686.qm@web57208.mail.re3.yahoo.com>
	<f84a38d30708021259lce91ea1pdc19e94b03c19995@mail.gmail.com>
Message-ID: <46B24A77.2090006@cisco.com>

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From d4rkf1ber at gmail.com  Thu Aug  2 17:33:42 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Thu, 2 Aug 2007 16:33:42 -0500
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
In-Reply-To: <000901c7d537$3a754530$af5fcf90$@com>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
	<000901c7d537$3a754530$af5fcf90$@com>
Message-ID: <a1b07ea60708021433oa62e30p9a7f56c5b481234@mail.gmail.com>

Actually 911 is not the issue, we do have local gateways with a couple of
pstn lines at each location strictly for 911.  So if someone in Houston
calls 911 it routes out the local gateway to reach the right psap.

Appreciate the response though.


On 8/2/07, Mark Holloway <mh at markholloway.com> wrote:
>
>  Hey man, had to skim through this one quick (busy day at work) but let me
> see if I'm getting this right.  You are assigning DID's from Dallas to users
> in Houston and San Antonio?  Yikes! How are you handling 911 for Houston/San
> Antonio users?  You should have an FXO on Houston's router and San Antonio's
> router with a 911 dial peer that displays a Houston or San Antonio TN to
> their local PSAP.  You are in major FCC violation for not providing local
> 911 access.  If Houston dials 911 and goes to a Dallas PSAP, that person is
> screwed.  That's a major law suit and a major FCC violation.  The reason
> Vonage can do it is they have the ability to give you a DID anywhere, but
> route 911 to your local PSAP.  You definitely need some FXO cards and 911
> dial peers in each of your remote locations.
>
>
>
>
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Dark Fiber
> *Sent:* Thursday, August 02, 2007 11:38 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Need some facts regarding DID's, Enterprise VoIP
> and LATAs
>
>
>
> In the last couple of weeks I have been dealing with e911/911 issues
> internally at work.  In trying to resolve and provide the best possible
> solution for 911 calling to all employees throughout the region I ended
> having to pull in our legal and regulatory folks to address some of these
> issues and what they felt we needed to provide at a minimum to our
> employees, so that we make sure we are completely covered.
>
>
>
> In doing so I begin to layout the network and phone system for them
> explaining what we have and where we are going and such.  Upon seeing this
> the regulatory folk begin telling me that this is wrong / illegal.
> Basically saying that the way we have deployed phones and Call Manager and
> such is wrong and needs to be corrected.  I was seriously taken aback by
> this, never did I question our phone deployment if you will, I mean heck
> this meeting was about 911 you know.
>
>
>
> The first thing they tell me is that DID's are for internal calling /
> usage, and not meant for external or incoming calling?  I was like blown
> away, this is completely OPPOSITE of what I have always thought and known.
> DID's are direct inward dialing, you can 100 DID's if you will on a PRI from
> your service provider and assign them to individuals internally so that
> users can have a direct inward number from the outside.  Heck even wikipedia
> "which I know is not the end all source of all knowledge" but fairly
> reliable states exactly what I have always known DID's as.
>
>
>
> Next they begin telling me I have to get circuits and DID's / numbers that
> correspond to each of my physical locations!
>
>
>
> Basically, right now we have various small locations spread out in
> different cities.  Say as an example Houston, Dallas and San Antonio.  Main
> office say is Dallas, my call managers and gateways are there in Dallas.  I
> get PRI's to terminate there and I have a large block of DID's all for
> Dallas.
>
>
>
> All ip phones are setup with four digit extension, and are tied to a DID
> from Dallas even though the phone and user may be in Houston or something.
> User in Houston places a call it goes out of Dallas, and incoming calls
> obviously go into Dallas then over our fiber to Houston.  Blah blah, nothing
> new there I know alot of places that do the same thing.
>
>
>
> Anyway, so they tell me I can't do this.  That legally I have to get
> circuits in each market, and provide those users numbers in that market.
>
>
>
> I just don't buy this.  I mean the past three employees I have been at and
> managed the call managers they all had similar setups and I am not the one
> who set them up so I know I was not the only one who believed it was
> perfectly acceptable to do things this way.
>
>
>
> I would love some facts to use to show that it is perfectly acceptable to
> do this.  I can't find anything from a legal perspective to substantiate
> what they said or what I believe.
>
>
>
> I even pointed out VoIP providers like Vonage and stuff, and said if it
> was illegal to do then don't you think these companies that base their
> business on just that sort of thing would exist?  That's one of the main
> selling points you can get a number from any market in the US pretty much no
> matter where you live.  Of course their answer was they are regulated
> differently.  And who knows maybe so.
>
>
>
> Any thoughts or arguments out there on this?  Would love to be able to
> point to some law or case or something regarding this rather then just
> thoughts and examples since they would have more weight for me to prove my
> point.  But I would love to hear anything at this point.
>
>
>
>
>
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From gusangjung at yahoo.com  Thu Aug  2 16:55:22 2007
From: gusangjung at yahoo.com (Gu Sang Jung)
Date: Thu, 2 Aug 2007 13:55:22 -0700 (PDT)
Subject: [cisco-voip] Cisco IP communicator with VPN issue
Message-ID: <90487.7689.qm@web52008.mail.re2.yahoo.com>

Hi guys.

We try to support remote-end user(tele-worker at home). 

The problem is that IP communication does not registered into Callmanager  with some wireless home router on the Cisco VPN clinet.
When it hooked up on the wired port, It worked fine with wired  connection on wireless home router . However, when It try to use wireless mode, it could not registered.
From PC(TCP 2000) port connectivty is fined to Callmanager server. 
Is there some solution how to fix it up? Cisco TAC looks like not so many helpful to this issue.

thanks

Gusang

Sr Tel Enginner broadcom






       
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From mh at markholloway.com  Thu Aug  2 17:52:00 2007
From: mh at markholloway.com (Mark Holloway)
Date: Thu, 2 Aug 2007 14:52:00 -0700
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
In-Reply-To: <a1b07ea60708021433oa62e30p9a7f56c5b481234@mail.gmail.com>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>	
	<000901c7d537$3a754530$af5fcf90$@com>
	<a1b07ea60708021433oa62e30p9a7f56c5b481234@mail.gmail.com>
Message-ID: <001001c7d54f$5b4e4fa0$11eaeee0$@com>

Out of rate center numbers are considered "dirty" and if your phone company
that you're getting the TN's from knew, they might make a big deal about it.


 

From: techguy at gmail.com [mailto:techguy at gmail.com] On Behalf Of Dark Fiber
Sent: Thursday, August 02, 2007 2:34 PM
To: Mark Holloway
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Need some facts regarding DID's, Enterprise VoIP
and LATAs

 

Actually 911 is not the issue, we do have local gateways with a couple of
pstn lines at each location strictly for 911.  So if someone in Houston
calls 911 it routes out the local gateway to reach the right psap.

 

Appreciate the response though.

 

On 8/2/07, Mark Holloway <mh at markholloway.com> wrote: 

Hey man, had to skim through this one quick (busy day at work) but let me
see if I'm getting this right.  You are assigning DID's from Dallas to users
in Houston and San Antonio?  Yikes! How are you handling 911 for Houston/San
Antonio users?  You should have an FXO on Houston's router and San Antonio's
router with a 911 dial peer that displays a Houston or San Antonio TN to
their local PSAP.  You are in major FCC violation for not providing local
911 access.  If Houston dials 911 and goes to a Dallas PSAP, that person is
screwed.  That's a major law suit and a major FCC violation.  The reason
Vonage can do it is they have the ability to give you a DID anywhere, but
route 911 to your local PSAP.  You definitely need some FXO cards and 911
dial peers in each of your remote locations. 

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dark Fiber
Sent: Thursday, August 02, 2007 11:38 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Need some facts regarding DID's, Enterprise VoIP and
LATAs 

 

In the last couple of weeks I have been dealing with e911/911 issues
internally at work.  In trying to resolve and provide the best possible
solution for 911 calling to all employees throughout the region I ended
having to pull in our legal and regulatory folks to address some of these
issues and what they felt we needed to provide at a minimum to our
employees, so that we make sure we are completely covered. 

 

In doing so I begin to layout the network and phone system for them
explaining what we have and where we are going and such.  Upon seeing this
the regulatory folk begin telling me that this is wrong / illegal.
Basically saying that the way we have deployed phones and Call Manager and
such is wrong and needs to be corrected.  I was seriously taken aback by
this, never did I question our phone deployment if you will, I mean heck
this meeting was about 911 you know. 

 

The first thing they tell me is that DID's are for internal calling / usage,
and not meant for external or incoming calling?  I was like blown away, this
is completely OPPOSITE of what I have always thought and known.  DID's are
direct inward dialing, you can 100 DID's if you will on a PRI from your
service provider and assign them to individuals internally so that users can
have a direct inward number from the outside.  Heck even wikipedia "which I
know is not the end all source of all knowledge" but fairly reliable states
exactly what I have always known DID's as. 

 

Next they begin telling me I have to get circuits and DID's / numbers that
correspond to each of my physical locations!

 

Basically, right now we have various small locations spread out in different
cities.  Say as an example Houston, Dallas and San Antonio.  Main office say
is Dallas, my call managers and gateways are there in Dallas.  I get PRI's
to terminate there and I have a large block of DID's all for Dallas. 

 

All ip phones are setup with four digit extension, and are tied to a DID
from Dallas even though the phone and user may be in Houston or something.
User in Houston places a call it goes out of Dallas, and incoming calls
obviously go into Dallas then over our fiber to Houston.  Blah blah, nothing
new there I know alot of places that do the same thing. 

 

Anyway, so they tell me I can't do this.  That legally I have to get
circuits in each market, and provide those users numbers in that market.  

 

I just don't buy this.  I mean the past three employees I have been at and
managed the call managers they all had similar setups and I am not the one
who set them up so I know I was not the only one who believed it was
perfectly acceptable to do things this way. 

 

I would love some facts to use to show that it is perfectly acceptable to do
this.  I can't find anything from a legal perspective to substantiate what
they said or what I believe.

 

I even pointed out VoIP providers like Vonage and stuff, and said if it was
illegal to do then don't you think these companies that base their business
on just that sort of thing would exist?  That's one of the main selling
points you can get a number from any market in the US pretty much no matter
where you live.  Of course their answer was they are regulated differently.
And who knows maybe so. 

 

Any thoughts or arguments out there on this?  Would love to be able to point
to some law or case or something regarding this rather then just thoughts
and examples since they would have more weight for me to prove my point.
But I would love to hear anything at this point. 

 

 

 

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From mh at markholloway.com  Thu Aug  2 16:58:24 2007
From: mh at markholloway.com (Mark Holloway)
Date: Thu, 2 Aug 2007 13:58:24 -0700
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
In-Reply-To: <14687.16406.qm@web57203.mail.re3.yahoo.com>
References: <46B22B09.90708@smalltime.com>
	<14687.16406.qm@web57203.mail.re3.yahoo.com>
Message-ID: <000901c7d547$de051da0$9a0f58e0$@com>

Although companies do this, it's not compliant.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Thursday, August 02, 2007 12:30 PM
To: Robert Kulagowski; Dark Fiber
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Need some facts regarding DID's, Enterprise VoIP
and LATAs

What you could do is, if management would be kosher
with this, have several dedicated 911 red phones with
straight POTS lines attached to them bypassing any 911
router business whatsoever. This is what one of our
customers did for several of their offices and they
seem to work well with the solution.

--- Robert Kulagowski <bob at smalltime.com> wrote:

> If you're going to provide for 911 at each location,
> then you're going 
> to need to get _something_ at that site.  It can be
> as simple as a 
> standard analog line that you connect to a FXO port
> on a voice gateway, 
> with appropriate routing in CM to send a 911 call
> out from that location.
> 
> PSAPs (Public Safety Answering Points) aren't
> necessarily linked to each 
> other, so if someone in Houston is having a heart
> attack, and all calls 
> go out from Dallas, there's nothing much that the
> PSAP in Dallas can do 
> about it.
> 
> You don't need to renumber, or give users in your
> other locations 
> numbers "DIDs" in those locations.  Just make sure
> that a call in 
> Houston goes out through Houston, and a call in
> Dallas goes out through 
> Dallas, etc.  Which pretty much means that you're
> going to need a 
> "circuit" at those locations.
> 
> One caveat:  you may be able to get someone like one
> of the E911 VoIP 
> service providers involved (www.911enable.com is
> one); you'd setup 
> something like a SIP trunk and route e911 calls to
> them.  You'd tell 
> them which DIDs are located in which city, and allow
> them to handle the 
> 911 call and send it to the correct PSAP.  (In which
> case 911 calls 
> wouldn't traverse your PRI - they'd go out over your
> ethernet to the 
> provider)
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
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Be a better Heartthrob. Get better relationship answers from someone who
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From Steven.Miller at metavante.com  Thu Aug  2 19:06:28 2007
From: Steven.Miller at metavante.com (Steven.Miller at metavante.com)
Date: Thu, 2 Aug 2007 18:06:28 -0500
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
In-Reply-To: <001001c7d54f$5b4e4fa0$11eaeee0$@com>
Message-ID: <OFA6BAC50C.55724867-ON8625732B.007E4A2A-8625732B.007EF052@metavante.com>

There is a regulatory battle going on about this between State Public 
Service Commisions and the FCC.  I have not found the FCC to state 
specifically what you are doing with your DID's is wrong but your State 
Public Service Commision may say something different. Minnesota PSC sued 
Vonage about this typeof issue but they lost. The court ruled that the 
feds have jusridiction. The Feds are concerned primarilly with 911 and 
CALEA wich is basically law enforcements ability to wire tap. If your 
legal dept can point to specific laws saying what you are doing with the 
did's is wrong I would like to see that and if they are state laws or 
Federal. 



"Mark Holloway" <mh at markholloway.com> 
Sent by: cisco-voip-bounces at puck.nether.net
08/02/2007 04:52 PM

To
"'Dark Fiber'" <d4rkf1ber at gmail.com>
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] Need some facts regarding DID's,       Enterprise VoIP 
and LATAs






Out of rate center numbers are considered ?dirty? and if your phone 
company that you?re getting the TN?s from knew, they might make a big deal 
about it. 
 
From: techguy at gmail.com [mailto:techguy at gmail.com] On Behalf Of Dark Fiber
Sent: Thursday, August 02, 2007 2:34 PM
To: Mark Holloway
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Need some facts regarding DID's, Enterprise VoIP 
and LATAs
 
Actually 911 is not the issue, we do have local gateways with a couple of 
pstn lines at each location strictly for 911.  So if someone in Houston 
calls 911 it routes out the local gateway to reach the right psap.
 
Appreciate the response though.

 
On 8/2/07, Mark Holloway <mh at markholloway.com> wrote: 
Hey man, had to skim through this one quick (busy day at work) but let me 
see if I'm getting this right.  You are assigning DID's from Dallas to 
users in Houston and San Antonio?  Yikes! How are you handling 911 for 
Houston/San Antonio users?  You should have an FXO on Houston's router and 
San Antonio's router with a 911 dial peer that displays a Houston or San 
Antonio TN to their local PSAP.  You are in major FCC violation for not 
providing local 911 access.  If Houston dials 911 and goes to a Dallas 
PSAP, that person is screwed.  That's a major law suit and a major FCC 
violation.  The reason Vonage can do it is they have the ability to give 
you a DID anywhere, but route 911 to your local PSAP.  You definitely need 
some FXO cards and 911 dial peers in each of your remote locations. 
 
 
 
From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net] On Behalf Of Dark Fiber
Sent: Thursday, August 02, 2007 11:38 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Need some facts regarding DID's, Enterprise VoIP and 
LATAs 
 
In the last couple of weeks I have been dealing with e911/911 issues 
internally at work.  In trying to resolve and provide the best possible 
solution for 911 calling to all employees throughout the region I ended 
having to pull in our legal and regulatory folks to address some of these 
issues and what they felt we needed to provide at a minimum to our 
employees, so that we make sure we are completely covered. 
 
In doing so I begin to layout the network and phone system for them 
explaining what we have and where we are going and such.  Upon seeing this 
the regulatory folk begin telling me that this is wrong / illegal. 
Basically saying that the way we have deployed phones and Call Manager and 
such is wrong and needs to be corrected.  I was seriously taken aback by 
this, never did I question our phone deployment if you will, I mean heck 
this meeting was about 911 you know. 
 
The first thing they tell me is that DID's are for internal calling / 
usage, and not meant for external or incoming calling?  I was like blown 
away, this is completely OPPOSITE of what I have always thought and known. 
 DID's are direct inward dialing, you can 100 DID's if you will on a PRI 
from your service provider and assign them to individuals internally so 
that users can have a direct inward number from the outside.  Heck even 
wikipedia "which I know is not the end all source of all knowledge" but 
fairly reliable states exactly what I have always known DID's as. 
 
Next they begin telling me I have to get circuits and DID's / numbers that 
correspond to each of my physical locations!
 
Basically, right now we have various small locations spread out in 
different cities.  Say as an example Houston, Dallas and San Antonio. Main 
office say is Dallas, my call managers and gateways are there in Dallas. I 
get PRI's to terminate there and I have a large block of DID's all for 
Dallas. 
 
All ip phones are setup with four digit extension, and are tied to a DID 
from Dallas even though the phone and user may be in Houston or something. 
 User in Houston places a call it goes out of Dallas, and incoming calls 
obviously go into Dallas then over our fiber to Houston.  Blah blah, 
nothing new there I know alot of places that do the same thing. 
 
Anyway, so they tell me I can't do this.  That legally I have to get 
circuits in each market, and provide those users numbers in that market. 
 
I just don't buy this.  I mean the past three employees I have been at and 
managed the call managers they all had similar setups and I am not the one 
who set them up so I know I was not the only one who believed it was 
perfectly acceptable to do things this way. 
 
I would love some facts to use to show that it is perfectly acceptable to 
do this.  I can't find anything from a legal perspective to substantiate 
what they said or what I believe.
 
I even pointed out VoIP providers like Vonage and stuff, and said if it 
was illegal to do then don't you think these companies that base their 
business on just that sort of thing would exist?  That's one of the main 
selling points you can get a number from any market in the US pretty much 
no matter where you live.  Of course their answer was they are regulated 
differently.  And who knows maybe so. 
 
Any thoughts or arguments out there on this?  Would love to be able to 
point to some law or case or something regarding this rather then just 
thoughts and examples since they would have more weight for me to prove my 
point.  But I would love to hear anything at this point. 
 
 
 _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From pfabian at telesphere.com  Thu Aug  2 21:23:13 2007
From: pfabian at telesphere.com (Pete Fabian)
Date: Thu, 2 Aug 2007 18:23:13 -0700
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP and LATAs
In-Reply-To: <000901c7d547$de051da0$9a0f58e0$@com>
Message-ID: <000901c7d56c$f79f1250$76d2a8c0@telesphereltd.com>

For 911 purposes we use 911Enable.  It's a third party company that we send
all 911 calls to and they send the calls to the correct psap based on
preconfigured addresses that you input during initial configuration.  You
can connect to them via sip trunk or by pstn.  You can tie any number to any
address and the call will be routed to the correct psap.  In some locations
where we can't get DIDs for our customers we even use 800 numbers.  They are
relatively inexpensive compared to bringing in pots lines and adding fxo and
dsps to remote routers

http://www.911enable.com/

pete

 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Holloway
Sent: Thursday, August 02, 2007 1:58 PM
To: 'Paul Choi'; 'Robert Kulagowski'; 'Dark Fiber'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Need some facts regarding DID's,Enterprise VoIP
and LATAs

Although companies do this, it's not compliant.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Thursday, August 02, 2007 12:30 PM
To: Robert Kulagowski; Dark Fiber
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Need some facts regarding DID's, Enterprise VoIP
and LATAs

What you could do is, if management would be kosher
with this, have several dedicated 911 red phones with
straight POTS lines attached to them bypassing any 911
router business whatsoever. This is what one of our
customers did for several of their offices and they
seem to work well with the solution.

--- Robert Kulagowski <bob at smalltime.com> wrote:

> If you're going to provide for 911 at each location,
> then you're going 
> to need to get _something_ at that site.  It can be
> as simple as a 
> standard analog line that you connect to a FXO port
> on a voice gateway, 
> with appropriate routing in CM to send a 911 call
> out from that location.
> 
> PSAPs (Public Safety Answering Points) aren't
> necessarily linked to each 
> other, so if someone in Houston is having a heart
> attack, and all calls 
> go out from Dallas, there's nothing much that the
> PSAP in Dallas can do 
> about it.
> 
> You don't need to renumber, or give users in your
> other locations 
> numbers "DIDs" in those locations.  Just make sure
> that a call in 
> Houston goes out through Houston, and a call in
> Dallas goes out through 
> Dallas, etc.  Which pretty much means that you're
> going to need a 
> "circuit" at those locations.
> 
> One caveat:  you may be able to get someone like one
> of the E911 VoIP 
> service providers involved (www.911enable.com is
> one); you'd setup 
> something like a SIP trunk and route e911 calls to
> them.  You'd tell 
> them which DIDs are located in which city, and allow
> them to handle the 
> 911 call and send it to the correct PSAP.  (In which
> case 911 calls 
> wouldn't traverse your PRI - they'd go out over your
> ethernet to the 
> provider)
> 
> _______________________________________________
> 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
_______________________________________________
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 Aug  2 22:55:26 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 2 Aug 2007 21:55:26 -0500
Subject: [cisco-voip] Question About the Message Store Wizard
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490007C92806@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490007C92806@DCEX2.DSMO.COM>
Message-ID: <5d093f9a0708021955j55149d47p63ffc0e807c6b498@mail.gmail.com>

Is the message store changing (different storage group?), if so, then
you need to run both the permissions wizard and the message store
configuration setup (Configuration Setup under
commserver/utilities)...

It is easy... just run it and choose the stores you want... make sure
the UnityDirSvcs account has exchange full control and the
UnityMsgStoreSvc has view only, but has send-as and receive-as rights
(so, it can't be a domain admin if it is Win2K3).


Jonathan

On 8/2/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
>
> We are migrating our Exchange mail stores to a SAN this weekend and they
> will be in a different place from where they currently reside.  We have been
> told by some experts that we need to run the permissions wizard and told by
> others that we only need to run the Unity Message Store Wizard in order to
> fix the mailboxes after the move.
>
> We are on Unity 4.2.1 and have a redundant failover server.  Our Exchange
> Servers are not clustered.  Does anyone have any experience running the
> Message Store Wizard?  I haven't heard much mention of it.
>
> 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
>
>

From MillerS at DicksteinShapiro.COM  Fri Aug  3 00:38:37 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Fri, 3 Aug 2007 00:38:37 -0400
Subject: [cisco-voip] Question About the Message Store Wizard
References: <B9B5B75DF9145F44A21111DD92CC490007C92806@DCEX2.DSMO.COM>
	<5d093f9a0708021955j55149d47p63ffc0e807c6b498@mail.gmail.com>
Message-ID: <B9B5B75DF9145F44A21111DD92CC490007C92812@DCEX2.DSMO.COM>

We keep seeing this prompt in the Permissions Wizard that says something
to the effect of "choose the container where you want to create AD
accounts".  This scares us, because TAC ran the permissions wizard a few
months ago and created duplicate accounts for everyone on the system and
it created MAJOR issues.  What exactly does it mean?  We are only trying
to run the permissions wizard on one folder where one mailbox resides -
as a test.  Please let me know if you have any insight.  Thank you for
your help! 


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: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Thursday, August 02, 2007 10:55 PM
To: Miller, Steve
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Question About the Message Store Wizard

Is the message store changing (different storage group?), if so, then
you need to run both the permissions wizard and the message store
configuration setup (Configuration Setup under
commserver/utilities)...

It is easy... just run it and choose the stores you want... make sure
the UnityDirSvcs account has exchange full control and the
UnityMsgStoreSvc has view only, but has send-as and receive-as rights
(so, it can't be a domain admin if it is Win2K3).


Jonathan

On 8/2/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
>
> We are migrating our Exchange mail stores to a SAN this weekend and
they
> will be in a different place from where they currently reside.  We
have been
> told by some experts that we need to run the permissions wizard and
told by
> others that we only need to run the Unity Message Store Wizard in
order to
> fix the mailboxes after the move.
>
> We are on Unity 4.2.1 and have a redundant failover server.  Our
Exchange
> Servers are not clustered.  Does anyone have any experience running
the
> Message Store Wizard?  I haven't heard much mention of it.
>
> 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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From ErasmuE4 at telkom.co.za  Fri Aug  3 02:01:39 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Fri, 3 Aug 2007 08:01:39 +0200
Subject: [cisco-voip] voicemail on a huntlist
In-Reply-To: <46B1F56D.50908@cisco.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F0182E74F@CNTRRA20-XCS01.telkom.co.za>


Hi Wes / Kris

 

The part I am not getting is that I don't see a tick box for voicemail
on the hunt pilot like one would see on a normal line where one set
the forwarding 

 

Only see  - use personal preferences and the fields for a DN and the
calling search space field .... So how do I actually point the thing
to voicemail

 

The personal pref tick - the way I understand will use the settings of
whoever extended the call originally. In my case all calls will come
from an attendant and I simply want to make sure that if a call hunts
through all options it goes to voicemail.

 

 

 

 

erik

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Thursday, August 02, 2007 5:17 PM
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] voicemail on a huntlist

 

CM: hunt pilot configuration. there are settings for max hunt time,
and hunt no answer destination.

/Wes

Erik Erasmus (E) wrote: 

I have a customer with a lot of hunt groups on call manager 4-2. They
complaint that for some hunt lists they have complains where customers
don't get to a person. I think these hunt groups are under staffed and
now want to suggest that they try using a general voicemail mailbox as
a last member. They have unity connection.

 

It is not clear to me how to implement this. I have done several
normal phones with Unity connection and voicemail works fine. But how
does one get voicemail on a hunt. Does it sit on the hunt pilot?.

 

The hunt pilot is DN 4455. Do I now create mailbox with extension 4455
and then how do I get the hunt to revert to voicemail after it has
exhausted the hunt group members. Basically in ccm - what do I do. I
also want to know how would multiple people then have access to this
one common mailbox. Does it need to be configured as a special mailbox
on connection and how would they dial the mailbox to clean it out.
When you dial the voicemail pilot normally your extension takes you to
your voicemail. Also - how would message waiting indication then works
or would one not use it and people simply have to randomly dial in a
check for messages.

 

Any help appreciated

 

 

 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

 



________________________________



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


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From tlaflamme at millerthomson.com  Fri Aug  3 02:34:06 2007
From: tlaflamme at millerthomson.com (Tammie Laflamme)
Date: Fri, 3 Aug 2007 02:34:06 -0400
Subject: [cisco-voip] CN=Tammie Laflamme/O=mtca is out of the office.
Message-ID: <OFBB3E5862.1695E932-ON8525732C.0024150A-8525732C.0024150A@millerthomson.ca>





I will be out of the office starting  08/03/2007 and will not return until
08/20/2007.

I will respond upon my return, but if this matter is urgent, please contact
Kendall Longley at 416-595-8541 (or internally at ext. 58541) for
assistance.
CONFIDENTIALITY: This e-mail message (including attachments, if any) is confidential and is intended only for the addressee. Any unauthorized use or disclosure is strictly prohibited. Disclosure of this e-mail to anyone other than the intended addressee does not constitute waiver of privilege. If you have received this communication in error, please notify us immediately and delete this. Thank you for your cooperation.  This message has not been encrypted.  Special arrangements can be made for encryption upon request.

Visit our website at www.millerthomson.com for information about our firm and the services we provide.

CONFIDENTIALIT?:  Ce message courriel (y compris les pi?ces jointes, le cas ?ch?ant) est confidentiel et destin? uniquement ? la personne ou  ? l'entit? ? qui il est adress?. Toute utilisation ou divulgation non permise est strictement interdite.  L'obligation de confidentialit? et de secret professionnel demeure malgr? toute divulgation.  Si vous avez re?u le pr?sent courriel et ses annexes par erreur, veuillez nous en informer imm?diatement et le d?truire.  Nous vous remercions de votre collaboration.  Le pr?sent message n'a pas ?t? crypt?.  Le cryptage est possible sur demande sp?ciale.

Pour tout renseignement au sujet des services offerts par notre cabinet, visitez notre site Web ? www.millerthomsonpouliot.com

From Marcus.McCutchen at unco.edu  Fri Aug  3 08:35:54 2007
From: Marcus.McCutchen at unco.edu (McCutchen, Marcus)
Date: Fri, 3 Aug 2007 06:35:54 -0600
Subject: [cisco-voip] CER Web Alert and IE 7.0
Message-ID: <838CFBC36FBC2B418A23C58AB31B015502F7B0C4@itex04.unco.edu>

Good morning,

 

Has anyone here had trouble using the Web Alert page for Cisco Emergency
Responder using I.E. 7.0?  

Works fine with IE 6.0 and IE 5.0 unfortunately everyone has been
upgraded to I.E. 7.0.

Any suggestions besides rolling back to IE 6.0 which we don't really
want to do?

 

Thanks in advance.

 

Marcus

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From ErasmuE4 at telkom.co.za  Fri Aug  3 08:45:28 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Fri, 3 Aug 2007 14:45:28 +0200
Subject: [cisco-voip] voicemail on a huntlist
In-Reply-To: <BD89624EE83BD349BD31A08B453DE2440197A8D1@uogexb201.cfs.uoguelph.ca>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F0182E7C3@CNTRRA20-XCS01.telkom.co.za>


Thanks 

Will try 

 

________________________________

From: Rasim Duric [mailto:rduric at uoguelph.ca] 
Sent: Friday, August 03, 2007 2:37 PM
To: Erik Erasmus (E)
Subject: RE: [cisco-voip] voicemail on a huntlist

 

In your Hunt List just add your Voice Mail Hunt group. This group
(Voice Mail Hunt Group) should be last in the list if you want Voice
mail to answer when you dial the Pilot DN and there is nobody to take
the call. 

 

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 Erik Erasmus
(E)
Sent: Friday, August 03, 2007 2:02 AM
To: Wes Sisk; Kris Seraphine
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] voicemail on a huntlist

 

Hi Wes / Kris

 

The part I am not getting is that I don't see a tick box for voicemail
on the hunt pilot like one would see on a normal line where one set
the forwarding 

 

Only see  - use personal preferences and the fields for a DN and the
calling search space field .... So how do I actually point the thing
to voicemail

 

The personal pref tick - the way I understand will use the settings of
whoever extended the call originally. In my case all calls will come
from an attendant and I simply want to make sure that if a call hunts
through all options it goes to voicemail.

 

 

 

 

erik

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Thursday, August 02, 2007 5:17 PM
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] voicemail on a huntlist

 

CM: hunt pilot configuration. there are settings for max hunt time,
and hunt no answer destination.

/Wes

Erik Erasmus (E) wrote: 

I have a customer with a lot of hunt groups on call manager 4-2. They
complaint that for some hunt lists they have complains where customers
don't get to a person. I think these hunt groups are under staffed and
now want to suggest that they try using a general voicemail mailbox as
a last member. They have unity connection.

 

It is not clear to me how to implement this. I have done several
normal phones with Unity connection and voicemail works fine. But how
does one get voicemail on a hunt. Does it sit on the hunt pilot?.

 

The hunt pilot is DN 4455. Do I now create mailbox with extension 4455
and then how do I get the hunt to revert to voicemail after it has
exhausted the hunt group members. Basically in ccm - what do I do. I
also want to know how would multiple people then have access to this
one common mailbox. Does it need to be configured as a special mailbox
on connection and how would they dial the mailbox to clean it out.
When you dial the voicemail pilot normally your extension takes you to
your voicemail. Also - how would message waiting indication then works
or would one not use it and people simply have to randomly dial in a
check for messages.

 

Any help appreciated

 

 

 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

 
 
 
 
 



________________________________



 
 
 
 
 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
  
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From Nick.Kassel at Charles-Stanley.co.uk  Fri Aug  3 08:46:43 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri, 3 Aug 2007 13:46:43 +0100
Subject: [cisco-voip] Inbound system access codes
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01452074@CSEXCH01.charles-stanley.co.uk>

I'm not too sure what this is called but I know it is a feature
available on some PBX's. What I would like to do is to be able to dial
into a DID which then provides an access code which when entered will
allow a dial tone and then be able to dial out of the PSTN. Does anyone
know if this is possible on Call Manager, we are running 5.1(1) 

********************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 Marcus.McCutchen at unco.edu  Fri Aug  3 08:49:40 2007
From: Marcus.McCutchen at unco.edu (McCutchen, Marcus)
Date: Fri, 3 Aug 2007 06:49:40 -0600
Subject: [cisco-voip] CER Web Alert and IE 7.0
In-Reply-To: <838CFBC36FBC2B418A23C58AB31B015502F7B0C4@itex04.unco.edu>
References: <838CFBC36FBC2B418A23C58AB31B015502F7B0C4@itex04.unco.edu>
Message-ID: <838CFBC36FBC2B418A23C58AB31B015502F7B0C6@itex04.unco.edu>

I answered my own question here.  It was the Default security settings
on IE 7.0 after adjusting the settings to the same as IE 6.0 and below
it worked!

 

Security Medium and Privacy Medium and it worked properly. 

 

Sorry to have bothered you with this question.

 

Marcus

 

From: McCutchen, Marcus 
Sent: Friday, August 03, 2007 6:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CER Web Alert and IE 7.0

 

Good morning,

 

Has anyone here had trouble using the Web Alert page for Cisco Emergency
Responder using I.E. 7.0?  

Works fine with IE 6.0 and IE 5.0 unfortunately everyone has been
upgraded to I.E. 7.0.

Any suggestions besides rolling back to IE 6.0 which we don't really
want to do?

 

Thanks in advance.

 

Marcus

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From ErasmuE4 at telkom.co.za  Fri Aug  3 08:55:10 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Fri, 3 Aug 2007 14:55:10 +0200
Subject: [cisco-voip] Inbound system access codes
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01452074@CSEXCH01.charles-stanley.co.uk>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F0182E7C6@CNTRRA20-XCS01.telkom.co.za>


The channel under the hood magazine cisco had online had a full case
study showing how to do a nice and secure setup to achieve this using
call manager and IPCC express

 

If I still have it I will mail it to you or you can also search for it
- but I think cisco had this nice publication well hidden for some
stupid reason - somewhere under the partner e learning section

 

Erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, August 03, 2007 2:47 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Inbound system access codes

 

I'm not too sure what this is called but I know it is a feature
available on some PBX's. What I would like to do is to be able to dial
into a DID which then provides an access code which when entered will
allow a dial tone and then be able to dial out of the PSTN. Does
anyone know if this is possible on Call Manager, we are running 5.1(1)


 


********************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. 

 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
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From pwalenta at wi.rr.com  Fri Aug  3 09:16:56 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri, 3 Aug 2007 08:16:56 -0500
Subject: [cisco-voip] Inbound system access codes
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F0182E7C6@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <017f01c7d5d0$934b2dc0$0232a8c0@pwnhome.com>

It's also possible to do this with H.323 gateways and Tcl.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus (E)
Sent: Friday, August 03, 2007 7:55 AM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Inbound system access codes



The channel under the hood magazine cisco had online had a full case study
showing how to do a nice and secure setup to achieve this using call manager
and IPCC express

 

If I still have it I will mail it to you or you can also search for it - but
I think cisco had this nice publication well hidden for some stupid reason -
somewhere under the partner e learning section

 

Erik erasmus

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, August 03, 2007 2:47 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Inbound system access codes

 

I'm not too sure what this is called but I know it is a feature available on
some PBX's. What I would like to do is to be able to dial into a DID which
then provides an access code which when entered will allow a dial tone and
then be able to dial out of the PSTN. Does anyone know if this is possible
on Call Manager, we are running 5.1(1) 

 



********************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. 

 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From Nick.Kassel at Charles-Stanley.co.uk  Fri Aug  3 09:46:43 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri, 3 Aug 2007 14:46:43 +0100
Subject: [cisco-voip] Inbound system access codes
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01452076@CSEXCH01.charles-stanley.co.uk>

Many thanks, that would be greatly appreciated.

 

________________________________

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za] 
Sent: 03 August 2007 14:12
To: Nick Kassel
Subject: RE: [cisco-voip] Inbound system access codes

 

Remember with most call manager you get a 5 seat ipcc free - and in a
small deployment this might do it for you

 

Let me try and get you the article first then you can see if it is what
you want anyway

 

erik

 

________________________________

From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Friday, August 03, 2007 2:59 PM
To: Erik Erasmus (E)
Subject: RE: [cisco-voip] Inbound system access codes

 

Thanks Erik, I'll have a search, do you know if this could only be done
with IPCC though?

 

________________________________

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za] 
Sent: 03 August 2007 13:55
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Inbound system access codes

 

The channel under the hood magazine cisco had online had a full case
study showing how to do a nice and secure setup to achieve this using
call manager and IPCC express

 

If I still have it I will mail it to you or you can also search for it -
but I think cisco had this nice publication well hidden for some stupid
reason - somewhere under the partner e learning section

 

Erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, August 03, 2007 2:47 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Inbound system access codes

 

I'm not too sure what this is called but I know it is a feature
available on some PBX's. What I would like to do is to be able to dial
into a DID which then provides an access code which when entered will
allow a dial tone and then be able to dial out of the PSTN. Does anyone
know if this is possible on Call Manager, we are running 5.1(1) 

 


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Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
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From Nick.Kassel at Charles-Stanley.co.uk  Fri Aug  3 09:47:56 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri, 3 Aug 2007 14:47:56 +0100
Subject: [cisco-voip] Inbound system access codes
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01452077@CSEXCH01.charles-stanley.co.uk>

Any idea how this is done Philip as we use H323 gateways.

 

________________________________

From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: 03 August 2007 14:17
To: 'Erik Erasmus (E)'; Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Inbound system access codes

 

It's also possible to do this with H.323 gateways and Tcl.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Friday, August 03, 2007 7:55 AM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Inbound system access codes

The channel under the hood magazine cisco had online had a full case
study showing how to do a nice and secure setup to achieve this using
call manager and IPCC express

 

If I still have it I will mail it to you or you can also search for it -
but I think cisco had this nice publication well hidden for some stupid
reason - somewhere under the partner e learning section

 

Erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, August 03, 2007 2:47 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Inbound system access codes

 

I'm not too sure what this is called but I know it is a feature
available on some PBX's. What I would like to do is to be able to dial
into a DID which then provides an access code which when entered will
allow a dial tone and then be able to dial out of the PSTN. Does anyone
know if this is possible on Call Manager, we are running 5.1(1) 

 


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PRINT**************************************************

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to or any action taken or omitted to be
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************************************************************************
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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
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Charles Stanley Sutherlands and Charles Stanley Securities are divisions of Charles Stanley & Co. Ltd  

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From pwalenta at wi.rr.com  Fri Aug  3 10:02:35 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri, 3 Aug 2007 09:02:35 -0500
Subject: [cisco-voip] Inbound system access codes
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01452077@CSEXCH01.charles-stanley.co.uk>
Message-ID: <019301c7d5d6$f3bebdb0$0232a8c0@pwnhome.com>

It's been discussed here before...but here's a link that details what's in
12.3 and how to use it:
 
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_administrati
on_guide_chapter09186a008022a7f8.html
 
There are several example scripts hidden somewhere on CCO, I just can't seem
to find them now.

  _____  

From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Friday, August 03, 2007 8:48 AM
To: Philip Walenta; Erik Erasmus (E); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Inbound system access codes



Any idea how this is done Philip as we use H323 gateways.

 

  _____  

From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: 03 August 2007 14:17
To: 'Erik Erasmus (E)'; Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Inbound system access codes

 

It's also possible to do this with H.323 gateways and Tcl.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus (E)
Sent: Friday, August 03, 2007 7:55 AM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Inbound system access codes

The channel under the hood magazine cisco had online had a full case study
showing how to do a nice and secure setup to achieve this using call manager
and IPCC express

 

If I still have it I will mail it to you or you can also search for it - but
I think cisco had this nice publication well hidden for some stupid reason -
somewhere under the partner e learning section

 

Erik erasmus

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, August 03, 2007 2:47 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Inbound system access codes

 

I'm not too sure what this is called but I know it is a feature available on
some PBX's. What I would like to do is to be able to dial into a DID which
then provides an access code which when entered will allow a dial tone and
then be able to dial out of the PSTN. Does anyone know if this is possible
on Call Manager, we are running 5.1(1) 

 



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PRINT**************************************************

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Access to this e-mail by any person other than the recipient is prohibited.
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You are advised that urgent, time sensitive and confidential communications
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accept that any instructions are deemed to have been given at the time the
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delivery receipt does not constitute acknowledgement or receipt by the
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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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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

 




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PRINT**************************************************

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You acknowledge that e-mails are not secure and you accept the risk of
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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.

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From bms314 at gmail.com  Fri Aug  3 10:15:54 2007
From: bms314 at gmail.com (Brian)
Date: Fri, 3 Aug 2007 09:15:54 -0500
Subject: [cisco-voip] "more" softkey timer
Message-ID: <7aaaf7f0708030715u16eb580byd0ea581af07c250d@mail.gmail.com>

Is there a way to adjust the timer for the "more" softkey?  I've checked
under Service Parameters>Call Manager, but nothing jumps out at me.  I'd
like to increase the timer so it does not go back to the first page of soft
keys quite as fast while in the Connected state.  Makes for easier training
too...

Thanks,
Brian
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From Nick.Kassel at Charles-Stanley.co.uk  Fri Aug  3 10:27:22 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri, 3 Aug 2007 15:27:22 +0100
Subject: [cisco-voip] Gateway PRI Calls
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01452078@CSEXCH01.charles-stanley.co.uk>

We currently have 2 x H323 gateways each with their own PRI circuit from
the same provider configured to fail over to one another should they
fail.

 

We now want to order another 2 x PRI circuits from a different provider
and locate one in each of the gateways so we have one circuit from each
provider in both gateways. We would like to use the new circuits for
outbound calls and use the existing circuits for inbound. 

 

However is it going to be possible to failover to the other vendors
circuit if say for instance all 60 out bound channels are in use.

 

 

 

 

 

 

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*******************************************************************************************************


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.

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From rsingleton at morsco.com  Fri Aug  3 10:49:15 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Fri, 03 Aug 2007 09:49:15 -0500
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP	and LATAs
In-Reply-To: <46B22B09.90708@smalltime.com>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
	<46B22B09.90708@smalltime.com>
Message-ID: <1186152555.23536.24.camel@robert.morsco.com>

On Thu, 2007-08-02 at 14:05 -0500, Robert Kulagowski wrote:
> If you're going to provide for 911 at each location, then you're going 
> to need to get _something_ at that site.  It can be as simple as a 
> standard analog line that you connect to a FXO port on a voice gateway, 
> with appropriate routing in CM to send a 911 call out from that location.

<snip>

> You don't need to renumber, or give users in your other locations 
> numbers "DIDs" in those locations.  Just make sure that a call in 
> Houston goes out through Houston, and a call in Dallas goes out through 
> Dallas, etc.  Which pretty much means that you're going to need a 
> "circuit" at those locations.

It can be as simple as an FXO on the local router connected to a single
local line at the site. CallManager and SRST will route 911 calls for
that Calling Search Space using that line. The important things to
comply with are that the service address of the number matches the
address that emergency services would be dispatched to and that that
line can answer a call back from the 911 operator.

> One caveat:  you may be able to get someone like one of the E911 VoIP 
> service providers involved (www.911enable.com is one); you'd setup 
> something like a SIP trunk and route e911 calls to them.  You'd tell 
> them which DIDs are located in which city, and allow them to handle the 
> 911 call and send it to the correct PSAP.  (In which case 911 calls 
> wouldn't traverse your PRI - they'd go out over your ethernet to the 
> provider)

We are doing something similar to this using a service provided by
http://www.dash911.com/ 

We have 71 branches in 21 different area codes, many with two or more
physical addresses. We also have a few partner companies that we are
'subletting' space on our CallManager. In all, there are 97 unique
external phone number masks from which a 911 call can be made. Our setup
involves a SIP trunk to an Asterisk box that does some real time
scripting. A user calls 911 and the call is sent to Asterisk. Asterisk
forwards the call to dash911, who then forwards the call (with the
proper information) to the appropriate PSAP. They require us to be able
to redirect the call to them automatically on busy, no answer, etc (thus
the Asterisk box) and if they cant determine the proper address and info
to send to the PSAP, they have a call center that manually transfers the
call if needed.

Robert



From klong at olotp.org  Fri Aug  3 10:52:58 2007
From: klong at olotp.org (klong at olotp.org)
Date: Fri, 03 Aug 2007 07:52:58 -0700
Subject: [cisco-voip] Please trim your posts
Message-ID: <20070803075258.97296067075fa8c76087192ed0e74e4e.67cb835bfd.wbe@email.secureserver.net>

Please trim your posts to include only the pertinent information or
info you are responding to.  Signatures, headers, etc. don't need
to appear multiple times in a message.

Thanks,

Kim

From ealeatherman at gmail.com  Fri Aug  3 11:07:14 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 3 Aug 2007 11:07:14 -0400
Subject: [cisco-voip] Reason Codes in IPCC Express
Message-ID: <94a1afde0708030807m52bf5583k37d2d9254f7eea1f@mail.gmail.com>

Hello,

We have several different call centers on our IPCC Express system. One of
them wants to implement reason codes now for logoff/notready. How do I set
this for only the team that wants it, or is that even possible?

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From pfabian at telesphere.com  Fri Aug  3 11:10:22 2007
From: pfabian at telesphere.com (Pete Fabian)
Date: Fri, 3 Aug 2007 08:10:22 -0700
Subject: [cisco-voip] Gateway PRI Calls
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01452078@CSEXCH01.charles-stanley.co.uk>
Message-ID: <004401c7d5e0$699dac40$34fea8c0@telesphereltd.com>

Are you looking for traffic to go from router1port1 to router2port1 to
router1port2 to router2port2, or would router1port1 to router1port2 to
router2port1 to router2port2 work?

 

If the second way works for you, you could just setup route lists/route
groups with router1 then router 2, for the pots dial-peers on the router
just preference port 1 with a lower preference then the dial-peer for port 2

 

 

If you really want the first way to work then you may be in for more work.
Off the top of my head you could probably setup the gateways as IPIPGW's and
failover via a voip dial-peer to the other router.  You would probably also
have to create some translation patterns on the routers to avoid routing
loops.  Don't know if that would even work but I am interested to see what
others have to say

 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, August 03, 2007 7:27 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Gateway PRI Calls

 

We currently have 2 x H323 gateways each with their own PRI circuit from the
same provider configured to fail over to one another should they fail.

 

We now want to order another 2 x PRI circuits from a different provider and
locate one in each of the gateways so we have one circuit from each provider
in both gateways. We would like to use the new circuits for outbound calls
and use the existing circuits for inbound. 

 

However is it going to be possible to failover to the other vendors circuit
if say for instance all 60 out bound channels are in use.

 

 

 

 

 

 

 



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From aelnagar at ACT-EG.COM  Fri Aug  3 11:14:46 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Fri, 3 Aug 2007 18:14:46 +0300
Subject: [cisco-voip] VMWI on analog phones with VG224
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A0FD@mail.ACT-EG.COM>

Hello all;

Did anyone succeded in running the VMWI with analog phones connected to the VG224, if yes please tell us which model that works with you (and which doesnot) and is it using the normal confiuration or required a workaround.

thanks

Ahmed Elnagar

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From matt at saskin.net  Fri Aug  3 11:20:33 2007
From: matt at saskin.net (Matthew Saskin)
Date: Fri, 03 Aug 2007 11:20:33 -0400
Subject: [cisco-voip] Cisco IP communicator with VPN issue
In-Reply-To: <90487.7689.qm@web52008.mail.re2.yahoo.com>
References: <90487.7689.qm@web52008.mail.re2.yahoo.com>
Message-ID: <46B347C1.2020101@saskin.net>

What version of IP communicator are you using?  If earlier than (2.0.1 I 
believe) you have to specify two different devices in callmanager, one 
using the users wired network card MAC address and one with the users 
wireless network card MAC address.

If using a newer version of IP communicator you can specify a single 
device name and configure it within IP comm.

-matt

Gu Sang Jung wrote:
> Hi guys.
> 
> We try to support remote-end user(tele-worker at home).
> 
> The problem is that IP communication does not registered into 
> Callmanager  with some wireless home router on the Cisco VPN clinet.
> When it hooked up on the wired port, It worked fine with wired  
> connection on wireless home router . However, when It try to use 
> wireless mode, it could not registered.
>  From PC(TCP 2000) port connectivty is fined to Callmanager server.
> Is there some solution how to fix it up? Cisco TAC looks like not so 
> many helpful to this issue.
> 
> thanks
> 
> Gusang
> 
> Sr Tel Enginner broadcom
> 
> 
> 
> ------------------------------------------------------------------------
> Be a better Globetrotter. Get better travel answers 
> <http://us.rd.yahoo.com/evt=48254/*http://answers.yahoo.com/dir/_ylc=X3oDMTI5MGx2aThyBF9TAzIxMTU1MDAzNTIEX3MDMzk2NTQ1MTAzBHNlYwNCQUJwaWxsYXJfTklfMzYwBHNsawNQcm9kdWN0X3F1ZXN0aW9uX3BhZ2U-?link=list&sid=396545469>from 
> someone who knows.
> Yahoo! Answers - Check it out.
> 
> 
> ------------------------------------------------------------------------
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From rsingleton at morsco.com  Fri Aug  3 11:30:12 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Fri, 03 Aug 2007 10:30:12 -0500
Subject: [cisco-voip] Need some facts regarding DID's,
	Enterprise VoIP	and LATAs
In-Reply-To: <a1b07ea60708021433oa62e30p9a7f56c5b481234@mail.gmail.com>
References: <a1b07ea60708021137w4fb61f53h4681e49d9096eb17@mail.gmail.com>
	<000901c7d537$3a754530$af5fcf90$@com>
	<a1b07ea60708021433oa62e30p9a7f56c5b481234@mail.gmail.com>
Message-ID: <1186155012.23536.41.camel@robert.morsco.com>

On Thu, 2007-08-02 at 16:33 -0500, Dark Fiber wrote:
> Actually 911 is not the issue, we do have local gateways with a couple
> of pstn lines at each location strictly for 911.  So if someone in
> Houston calls 911 it routes out the local gateway to reach the right
> psap.

Ah, well, lets look at that, then.. :)

Ignoring my locations without SRST (and that's actually most of them),
outbound calls from any location go out a VoIP LD carrier or either of
two PRIs, depending on the NPANXX dialed. While the cheap VoIP carrier
is very cheap, free is better, so calls to the Dallas-Fort Worth or
Houston areas are routed out those PRIs, as well as toll free numbers.
We also route all fax calls out the PRIs because, while the VoIP carrier
can handle the fax call, there is enough variation in propagation delays
and such to make fax calling less than acceptably reliable throught he
VoIP LD carrier.

Inbound calls are similarly gathered from all over. Anywhere our
provider has a switch, they can provide local numbers in that market and
we either forward or port as needed our locations' numbers into their
switch. They route it to the appropriate PRI. Each regional PRI is
backhauled to our co-lo for the convenience of connecting them all to
ports on one 6608 blade in a 6509 chassis. We have contemplated putting
them in major locations in each of those markets, but at this point,
this is more elegant.

The exceptions to this architecture are the places where our provider
does not have a local switch. In these locations, we use FXO ports on
the local router and on most (but not all) of those routers, we are also
using SRST. We handle incoming calls on the local gateway and outgoing
calls on the WAN. This lets us use a smaller number of local lines by
not using them for outbound calling.

Now... Is all that legal? As far as I know. I'd think that our provider
would object if there were a legal issue...

Robert



From aelnagar at ACT-EG.COM  Fri Aug  3 11:35:24 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Fri, 3 Aug 2007 18:35:24 +0300
Subject: [cisco-voip] Duplicate CDR globalCallID_callID
References: <46B03F82.9874.0039.0@mtb.com> <46B094D6.1000601@cisco.com>
	<46B0892E.9874.0039.0@mtb.com>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A0FE@mail.ACT-EG.COM>

yes I have this problem also, it is not a conference calls and it is not even related calls at all, but has the same globalcallID_callID, pretty strange!!!
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Engineer Specialist
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of STEVEN CASPER
Sent: Wed 01-Aug-07 8:22 PM
To: Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Duplicate CDR globalCallID_callID


I did but I may be blind. I see no reference to this particular field.
 
Steve

>>> "Wes Sisk" <wsisk at cisco.com> 8/1/2007 10:12 AM >>>
did you check the conference section of the CDR definition doc?
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_programming_usage_guide09186a00806ed410.html#wp61027

/Wes

STEVEN CASPER wrote: 

	Do CDR records involved in Conference calls or other modifications use the same globalCallID_callID as the originating call? I am seeing duplicates on the same server and Call Manager has not been restarted in a while. Any other thoughts?
	 
	 
	Thanks,
	
	Steve Casper
	Voice Technologies
	M&T Bank
	(410) 347-6026 

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From tennille.spence at gsa.gov  Fri Aug  3 12:34:37 2007
From: tennille.spence at gsa.gov (tennille.spence at gsa.gov)
Date: Fri, 3 Aug 2007 11:34:37 -0500
Subject: [cisco-voip] Pulling Phone Serial Numbers from Call Manager
In-Reply-To: <1186152555.23536.24.camel@robert.morsco.com>
Message-ID: <OF32FA6E9E.8D63EB11-ON8625732C.005AEE43-8625732C.005B0F91@gsa.gov>

Is anyone aware of a quick way to pull serial and model number of all
phones from call manager?



From lelio at uoguelph.ca  Fri Aug  3 12:38:47 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 3 Aug 2007 12:38:47 -0400
Subject: [cisco-voip] Pulling Phone Serial Numbers from Call Manager
References: <OF32FA6E9E.8D63EB11-ON8625732C.005AEE43-8625732C.005B0F91@gsa.gov>
Message-ID: <01e601c7d5ec$c4f5a6e0$47196883@cfs.uoguelph.ca>

credits to Wes. however, cmreports.com has a phone survey tool for under 100 
bucks. i bought it, but couldn't get it working so i can't comment on it. i 
hope to have it working soon since a bugfix was sent to me. others who have 
used it liked it.

i believe both options require the phones to be plugged in since they're 
getting the S/N from the webpage of the phone.

----- Original Message ----- 
From: "Wes Sisk" <wsisk at cisco.com>
To: "Marcin Nowacki" <Marcin_Nowacki at sevenet.pl>
Cc: <cisco-voip at puck.nether.net>
Sent: Sunday, January 02, 2005 10:15 AM
Subject: Re: [cisco-voip] IP Phone serial numbers


> No automated way built into CM, but pretty easy:
>
> 1. use 'mmfspy a' to ge the IP of all phones registered to CM
> 2. use 'wget', an open source utility, to http get all of the web pages
> from the phones.
> 3. use 'grep' to parse the files and pick out the serial numbers
>
> /Wes

--------------------------------------------------------------------------------
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
----- Original Message ----- 
From: <tennille.spence at gsa.gov>
To: "'cisco-voip at puck-nether.net Voip'" <cisco-voip at puck.nether.net>
Sent: Friday, August 03, 2007 12:34 PM
Subject: [cisco-voip] Pulling Phone Serial Numbers from Call Manager


> Is anyone aware of a quick way to pull serial and model number of all
> phones from call manager?
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From rsingleton at morsco.com  Fri Aug  3 13:22:38 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Fri, 03 Aug 2007 12:22:38 -0500
Subject: [cisco-voip] Pulling Phone Serial Numbers from Call Manager
In-Reply-To: <OF32FA6E9E.8D63EB11-ON8625732C.005AEE43-8625732C.005B0F91@gsa.gov>
References: <OF32FA6E9E.8D63EB11-ON8625732C.005AEE43-8625732C.005B0F91@gsa.gov>
Message-ID: <1186161758.23536.70.camel@robert.morsco.com>

On Fri, 2007-08-03 at 11:34 -0500, tennille.spence at gsa.gov wrote:
> Is anyone aware of a quick way to pull serial and model number of all
> phones from call manager?

So far as I have ever been able to determine, CallManager does not care
about the serial numbers and thus does not have them.

You can, however, get them from the phones themselves.

I do this from my Linux workstation with a collection of command line
tools that are part of most basic Linux installations, most of which are
also available in either packages of *nix utilities for Windows or as
standalone binaries.

For my particular uses, I have a script called 'inv' that contains the
follwoing:

----------
#!/bin/bash

for x in `nmap -sP 172.26,28.$1.2-254 | awk '{if ($1 ~ "^Host") print
$2}'`; do lynx -dump $x;done | egrep 'DN|Serial|Model|Product' | awk
'{printf("%s %s %s\n",$1,$2,$3)}' | sed 's/Phone/\nPhone/g'
-----------

Usage is "inv XX" where XX is the branch number on my WAN.

I will gladly give anyone the details for each of these commands, but
I'll keep it short here.

The "for x .... ; do" loops through a list of IP addresses for
telephones. nmap gets the list of IPs and awk filters out the lines from
nmap that don't matter to us.

lynx is a text based web browswer. The -dump option outputs the text of
the webpages. I prefer this over wget because it's easier for sort and
filter than wget's output, which is raw html.

egrep takes lynx's output and filter any lines that contain any of the
patterns listed between the '' quotes. It is here that clever pattern
choices can make or break the script. These lines were carefully chosen
to match the output of 7940 and 7960 telephones, without matching the
lines returned by ATA-186/8. The web page on the ATA is substantially
different than for the 79XX telephones. In *my* case, I do not want
anything but 79XX phones listed.

The 2nd awk just prints the output pretty. It is at this point where I
could make it output a csv or some other format.

The sed at the end puts an extra blank line to separate phones. Upon
reflection, I could probably do this with the awk line, but I presume I
didn't for some good reason. :)

The final output for my branch 10 looks like this:

[robert at robert ~]$ inv 10

Phone DN 2010550
Serial Number FCH11048ZZK
Model Number CP-7960G

Phone DN 2010763
Serial Number FCH10409JYA
Model Number CP-7960G

Phone DN 2010762
Serial Number FCH1104916A
Model Number CP-7960G

Phone DN 2010500
Serial Number FCH110491ML
Model Number CP-7960G

Phone DN 2010761
Serial Number FCH110491PY
Model Number CP-7960G

Phone DN 2010510
Serial Number FCH11048ZSM
Model Number CP-7960G

Phone DN 2010750
Serial Number FCH11039TQL
Model Number CP-7960G

Phone DN 2010520
Serial Number FCH11049007
Model Number CP-7960G

Phone DN 2010710
Serial Number FCH11048MU9
Model Number CP-7960G

Phone DN 2010701
Serial Number FCH11039RE2
Model Number CP-7960G

Phone DN 2010703
Serial Number FCH11048LQE
Model Number CP-7960G

Phone DN 2010702
Serial Number FCH11048NQD
Model Number CP-7960G

Phone DN 2010501
Serial Number FCH10509SXY
Model Number CP-7960G

Phone DN 2010760
Serial Number FCH11039BQM
Model Number CP-7960G
[robert at robert ~]$


Robert


From dwolgas1 at rochester.rr.com  Fri Aug  3 15:38:04 2007
From: dwolgas1 at rochester.rr.com (Dave Wolgast)
Date: Fri, 3 Aug 2007 15:38:04 -0400
Subject: [cisco-voip] Polycom SE200 H.323 Gatekeeper w/ CCM 4.1(3)
Message-ID: <e4491a9c0708031238q2399b2a1n5e0519ef0c5fc25@mail.gmail.com>

I'm hoping someone out there has some experience with this platform...I have
established a Gatekeeper-controlled H.323 trunk with the SE200, and can now
make audio calls to my Polycom VSX endpoints.  What I don't know how to do
is to set up dialing in the other direction (VSX -> CCM endpoints).

The system was configured with a single site.  Do I need to add a site in
order to make this happen?  Where do I put dial-plan information (If 1000 is
dialed, send to the H.323 trunk, etc...).

Any help is greatly appreciated.

-- 
Dave Wolgast
Livonia, NY
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From rhlphp at hotmail.com  Fri Aug  3 15:47:21 2007
From: rhlphp at hotmail.com (Vlad Liapkov)
Date: Fri, 3 Aug 2007 15:47:21 -0400
Subject: [cisco-voip] IPIPGW H323-H323 IVR, DTMF, Digits not collected
Message-ID: <BAY117-W195DEF9F506FB0F2242DEED1EA0@phx.gbl>

Hi. I have 2821 configured as ipipgw h323-h323 with app_debitcard tcl script. Everything working fine. Voice prompts played, but when script asking for pin number - digits do not get collected, no response when * or # pressed either. Script times out 3 times and disconnects. I have tried different dtmf-relay settings, nothing works. When application is removed from dial-peer and call sent directly to termination gateway, dtmf passed through on both legs and working fine when accessing voice mail on termination end. System image file is "flash:c2800nm-adventerprisek9_ivs-mz.124-11.T.bin" dial-peer voice 1 voip service debitcard incoming called-number . dtmf-relay h245-alphanumeric codec g711ulaw ! dial-peer voice 2 voip destination-pattern . session target ipv4:10.10.10.10 dtmf-relay rtp-nte codec g711ulaw I can not find any solution or similar problem anywhere. Please help. 
_________________________________________________________________
News, entertainment and everything you care about at Live.com. Get it now!
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From matt at saskin.net  Fri Aug  3 16:23:14 2007
From: matt at saskin.net (Matthew Saskin)
Date: Fri, 03 Aug 2007 16:23:14 -0400
Subject: [cisco-voip] Polycom SE200 H.323 Gatekeeper w/ CCM 4.1(3)
In-Reply-To: <e4491a9c0708031238q2399b2a1n5e0519ef0c5fc25@mail.gmail.com>
References: <e4491a9c0708031238q2399b2a1n5e0519ef0c5fc25@mail.gmail.com>
Message-ID: <46B38EB2.5050700@saskin.net>

If the Polycom VSX endpoints are relying on the gatekeeper, you'd have 
to put dial plan information into the gatekeeper config:

gatekeeper
   zone local ......
   zone local callmanager ....
   zone prefix callmanager 9*

This will make the gatekeeper send ARQs that begin with the digit 9 
towards the "callmanager" zone, which is the zone the ccm trunk 
registers itself to.  I'm not sure of any other way to do this since CCM 
doesn't provide the ability to register it's dial plan (or parts of the 
dial plan) with a gatekeeper like voice gateways do.

-matt

Dave Wolgast wrote:
> I'm hoping someone out there has some experience with this platform...I 
> have established a Gatekeeper-controlled H.323 trunk with the SE200, and 
> can now make audio calls to my Polycom VSX endpoints.  What I don't know 
> how to do is to set up dialing in the other direction (VSX -> CCM 
> endpoints).
>  
> The system was configured with a single site.  Do I need to add a site 
> in order to make this happen?  Where do I put dial-plan information (If 
> 1000 is dialed, send to the H.323 trunk, etc...).
>  
> Any help is greatly appreciated.
> 
> -- 
> Dave Wolgast
> Livonia, NY
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From pwalenta at wi.rr.com  Fri Aug  3 17:02:12 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri, 3 Aug 2007 16:02:12 -0500
Subject: [cisco-voip] Polycom SE200 H.323 Gatekeeper w/ CCM 4.1(3)
In-Reply-To: <e4491a9c0708031238q2399b2a1n5e0519ef0c5fc25@mail.gmail.com>
Message-ID: <021301c7d611$90ab6c10$0232a8c0@pwnhome.com>

Hi Dave, I work for Polycom, and the SE200 is one of the products I deal
with daily.
 
I'm currently in the middle of writing a white paper for Polycom on this
subject and hope to have it ready to go in about a week or so.
 
I will contact you off-list then.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dave Wolgast
Sent: Friday, August 03, 2007 2:38 PM
To: Cisco VOIP Newsletter - puck.nether.net
Subject: [cisco-voip] Polycom SE200 H.323 Gatekeeper w/ CCM 4.1(3)


I'm hoping someone out there has some experience with this platform...I have
established a Gatekeeper-controlled H.323 trunk with the SE200, and can now
make audio calls to my Polycom VSX endpoints.  What I don't know how to do
is to set up dialing in the other direction (VSX -> CCM endpoints). 
 
The system was configured with a single site.  Do I need to add a site in
order to make this happen?  Where do I put dial-plan information (If 1000 is
dialed, send to the H.323 trunk, etc...).
 
Any help is greatly appreciated.

-- 
Dave Wolgast
Livonia, NY 
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From jsteinberg at gmail.com  Fri Aug  3 17:46:22 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 3 Aug 2007 17:46:22 -0400
Subject: [cisco-voip] "more" softkey timer
In-Reply-To: <7aaaf7f0708030715u16eb580byd0ea581af07c250d@mail.gmail.com>
References: <7aaaf7f0708030715u16eb580byd0ea581af07c250d@mail.gmail.com>
Message-ID: <f1b65f880708031446u66764645k734f3b2569e5f57a@mail.gmail.com>

I see this field on the phone's device page with CCM 4.1.3sr5b and 7941G
phone.


On 8/3/07, Brian <bms314 at gmail.com> wrote:
>
> Is there a way to adjust the timer for the "more" softkey?  I've checked
> under Service Parameters>Call Manager, but nothing jumps out at me.  I'd
> like to increase the timer so it does not go back to the first page of soft
> keys quite as fast while in the Connected state.  Makes for easier training
> too...
>
> Thanks,
> Brian
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From PRuttman at foley.com  Fri Aug  3 20:03:29 2007
From: PRuttman at foley.com (Ruttman, Pete G.)
Date: Fri, 3 Aug 2007 19:03:29 -0500
Subject: [cisco-voip] IOS conferencing question
In-Reply-To: <2B3745BC9D31514381A6FCC5E72D937301FAB83A@SMOMAIL.jhacorp.com>
Message-ID: <5DB17A1560B7EE4186747DF43A6D4C6C6EC49A@WIMKEEX07.foleylaw.com>

Okay.  Let me rework that question.  When I select g729 in Callmanager
what version of g729 is actually used?

________________________________

From: Chris D'amore [mailto:CDAmore at jackhenry.com] 
Sent: Friday, August 03, 2007 2:00 PM
To: Ruttman, Pete G.
Subject: RE: [cisco-voip] IOS conferencing question



You should not use g729b at all in your network. This is because VAD is
enabled in this codec. It can't be disabled with using g729b.

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Pete
G.
Sent: Wednesday, August 01, 2007 5:30 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IOS conferencing question

 

We have 3845 routers with NM-HDV2 modules at each site to support local
PRIs and conferencing.  We have PVDM-64 installed on the HDV2 modules.
I am setting up conferencing on the routers and I want to eliminate
unnecessary codec from the list of codecs supported.  We use 7960 and
790x phones.  Callmanager is configured for calls over the WAN to be
g.729 and we use g.711 internally to each office.   A user at any site
can envoke local conference resource to conferenc PSTN calls,  local
phones and phones located across the WAN.

I can leave the following config but it wastes DSP resources: 

dspfarm profile 1 conference 
 description office Conference 1 
 codec g711ulaw 
 codec g729r8 
 codec g711alaw 
 codec g729ar8 
 codec g729abr8 
 codec g729br8 
 associate application SCCP 
 maximum sessions 4 

 

What can I remove from the codec list? 

Thanks, 

Pete Ruttman 


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.

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.

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.

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From jim at tgasolutions.com  Fri Aug  3 22:05:24 2007
From: jim at tgasolutions.com (Jim McBurnett)
Date: Fri, 3 Aug 2007 22:05:24 -0400
Subject: [cisco-voip] Pulling Phone Serial Numbers from Call Manager
In-Reply-To: <1186161758.23536.70.camel@robert.morsco.com>
References: <OF32FA6E9E.8D63EB11-ON8625732C.005AEE43-8625732C.005B0F91@gsa.gov>
	<1186161758.23536.70.camel@robert.morsco.com>
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC370AD1@exchange.TGASolutions.local>

There is a tool at CMreports.com that does this for about $50.
Jim

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Singleton
Sent: Friday, August 03, 2007 1:23 PM
To: 'cisco-voip at puck-nether.net Voip'
Subject: Re: [cisco-voip] Pulling Phone Serial Numbers from Call Manager

On Fri, 2007-08-03 at 11:34 -0500, tennille.spence at gsa.gov wrote:
> Is anyone aware of a quick way to pull serial and model number of all
> phones from call manager?

So far as I have ever been able to determine, CallManager does not care
about the serial numbers and thus does not have them.

You can, however, get them from the phones themselves.

I do this from my Linux workstation with a collection of command line
tools that are part of most basic Linux installations, most of which are
also available in either packages of *nix utilities for Windows or as
standalone binaries.

For my particular uses, I have a script called 'inv' that contains the
follwoing:

----------
#!/bin/bash

for x in `nmap -sP 172.26,28.$1.2-254 | awk '{if ($1 ~ "^Host") print
$2}'`; do lynx -dump $x;done | egrep 'DN|Serial|Model|Product' | awk
'{printf("%s %s %s\n",$1,$2,$3)}' | sed 's/Phone/\nPhone/g'
-----------

Usage is "inv XX" where XX is the branch number on my WAN.

I will gladly give anyone the details for each of these commands, but
I'll keep it short here.

The "for x .... ; do" loops through a list of IP addresses for
telephones. nmap gets the list of IPs and awk filters out the lines from
nmap that don't matter to us.

lynx is a text based web browswer. The -dump option outputs the text of
the webpages. I prefer this over wget because it's easier for sort and
filter than wget's output, which is raw html.

egrep takes lynx's output and filter any lines that contain any of the
patterns listed between the '' quotes. It is here that clever pattern
choices can make or break the script. These lines were carefully chosen
to match the output of 7940 and 7960 telephones, without matching the
lines returned by ATA-186/8. The web page on the ATA is substantially
different than for the 79XX telephones. In *my* case, I do not want
anything but 79XX phones listed.

The 2nd awk just prints the output pretty. It is at this point where I
could make it output a csv or some other format.

The sed at the end puts an extra blank line to separate phones. Upon
reflection, I could probably do this with the awk line, but I presume I
didn't for some good reason. :)

The final output for my branch 10 looks like this:

[robert at robert ~]$ inv 10

Phone DN 2010550
Serial Number FCH11048ZZK
Model Number CP-7960G

Phone DN 2010763
Serial Number FCH10409JYA
Model Number CP-7960G

Phone DN 2010762
Serial Number FCH1104916A
Model Number CP-7960G

Phone DN 2010500
Serial Number FCH110491ML
Model Number CP-7960G

Phone DN 2010761
Serial Number FCH110491PY
Model Number CP-7960G

Phone DN 2010510
Serial Number FCH11048ZSM
Model Number CP-7960G

Phone DN 2010750
Serial Number FCH11039TQL
Model Number CP-7960G

Phone DN 2010520
Serial Number FCH11049007
Model Number CP-7960G

Phone DN 2010710
Serial Number FCH11048MU9
Model Number CP-7960G

Phone DN 2010701
Serial Number FCH11039RE2
Model Number CP-7960G

Phone DN 2010703
Serial Number FCH11048LQE
Model Number CP-7960G

Phone DN 2010702
Serial Number FCH11048NQD
Model Number CP-7960G

Phone DN 2010501
Serial Number FCH10509SXY
Model Number CP-7960G

Phone DN 2010760
Serial Number FCH11039BQM
Model Number CP-7960G
[robert at robert ~]$


Robert

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

From aizhar at ccbootcamp.com  Fri Aug  3 22:11:19 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Fri, 3 Aug 2007 19:11:19 -0700
Subject: [cisco-voip] IOS conferencing question
	[bcc][faked-from][html-rem]
References: <5DB17A1560B7EE4186747DF43A6D4C6C6EC49A@WIMKEEX07.foleylaw.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C366356111B@nlimail.nlidomain>

g729r8
 
-------------------------------------------
thanks,
Avner Izhar
CCIE#15999 (Voice), CCVP.
Technical Instructor
CCBOOTCAMP - A Cisco Sponsored Organization (SO)
Email: aizhar at ccbootcamp.com <mailto: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

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ruttman, Pete G.
Sent: Fri 8/3/2007 5:03 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS conferencing question [bcc][faked-from][html-rem]



Okay.  Let me rework that question.  When I select g729 in Callmanager
what version of g729 is actually used?

________________________________

From: Chris D'amore [mailto:CDAmore at jackhenry.com]
Sent: Friday, August 03, 2007 2:00 PM
To: Ruttman, Pete G.
Subject: RE: [cisco-voip] IOS conferencing question



You should not use g729b at all in your network. This is because VAD is
enabled in this codec. It can't be disabled with using g729b.





From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Pete
G.
Sent: Wednesday, August 01, 2007 5:30 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IOS conferencing question



We have 3845 routers with NM-HDV2 modules at each site to support local
PRIs and conferencing.  We have PVDM-64 installed on the HDV2 modules.
I am setting up conferencing on the routers and I want to eliminate
unnecessary codec from the list of codecs supported.  We use 7960 and
790x phones.  Callmanager is configured for calls over the WAN to be
g.729 and we use g.711 internally to each office.   A user at any site
can envoke local conference resource to conferenc PSTN calls,  local
phones and phones located across the WAN.

I can leave the following config but it wastes DSP resources:

dspfarm profile 1 conference
 description office Conference 1
 codec g711ulaw
 codec g729r8
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729br8
 associate application SCCP
 maximum sessions 4



What can I remove from the codec list?

Thanks,

Pete Ruttman


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.

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.

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.

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


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From jsteinberg at gmail.com  Sat Aug  4 10:26:02 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Sat, 4 Aug 2007 10:26:02 -0400
Subject: [cisco-voip] IOS conferencing question
In-Reply-To: <5DB17A1560B7EE4186747DF43A6D4C6C6EC49A@WIMKEEX07.foleylaw.com>
References: <2B3745BC9D31514381A6FCC5E72D937301FAB83A@SMOMAIL.jhacorp.com>
	<5DB17A1560B7EE4186747DF43A6D4C6C6EC49A@WIMKEEX07.foleylaw.com>
Message-ID: <f1b65f880708040726p5ec74925idfe5e382b9ce8186@mail.gmail.com>

depends on the capabilities of your endpoints and the endpoints codec
priority list.

CCM has a service parameter where CCM will strip g729b thus preventing it
from being used in your network.  I usually turn this parameter on during
installs to prevent the use of silence suppression.

Look for ccm service parameter "strip g729 annex B"

Justin


On 8/3/07, Ruttman, Pete G. <PRuttman at foley.com> wrote:
>
>  Okay.  Let me rework that question.  When I select g729 in Callmanager
> what version of g729 is actually used?
>  ------------------------------
> *From:* Chris D'amore [mailto:CDAmore at jackhenry.com]
> *Sent:* Friday, August 03, 2007 2:00 PM
> *To:* Ruttman, Pete G.
> *Subject:* RE: [cisco-voip] IOS conferencing question
>
>  You should not use g729b at all in your network. This is because VAD is
> enabled in this codec. It can't be disabled with using g729b.
>
>
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ruttman, Pete G.
> *Sent:* Wednesday, August 01, 2007 5:30 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] IOS conferencing question
>
>
>
> We have 3845 routers with NM-HDV2 modules at each site to support local
> PRIs and conferencing.  We have PVDM-64 installed on the HDV2 modules.  I am
> setting up conferencing on the routers and I want to eliminate unnecessary
> codec from the list of codecs supported.  We use 7960 and 790x phones.
> Callmanager is configured for calls over the WAN to be g.729 and we use
> g.711 internally to each office.   A user at any site can envoke local
> conference resource to conferenc PSTN calls,  local phones and phones
> located across the WAN.
>
> I can leave the following config but it wastes DSP resources:
>
> dspfarm profile 1 conference
>  description office Conference 1
>  codec g711ulaw
>  codec g729r8
>  codec g711alaw
>  codec g729ar8
>  codec g729abr8
>  codec g729br8
>  associate application SCCP
>  maximum sessions 4
>
>
>
> What can I remove from the codec list?
>
> Thanks,
>
> Pete Ruttman
>
>
> 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.
>
> 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.
>
>
> 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.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From PRuttman at foley.com  Sat Aug  4 14:24:33 2007
From: PRuttman at foley.com (Ruttman, Pete G.)
Date: Sat, 4 Aug 2007 13:24:33 -0500
Subject: [cisco-voip] IOS conferencing question
In-Reply-To: <f1b65f880708040726p5ec74925idfe5e382b9ce8186@mail.gmail.com>
Message-ID: <5DB17A1560B7EE4186747DF43A6D4C6C6EC4B3@WIMKEEX07.foleylaw.com>

Our endpoints are 7960, 790x and 793x.

 

________________________________

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Saturday, August 04, 2007 9:26 AM
To: Ruttman, Pete G.
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS conferencing question

 

depends on the capabilities of your endpoints and the endpoints codec
priority list.

CCM has a service parameter where CCM will strip g729b thus preventing
it from being used in your network.  I usually turn this parameter on
during installs to prevent the use of silence suppression. 

Look for ccm service parameter "strip g729 annex B"

Justin



On 8/3/07, Ruttman, Pete G. <PRuttman at foley.com> wrote:

Okay.  Let me rework that question.  When I select g729 in Callmanager
what version of g729 is actually used?

________________________________

From: Chris D'amore [mailto:CDAmore at jackhenry.com] 
Sent: Friday, August 03, 2007 2:00 PM
To: Ruttman, Pete G.
Subject: RE: [cisco-voip] IOS conferencing question

You should not use g729b at all in your network. This is because VAD is
enabled in this codec. It can't be disabled with using g729b.

 

 

From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Ruttman, Pete
G.
Sent: Wednesday, August 01, 2007 5:30 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IOS conferencing question

 

We have 3845 routers with NM-HDV2 modules at each site to support local
PRIs and conferencing.  We have PVDM-64 installed on the HDV2 modules.
I am setting up conferencing on the routers and I want to eliminate
unnecessary codec from the list of codecs supported.  We use 7960 and
790x phones.  Callmanager is configured for calls over the WAN to be
g.729 and we use g.711 internally to each office.   A user at any site
can envoke local conference resource to conferenc PSTN calls,  local
phones and phones located across the WAN.

I can leave the following config but it wastes DSP resources: 

dspfarm profile 1 conference 
 description office Conference 1 
 codec g711ulaw 
 codec g729r8 
 codec g711alaw 
 codec g729ar8 
 codec g729abr8 
 codec g729br8 
 associate application SCCP 
 maximum sessions 4 

 

What can I remove from the codec list? 

Thanks, 

Pete Ruttman 


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.

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.


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. 


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

 


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.

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From jonvoip at gmail.com  Sun Aug  5 21:30:03 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 5 Aug 2007 20:30:03 -0500
Subject: [cisco-voip] VMWI on analog phones with VG224
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED88A0FD@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED88A0FD@mail.ACT-EG.COM>
Message-ID: <5d093f9a0708051830me053d21o483d1ce4497f85f8@mail.gmail.com>

I got it to work with just about any pots phone...

It isn't really MWI tho... it is actually stutter tone (so when they
pick up the receiver they hear a bizarre beeping).

Make sure you are running like 12.4.10 on the VG224...



Jonathan

On 8/3/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>
>
> Hello all;
>
> Did anyone succeded in running the VMWI with analog phones connected to the
> VG224, if yes please tell us which model that works with you (and which
> doesnot) and is it using the normal confiuration or required a workaround.
>
> thanks
>
> Ahmed Elnagar
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From aelnagar at ACT-EG.COM  Mon Aug  6 05:31:51 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Mon, 6 Aug 2007 12:31:51 +0300
Subject: [cisco-voip] VMWI on analog phones with VG224
References: <88C8D019DA547C4BABECB9A806ED22ED88A0FD@mail.ACT-EG.COM>
	<5d093f9a0708051830me053d21o483d1ce4497f85f8@mail.gmail.com>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A10B@mail.ACT-EG.COM>

yes I got the stutter tone working, but I need the visual indication VMWI not the Audio indication.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Engineer Specialist
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Jonathan Charles [mailto:jonvoip at gmail.com]
Sent: Mon 06-Aug-07 4:30 AM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VMWI on analog phones with VG224



I got it to work with just about any pots phone...

It isn't really MWI tho... it is actually stutter tone (so when they
pick up the receiver they hear a bizarre beeping).

Make sure you are running like 12.4.10 on the VG224...



Jonathan

On 8/3/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>
>
> Hello all;
>
> Did anyone succeded in running the VMWI with analog phones connected to the
> VG224, if yes please tell us which model that works with you (and which
> doesnot) and is it using the normal confiuration or required a workaround.
>
> thanks
>
> Ahmed Elnagar
> _______________________________________________
> 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  Mon Aug  6 12:00:21 2007
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Tue, 7 Aug 2007 01:00:21 +0900
Subject: [cisco-voip] Unity Connection "Live Record Messages" Feature
Message-ID: <bbe09c1d0708060900q73fb4b0ao40fc98ea91e3122f@mail.gmail.com>

Does anyone know whether unity connection 1.x or 2.x "Live Record
Messages" feature works with Call Manager Express?

Thanks,
Pere

From SBarron at stinson.com  Mon Aug  6 12:30:24 2007
From: SBarron at stinson.com (Barron, Stephen M.)
Date: Mon, 6 Aug 2007 11:30:24 -0500
Subject: [cisco-voip] Unity plays system undeliverable messages from
	Exchange mailbox
Message-ID: <319A08ED1241E841AD9ECB0138F3B0880537F224@FIRMEXCH01.Our-Firm.com>

Our Unity 4.0(4) has started picking up users' undeliverable messages in
their exchange box and since they are read, they can be played through
the saved messages in the telephone interface of Unity.  Does anyone
have any Ideas of how to suppress these type of messages from playing in
Unity?  Short of deleting them I guess.
 
Thanks
 

Stephen 

Network Engineer
 
 
This communication is from a law firm and may contain confidential and/or privileged information. If it has been sent to you in error, please contact the sender for instructions concerning return or destruction, and do not use or disclose the contents to others.
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From svoll.voip at gmail.com  Mon Aug  6 14:12:13 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 6 Aug 2007 11:12:13 -0700
Subject: [cisco-voip] 9911 comming in bound
Message-ID: <f84a38d30708061112g5d9095c8i2ef1a043125aecbb@mail.gmail.com>

OK--

help me think through this one.

9.911 is used as E911 in my voice network.

I'm moving a Fax line 606-9911 over to a BRI on my Voice network then using
a ATA to attach to the Fax machine.

how do I do this?

do I have to create a new CSS just for the VGW and make the ATA extension
9911 only apart of that VGW's CSS?

Then leave the current CSS for all the phones at that site.  then the ATA
will also have the normal CSS for the site.

Am I thinking correctly?

Thanks

Scott
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From dustin.s.fowler at gmail.com  Mon Aug  6 14:17:23 2007
From: dustin.s.fowler at gmail.com (Dustin S Fowler)
Date: Mon, 6 Aug 2007 14:17:23 -0400
Subject: [cisco-voip] 9911 comming in bound
In-Reply-To: <f84a38d30708061112g5d9095c8i2ef1a043125aecbb@mail.gmail.com>
References: <f84a38d30708061112g5d9095c8i2ef1a043125aecbb@mail.gmail.com>
Message-ID: <f2d16ce0708061117i22e77ffcxa2cdc69448cab0b1@mail.gmail.com>

The best way I have found is create set the VGW to a new CSS and put 9911 in
a translation pattern with a partition on the VGW can call.
This way only the VGW will be able to hit the ATA through a translation
pattern and the rest of the users while 9.911 get the authorites.


On 8/6/07, Scott Voll <svoll.voip at gmail.com> wrote:
>
> OK--
>
> help me think through this one.
>
> 9.911 is used as E911 in my voice network.
>
> I'm moving a Fax line 606-9911 over to a BRI on my Voice network then
> using a ATA to attach to the Fax machine.
>
> how do I do this?
>
> do I have to create a new CSS just for the VGW and make the ATA extension
> 9911 only apart of that VGW's CSS?
>
> Then leave the current CSS for all the phones at that site.  then the ATA
> will also have the normal CSS for the site.
>
> Am I thinking correctly?
>
> 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 dpriebe at theriverbank.com  Mon Aug  6 17:46:08 2007
From: dpriebe at theriverbank.com (Duane Priebe)
Date: Mon, 6 Aug 2007 16:46:08 -0500
Subject: [cisco-voip] Audio Prompt Pointers...
Message-ID: <43D799CBE4331D44999AB7C9CD67F302033BA34A@NTEXCHANGE.riverbank.com>


I've been mixing and converting various G.729 audio prompts for a
project I've been working. The prompts sound great until I drop them
into call manager and/or IPCC prompt management then they sound quite
distorted.  Anyone have some recording tune up tips or something that
might be able to help me create these prompts better quality?  Thanks!

Duane
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From wsisk at cisco.com  Mon Aug  6 19:14:48 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 06 Aug 2007 19:14:48 -0400
Subject: [cisco-voip] Audio Prompt Pointers...
In-Reply-To: <43D799CBE4331D44999AB7C9CD67F302033BA34A@NTEXCHANGE.riverbank.com>
References: <43D799CBE4331D44999AB7C9CD67F302033BA34A@NTEXCHANGE.riverbank.com>
Message-ID: <46B7AB68.1020108@cisco.com>

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From Kenneth.Sinfield at didata.com.au  Tue Aug  7 05:36:31 2007
From: Kenneth.Sinfield at didata.com.au (Kenneth Sinfield)
Date: Tue, 7 Aug 2007 19:36:31 +1000
Subject: [cisco-voip]  VMWI on analog phones with VG224
Message-ID: <9BF888E063AF4D4590836EE9F01A61B001638273@AUNDDVSEXC01.au.didata.local>

Hi Ahmed,

 

I have VMWI working on an ISR router with FXS ports (VMWI will work on a
VG224 as well). Basically below are the requirements:

 

*         The Cisco Gateway must run IOS version IP Voice T strain (not
SP Services on ISRs) 12.4(6)T or later (I tested 12.4(11)T2)

*         Cisco CallManager version must be version 4.2 or later (I
tested CallManager 5.1.2)

*         Any version of Unity (I tested Unity Connection 1.2)

*         Must use SCCP(Skinny) as the gateway protocol, and NOT MGCP

 

Here is an output from the gateway(MWI refers to stutter tone VMWI needs
to be "ON"):

 

pstn#show stcapp device voice-port 0/3/3

Port Identifier:  0/3/3

Device Type:      ALG 

Device Id:        4

Device Name:      AN17598075F0183

Modem Capability: None

Device State:     IS

Diagnostic:       None

Directory Number: 1005

Dial Peer(s):     101 

Dialtone after remote onhook feature: activated

Last Event:       STCAPP_CC_EV_CALL_DISCONNECT_DONE

Line State:       IDLE

Hook State:       ONHOOK

mwi:              ENABLE

vmwi:             ON

PLAR:             DISABLE

Number of CCBs:   0

Global call info:

    Total CCB count      = 0

    Total call leg count = 0

 

Thanks,

Ken.

 

Kenneth Sinfield

Systems Engineer

Dimension Data, Australia

139 Coronation Drive, Milton

Queensland, Australia, 4064

Phone: +61732921968

 


******************************************************************************
 - NOTICE FROM DIMENSION DATA AUSTRALIA
This message is confidential, and may contain proprietary or legally privileged information.  If you have received this email in error, please notify the sender and delete it immediately.

Internet communications are not secure. You should scan this message and any attachments for viruses.  Under no circumstances do we accept liability for any loss or damage which may result from your receipt of this message or any attachments.
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From PRuttman at foley.com  Tue Aug  7 10:03:50 2007
From: PRuttman at foley.com (Ruttman, Pete G.)
Date: Tue, 7 Aug 2007 09:03:50 -0500
Subject: [cisco-voip] IOS conferencing question
	[bcc][faked-from][html-rem]
In-Reply-To: <E76ACA8C13EF1847BC69538B522C366356111B@nlimail.nlidomain>
Message-ID: <5DB17A1560B7EE4186747DF43A6D4C6C6EC4C3@WIMKEEX07.foleylaw.com>

How would I see this is the case?  Traces in CCM?  Debugs on the
gateway?  
 
Pete

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Friday, August 03, 2007 9:11 PM
To: Ruttman, Pete G.; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IOS conferencing question
[bcc][faked-from][html-rem]


g729r8
 
-------------------------------------------
thanks,
Avner Izhar
CCIE#15999 (Voice), CCVP.
Technical Instructor
CCBOOTCAMP - A Cisco Sponsored Organization (SO)
Email: aizhar at ccbootcamp.com <mailto: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

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ruttman, Pete G.
Sent: Fri 8/3/2007 5:03 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS conferencing question
[bcc][faked-from][html-rem]



Okay.  Let me rework that question.  When I select g729 in Callmanager
what version of g729 is actually used?

________________________________

From: Chris D'amore [mailto:CDAmore at jackhenry.com]
Sent: Friday, August 03, 2007 2:00 PM
To: Ruttman, Pete G.
Subject: RE: [cisco-voip] IOS conferencing question



You should not use g729b at all in your network. This is because VAD is
enabled in this codec. It can't be disabled with using g729b.





From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Pete
G.
Sent: Wednesday, August 01, 2007 5:30 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IOS conferencing question



We have 3845 routers with NM-HDV2 modules at each site to support local
PRIs and conferencing.  We have PVDM-64 installed on the HDV2 modules.
I am setting up conferencing on the routers and I want to eliminate
unnecessary codec from the list of codecs supported.  We use 7960 and
790x phones.  Callmanager is configured for calls over the WAN to be
g.729 and we use g.711 internally to each office.   A user at any site
can envoke local conference resource to conferenc PSTN calls,  local
phones and phones located across the WAN.

I can leave the following config but it wastes DSP resources:

dspfarm profile 1 conference
 description office Conference 1
 codec g711ulaw
 codec g729r8
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729br8
 associate application SCCP
 maximum sessions 4



What can I remove from the codec list?

Thanks,

Pete Ruttman


The preceding email message may be confidential or protected by the
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Legal advice contained in the preceding message is solely for the
benefit of the Foley & Lardner LLP client(s) represented by the Firm in
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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
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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
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_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip



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.

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From d4rkf1ber at gmail.com  Tue Aug  7 10:07:34 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Tue, 7 Aug 2007 09:07:34 -0500
Subject: [cisco-voip] IPCC usage at remote locations?
Message-ID: <a1b07ea60708070707l59d8a807id03dba9309d2735c@mail.gmail.com>

A few months back I deployed IPCC express internally for our help desk, few
little hiccups and learning curve but got it all working and its been
humming along fine for some time now.

I was just asked to look into what it would take to leverage IPCC at a
remote location that has their own call manager deployment.  Basically its
another company, our management wants to provide on site support people
there to help them out with some big projects I guess.

Connectivity, bandwidth and all is not an issue, plenty of that between the
two.  What I am trying to figure out is if there is a way to leverage our
IPCC with their Callmanagers and phones, or if I am just going to have to
deploy phones there that will connect with our callmanagers in order to
leverage IPCC there.

I don't think you can use IPCC servers to connect to different CallManager
clusters since you have to specify in IPCC the JTAPI provider and the CRS
LDAP Server info which in our case points to our primary callmanager.

It would seem the only viable option would probably be to just deploy phones
at the remote site that connect back to our callmanager.

But I know there are plenty of experts out there who have seen and heard
many situations and scenarios and have different ideas on ways to accomplish
things.

Thanks
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From jonvoip at gmail.com  Tue Aug  7 10:31:36 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 7 Aug 2007 09:31:36 -0500
Subject: [cisco-voip] IPCC usage at remote locations?
In-Reply-To: <a1b07ea60708070707l59d8a807id03dba9309d2735c@mail.gmail.com>
References: <a1b07ea60708070707l59d8a807id03dba9309d2735c@mail.gmail.com>
Message-ID: <5d093f9a0708070731v123ea840p47052be564eac9bc@mail.gmail.com>

The problem you will run into is that the CTI Manager and JTAPI
connection have to be to the same CallManager cluster.

It would probably be cheaper to get them their own IPCC server/cluster...

Alternatively, you could dump their CCM cluster, and just have them join yours.



Jonathan

On 8/7/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
> A few months back I deployed IPCC express internally for our help desk, few
> little hiccups and learning curve but got it all working and its been
> humming along fine for some time now.
>
> I was just asked to look into what it would take to leverage IPCC at a
> remote location that has their own call manager deployment.  Basically its
> another company, our management wants to provide on site support people
> there to help them out with some big projects I guess.
>
> Connectivity, bandwidth and all is not an issue, plenty of that between the
> two.  What I am trying to figure out is if there is a way to leverage our
> IPCC with their Callmanagers and phones, or if I am just going to have to
> deploy phones there that will connect with our callmanagers in order to
> leverage IPCC there.
>
> I don't think you can use IPCC servers to connect to different CallManager
> clusters since you have to specify in IPCC the JTAPI provider and the CRS
> LDAP Server info which in our case points to our primary callmanager.
>
> It would seem the only viable option would probably be to just deploy phones
> at the remote site that connect back to our callmanager.
>
> But I know there are plenty of experts out there who have seen and heard
> many situations and scenarios and have different ideas on ways to accomplish
> things.
>
> Thanks
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From aizhar at ccbootcamp.com  Tue Aug  7 11:16:33 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Tue, 7 Aug 2007 08:16:33 -0700
Subject: [cisco-voip] IOS conferencing
	question[bcc][faked-from][html-rem]
References: <5DB17A1560B7EE4186747DF43A6D4C6C6EC4C3@WIMKEEX07.foleylaw.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663561129@nlimail.nlidomain>

You can see that on the gateway, by typing 'show call active voice brief".
 
BTW, call manager's g729 allow all the flavors of g729, it is up to the endpoints to agree on what codec will they use, now since the ios gateways will default to g729r8 on their voip dial-peers, that would most likely be the selected codec.
 
-------------------------------------------
thanks,
Avner Izhar
CCIE#15999 (Voice), CCVP.
Technical Instructor
CCBOOTCAMP - A Cisco Sponsored Organization (SO)
Email: aizhar at ccbootcamp.com <mailto: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

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ruttman, Pete G.
Sent: Tue 8/7/2007 7:03 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS conferencing question[bcc][faked-from][html-rem]


How would I see this is the case?  Traces in CCM?  Debugs on the gateway?  
 
Pete

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Friday, August 03, 2007 9:11 PM
To: Ruttman, Pete G.; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IOS conferencing question [bcc][faked-from][html-rem]


g729r8
 
-------------------------------------------
thanks,
Avner Izhar
CCIE#15999 (Voice), CCVP.
Technical Instructor
CCBOOTCAMP - A Cisco Sponsored Organization (SO)
Email: aizhar at ccbootcamp.com <mailto: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

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ruttman, Pete G.
Sent: Fri 8/3/2007 5:03 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS conferencing question [bcc][faked-from][html-rem]



Okay.  Let me rework that question.  When I select g729 in Callmanager
what version of g729 is actually used?

________________________________

From: Chris D'amore [mailto:CDAmore at jackhenry.com]
Sent: Friday, August 03, 2007 2:00 PM
To: Ruttman, Pete G.
Subject: RE: [cisco-voip] IOS conferencing question



You should not use g729b at all in your network. This is because VAD is
enabled in this codec. It can't be disabled with using g729b.





From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Pete
G.
Sent: Wednesday, August 01, 2007 5:30 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IOS conferencing question



We have 3845 routers with NM-HDV2 modules at each site to support local
PRIs and conferencing.  We have PVDM-64 installed on the HDV2 modules.
I am setting up conferencing on the routers and I want to eliminate
unnecessary codec from the list of codecs supported.  We use 7960 and
790x phones.  Callmanager is configured for calls over the WAN to be
g.729 and we use g.711 internally to each office.   A user at any site
can envoke local conference resource to conferenc PSTN calls,  local
phones and phones located across the WAN.

I can leave the following config but it wastes DSP resources:

dspfarm profile 1 conference
 description office Conference 1
 codec g711ulaw
 codec g729r8
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729br8
 associate application SCCP
 maximum sessions 4



What can I remove from the codec list?

Thanks,

Pete Ruttman


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
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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.

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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.

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From pcaltagirone at comporium.net  Tue Aug  7 12:19:51 2007
From: pcaltagirone at comporium.net (pcaltagirone)
Date: Tue, 7 Aug 2007 12:19:51 -0400
Subject: [cisco-voip] IPCCX agent roll time
Message-ID: <003401c7d90e$c8966ee0$c803010a@na.didata.local>

Does anybody know where in IPCCX or CM can you find the setting for the amount of time or rings before an unanswered call will roll to the next agent?

Thanks in advance!

Paul
Charlotte,NC
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From Johnny.Crothers at vanco.co.uk  Tue Aug  7 12:24:38 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Tue, 7 Aug 2007 17:24:38 +0100
Subject: [cisco-voip] IPCCX agent roll time
In-Reply-To: <003401c7d90e$c8966ee0$c803010a@na.didata.local>
Message-ID: <A56BD242744BF540B15D1672E0E3E8750193DF89@LON-INT-MSGVS01.internal.vancocorp.com>

Inside the script you will need to change the properties of the "Select
Resource" step, defaults to 12 seconds.

 

Johnny Crothers

IPT Specialist

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of pcaltagirone
Sent: 07 August 2007 17:20
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCCX agent roll time

 

Does anybody know where in IPCCX or CM can you find the setting for the
amount of time or rings before an unanswered call will roll to the next
agent?

 

Thanks in advance!

 

Paul

Charlotte,NC



**********************************************************************
Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient.  If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with the content of this email please e-mail to: e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
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From pcaltagirone at comporium.net  Tue Aug  7 12:42:47 2007
From: pcaltagirone at comporium.net (pcaltagirone)
Date: Tue, 7 Aug 2007 12:42:47 -0400
Subject: [cisco-voip] IPCCX agent roll time
References: <A56BD242744BF540B15D1672E0E3E8750193DF89@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <004601c7d911$fc480c00$c803010a@na.didata.local>

Thanks Johny.....I'll take a look at that and re-post

Paul

Charlotte,NC

  ----- Original Message ----- 
  From: Johnny Crothers c 
  To: pcaltagirone ; cisco-voip at puck.nether.net 
  Sent: Tuesday, August 07, 2007 12:24 PM
  Subject: RE: [cisco-voip] IPCCX agent roll time


  Inside the script you will need to change the properties of the "Select Resource" step, defaults to 12 seconds.



  Johnny Crothers

  IPT Specialist


------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of pcaltagirone
  Sent: 07 August 2007 17:20
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] IPCCX agent roll time



  Does anybody know where in IPCCX or CM can you find the setting for the amount of time or rings before an unanswered call will roll to the next agent?



  Thanks in advance!



  Paul

  Charlotte,NC



  **********************************************************************
  Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient. If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

  If you have received this email in error, or if you are concerned with the content of this email please e-mail to: e-security.support at vanco.info

  The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

  Vanco UK Ltd Registered in England No: 2296733 Registered Office: John Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

  Please consider the environment before printing this e-mail
  **********************************************************************
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From pcaltagirone at comporium.net  Tue Aug  7 13:01:09 2007
From: pcaltagirone at comporium.net (pcaltagirone)
Date: Tue, 7 Aug 2007 13:01:09 -0400
Subject: [cisco-voip] 2851 gateway Error: DeviceUnregistered Reason Code 9
Message-ID: <005701c7d914$8d9f7740$c803010a@na.didata.local>

I have a newly deployed 2851 voice gateway connected back to CM via two dedicated T-1 circuits and am having issues with the FXO/FXS ports unregistering 4-5 times daily......All VoIP control QoS is in place....policy maps are being hit? I can also see a handful of phones doing the same during these times?

DeviceUnregistered Reason Code 9 points to keepalives but should not be an issue here??

Thanks in avance!

Paul
Charlotte,NC
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From Jeremy.Rogers at wmg.com  Tue Aug  7 15:03:19 2007
From: Jeremy.Rogers at wmg.com (Rogers, Jeremy (WMG))
Date: Tue, 7 Aug 2007 20:03:19 +0100 
Subject: [cisco-voip] Speakerphone cuts out
Message-ID: <4DA11A95A4F28944B0B89404EF69A10B0348E856@wmggsex25.wmg.com>

I have a user who is having a random issue.  He has an IP phone at his house
behind a hardware firewall.  On occasion he said the speakerphone would cut
out and that there was barely any sound coming out, but if he picked up the
receiver he could hear fine.  We thought it was a VPN issue, but the same
thing just happened to his office phone.  Has anyone seen this before?

 

Jeremy Rogers

Senior VoIP Engineer

Warner Music Group

 

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From Jeremy.Rogers at wmg.com  Tue Aug  7 16:01:20 2007
From: Jeremy.Rogers at wmg.com (Rogers, Jeremy (WMG))
Date: Tue, 7 Aug 2007 21:01:20 +0100 
Subject: [cisco-voip] FW:  Speakerphone cuts out
Message-ID: <4DA11A95A4F28944B0B89404EF69A10B0348E858@wmggsex25.wmg.com>

Let me add that I just found out when the speakerphone cuts out, so does the
ringer.

 

Jeremy Rogers

Senior VoIP Engineer

Warner Music Group

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rogers, Jeremy
(WMG)
Sent: Tuesday, August 07, 2007 3:03 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Speakerphone cuts out

 

I have a user who is having a random issue.  He has an IP phone at his house
behind a hardware firewall.  On occasion he said the speakerphone would cut
out and that there was barely any sound coming out, but if he picked up the
receiver he could hear fine.  We thought it was a VPN issue, but the same
thing just happened to his office phone.  Has anyone seen this before?

 

Jeremy Rogers

Senior VoIP Engineer

Warner Music Group

 

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From manuel.cristobal at us.didata.com  Tue Aug  7 16:02:23 2007
From: manuel.cristobal at us.didata.com (Manuel Cristobal (US))
Date: Tue, 7 Aug 2007 16:02:23 -0400
Subject: [cisco-voip] Qsig with 3Com NBX 5000
Message-ID: <C1FE15183DA37645BC0633BC604E44F0065C4FEF@USNAEXCH.na.didata.local>

Anyone has successfully deployed a QSIG trunk between a 3com NBX 5000
PBX and a Cisco 2800 gateway via a T1 card?

If so, please share the pain.

 

 

Thanks. 

Regards,

 

Manuel Cristobal

Solutions Architect

Dimension Data

Office 954 757 0737

Cell  954 495 0274

manuel.cristobal at us.didata.com <mailto:manuel.cristobal at us.didata.com> 

 

Visit us on the web at http://www.dimensiondata.com/na
<http://www.dimensiondata.com/na>  

 




-----------------------------------------
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 asobihoudai at yahoo.com  Tue Aug  7 16:12:36 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 7 Aug 2007 13:12:36 -0700 (PDT)
Subject: [cisco-voip] FW:  Speakerphone cuts out
In-Reply-To: <4DA11A95A4F28944B0B89404EF69A10B0348E858@wmggsex25.wmg.com>
Message-ID: <819468.67130.qm@web57208.mail.re3.yahoo.com>

Do you see anything odd in CCM traces? 

What about the phone status page? 

Do you see any abnormalities on the diagnostic page
for the phone? [question mark button twice during
conversation]

Any dropped packets?


--- "Rogers, Jeremy (WMG)" <Jeremy.Rogers at wmg.com>
wrote:

> Let me add that I just found out when the
> speakerphone cuts out, so does the
> ringer.
> 
>  
>   _____  
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Rogers, Jeremy
> (WMG)
> Sent: Tuesday, August 07, 2007 3:03 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Speakerphone cuts out
> 
>  
> 
> I have a user who is having a random issue.  He has
> an IP phone at his house
> behind a hardware firewall.  On occasion he said the
> speakerphone would cut
> out and that there was barely any sound coming out,
> but if he picked up the
> receiver he could hear fine.  We thought it was a
> VPN issue, but the same
> thing just happened to his office phone.  Has anyone
> seen this before?
> 
>  
> 
> Jeremy Rogers
> 
> Senior VoIP Engineer
> 
> Warner Music Group
> 



      ____________________________________________________________________________________
Shape Yahoo! in your own image.  Join our Network Research Panel today!   http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 



From Jeremy.Rogers at wmg.com  Tue Aug  7 16:30:39 2007
From: Jeremy.Rogers at wmg.com (Rogers, Jeremy (WMG))
Date: Tue, 7 Aug 2007 21:30:39 +0100 
Subject: [cisco-voip] FW:  Speakerphone cuts out
Message-ID: <4DA11A95A4F28944B0B89404EF69A10B0348E85A@wmggsex25.wmg.com>

No dropped packets, no errors on the diagnostics page.  I don't see anything
in the traces.  I can hear him talking into the mic, but the speaker is not
working on his end.

Jeremy Rogers
Senior VoIP Engineer
Warner Music Group
75 Rockefeller Plaza
New York, NY 10019
212-275-2183
-----Original Message-----
From: Paul Choi [mailto:asobihoudai at yahoo.com] 
Sent: Tuesday, August 07, 2007 4:13 PM
To: Rogers, Jeremy (WMG); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] FW: Speakerphone cuts out

Do you see anything odd in CCM traces? 

What about the phone status page? 

Do you see any abnormalities on the diagnostic page
for the phone? [question mark button twice during
conversation]

Any dropped packets?


--- "Rogers, Jeremy (WMG)" <Jeremy.Rogers at wmg.com>
wrote:

> Let me add that I just found out when the
> speakerphone cuts out, so does the
> ringer.
> 
>  
>   _____  
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Rogers, Jeremy
> (WMG)
> Sent: Tuesday, August 07, 2007 3:03 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Speakerphone cuts out
> 
>  
> 
> I have a user who is having a random issue.  He has
> an IP phone at his house
> behind a hardware firewall.  On occasion he said the
> speakerphone would cut
> out and that there was barely any sound coming out,
> but if he picked up the
> receiver he could hear fine.  We thought it was a
> VPN issue, but the same
> thing just happened to his office phone.  Has anyone
> seen this before?
> 
>  
> 
> Jeremy Rogers
> 
> Senior VoIP Engineer
> 
> Warner Music Group
> 



 
____________________________________________________________________________
________
Shape Yahoo! in your own image.  Join our Network Research Panel today!
http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 

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From mh at markholloway.com  Tue Aug  7 16:50:20 2007
From: mh at markholloway.com (Mark Holloway)
Date: Tue, 7 Aug 2007 13:50:20 -0700
Subject: [cisco-voip] FW:  Speakerphone cuts out
In-Reply-To: <4DA11A95A4F28944B0B89404EF69A10B0348E85A@wmggsex25.wmg.com>
References: <4DA11A95A4F28944B0B89404EF69A10B0348E85A@wmggsex25.wmg.com>
Message-ID: <006c01c7d934$91a8bcf0$b4fa36d0$@com>

What kind of speakerphone?  I had to troubleshoot a similar issue.  The
problem was the default gain and attenuation setting of the particular
speaker phone (Polycom) were not compatible with the Cisco AS5400.  I had to
make adjustments on the Polycom so the center clipper values matched better.
Otherwise, it's "choppy audio". 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rogers, Jeremy
(WMG)
Sent: Tuesday, August 07, 2007 1:31 PM
To: 'Paul Choi'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] FW: Speakerphone cuts out

 

No dropped packets, no errors on the diagnostics page.  I don't see anything
in the traces.  I can hear him talking into the mic, but the speaker is not
working on his end.

Jeremy Rogers 
Senior VoIP Engineer 
Warner Music Group 
75 Rockefeller Plaza 
New York, NY 10019 
212-275-2183 
-----Original Message----- 
From: Paul Choi [mailto:asobihoudai at yahoo.com] 
Sent: Tuesday, August 07, 2007 4:13 PM 
To: Rogers, Jeremy (WMG); cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] FW: Speakerphone cuts out 

Do you see anything odd in CCM traces? 

What about the phone status page? 

Do you see any abnormalities on the diagnostic page 
for the phone? [question mark button twice during 
conversation] 

Any dropped packets? 

 

--- "Rogers, Jeremy (WMG)" <Jeremy.Rogers at wmg.com> 
wrote: 

> Let me add that I just found out when the 
> speakerphone cuts out, so does the 
> ringer. 
> 
>  
>   _____  
> 
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On 
> Behalf Of Rogers, Jeremy 
> (WMG) 
> Sent: Tuesday, August 07, 2007 3:03 PM 
> To: cisco-voip at puck.nether.net 
> Subject: [cisco-voip] Speakerphone cuts out 
> 
>  
> 
> I have a user who is having a random issue.  He has 
> an IP phone at his house 
> behind a hardware firewall.  On occasion he said the 
> speakerphone would cut 
> out and that there was barely any sound coming out, 
> but if he picked up the 
> receiver he could hear fine.  We thought it was a 
> VPN issue, but the same 
> thing just happened to his office phone.  Has anyone 
> seen this before? 
> 
>  
> 
> Jeremy Rogers 
> 
> Senior VoIP Engineer 
> 
> Warner Music Group 
> 

 

 
____________________________________________________________________________
________ 
Shape Yahoo! in your own image.  Join our Network Research Panel today!
http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 

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From Jeremy.Rogers at wmg.com  Tue Aug  7 16:53:02 2007
From: Jeremy.Rogers at wmg.com (Rogers, Jeremy (WMG))
Date: Tue, 7 Aug 2007 21:53:02 +0100 
Subject: [cisco-voip] FW:  Speakerphone cuts out
Message-ID: <4DA11A95A4F28944B0B89404EF69A10B0348E85B@wmggsex25.wmg.com>

This is just a 7960 plugged into a POE 4507 switch.  

 

Jeremy Rogers

Senior VoIP Engineer

Warner Music Group

75 Rockefeller Plaza

New York, NY 10019

212-275-2183

  _____  

From: Mark Holloway [mailto:mh at markholloway.com] 
Sent: Tuesday, August 07, 2007 4:50 PM
To: Rogers, Jeremy (WMG); 'Paul Choi'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] FW: Speakerphone cuts out

 

What kind of speakerphone?  I had to troubleshoot a similar issue.  The
problem was the default gain and attenuation setting of the particular
speaker phone (Polycom) were not compatible with the Cisco AS5400.  I had to
make adjustments on the Polycom so the center clipper values matched better.
Otherwise, it's "choppy audio". 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rogers, Jeremy
(WMG)
Sent: Tuesday, August 07, 2007 1:31 PM
To: 'Paul Choi'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] FW: Speakerphone cuts out

 

No dropped packets, no errors on the diagnostics page.  I don't see anything
in the traces.  I can hear him talking into the mic, but the speaker is not
working on his end.

Jeremy Rogers 
Senior VoIP Engineer 
Warner Music Group 
75 Rockefeller Plaza 
New York, NY 10019 
212-275-2183 
-----Original Message----- 
From: Paul Choi [mailto:asobihoudai at yahoo.com <mailto:asobihoudai at yahoo.com>
] 
Sent: Tuesday, August 07, 2007 4:13 PM 
To: Rogers, Jeremy (WMG); cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] FW: Speakerphone cuts out 

Do you see anything odd in CCM traces? 

What about the phone status page? 

Do you see any abnormalities on the diagnostic page 
for the phone? [question mark button twice during 
conversation] 

Any dropped packets? 

 

--- "Rogers, Jeremy (WMG)" <Jeremy.Rogers at wmg.com> 
wrote: 

> Let me add that I just found out when the 
> speakerphone cuts out, so does the 
> ringer. 
> 
>  
>   _____  
> 
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On 
> Behalf Of Rogers, Jeremy 
> (WMG) 
> Sent: Tuesday, August 07, 2007 3:03 PM 
> To: cisco-voip at puck.nether.net 
> Subject: [cisco-voip] Speakerphone cuts out 
> 
>  
> 
> I have a user who is having a random issue.  He has 
> an IP phone at his house 
> behind a hardware firewall.  On occasion he said the 
> speakerphone would cut 
> out and that there was barely any sound coming out, 
> but if he picked up the 
> receiver he could hear fine.  We thought it was a 
> VPN issue, but the same 
> thing just happened to his office phone.  Has anyone 
> seen this before? 
> 
>  
> 
> Jeremy Rogers 
> 
> Senior VoIP Engineer 
> 
> Warner Music Group 
> 

 

 
____________________________________________________________________________
________ 
Shape Yahoo! in your own image.  Join our Network Research Panel today!
http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7
<http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7>  

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From lmarasci at cisco.com  Tue Aug  7 16:54:26 2007
From: lmarasci at cisco.com (Louis Marascio (lmarasci))
Date: Tue, 7 Aug 2007 13:54:26 -0700
Subject: [cisco-voip] FW:  Speakerphone cuts out
In-Reply-To: <4DA11A95A4F28944B0B89404EF69A10B0348E85A@wmggsex25.wmg.com>
Message-ID: <6EE7932CC71C8D49A20C0E639BCD21650270DADB@xmb-sjc-21c.amer.cisco.com>

Make sure his hand set is not coming off hook slightly, i.e. if he bumps
the desk or something. If you start speakerphone w/ the handset down,
and then if the handset lifts, speakerphone will become disabled but the
call will still be connected. He won't hear anything since audio is now
flowing from the handset, but you will likely still hear him since the
handset will pick him up as he talks. It's easy enough to verify, when
it happens, ask him if his speakerphone button goes dark (i.e., isn't
lit anymore).
 
Louis

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rogers, Jeremy
(WMG)
Sent: Tuesday, August 07, 2007 3:31 PM
To: 'Paul Choi'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] FW: Speakerphone cuts out



No dropped packets, no errors on the diagnostics page.  I don't see
anything in the traces.  I can hear him talking into the mic, but the
speaker is not working on his end.

Jeremy Rogers 
Senior VoIP Engineer 
Warner Music Group 
75 Rockefeller Plaza 
New York, NY 10019 
212-275-2183 
-----Original Message----- 
From: Paul Choi [mailto:asobihoudai at yahoo.com] 
Sent: Tuesday, August 07, 2007 4:13 PM 
To: Rogers, Jeremy (WMG); cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] FW: Speakerphone cuts out 

Do you see anything odd in CCM traces? 

What about the phone status page? 

Do you see any abnormalities on the diagnostic page 
for the phone? [question mark button twice during 
conversation] 

Any dropped packets? 


--- "Rogers, Jeremy (WMG)" <Jeremy.Rogers at wmg.com> 
wrote: 

> Let me add that I just found out when the 
> speakerphone cuts out, so does the 
> ringer. 
> 
>  
>   _____  
> 
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On 
> Behalf Of Rogers, Jeremy 
> (WMG) 
> Sent: Tuesday, August 07, 2007 3:03 PM 
> To: cisco-voip at puck.nether.net 
> Subject: [cisco-voip] Speakerphone cuts out 
> 
>  
> 
> I have a user who is having a random issue.  He has 
> an IP phone at his house 
> behind a hardware firewall.  On occasion he said the 
> speakerphone would cut 
> out and that there was barely any sound coming out, 
> but if he picked up the 
> receiver he could hear fine.  We thought it was a 
> VPN issue, but the same 
> thing just happened to his office phone.  Has anyone 
> seen this before? 
> 
>  
> 
> Jeremy Rogers 
> 
> Senior VoIP Engineer 
> 
> Warner Music Group 
> 



 
________________________________________________________________________
____________ 
Shape Yahoo! in your own image.  Join our Network Research Panel today!
http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 

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From tman701 at gmail.com  Tue Aug  7 18:47:51 2007
From: tman701 at gmail.com (Joel Perez)
Date: Tue, 7 Aug 2007 18:47:51 -0400
Subject: [cisco-voip] 2851-intertel
In-Reply-To: <20070725045454.GB6783@geeks.org>
Message-ID: <46b8f6b7.4201360a.7016.1f2d@mx.google.com>

Hey guys,

Here is an update on my original question.

We used an VWIC2-2MFT-T1/E1 for the link between my 2851 and the Intertel. I
don't know what version of software the Intertel was running, but it's a
pretty recent one.
It was simpler than I thought on the cisco side, don't really know about the
intertel side of things though. Below I pasted the config for the T-1:

network-clock-participate wic 0

controller T1 0/0/1
 framing esf
 linecode ami
 cablelength short 133
 ds0-group 0 timeslots 1-24 type e&m-wink-start

dial-peer voice 110 pots
 description "Intertel"
 destination-pattern 153..
 port 0/0/1:0
 forward-digits 5

Now it is still under testing but so far I have had it up 1 week and no
issues at all. Might tweak it if anything comes up but so far nothing has.
Keeping fingers crossed.

Joel P

On Fri, Jul 20, 2007 at 03:48:05PM -0400, Joel Perez wrote:
> I was wondering if anybody out there has ever made an INtertel work with a
> 2851?
> Here is the scenario, I have 2 offices, One of them is brand new and is
> going to be using a 2811 running CME/CUE. There is an existing office that
> is running an INtertel PBX. I have never dealt with one of those so I cant
> give any specifics on it. We want 5 digit dialing between office A and
> office B. Both offices are connected to each other over an MPLS cloud. 
> Since the INtertel cant do SIP we are adding a 2851 in between the
Intertel
> and the 2811 at office A. So I will send calls that match a certain
> dial-peer to the 2851 via the MPLS cloud and then the 2851 will send it to
> the INtertel TDM via a T-1. 
> My question is if anyone has ever gotten an Intertel to work with a 2851
in
> this fashion before? The rest of the set up is not a problem, just that
part
> with the Intertel. There will be guys on site that are Intertel experts,
but
> im not sure how far does their Cisco knowledge go.



From jonvoip at gmail.com  Tue Aug  7 20:19:39 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 7 Aug 2007 19:19:39 -0500
Subject: [cisco-voip] CCM6 upgrade from 5...
Message-ID: <5d093f9a0708071719v3a983cdcs27227cb4c3328f59@mail.gmail.com>

So, anyone done it yet?

I did it on a test box, it took forever, but it worked... looks pretty cool.

Ummm... CCM service won't start, getting the following:

Aug 7 19:21:33 CCMPUB Error Cisco CallManager : 0: Aug 08 00:21:33.928
UTC : %CCM_CALLMANAGER-CALLMANAGER-3-NoFeatureLicense: No feature
license found. Cluster ID:StandAloneCluster Node ID:CCMPUB

So, the CCM service starts and then dies 25 seconds later.

I checked the licenses, I am still licensed, single node, 50 device
counts (demo lab license)...

But there is a new field...

Software License Version, and License Server...?

This is not on the CCM 5.1.2 box.

What does this mean?



Jonathan

From nathan.a.reeves at gmail.com  Wed Aug  8 00:10:18 2007
From: nathan.a.reeves at gmail.com (Nathan Reeves)
Date: Wed, 8 Aug 2007 12:10:18 +0800
Subject: [cisco-voip] CCM6 upgrade from 5...
Message-ID: <62833d8a0708072110s12e8df22wc124e52273e3f1ea@mail.gmail.com>

Do you have to upgrade the license file for the version change?  Was
looking through the Cisco site this morning on licensing and it
appears that there may be licensing based on the Major number of the
release.  You may have to get an updated license file to cover the
change from V5 to V6.

Should be mucking around with V6 next week once I've started a new
job, might find out more then.

Cheers

Nathan

From sergej.pioch at badenit.de  Wed Aug  8 01:55:30 2007
From: sergej.pioch at badenit.de (Pioch Sergej)
Date: Wed, 8 Aug 2007 07:55:30 +0200
Subject: [cisco-voip] FW:  Speakerphone cuts out
Message-ID: <00485976750F6340BC077D538603B2E181CD5C@bit123.badenit.intern>

We had a similar issue. Which firmware is currently installed on the
phone? We encountered this behavior after updating a CCM 4.3 with the
newest service release (4.2(3)SR2b) which also contained a devicepack
newer than 4.2(3.20).

 

In our lab we could reproduce this behavior. The following occurred in
conjunction with the pickup function. If the user presses the pickup
button and immediately lifts the headset, he will not hear the other
side speaking, but the other party hears loud and clear. If the user
presses the pickup button and waits for the display to come ready, means
showing all information, the phone is ringing for one time and he now
lifts the headset everything works fine. The behavior of the phone is
completely different with an older firmware version.

 

Best regards

 

 

S. Pioch

badenIT GmbH

 

> From: cisco-voip-bounces at puck.nether.net 

> [mailto:cisco-voip-bounces at puck.nether.net] On 

> Behalf Of Rogers, Jeremy 

> (WMG) 

> Sent: Tuesday, August 07, 2007 3:03 PM 

> To: cisco-voip at puck.nether.net 

> Subject: [cisco-voip] Speakerphone cuts out 

> 

>  

> 

> I have a user who is having a random issue.  He has 

> an IP phone at his house 

> behind a hardware firewall.  On occasion he said the 

> speakerphone would cut 

> out and that there was barely any sound coming out, 

> but if he picked up the 

> receiver he could hear fine.  We thought it was a 

> VPN issue, but the same 

> thing just happened to his office phone.  Has anyone 

> seen this before? 

> 

>  

> 

> Jeremy Rogers 

> 

> Senior VoIP Engineer 

> 

> Warner Music Group 

> 

 

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From patrick.diener at gmail.com  Wed Aug  8 03:38:51 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Wed, 8 Aug 2007 09:38:51 +0200
Subject: [cisco-voip] Cisco srtp CM 4.2.3
In-Reply-To: <000701c7d439$d4b31900$661b1b0a@cs.myharris.net>
References: <mailman.9003.1185969334.1529.cisco-voip@puck.nether.net>
	<000701c7d439$d4b31900$661b1b0a@cs.myharris.net>
Message-ID: <c0deb5b80708080038r1f41ef15q4148dc525215563d@mail.gmail.com>

If you want to read up on this topic the "CallManager Secuity Guide"
is the place to start:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_book09186a00806cd596.html

Regards
Patrick

On 8/1/07, cisco.voip <cisco.voip at verizon.net> wrote:
> Has anyone implemented  SRTP on Call Manager 4.2.3?
> How much of a hassle is it, and which server is recommended for the
> certifcates  other than using the call manager.
>
> Can it be implemented on a phoen by phone basis?
>
> Thanks,
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From Johnny.Crothers at vanco.co.uk  Wed Aug  8 06:51:26 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Wed, 8 Aug 2007 11:51:26 +0100
Subject: [cisco-voip]  Fax Machines and ATA's
Message-ID: <A56BD242744BF540B15D1672E0E3E8750193E141@LON-INT-MSGVS01.internal.vancocorp.com>

Has anyone had any issues with ATA 186's and Panafax UF series fax
machines? Cannot send or receive, although when ringing them, appears to
be presented just nothing happens on the fax. Analogue phones and other
models of fax's work fine.

 

Don't think it's a baud issue as the fax doesn't even recognise that
there is a call being presented.

 



**********************************************************************
Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient.  If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

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The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

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From ErasmuE4 at telkom.co.za  Wed Aug  8 07:04:24 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 8 Aug 2007 13:04:24 +0200
Subject: [cisco-voip] Fax Machines and ATA's
In-Reply-To: <A56BD242744BF540B15D1672E0E3E8750193E141@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F0182EA72@CNTRRA20-XCS01.telkom.co.za>


The group had several discussions on fax before

 

One thing I picked up was that you sometimes need to disable ECM and
try lower speeds

 

 

 

erik

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johnny
Crothers (c)
Sent: Wednesday, August 08, 2007 12:51 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Fax Machines and ATA's

 

Has anyone had any issues with ATA 186's and Panafax UF series fax
machines? Cannot send or receive, although when ringing them, appears
to be presented just nothing happens on the fax. Analogue phones and
other models of fax's work fine.

 

Don't think it's a baud issue as the fax doesn't even recognise that
there is a call being presented.

 



**********************************************************************
Any opinions expressed in the email are those of the individual and
not necessarily the company. This email and any files transmitted with
it are confidential and solely for the use of the intended recipient.
If you are not the intended recipient or the person responsible for
delivering it to the intended recipient, be advised that you have
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If you have received this email in error, or if you are concerned with
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From aelnagar at ACT-EG.COM  Wed Aug  8 07:42:22 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Wed, 8 Aug 2007 14:42:22 +0300
Subject: [cisco-voip] VMWI on analog phones with VG224
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED0110A3F1@mail.ACT-EG.COM>

Hello kenneth;

Thanks for your reply, the problem was that the VG224 only supports FSK
with 24 or 48 volt and the analog phone that I was testing was using 90
volt by default, this phone works well with the VG248 as it supports
this high voltage but not the VG224.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kenneth
Sinfield
Sent: Tuesday, August 07, 2007 12:37 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VMWI on analog phones with VG224

 

Hi Ahmed,

 

I have VMWI working on an ISR router with FXS ports (VMWI will work on a
VG224 as well). Basically below are the requirements:

 

*         The Cisco Gateway must run IOS version IP Voice T strain (not
SP Services on ISRs) 12.4(6)T or later (I tested 12.4(11)T2)

*         Cisco CallManager version must be version 4.2 or later (I
tested CallManager 5.1.2)

*         Any version of Unity (I tested Unity Connection 1.2)

*         Must use SCCP(Skinny) as the gateway protocol, and NOT MGCP

 

Here is an output from the gateway(MWI refers to stutter tone VMWI needs
to be "ON"):

 

pstn#show stcapp device voice-port 0/3/3

Port Identifier:  0/3/3

Device Type:      ALG 

Device Id:        4

Device Name:      AN17598075F0183

Modem Capability: None

Device State:     IS

Diagnostic:       None

Directory Number: 1005

Dial Peer(s):     101 

Dialtone after remote onhook feature: activated

Last Event:       STCAPP_CC_EV_CALL_DISCONNECT_DONE

Line State:       IDLE

Hook State:       ONHOOK

mwi:              ENABLE

vmwi:             ON

PLAR:             DISABLE

Number of CCBs:   0

Global call info:

    Total CCB count      = 0

    Total call leg count = 0

 

Thanks,

Ken.

 

Kenneth Sinfield

Systems Engineer

Dimension Data, Australia

139 Coronation Drive, Milton

Queensland, Australia, 4064

Phone: +61732921968

 

************************************************************************
******

- NOTICE FROM DIMENSION DATA AUSTRALIA

This message is confidential, and may contain proprietary or legally
privileged information. If you have received this email in error, please
notify the sender and delete it immediately.

 

Internet communications are not secure. You should scan this message and
any attachments for viruses. Under no circumstances do we accept
liability for any loss or damage which may result from your receipt of
this message or any attachments.

************************************************************************
******

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From aelnagar at ACT-EG.COM  Wed Aug  8 07:45:04 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Wed, 8 Aug 2007 14:45:04 +0300
Subject: [cisco-voip] Securing CM
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED0110A3F3@mail.ACT-EG.COM>

Hello all;

 

For all running CM how do you secure it from viruses attacks, how do you
secure its database from corruption. If anyone has best practice for
securing CM please share with us.

 

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From CarlosOrtiz at bayviewfinancial.com  Wed Aug  8 08:56:07 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 8 Aug 2007 08:56:07 -0400
Subject: [cisco-voip] IPCC Express IP address change
Message-ID: <OFCDFCF057.5D6576F8-ON85257331.0045BF5C-85257331.00470E68@bftg.com>

We are looking to move our IPCC server and change the IP address.  What is 
the quickest way to make this change for all IPCC agents?  Do we need to 
do some type of SMS push or something?

Carlos
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From Matt.Slaga at us.didata.com  Wed Aug  8 10:40:19 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 8 Aug 2007 10:40:19 -0400
Subject: [cisco-voip] IPCC Express IP address change
In-Reply-To: <OFCDFCF057.5D6576F8-ON85257331.0045BF5C-85257331.00470E68@bftg.com>
References: <OFCDFCF057.5D6576F8-ON85257331.0045BF5C-85257331.00470E68@bftg.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0065FB04D@USNAEXCH.na.didata.local>

This depends on how the clients were installed.  Did you point to IPCC
Express IP address or server name?  If server name, you should be in
good shape except for recording registry entries will need to be
modified or the agent software reinstalled.  If they are pointing to the
server IP Address, you will be forced to reinstall the agents.  

 

Not sure about SMS packages for this.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Wednesday, August 08, 2007 8:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Express IP address change

 


We are looking to move our IPCC server and change the IP address.  What
is the quickest way to make this change for all IPCC agents?  Do we need
to do some type of SMS push or something? 

Carlos




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From Chris.Ellington at inin.com  Wed Aug  8 10:44:15 2007
From: Chris.Ellington at inin.com (Ellington, Chris)
Date: Wed, 8 Aug 2007 10:44:15 -0400
Subject: [cisco-voip] TLS/SRTP
Message-ID: <8F9F1C6AA1D6834EACC379C8821533A605771CB9@i3mail1.i3domain.inin.com>

Anyone using TLS/SRTP and if so, can you share experiences?

Particularly, I'm looking for a way to decode packet captures to replay
audio streams but any experiences with TLS & SRTP is appreciated!

Thanks,

chris

Christopher Ellington | VoIP/SIP Engineer 
phone & fax +1.317.715.8578 | chris.ellington at inin.com
CCIE #6814


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From moody at arizona.edu  Wed Aug  8 10:14:45 2007
From: moody at arizona.edu (Walt Moody)
Date: Wed, 08 Aug 2007 07:14:45 -0700
Subject: [cisco-voip] Fax Machines and ATA's
In-Reply-To: <A56BD242744BF540B15D1672E0E3E8750193E141@LON-INT-MSGVS01.internal.vancocorp.com>
References: <A56BD242744BF540B15D1672E0E3E8750193E141@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <46B9CFD5.1020407@arizona.edu>

Since the fax is not answering the call, the distinctive ring detector
is probably not matched to the ATA's ring pattern.  Check this in the
fax's setup menu.

-walt

Johnny Crothers (c) wrote:
> Has anyone had any issues with ATA 186?s and Panafax UF series fax 
> machines? Cannot send or receive, although when ringing them, appears to 
> be presented just nothing happens on the fax. Analogue phones and other 
> models of fax?s work fine.
> 
>  
> 
> Don?t think it?s a baud issue as the fax doesn?t even recognise that 
> there is a call being presented.
> 
>  
> 
> 
> 
> **********************************************************************
> Any opinions expressed in the email are those of the individual and not 
> necessarily the company. This email and any files transmitted with it 
> are confidential and solely for the use of the intended recipient. If 
> you are not the intended recipient or the person responsible for 
> delivering it to the intended recipient, be advised that you have 
> received this email in error and that any dissemination, distribution, 
> copying or use is strictly prohibited.
> 
> If you have received this email in error, or if you are concerned with 
> the content of this email please e-mail to: e-security.support at vanco.info
> 
> The contents of an attachment to this e-mail may contain software 
> viruses which could damage your own computer system. While the sender 
> has taken every reasonable precaution to minimise this risk, we cannot 
> accept liability for any damage which you sustain as a result of 
> software viruses. You should carry out your own virus checks before 
> opening any attachments to this e-mail.
> 
> Vanco UK Ltd Registered in England No: 2296733 Registered Office: John 
> Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX
> 
> Please consider the environment before printing this e-mail
> **********************************************************************
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From wsisk at cisco.com  Wed Aug  8 11:25:22 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 08 Aug 2007 11:25:22 -0400
Subject: [cisco-voip] Who runs the "Tools & Resources" Software News
 thing on CCO?
In-Reply-To: <46AF9235.1020104@cisco.com>
References: <46AF8E30.30404@smalltime.com> <46AF9235.1020104@cisco.com>
Message-ID: <46B9E062.6060109@cisco.com>

CM6.0 and Unity 5.0 are available in the subscription tool.

CM 6.0(1a) posted last night and I approved the news article today so 
notification will go out tonight if you are subscribed for CM6.

/Wes

Wes Sisk wrote:
> I don't run all of it, but I do have some influence to that tool.  
> Working on it.  Usual ETA is 4 weeks. /Wes
>
> Robert Kulagowski wrote:
>> http://www.cisco.com/cgi-bin/Software/Newsbuilder/Builder/VOICE.cgi
>>
>> I've signed up, but it doesn't seem to send me messages anymore.  
>> Also, it's way out of date.
>> "4:50 PM 3/29/2006"
>>
>> Doesn't have CM 6.0
>> Doesn't have Unity 5.0
>>
>> Checkboxes for versions seem to be in random order.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>   
>

From wsisk at cisco.com  Wed Aug  8 11:33:34 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 08 Aug 2007 11:33:34 -0400
Subject: [cisco-voip] Securing CM
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED0110A3F3@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED0110A3F3@mail.ACT-EG.COM>
Message-ID: <46B9E24E.1030609@cisco.com>

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From CarlosOrtiz at bayviewfinancial.com  Wed Aug  8 11:36:29 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 8 Aug 2007 11:36:29 -0400
Subject: [cisco-voip] IPCC Express IP address change
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0065FB04D@USNAEXCH.na.didata.local>
Message-ID: <OFECA2FFA9.216A714A-ON85257331.0055B9B8-85257331.0055BD00@bftg.com>

I didn't personally do it  but from talking to the staff it looks like it 
has been a mixture of installations.  According to TAC all I need to do 
for each client is run the Post Install.exe files found in c:\program 
files\cisco\desktop\bin.

Why this is not an executable file that can just be executed with a switch 
and change the IP....such as postinstall.exe /10.10.10.1 or 
postinstall.exe /NEW-IPCC-SERVER-NAME  is beyond me........



"Matt Slaga \(US\)" <Matt.Slaga at us.didata.com> 
08/08/2007 10:40 AM

To
<CarlosOrtiz at bayviewfinancial.com>, <cisco-voip at puck.nether.net>
cc

Subject
RE: [cisco-voip] IPCC Express IP address change






This depends on how the clients were installed.  Did you point to IPCC 
Express IP address or server name?  If server name, you should be in good 
shape except for recording registry entries will need to be modified or 
the agent software reinstalled.  If they are pointing to the server IP 
Address, you will be forced to reinstall the agents. 
 
Not sure about SMS packages for this.
 
 
 
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Wednesday, August 08, 2007 8:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Express IP address change
 

We are looking to move our IPCC server and change the IP address.  What is 
the quickest way to make this change for all IPCC agents?  Do we need to 
do some type of SMS push or something? 

Carlos

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  Wed Aug  8 11:41:42 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 08 Aug 2007 11:41:42 -0400
Subject: [cisco-voip] CCM6 upgrade from 5...
In-Reply-To: <5d093f9a0708071719v3a983cdcs27227cb4c3328f59@mail.gmail.com>
References: <5d093f9a0708071719v3a983cdcs27227cb4c3328f59@mail.gmail.com>
Message-ID: <46B9E436.6090004@cisco.com>

You have to get a 'software feature license' after upgrade to re-enable 
the features you had licensed in 5.x:

http://www.cisco.com/en/US/customer/docs/voice_ip_comm/cucm/rel_notes/6_0_1/601cmrn.html#wp45773

/Wes

Jonathan Charles wrote:
> So, anyone done it yet?
>
> I did it on a test box, it took forever, but it worked... looks pretty cool.
>
> Ummm... CCM service won't start, getting the following:
>
> Aug 7 19:21:33 CCMPUB Error Cisco CallManager : 0: Aug 08 00:21:33.928
> UTC : %CCM_CALLMANAGER-CALLMANAGER-3-NoFeatureLicense: No feature
> license found. Cluster ID:StandAloneCluster Node ID:CCMPUB
>
> So, the CCM service starts and then dies 25 seconds later.
>
> I checked the licenses, I am still licensed, single node, 50 device
> counts (demo lab license)...
>
> But there is a new field...
>
> Software License Version, and License Server...?
>
> This is not on the CCM 5.1.2 box.
>
> What does this mean?
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From rsingleton at morsco.com  Wed Aug  8 12:12:07 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Wed, 08 Aug 2007 11:12:07 -0500
Subject: [cisco-voip] Fax Machines and ATA's
In-Reply-To: <A56BD242744BF540B15D1672E0E3E8750193E141@LON-INT-MSGVS01.internal.vancocorp.com>
References: <A56BD242744BF540B15D1672E0E3E8750193E141@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <1186589527.18834.46.camel@robert.morsco.com>

On Wed, 2007-08-08 at 11:51 +0100, Johnny Crothers (c) wrote:
> Has anyone had any issues with ATA 186?s and Panafax UF series fax
> machines? Cannot send or receive, although when ringing them, appears
> to be presented just nothing happens on the fax. Analogue phones and
> other models of fax?s work fine.
> 
>  
> 
> Don?t think it?s a baud issue as the fax doesn?t even recognise that
> there is a call being presented.

There is one thing that I actually thought of when I was re-reading this
post and it might explain it better than any of the thoughts below. Is
the RJ11 line cord in the "line" jack or the "phone" jack? Sometimes it
matters.

Though I haven't had *this* problem with any of our faxes, I understand
that some faxes can be sensitive to the ring voltage from the ATA, which
may not be be high enough. This kind of problem usually only presents
itself when using an ATA to drive a house full of bell ringers (that's
where that ringer equivalency rating comes in). Bell ringers draw more
current than electronic ringers and an ATA (and most ISDN modems, BTW)
are much more current limited than your LEC, so the voltage drops, per
Ohm's law, and then electronic ringers don't see ring voltage.

The ringing cadence can be an issue, too... if the ATA is doing two
short rings, a stubborn fax machine may not see it.

Finally, it's incredibly rare, but some devices can be polarity
sensitive. The "standard" RJ11 cord flips polarity from one end to the
other, but many people make their own cords, often without a flip or
maybe without regard to polarity at all and many times, and an RJ11 data
patch cord (a rare bird these days, too) will specifically *not* flip
polarity. Almost nothing modern cares what the polarity is, but it's an
easy thing to try...

Robert




From CarlosOrtiz at bayviewfinancial.com  Wed Aug  8 14:00:46 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 8 Aug 2007 14:00:46 -0400
Subject: [cisco-voip] IPCC Express IP address change
In-Reply-To: <OFECA2FFA9.216A714A-ON85257331.0055B9B8-85257331.0055BD00@bftg.com>
Message-ID: <OF4A64DE64.708699EB-ON85257331.0062C1F6-85257331.0062F28B@bftg.com>

Latest update on this from TAC in case anyone runs into this situation. 
You can remotely edit the registry of all the agent PC's under the 
following key:

HKEY_LOCAL_MACHINE\SOFTWARE\Spanlink\Site Setup\LDAP Host 1

Set to the new IP address and you are good to go.  We are going to use a 
batch tool with Dameware to make the change.

Carlos



CarlosOrtiz at bayviewfinancial.com 
Sent by: cisco-voip-bounces at puck.nether.net
08/08/2007 11:36 AM

To
"Matt Slaga \(US\)" <Matt.Slaga at us.didata.com>
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] IPCC Express IP address change







I didn't personally do it  but from talking to the staff it looks like it 
has been a mixture of installations.  According to TAC all I need to do 
for each client is run the Post Install.exe files found in c:\program 
files\cisco\desktop\bin. 

Why this is not an executable file that can just be executed with a switch 
and change the IP....such as postinstall.exe /10.10.10.1 or 
postinstall.exe /NEW-IPCC-SERVER-NAME  is beyond me........ 


"Matt Slaga \(US\)" <Matt.Slaga at us.didata.com> 
08/08/2007 10:40 AM 


To
<CarlosOrtiz at bayviewfinancial.com>, <cisco-voip at puck.nether.net> 
cc

Subject
RE: [cisco-voip] IPCC Express IP address change








This depends on how the clients were installed.  Did you point to IPCC 
Express IP address or server name?  If server name, you should be in good 
shape except for recording registry entries will need to be modified or 
the agent software reinstalled.  If they are pointing to the server IP 
Address, you will be forced to reinstall the agents.   
  
Not sure about SMS packages for this. 
  
  
  
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Wednesday, August 08, 2007 8:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Express IP address change 
  

We are looking to move our IPCC server and change the IP address.  What is 
the quickest way to make this change for all IPCC agents?  Do we need to 
do some type of SMS push or something? 

Carlos 

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 Joe.Quigley at braunlift.com  Wed Aug  8 13:46:21 2007
From: Joe.Quigley at braunlift.com (Joe Quigley)
Date: Wed, 8 Aug 2007 13:46:21 -0400
Subject: [cisco-voip] Looking for the kindest way to find the top callers
	into IPCC
Message-ID: <8706B9F854CD7B4E94E11A373FE89E74015E9A3E@BRAUNEXCHANGE.braunlift.com>

I have been requested to find out who the top callers are into IPCC, and
I am finding out the CRS Historical Reporting isn't going to be the
answer.

 

I have CCM 4.1(3) and IPCCX4.0(5) in the scenario. The report aims to
provide who the top callers are in to our Product Support queues to
determine which of our product dealers may need more training.

 

If anyone has been down this road before, I would love to hear their
experience and advice. Thanks!

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From WOCAMPO at broward.org  Wed Aug  8 15:24:52 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Wed, 8 Aug 2007 15:24:52 -0400
Subject: [cisco-voip] Looking for the kindest way to find the top
	callersinto IPCC
In-Reply-To: <8706B9F854CD7B4E94E11A373FE89E74015E9A3E@BRAUNEXCHANGE.braunlift.com>
References: <8706B9F854CD7B4E94E11A373FE89E74015E9A3E@BRAUNEXCHANGE.braunlift.com>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA450@UVEXVS01.bc.broward.cty>

You are in the right track, if you just want to know who are the top
callers. Historical report will give you that information.

 

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

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Quigley
Sent: Wednesday, August 08, 2007 1:46 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Looking for the kindest way to find the top
callersinto IPCC

 

I have been requested to find out who the top callers are into IPCC, and
I am finding out the CRS Historical Reporting isn't going to be the
answer.

 

I have CCM 4.1(3) and IPCCX4.0(5) in the scenario. The report aims to
provide who the top callers are in to our Product Support queues to
determine which of our product dealers may need more training.

 

If anyone has been down this road before, I would love to hear their
experience and advice. Thanks!

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From wsisk at cisco.com  Wed Aug  8 16:19:30 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 08 Aug 2007 16:19:30 -0400
Subject: [cisco-voip] check your versions of VNC
Message-ID: <46BA2552.3050504@cisco.com>

http://www.purdue.edu/securepurdue/steam/newsDetail.cfm?NewsID=59

From lelio at uoguelph.ca  Wed Aug  8 17:14:55 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 8 Aug 2007 17:14:55 -0400
Subject: [cisco-voip] cisco forums down?
Message-ID: <004e01c7da01$2b1271c0$47196883@cfs.uoguelph.ca>

 is it just me or are the cisco forums down?

--------------------------------------------------------------------------------
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 pwalenta at wi.rr.com  Wed Aug  8 18:01:00 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Wed, 8 Aug 2007 17:01:00 -0500
Subject: [cisco-voip] cisco forums down?
In-Reply-To: <004e01c7da01$2b1271c0$47196883@cfs.uoguelph.ca>
Message-ID: <008501c7da07$9c04e060$3b3812ac@pwnhome.com>

CCO had some sort of crash a few hours ago.  They are slowly bringing
everything back.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, August 08, 2007 4:15 PM
To: cisco voip list
Subject: [cisco-voip] cisco forums down?


 is it just me or are the cisco forums down?
 
----------------------------------------------------------------------------
----
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 patrick.diener at gmail.com  Wed Aug  8 18:57:08 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Thu, 9 Aug 2007 00:57:08 +0200
Subject: [cisco-voip] TLS/SRTP
In-Reply-To: <8F9F1C6AA1D6834EACC379C8821533A605771CB9@i3mail1.i3domain.inin.com>
References: <8F9F1C6AA1D6834EACC379C8821533A605771CB9@i3mail1.i3domain.inin.com>
Message-ID: <c0deb5b80708081557s4b1e6396x4cfdbf959593d784@mail.gmail.com>

Hi Chris

if you want to capture and decode encrypted signaling and voice
traffic in a Unified CM 5.X environment this is the way to go:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a00806b7b4a.html#wp1090938

Regards
Patrick

On 8/8/07, Ellington, Chris <Chris.Ellington at inin.com> wrote:
>
>
>
> Anyone using TLS/SRTP and if so, can you share experiences?
>
> Particularly, I'm looking for a way to decode packet captures to replay
> audio streams but any experiences with TLS & SRTP is appreciated!
>
> Thanks,
>
> chris
>
> Christopher Ellington | VoIP/SIP Engineer
>  phone & fax +1.317.715.8578 | chris.ellington at inin.com
> CCIE #6814
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jefflin.choi at gmail.com  Wed Aug  8 21:02:15 2007
From: jefflin.choi at gmail.com (Jefflin Choi)
Date: Thu, 9 Aug 2007 09:02:15 +0800
Subject: [cisco-voip] 3rd party SIP phone using FAC?
Message-ID: <5c72afa80708081802t3aa3314ag280a2ee2ff169d1c@mail.gmail.com>

Hi all,

It seems that FAC/CMC is not working on 3rd party SIP Phone.

Any workaround to have long distance reporting per user on CDR with 3rd
party SIP phone?

Using CCM 5.0.4.

Thanks,
Jeff
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From aelnagar at ACT-EG.COM  Thu Aug  9 02:12:03 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Thu, 9 Aug 2007 09:12:03 +0300
Subject: [cisco-voip] cisco forums down?
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED0110A5CC@mail.ACT-EG.COM>

Just you :-) 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Thursday, August 09, 2007 12:15 AM
To: cisco voip list
Subject: [cisco-voip] cisco forums down?

 

 is it just me or are the cisco forums down?

 

------------------------------------------------------------------------
--------
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 aelnagar at ACT-EG.COM  Thu Aug  9 06:05:33 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Thu, 9 Aug 2007 13:05:33 +0300
Subject: [cisco-voip] CM Installation
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED0110A7FC@mail.ACT-EG.COM>

Hello all;

 

Is there is an option to install Cisco Callmanager from scratch without
using a monitor or a keyboard, just from my laptop. Can this be done?

 

Regards,

Ahmed Elnagar

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From manoj.kalpage at gmail.com  Thu Aug  9 08:41:27 2007
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Thu, 9 Aug 2007 21:41:27 +0900
Subject: [cisco-voip] IPCC Express V4.0.1
Message-ID: <bbe09c1d0708090541mb479e7pc01c3341a574ca95@mail.gmail.com>

I am looking for  IPCC demo license for my CCIE voice lab. Does anyone
know where I can get them?

Thanks,
Pere

From wsisk at cisco.com  Thu Aug  9 10:07:40 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 09 Aug 2007 10:07:40 -0400
Subject: [cisco-voip] CM Installation
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED0110A7FC@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED0110A7FC@mail.ACT-EG.COM>
Message-ID: <46BB1FAC.5060001@cisco.com>

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From aelnagar at ACT-EG.COM  Thu Aug  9 10:18:01 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Thu, 9 Aug 2007 17:18:01 +0300
Subject: [cisco-voip] CM Installation
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED01134287@mail.ACT-EG.COM>

But seems it is not supported for the 4.x train, what about the ILO port
on HP servers, could it help in this??

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Thursday, August 09, 2007 5:08 PM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM Installation

 

'answer file' feature
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_note09
186a008079a217.html#wp350593

/Wes

Ahmed Elnagar wrote: 

Hello all;

 

Is there is an option to install Cisco Callmanager from scratch without
using a monitor or a keyboard, just from my laptop. Can this be done?

 

Regards,

Ahmed Elnagar

 



________________________________



 
_______________________________________________
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 Aug  9 10:21:50 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 9 Aug 2007 09:21:50 -0500
Subject: [cisco-voip] CM Installation
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED01134287@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED01134287@mail.ACT-EG.COM>
Message-ID: <5d093f9a0708090721t421181cdhd4f2964040346541@mail.gmail.com>

I have installed CCM and Unity completely thru VNC...

The Unity install thru ILO has failed in the past... (Cisco pointed to
it being because of ILO, but I didn't believe it).


Jonathan

On 8/9/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>
>
>
> But seems it is not supported for the 4.x train, what about the ILO port on
> HP servers, could it help in this??
>
>
>
>  ________________________________
>
>
> From: Wes Sisk [mailto:wsisk at cisco.com]
>  Sent: Thursday, August 09, 2007 5:08 PM
>  To: Ahmed Elnagar
>  Cc: cisco-voip at puck.nether.net
>  Subject: Re: [cisco-voip] CM Installation
>
>
>
>
> 'answer file' feature
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_note09186a008079a217.html#wp350593
>
>  /Wes
>
>  Ahmed Elnagar wrote:
>
> Hello all;
>
>
>
> Is there is an option to install Cisco Callmanager from scratch without
> using a monitor or a keyboard, just from my laptop. Can this be done?
>
>
>
> Regards,
>
> Ahmed Elnagar
>
>
> ________________________________
>
>
>
>
> _______________________________________________
> 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  Thu Aug  9 10:25:18 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Thu, 9 Aug 2007 17:25:18 +0300
Subject: [cisco-voip] CM Installation
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED01134295@mail.ACT-EG.COM>

Yes I know the VNC could solve the application part (CallManager or
Unity). But what I need is the operating system part, I face this
problem in sites when I have to reinstall everything and I don't have a
screen and a keyboard nearby (strange situation I know but I faced it)

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Thursday, August 09, 2007 5:22 PM
To: Ahmed Elnagar
Cc: Wes Sisk; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM Installation

I have installed CCM and Unity completely thru VNC...

The Unity install thru ILO has failed in the past... (Cisco pointed to
it being because of ILO, but I didn't believe it).


Jonathan

On 8/9/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>
>
>
> But seems it is not supported for the 4.x train, what about the ILO
port on
> HP servers, could it help in this??
>
>
>
>  ________________________________
>
>
> From: Wes Sisk [mailto:wsisk at cisco.com]
>  Sent: Thursday, August 09, 2007 5:08 PM
>  To: Ahmed Elnagar
>  Cc: cisco-voip at puck.nether.net
>  Subject: Re: [cisco-voip] CM Installation
>
>
>
>
> 'answer file' feature
>
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_note09
186a008079a217.html#wp350593
>
>  /Wes
>
>  Ahmed Elnagar wrote:
>
> Hello all;
>
>
>
> Is there is an option to install Cisco Callmanager from scratch
without
> using a monitor or a keyboard, just from my laptop. Can this be done?
>
>
>
> Regards,
>
> Ahmed Elnagar
>
>
> ________________________________
>
>
>
>
> _______________________________________________
> 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 Aug  9 10:28:24 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 9 Aug 2007 09:28:24 -0500
Subject: [cisco-voip] CCM6 upgrade from 5...
In-Reply-To: <46B9E436.6090004@cisco.com>
References: <5d093f9a0708071719v3a983cdcs27227cb4c3328f59@mail.gmail.com>
	<46B9E436.6090004@cisco.com>
Message-ID: <5d093f9a0708090728w41ab5a4ej821994484023cc61@mail.gmail.com>

OK, so there is no leeway, no temp license to keep them up and running
for 7 days or so to get a feature license?

I think there should be a giant warning here, if someone were to
upgrade to 6 without knowing this, the net affect would be to take out
their Call Managers.

Just so everyone else is aware, after the upgrade of the pub, the CCM
service is taken out (will not start), until they get a new feature
license for 6. If they have a pub and sub, the pub will work fine, and
not scream any particular warning. When they do the sub and it
reboots, all of their phones are going down, and staying down until
the feature license is applied to the Pub.

The funny thing is that the phones got new code when they tried to
come up (which would be expected after an upgrade...), so, you would
have to reboot off of the old partition to get your system back up
until you got the license.

This is scary.


Jonathan

On 8/8/07, Wes Sisk <wsisk at cisco.com> wrote:
> You have to get a 'software feature license' after upgrade to re-enable
> the features you had licensed in 5.x:
>
> http://www.cisco.com/en/US/customer/docs/voice_ip_comm/cucm/rel_notes/6_0_1/601cmrn.html#wp45773
>
> /Wes
>
> Jonathan Charles wrote:
> > So, anyone done it yet?
> >
> > I did it on a test box, it took forever, but it worked... looks pretty cool.
> >
> > Ummm... CCM service won't start, getting the following:
> >
> > Aug 7 19:21:33 CCMPUB Error Cisco CallManager : 0: Aug 08 00:21:33.928
> > UTC : %CCM_CALLMANAGER-CALLMANAGER-3-NoFeatureLicense: No feature
> > license found. Cluster ID:StandAloneCluster Node ID:CCMPUB
> >
> > So, the CCM service starts and then dies 25 seconds later.
> >
> > I checked the licenses, I am still licensed, single node, 50 device
> > counts (demo lab license)...
> >
> > But there is a new field...
> >
> > Software License Version, and License Server...?
> >
> > This is not on the CCM 5.1.2 box.
> >
> > What does this mean?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From jonvoip at gmail.com  Thu Aug  9 10:31:01 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 9 Aug 2007 09:31:01 -0500
Subject: [cisco-voip] CM Installation
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED01134295@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED01134295@mail.ACT-EG.COM>
Message-ID: <5d093f9a0708090731p1b11bad3s6bf621edf5a1e2d0@mail.gmail.com>

It isn't strange at all... a lot of people don't have KVM connected to
their CCMs...

Your only real option is ILO (or whatever IBM equivalent there is) or
to get a KVM hooked up for OS and then install VNC and go from there.


J

On 8/9/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
> Yes I know the VNC could solve the application part (CallManager or
> Unity). But what I need is the operating system part, I face this
> problem in sites when I have to reinstall everything and I don't have a
> screen and a keyboard nearby (strange situation I know but I faced it)
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Thursday, August 09, 2007 5:22 PM
> To: Ahmed Elnagar
> Cc: Wes Sisk; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CM Installation
>
> I have installed CCM and Unity completely thru VNC...
>
> The Unity install thru ILO has failed in the past... (Cisco pointed to
> it being because of ILO, but I didn't believe it).
>
>
> Jonathan
>
> On 8/9/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
> >
> >
> >
> >
> > But seems it is not supported for the 4.x train, what about the ILO
> port on
> > HP servers, could it help in this??
> >
> >
> >
> >  ________________________________
> >
> >
> > From: Wes Sisk [mailto:wsisk at cisco.com]
> >  Sent: Thursday, August 09, 2007 5:08 PM
> >  To: Ahmed Elnagar
> >  Cc: cisco-voip at puck.nether.net
> >  Subject: Re: [cisco-voip] CM Installation
> >
> >
> >
> >
> > 'answer file' feature
> >
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_note09
> 186a008079a217.html#wp350593
> >
> >  /Wes
> >
> >  Ahmed Elnagar wrote:
> >
> > Hello all;
> >
> >
> >
> > Is there is an option to install Cisco Callmanager from scratch
> without
> > using a monitor or a keyboard, just from my laptop. Can this be done?
> >
> >
> >
> > Regards,
> >
> > Ahmed Elnagar
> >
> >
> > ________________________________
> >
> >
> >
> >
> > _______________________________________________
> > 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 David.Sullivan at barnet.ac.uk  Thu Aug  9 10:30:45 2007
From: David.Sullivan at barnet.ac.uk (David Sullivan)
Date: Thu, 9 Aug 2007 15:30:45 +0100
Subject: [cisco-voip] CM Installation
In-Reply-To: <5d093f9a0708090721t421181cdhd4f2964040346541@mail.gmail.com>
Message-ID: <EDB47A2F10617E4095AE70CBBC723F611E2F45@oak.bc-ad.barnet.ac.uk>

cisco-voip-bounces at puck.nether.net wrote:
> I have installed CCM and Unity completely thru VNC...
> 
> The Unity install thru ILO has failed in the past... (Cisco
> pointed to it being because of ILO, but I didn't believe it).
> 

I'd find it hard to believe, the only potential oddity about ilo is the
media redirection pretends to be a USB cd/floppy drive which might be a
problem though I can't say I've had any problems installing any using
it, you could of course put the CD in the server's physical drive and
just use the ILO for keyboard video and mouse.

David Sullivan
IT Services
Barnet College
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From Chris.Ellington at inin.com  Thu Aug  9 11:12:24 2007
From: Chris.Ellington at inin.com (Ellington, Chris)
Date: Thu, 9 Aug 2007 11:12:24 -0400
Subject: [cisco-voip] cisco-voip Digest, Vol 54, Issue 38
In-Reply-To: <mailman.10162.1186669100.1529.cisco-voip@puck.nether.net>
References: <mailman.10162.1186669100.1529.cisco-voip@puck.nether.net>
Message-ID: <8F9F1C6AA1D6834EACC379C8821533A605771CD2@i3mail1.i3domain.inin.com>

After reading the relevant sections of the guide, I have three follow on
questions:

1) Anyone know what "packet capture" does on the server?  From reading,
it appears that it merely creates the traces but since it indicates it
can cause fairly high CPU I want to make sure it's not doing something
more

2) What magic does TAC perform/what tool is used to decrypt?  It would
be nice to decrypt without opening a case and waiting on TAC

3) What about 4.x?  I'm searching CCO now for this answer but if someone
knows off the top of their head...

Thanks,

Chris

>
>Hi Chris
>
>if you want to capture and decode encrypted signaling and voice
>traffic in a Unified CM 5.X environment this is the way to go:
>http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooti
ng_guide_chapter09186a00806b7b4a.html#wp>1090938
>
>Regards
>Patrick
>
>On 8/8/07, Ellington, Chris <Chris.Ellington at inin.com> wrote:
>>
>>
>>
>> Anyone using TLS/SRTP and if so, can you share experiences?
>>
>> Particularly, I'm looking for a way to decode packet captures to
replay
>> audio streams but any experiences with TLS & SRTP is appreciated!
>>
>> Thanks,
>>
>> chris
>>
>> Christopher Ellington | VoIP/SIP Engineer
>>  phone & fax +1.317.715.8578 | chris.ellington at inin.com
>> CCIE #6814
>>



From wsisk at cisco.com  Thu Aug  9 11:30:31 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 09 Aug 2007 11:30:31 -0400
Subject: [cisco-voip] CM Installation
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED01134295@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED01134295@mail.ACT-EG.COM>
Message-ID: <46BB3317.5080004@cisco.com>

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From asobihoudai at yahoo.com  Thu Aug  9 11:37:29 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 9 Aug 2007 08:37:29 -0700 (PDT)
Subject: [cisco-voip] CM Installation
In-Reply-To: <46BB3317.5080004@cisco.com>
Message-ID: <655460.26041.qm@web57212.mail.re3.yahoo.com>

On that note, I'd just like to say

STAY THE HELL AWAY FROM AVOCENT!

We installed an IP-KVM and everytime we'd try to
remotely administer the CCM/Unity/CRS boxen attached
to them, random weird crap would happen. The KVM would
turn the power off or reboot the boxes. Beware of
avocent. Their support sucks eggs too unless you're
willing to buy into their "installation support scam"
for several thousands of dollars.

--- Wes Sisk <wsisk at cisco.com> wrote:


---------------------------------
  You need an IP-KVM then, or iLo. /wes

Ahmed Elnagar wrote:  
Yes I know the VNC could solve the application part
(CallManager orUnity). But what I need is the
operating system part, I face thisproblem in sites
when I have to reinstall everything and I don't have
ascreen and a keyboard nearby (strange situation I
know but I faced it)-----Original Message-----From:
Jonathan Charles [mailto:jonvoip at gmail.com] Sent:
Thursday, August 09, 2007 5:22 PMTo: Ahmed ElnagarCc:
Wes Sisk; cisco-voip at puck.nether.netSubject: Re:
[cisco-voip] CM InstallationI have installed CCM and
Unity completely thru VNC...The Unity install thru ILO
has failed in the past... (Cisco pointed toit being
because of ILO, but I didn't believe it).JonathanOn
8/9/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:  
      
But seems it is not supported for the 4.x train, what
about the ILO    
    
port on  
      
HP servers, could it help in this??
________________________________From: Wes Sisk
[mailto:wsisk at cisco.com] Sent: Thursday, August 09,
2007 5:08 PM To: Ahmed Elnagar Cc:
cisco-voip at puck.nether.net Subject: Re: [cisco-voip]
CM Installation'answer file' feature    
    
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_note09186a008079a217.html#wp350593
 
      
 /Wes Ahmed Elnagar wrote:Hello all;Is there is an
option to install Cisco Callmanager from scratch    
    
without  
      
using a monitor or a keyboard, just from my laptop.
Can this be done?Regards,Ahmed
Elnagar_______________________________________________________________________________cisco-voip
mailing
listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip_______________________________________________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



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

From gokhanciscottl at yahoo.com  Thu Aug  9 10:42:40 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Thu, 9 Aug 2007 07:42:40 -0700 (PDT)
Subject: [cisco-voip] fxo ports and fct   CCM 4.1.3
Message-ID: <424474.60469.qm@web39613.mail.mud.yahoo.com>

Hi
i need your help about 2 problems below

1.   I got CCM 4.1.3 and 2851 gateway,
there is one VIC2-4FXO  card and  FCT devices  which are connected to these FXO ports for GSM calls

the problem is that sometimes FCT s are locked. And the customer wanna see which one is locked
Is it possible to see fxo port status from callmanager . I can check it from gateway by  executing  ' show voice port summary'   but I dont know how to see from CCM webpage,i wanna see if the port is really used at that moment. 



2.Also another thing ;  when i call GSM  from fct devices , sometimes ip phone displays that it is connected and the connected time is increasing but actually the call is not set up. I wait 50-55 seconds but there is no even ring tone. 
And then i  close the phone adn call again.


thanks  a lot


       
____________________________________________________________________________________
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  
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From WOCAMPO at broward.org  Thu Aug  9 11:58:01 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Thu, 9 Aug 2007 11:58:01 -0400
Subject: [cisco-voip] fxo ports and fct   CCM 4.1.3
In-Reply-To: <424474.60469.qm@web39613.mail.mud.yahoo.com>
References: <424474.60469.qm@web39613.mail.mud.yahoo.com>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA459@UVEXVS01.bc.broward.cty>

You need to launch the RTMT,  It will display the status of the port in
real time. 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administra
tion_guide_chapter09186a008062a98c.html#wp28847

 

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

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of gokhan senol
Sent: Thursday, August 09, 2007 10:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] fxo ports and fct CCM 4.1.3

 

Hi

i need your help about 2 problems below

 

1.   I got CCM 4.1.3 and 2851 gateway,

there is one VIC2-4FXO  card and  FCT devices  which are connected to
these FXO ports for GSM calls

 

the problem is that sometimes FCT s are locked. And the customer wanna
see which one is locked

Is it possible to see fxo port status from callmanager . I can check it
from gateway by  executing  ' show voice port summary'   but I dont know
how to see from CCM webpage,i wanna see if the port is really used at
that moment. 

 

 

 

2.Also another thing ;  when i call GSM  from fct devices , sometimes ip
phone displays that it is connected and the connected time is increasing
but actually the call is not set up. I wait 50-55 seconds but there is
no even ring tone. 

And then i  close the phone adn call again.

 

 

thanks  a lot

 

________________________________

Take the Internet to Go: Yahoo!Go puts the Internet in your pocket:
<http://us.rd.yahoo.com/evt=48253/*http:/mobile.yahoo.com/go?refer=1GNXI
C>  mail, news, photos & more. 

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From c3voip at nc.rr.com  Thu Aug  9 13:20:07 2007
From: c3voip at nc.rr.com (c3voip)
Date: Thu, 9 Aug 2007 13:20:07 -0400
Subject: [cisco-voip] Endpoint Gatekeeper and RASAggregator Trunk
Message-ID: <000001c7daa9$8881e1d0$9985a570$@rr.com>

Does anyone have any experience with using endpoint gatekeepers with a
RASAggregator trunk to the CCM to do call admission?

 

I am trying to set this up in my lab according to the IP Video Telephony
SRND for CCM 4.1, but I am not having any luck.

 

I have everything setup as they describe in the SRND, but I keep getting RAS
reject messages stating "rejectReason resourceUnavailable : NULL"

 

I have a simple single cluster with one Device Pool, one CSS, one partition,
one 7985, one H.323 Client, one gatekeeper with one zone, just the default
tech prefix, and one H.323 endpoint.

 

The H.323 endpoint is registered to the gatekeeper and the CCM establishes
the RasAggregator trunk to the gatekeeper.  What am I missing?

 

Is there something missing from the SRND that I need to configure?

 

Thanks,

-C

 

 

 

 

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From c3voip at nc.rr.com  Thu Aug  9 13:39:27 2007
From: c3voip at nc.rr.com (c3voip)
Date: Thu, 9 Aug 2007 13:39:27 -0400
Subject: [cisco-voip] Endpoint Gatekeeper and RASAggregator Trunk
In-Reply-To: <000001c7daa9$8881e1d0$9985a570$@rr.com>
References: <000001c7daa9$8881e1d0$9985a570$@rr.com>
Message-ID: <000b01c7daac$3c1a38d0$b44eaa70$@rr.com>

Nevermind, I reloaded the gatekeeper, rebooted my H.323 endpoint, and reset
the gatekeeper from CCM and now it is working.

 

I should have tried that first.

 

Thanks,

-C

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of c3voip
Sent: Thursday, August 09, 2007 1:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Endpoint Gatekeeper and RASAggregator Trunk

 

Does anyone have any experience with using endpoint gatekeepers with a
RASAggregator trunk to the CCM to do call admission?

 

I am trying to set this up in my lab according to the IP Video Telephony
SRND for CCM 4.1, but I am not having any luck.

 

I have everything setup as they describe in the SRND, but I keep getting RAS
reject messages stating "rejectReason resourceUnavailable : NULL"

 

I have a simple single cluster with one Device Pool, one CSS, one partition,
one 7985, one H.323 Client, one gatekeeper with one zone, just the default
tech prefix, and one H.323 endpoint.

 

The H.323 endpoint is registered to the gatekeeper and the CCM establishes
the RasAggregator trunk to the gatekeeper.  What am I missing?

 

Is there something missing from the SRND that I need to configure?

 

Thanks,

-C

 

 

 

 

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From Scott.Crosby at apptis.com  Thu Aug  9 13:50:32 2007
From: Scott.Crosby at apptis.com (Scott Crosby)
Date: Thu, 9 Aug 2007 10:50:32 -0700
Subject: [cisco-voip] Endpoint Gatekeeper and RASAggregator Trunk
In-Reply-To: <000001c7daa9$8881e1d0$9985a570$@rr.com>
References: <000001c7daa9$8881e1d0$9985a570$@rr.com>
Message-ID: <1DD380DD69341B48A16068FAD379D4950297B74C@SE-EXCH-01.intelltech.apptis.inc>

There's one sneaky you are probably missing, in the SRND, there is a
small little sentence that talks about setting a CCM Service value
related to H.323. The option is called "Send Product ID and Version ID*"
and it needs to be set to "true". Also the extension and the E.164
address need the same and what's registered with the Gatekeeper,
otherwise CCM will refuse the call.

 

--Scott

 

--Scott

 

 

Scott Crosby

Advanced Technology Architect

 

CCNP, CCDA, CCVP, MCSE

Cisco Technical Expert - Cisco Unity

Twisted Pair Solutions, Inc. - WAVE Certified Engineer

 

Mobile: 253-227-4095

Black Berry: 253-332-3840

Desk: 253-867-1820

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of c3voip
Sent: Thursday, August 09, 2007 1:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Endpoint Gatekeeper and RASAggregator Trunk

 

Does anyone have any experience with using endpoint gatekeepers with a
RASAggregator trunk to the CCM to do call admission?

 

I am trying to set this up in my lab according to the IP Video Telephony
SRND for CCM 4.1, but I am not having any luck.

 

I have everything setup as they describe in the SRND, but I keep getting
RAS reject messages stating "rejectReason resourceUnavailable : NULL"

 

I have a simple single cluster with one Device Pool, one CSS, one
partition, one 7985, one H.323 Client, one gatekeeper with one zone,
just the default tech prefix, and one H.323 endpoint.

 

The H.323 endpoint is registered to the gatekeeper and the CCM
establishes the RasAggregator trunk to the gatekeeper.  What am I
missing?

 

Is there something missing from the SRND that I need to configure?

 

Thanks,

-C

 

 

 

 

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From c3voip at nc.rr.com  Thu Aug  9 14:04:28 2007
From: c3voip at nc.rr.com (c3voip)
Date: Thu, 9 Aug 2007 14:04:28 -0400
Subject: [cisco-voip] Endpoint Gatekeeper and RASAggregator Trunk
In-Reply-To: <1DD380DD69341B48A16068FAD379D4950297B74C@SE-EXCH-01.intelltech.apptis.inc>
References: <000001c7daa9$8881e1d0$9985a570$@rr.com>
	<1DD380DD69341B48A16068FAD379D4950297B74C@SE-EXCH-01.intelltech.apptis.inc>
Message-ID: <001901c7daaf$ba8fd9b0$2faf8d10$@rr.com>

That must have been it, because I noticed that the RasAggregator had
registered as type VOIP-GW at first.  I then noticed that line in the SRND
and changed it to true, but the type never changed even after resetting the
gatekeeper a couple times from CCM.  Finally after reloading the gatekeeper
the trunk registered as type H323-GW and now works!

 

Thanks Scott!

 

-C

 

From: Scott Crosby [mailto:Scott.Crosby at apptis.com] 
Sent: Thursday, August 09, 2007 1:51 PM
To: c3voip; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Endpoint Gatekeeper and RASAggregator Trunk

 

There's one sneaky you are probably missing, in the SRND, there is a small
little sentence that talks about setting a CCM Service value related to
H.323. The option is called "Send Product ID and Version ID*" and it needs
to be set to "true". Also the extension and the E.164 address need the same
and what's registered with the Gatekeeper, otherwise CCM will refuse the
call.

 

--Scott

 

--Scott

 

 

Scott Crosby

Advanced Technology Architect

 

CCNP, CCDA, CCVP, MCSE

Cisco Technical Expert - Cisco Unity

Twisted Pair Solutions, Inc. - WAVE Certified Engineer

 

Mobile: 253-227-4095

Black Berry: 253-332-3840

Desk: 253-867-1820

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of c3voip
Sent: Thursday, August 09, 2007 1:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Endpoint Gatekeeper and RASAggregator Trunk

 

Does anyone have any experience with using endpoint gatekeepers with a
RASAggregator trunk to the CCM to do call admission?

 

I am trying to set this up in my lab according to the IP Video Telephony
SRND for CCM 4.1, but I am not having any luck.

 

I have everything setup as they describe in the SRND, but I keep getting RAS
reject messages stating "rejectReason resourceUnavailable : NULL"

 

I have a simple single cluster with one Device Pool, one CSS, one partition,
one 7985, one H.323 Client, one gatekeeper with one zone, just the default
tech prefix, and one H.323 endpoint.

 

The H.323 endpoint is registered to the gatekeeper and the CCM establishes
the RasAggregator trunk to the gatekeeper.  What am I missing?

 

Is there something missing from the SRND that I need to configure?

 

Thanks,

-C

 

 

 

 

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From asobihoudai at yahoo.com  Thu Aug  9 16:11:53 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 9 Aug 2007 13:11:53 -0700 (PDT)
Subject: [cisco-voip] question about CCME label/description/name functions
In-Reply-To: <001901c7daaf$ba8fd9b0$2faf8d10$@rr.com>
Message-ID: <917663.51296.qm@web57211.mail.re3.yahoo.com>

My understanding is that the description is the text
displayed on the phone, the name is what shows up when
someone calls that ephone-dn and label I'm not sure
about. Can anyone please clarify this? I'm a bit
confused here since I don't have a lab to work with at
the moment.

Paul


      ____________________________________________________________________________________
Fussy? Opinionated? Impossible to please? Perfect.  Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 


From asobihoudai at yahoo.com  Thu Aug  9 17:32:25 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 9 Aug 2007 14:32:25 -0700 (PDT)
Subject: [cisco-voip] question about CCME label/description/name
	functions
In-Reply-To: <917663.51296.qm@web57211.mail.re3.yahoo.com>
Message-ID: <253286.49836.qm@web57204.mail.re3.yahoo.com>

nevermind, I found what I was looking for...

duh.


      ____________________________________________________________________________________
Luggage? GPS? Comic books? 
Check out fitting gifts for grads at Yahoo! Search
http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz

From gokhanciscottl at yahoo.com  Fri Aug 10 04:19:08 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Fri, 10 Aug 2007 01:19:08 -0700 (PDT)
Subject: [cisco-voip] ccm 4.1.3  Windows VISTA tsp.exe version
Message-ID: <836994.29314.qm@web39604.mail.mud.yahoo.com>

Hello

Do you know which TSP.exe version should i use for   CCM 4.1.3  Microsoft VISTA

thanks


       
____________________________________________________________________________________
Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games.
http://sims.yahoo.com/  
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From jason.aarons at us.didata.com  Fri Aug 10 08:44:53 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 10 Aug 2007 08:44:53 -0400
Subject: [cisco-voip] VMWI on analog phones with VG224
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED0110A3F1@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED0110A3F1@mail.ACT-EG.COM>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00667E9F0@USNAEXCH.na.didata.local>

What make/model phone or region/country uses 90 volt phones? I did a
google search and see some intercom/hospital phones. I had never heard
of 90 volt until your posting, something to watch out for! -jason

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Wednesday, August 08, 2007 7:42 AM
To: Kenneth Sinfield (AU); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VMWI on analog phones with VG224

 

Hello kenneth;

Thanks for your reply, the problem was that the VG224 only supports FSK
with 24 or 48 volt and the analog phone that I was testing was using 90
volt by default, this phone works well with the VG248 as it supports
this high voltage but not the VG224.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kenneth
Sinfield
Sent: Tuesday, August 07, 2007 12:37 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VMWI on analog phones with VG224

 

Hi Ahmed,

 

I have VMWI working on an ISR router with FXS ports (VMWI will work on a
VG224 as well). Basically below are the requirements:

 

*         The Cisco Gateway must run IOS version IP Voice T strain (not
SP Services on ISRs) 12.4(6)T or later (I tested 12.4(11)T2)

*         Cisco CallManager version must be version 4.2 or later (I
tested CallManager 5.1.2)

*         Any version of Unity (I tested Unity Connection 1.2)

*         Must use SCCP(Skinny) as the gateway protocol, and NOT MGCP

 

Here is an output from the gateway(MWI refers to stutter tone VMWI needs
to be "ON"):

 

pstn#show stcapp device voice-port 0/3/3

Port Identifier:  0/3/3

Device Type:      ALG 

Device Id:        4

Device Name:      AN17598075F0183

Modem Capability: None

Device State:     IS

Diagnostic:       None

Directory Number: 1005

Dial Peer(s):     101 

Dialtone after remote onhook feature: activated

Last Event:       STCAPP_CC_EV_CALL_DISCONNECT_DONE

Line State:       IDLE

Hook State:       ONHOOK

mwi:              ENABLE

vmwi:             ON

PLAR:             DISABLE

Number of CCBs:   0

Global call info:

    Total CCB count      = 0

    Total call leg count = 0

 

Thanks,

Ken.

 

Kenneth Sinfield

Systems Engineer

Dimension Data, Australia

139 Coronation Drive, Milton

Queensland, Australia, 4064

Phone: +61732921968

 

************************************************************************
******

- NOTICE FROM DIMENSION DATA AUSTRALIA

This message is confidential, and may contain proprietary or legally
privileged information. If you have received this email in error, please
notify the sender and delete it immediately.

 

Internet communications are not secure. You should scan this message and
any attachments for viruses. Under no circumstances do we accept
liability for any loss or damage which may result from your receipt of
this message or any attachments.

************************************************************************
******




-----------------------------------------
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From akoul at dataways.gr  Fri Aug 10 09:05:40 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri, 10 Aug 2007 16:05:40 +0300
Subject: [cisco-voip] Cisco UC500 for SmallBusiness and Configuration
	Assistant
Message-ID: <46BC62A4.4010303@dataways.gr>

Hi,
i have just gotten my first UC500 with 2BRI ports and i was curious to 
see if i could make it work using the configuration assistant!!
Well it sucked! I couldn't make it work for other compand type than 
u-law so i have to give the command from the cli. Whatever region i put 
in the tab "Device Parameters" of the Telephony/voice menu,  the voice 
ports get NO CP-tone or compand type other than US.
Also, it can't configure incoming calls to be routed to AutoAttendant. 
In fact, no translation patterns are configured on the voice ports.
Has anyone make it work without using CLI? :-)

And a quiz:
Why using the Configuration ASS(istant)? :-(


Cheers
Anthony

From senccie at gmail.com  Fri Aug 10 10:01:12 2007
From: senccie at gmail.com (senthil natarajan)
Date: Fri, 10 Aug 2007 10:01:12 -0400
Subject: [cisco-voip] extra caller id in CME call transfer
Message-ID: <be16fc90708100701i3a42ef85sbeedcc9a122621e7@mail.gmail.com>

Hi,

I have CME 3.3 on a Cisco 2600 gateway.

When a IP phone user transfers an external call to another IP phone user,
caller id gets garbled. System automatically appends alphanumeric charcters
, for example John smith5EFES2s45SDES

Current image on the router - c2600-spservicesk9-mz.124-16.bin
Affected phone endpoint - 7960 and 7940
Firmware - P00307020200

Anyone seen this?

Thanks
Senthil
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From jason.aarons at us.didata.com  Fri Aug 10 14:15:05 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 10 Aug 2007 14:15:05 -0400
Subject: [cisco-voip] Moving users from one Unity UM 4.2 server to another
Message-ID: <C1FE15183DA37645BC0633BC604E44F00667F01B@USNAEXCH.na.didata.local>

Customer wants to move subscribers in Windows Domain A to Windows Domain
B on a new Unity UM 4.2 server. These are separate AD forests and no
trusts.

 

 How can I view/export the subscribers Unity passwords when they press
the Messages button.  I figure I can export their Greetings using
DiRT/Bulk Edit tool and move them, but what can be done about other
settings? 

 

I'm thinking I'm better off just creating a new subscriber and let them
go thru enrollment and re-record their greeting.  Trying to "migrate"
them doesn't seem to be worth the headache.

 




-----------------------------------------
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 carneyconsulting at comcast.net  Fri Aug 10 19:47:59 2007
From: carneyconsulting at comcast.net (John Carney)
Date: Fri, 10 Aug 2007 19:47:59 -0400
Subject: [cisco-voip] problem with 3745 configuration and FXO disconnect
Message-ID: <000001c7dba8$e2234760$6d01a8c0@seyboldcarney1>

I'm having a really strange problem with my Cisco 3745.

 

It's a lab machine, setup at home, so using FXO ports.

 

It's a 3745, Vic2-2FXO card.

AIM-CUE installed

 

Config is below. Here is the problem:

 

When a call comes from the outside, caller hears the phone ringing.

Watching the phones inside, the phone rings 1 - 2 times, then stops

Caller continues to hear ringing

Phone inside starts ringing again, 

Phone can be answered either time as it shows an incoming call

 

First time the caller-id shows the caller's phone number

Second time the caller-id shows the trunk from the ephone-dn

If it goes to voicemail, phone indicates 2 missed calls

 

I've tried a lot of different configs, but can't figure this one out. Anyone
have similar problems or have suggestions?

 

Thanks.

jc

 

3745-CME#sh config

Using 8209 out of 155640 bytes

!

! Last configuration change at 01:26:51 UTC Wed Mar 6 2002 by 

! NVRAM config last updated at 01:26:54 UTC Wed Mar 6 2002 by 

!

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname 3745-CME

!

boot-start-marker

boot system flash slot0:c3745-ipvoice-mz.124-9.T4.bin

boot-end-marker

!

logging buffered 51200 warnings

 

no aaa new-model

!

resource policy

!

no network-clock-participate slot 1

ip cef

!

!

no ip domain lookup

ip domain name yourdomain.com

ip name-server 68.87.75.194

ip name-server 68.82.0.5

ip name-server 68.82.0.6

voice-card 1

 dspfarm

 dsp services dspfarm

!

username cisco password 0 cisco

!

interface FastEthernet0/0

 no ip address

 duplex auto

 speed auto

!

interface FastEthernet0/0.100

 description data vlan trunk to 3550

 encapsulation dot1Q 100

 ip address 192.168.1.11 255.255.255.0

!

interface FastEthernet0/0.200

 description voice vlan trunk to 3550

 encapsulation dot1Q 200

 ip address 192.168.2.11 255.255.255.0

!

interface FastEthernet0/0.300

 description internet vlan trunk to 3550

 encapsulation dot1Q 300

 ip address 192.168.3.11 255.255.255.0

!

interface Service-Engine0/0

 ip unnumbered FastEthernet0/0.200

 service-module ip address 192.168.2.99 255.255.255.0

 service-module ip default-gateway 192.168.2.11

!

interface FastEthernet0/1

 no ip address

 shutdown

 duplex auto

 speed auto

!

ip route 0.0.0.0 0.0.0.0 192.168.2.25

ip route 192.168.2.99 255.255.255.255 Service-Engine0/0

!

ip http server

ip http authentication local

ip http timeout-policy idle 5 life 86400 requests 10000

ip http path flash:

!

control-plane

!

!

voice-port 1/0/0

 input gain 9

 output attenuation -3

 timeouts call-disconnect 3

 connection plar opx 6109422241

 caller-id enable

!

voice-port 1/0/1

!

!

!

sccp local FastEthernet0/0.200

sccp ccm 192.168.2.11 identifier 1

sccp

!

sccp ccm group 1

 associate ccm 1 priority 1

 associate profile 1 register MTP0011bbcf4770

 associate profile 2 register CFB0011bbcf4770

!

dspfarm profile 1 transcode

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 maximum sessions 3

 associate application SCCP

!

dspfarm profile 2 conference

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 codec g729br8

 maximum sessions 2

 associate application SCCP

!

!

dial-peer voice 2 pots

 destination-pattern 2241

 incoming called-number .

 port 1/0/0

 forward-digits 0

!

dial-peer voice 16 voip

 destination-pattern 400.

 session protocol sipv2

 session target ipv4:192.168.2.99

 incoming called-number .

 dtmf-relay sip-notify

 codec g711ulaw

!

dial-peer voice 3 pots

 destination-pattern 911

 port 1/0/0

 forward-digits 3

!

dial-peer voice 5 pots

 destination-pattern 610.......

 port 1/0/0

 forward-digits 10

!

dial-peer voice 6 pots

 destination-pattern 1[2-9].........

 port 1/0/0

 forward-digits 11

!

dial-peer voice 7 pots

 destination-pattern [2-8]11

 port 1/0/0

 forward-digits 3

!

dial-peer voice 8 pots

 destination-pattern 0T

 port 1/0/0

 forward-digits all

!

!

!

telephony-service

 fxo hook-flash

 load 7920 cmterm_7920.4.0-03-01

 load 7970 term70.7-0-3-0s

 max-ephones 24

 max-dn 48

 ip source-address 192.168.2.11 port 2000

 auto assign 1 to 12

  sdspfarm units 2

 sdspfarm transcode sessions 3

 sdspfarm tag 1 MTP0011bbcf4770

 sdspfarm tag 2 CFB0011bbcf4770

 cnf-file perphone

 time-zone 12

 voicemail 4001

 mwi relay

 max-conferences 8 gain -6

 call-park system redirect

 moh music-on-hold.au

 dn-webedit

 time-webedit

 transfer-system full-consult

 create cnf-files version-stamp 7960 Mar 05 2002 07:44:44

!

!

ephone-template  1

 softkeys hold  Resume

 softkeys idle  Newcall

 softkeys seized  Endcall Pickup Redial

 softkeys alerting  Callback Endcall

 softkeys connected  Endcall Flash Park Hold

!

!

ephone-template  2

 softkeys hold  Resume

 softkeys idle  Newcall

 softkeys seized  Endcall Pickup

 softkeys connected  Flash Park

!

!

ephone-dn  1  dual-line

 number 201

!

!

ephone-dn  2  dual-line

 number 202

!

!

ephone-dn  3  dual-line

 number 203

!

ephone-dn  4  dual-line

 number 204

!

!

ephone-dn  5  dual-line

 number 205

!

!

ephone-dn  6  dual-line

 number 206

!

!

ephone-dn  8  dual-line

 number 208

!

!

ephone-dn  9  dual-line

 number 209

!

!

ephone-dn  10  dual-line

 number 210

!

!

ephone-dn  11  dual-line

 number 211

!

!

ephone-dn  15

 number 999

 name Paging

 paging ip 239.1.1.1 port 2000

!

!

ephone-dn  24  dual-line

 number 99

 park-slot timeout 60 limit 3 recall

!

!

ephone-dn  32

 number 800..........

 mwi on

!

!

ephone-dn  33

 number 801..........

 mwi off

!

!

ephone-dn  46

 number 6109422241

 label 2241

 call-forward busy 4001

 call-forward noan 4001 timeout 10

 hold-alert 90 shared

 trunk 2241

!

!

ephone  1

 no multicast-moh

 mac-address 001B.D433.1917

 paging-dn 15

 type 7970

 button  1:46 2:1

!

!

!

ephone  2

 no multicast-moh

 mac-address 001B.D432.CBF7

 paging-dn 15

 type 7970

 button  1:46 2:2

!

!

!

ephone  3

 no multicast-moh

 mac-address 001B.D432.D301

 paging-dn 15

 type 7970

 button  1:3

!

!

!

ephone  4

 no multicast-moh

 mac-address 0019.0633.99E0

 paging-dn 15

 type 7920

 button  1:46 2:4

!

!

!

ephone  5

 no multicast-moh

 mac-address 001B.D433.1A4C

 paging-dn 15

 type 7970

 button  1:5

!

!

!

ephone  6

 no multicast-moh

 mac-address 0016.C852.7B4C

 paging-dn 15

 type 7970

 button  1:46 2:6

!

!

!

ephone  8

 no multicast-moh

 mac-address 0019.E78F.D28A

 ephone-template 1

 paging-dn 15

 type 7970

 button  1:46 2:8

!

!

!

ephone  9

 no multicast-moh

 mac-address 000D.288E.6485

 ephone-template 2

 paging-dn 15

 type 7920

 button  1:46 2:9

!

!

!

ephone  10

 no multicast-moh

 mac-address 001B.D433.1945

 ephone-template 1

 speed-dial 3 6109422241#

 paging-dn 15

 type 7970

 button  1:46 2:10

!

!

!

ephone  11

 no multicast-moh

!

!

!

ephone  12

 no multicast-moh

!

!

!

ephone  14

!

!

!

line con 0

 login local

line 26

 no activation-character

 no exec

 transport preferred none

 transport input all

 transport output pad telnet rlogin lapb-ta mop udptn v120

line aux 0

line vty 0 4

 privilege level 15

 password cisco

 login local

 transport input telnet

line vty 5 15

 privilege level 15

 login local

 transport input telnet

!

ntp clock-period 17180031

ntp source FastEthernet0/0.200

ntp master 3

ntp server 132.163.4.103

ntp server 67.128.71.75

!

end

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From aelnagar at ACT-EG.COM  Sat Aug 11 00:06:19 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Sat, 11 Aug 2007 07:06:19 +0300
Subject: [cisco-voip] VMWI on analog phones with VG224
References: <88C8D019DA547C4BABECB9A806ED22ED0110A3F1@mail.ACT-EG.COM>
	<C1FE15183DA37645BC0633BC604E44F00667E9F0@USNAEXCH.na.didata.local>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A113@mail.ACT-EG.COM>

hello jason;
 
The phone is working normally for all of the normal options (off hook, on hook, dialing, answering calls,...etc) the problem is with the VMWI signal that is maybe sent 90 volt (or even in some models higher) it is an FSK signal with 90 v that signals the analog phone to light the lamp whenever the user has a voice mail. in VG248 it supports the 90 v signal but for the VG224 it doesnot, so it is not affecting the over all operation of the phone just the VMWI is the affected feature.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Engineer Specialist
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Fri 10-Aug-07 3:44 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VMWI on analog phones with VG224



What make/model phone or region/country uses 90 volt phones? I did a google search and see some intercom/hospital phones. I had never heard of 90 volt until your posting, something to watch out for! -jason

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Wednesday, August 08, 2007 7:42 AM
To: Kenneth Sinfield (AU); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VMWI on analog phones with VG224

 

Hello kenneth;

Thanks for your reply, the problem was that the VG224 only supports FSK with 24 or 48 volt and the analog phone that I was testing was using 90 volt by default, this phone works well with the VG248 as it supports this high voltage but not the VG224.

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kenneth Sinfield
Sent: Tuesday, August 07, 2007 12:37 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VMWI on analog phones with VG224

 

Hi Ahmed,

 

I have VMWI working on an ISR router with FXS ports (VMWI will work on a VG224 as well). Basically below are the requirements:

 

?         The Cisco Gateway must run IOS version IP Voice T strain (not SP Services on ISRs) 12.4(6)T or later (I tested 12.4(11)T2)

?         Cisco CallManager version must be version 4.2 or later (I tested CallManager 5.1.2)

?         Any version of Unity (I tested Unity Connection 1.2)

?         Must use SCCP(Skinny) as the gateway protocol, and NOT MGCP

 

Here is an output from the gateway(MWI refers to stutter tone VMWI needs to be "ON"):

 

pstn#show stcapp device voice-port 0/3/3

Port Identifier:  0/3/3

Device Type:      ALG 

Device Id:        4

Device Name:      AN17598075F0183

Modem Capability: None

Device State:     IS

Diagnostic:       None

Directory Number: 1005

Dial Peer(s):     101 

Dialtone after remote onhook feature: activated

Last Event:       STCAPP_CC_EV_CALL_DISCONNECT_DONE

Line State:       IDLE

Hook State:       ONHOOK

mwi:              ENABLE

vmwi:             ON

PLAR:             DISABLE

Number of CCBs:   0

Global call info:

    Total CCB count      = 0

    Total call leg count = 0

 

Thanks,

Ken.

 

Kenneth Sinfield

Systems Engineer

Dimension Data, Australia

139 Coronation Drive, Milton

Queensland, Australia, 4064

Phone: +61732921968

 

******************************************************************************

- NOTICE FROM DIMENSION DATA AUSTRALIA

This message is confidential, and may contain proprietary or legally privileged information. If you have received this email in error, please notify the sender and delete it immediately.

 

Internet communications are not secure. You should scan this message and any attachments for viruses. Under no circumstances do we accept liability for any loss or damage which may result from your receipt of this message or any attachments.

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From Nick.Kassel at Charles-Stanley.co.uk  Mon Aug 13 02:53:32 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 13 Aug 2007 07:53:32 +0100
Subject: [cisco-voip] Recieved Calls List Digits Lose Prefix
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01452095@CSEXCH01.charles-stanley.co.uk>

Hi Mika

I can't remember the exactly but I think it may have been an issue the
IOS version, what IOS are you using?


-----Original Message-----
From: mika at loukola.com [mailto:mika at loukola.com] 
Sent: 12 August 2007 22:44
To: Nick Kassel
Subject: Re: Recieved Calls List Digits Lose Prefix

Hi Nick,

I saw the email thread regarding the issue with gateway translation-rule
is being dropped once the call is answered.
  http://www.threebit.net/mail-archive/cisco-voip/msg04775.html

I have the same program with my CallManager setup.

Did you find a solution to this proglem?
(I didn't see the solution as part of the thread)

Kind regards,

Mika Loukola


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From gokhanciscottl at yahoo.com  Mon Aug 13 04:18:37 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Mon, 13 Aug 2007 01:18:37 -0700 (PDT)
Subject: [cisco-voip] fxo discoonect issu on mgcp gateway
Message-ID: <859711.98417.qm@web39605.mail.mud.yahoo.com>

helo

my config on mgcp gateway is below.  FCT devices are connected to these fxo ports
Mostly i face to discconnect issue. even i on-hook the phone the port is still looks up.  Gateway does not close the port.

This mostly happens on voice port 0/1/2 and 0/1/3 but not on port 0/3/0

the difference between ports are below,

i route the incoming calls into the FCT  to the IVR(attendant DN)  and for that  i need      no battery-reversal   command  on voice-port 0/1/2 and 0/1/3 (they are ericsson  fct)
If i dont use no battery-reversal   command then the calls can not be routed to the IVR.

But for routing incoming calls into voice-port 0/3/0  i dont neet this command(it is different vendor FCT)


So what can i do to handle this issue. WHich command i can add on voice-port 0/1/2 and 0/1/3 so the mgcp gateway  can close the port after call is finished.



voice-port 0/1/2
 no battery-reversal
 timing hookflash-out 500
 description 9203441
!
voice-port 0/1/3
 no battery-reversal
 timing hookflash-out 500
 description 9203442
!
voice-port 0/3/0
 timing hookflash-out 500
 description avea 0554 704 66 64


      ____________________________________________________________________________________
Fussy? Opinionated? Impossible to please? Perfect.  Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 
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From voip at mrga.ch  Mon Aug 13 05:37:34 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Mon, 13 Aug 2007 11:37:34 +0200
Subject: [cisco-voip] Phone 7960 / 7961
Message-ID: <1186997854.46c0265ec59da@www.mail2web.ch>

Hi all

we run a CM 4.2(3) with IP Phones 7960.
We now ordered new phones 7961. If I have to replace an old 7960 phone is it
possible to upgrade the phone to the new model (7961) or do I have to delete
the old config and set up a new phone with model type 7961.

Thanks for help
Reto




From pwalenta at wi.rr.com  Mon Aug 13 07:41:40 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Mon, 13 Aug 2007 06:41:40 -0500
Subject: [cisco-voip] Phone 7960 / 7961
In-Reply-To: <1186997854.46c0265ec59da@www.mail2web.ch>
Message-ID: <000601c7dd9e$eab28dd0$0232a8c0@pwnhome.com>

You need to delete the old phone.  There's no way to change the device type
once it is in the system.

The good news is, the line won't get deleted, so once you add it to the new
phone, all its settings will come back. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Reto Gassmann
Sent: Monday, August 13, 2007 4:38 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Phone 7960 / 7961

Hi all

we run a CM 4.2(3) with IP Phones 7960.
We now ordered new phones 7961. If I have to replace an old 7960 phone is it
possible to upgrade the phone to the new model (7961) or do I have to delete
the old config and set up a new phone with model type 7961.

Thanks for help
Reto



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


From jonvoip at gmail.com  Mon Aug 13 09:13:59 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 13 Aug 2007 08:13:59 -0500
Subject: [cisco-voip] Unity 5 Schema Additions
Message-ID: <5d093f9a0708130613i3789cd80rbb74c05ede71b430@mail.gmail.com>

We are about to install a Unity 5 cluster for a customer. That
customer wants a detailed list of the Schema updates for Unity 5 and
why each are needed (IOW, what changes are made and why).

I can't find this on Cisco's web site for Unity 5, does such a document exist?



Jonathan

From jason.aarons at us.didata.com  Mon Aug 13 09:44:55 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 13 Aug 2007 09:44:55 -0400
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact Center
	Express 5.0(1) (UCCX)
Message-ID: <C1FE15183DA37645BC0633BC604E44F00667F7A3@USNAEXCH.na.didata.local>

This server MCS-7845-H2-CCX has 4 hard drives, is it two Raid 1 mirrors,
or a single Raid 5 array?

 

The customer is  providing their own hardware and setup a single Raid 5
array.

 

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_shee
t0900aecd80587390.html

 

Jason Aarons

Consultant

 

Dimension Data North America

E-mail ::jason.aarons at us.didata.com

Mobile :: 904.338.3245 e-fax :: 904.339.9696

                      

http://www.dimensiondata.com <http://www.us.didata.com> 

Plan.Build.Support & Manage

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 




-----------------------------------------
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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From rduric at uoguelph.ca  Mon Aug 13 09:50:13 2007
From: rduric at uoguelph.ca (Rasim Duric)
Date: Mon, 13 Aug 2007 09:50:13 -0400
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
	CenterExpress 5.0(1) (UCCX)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00667F7A3@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00667F7A3@USNAEXCH.na.didata.local>
Message-ID: <BD89624EE83BD349BD31A08B453DE24401B03858@uogexb201.cfs.uoguelph.ca>

Two RAID 1 mirrors.

 

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 Jason Aarons
(US)
Sent: Monday, August 13, 2007 9:45 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
CenterExpress 5.0(1) (UCCX)

 

This server MCS-7845-H2-CCX has 4 hard drives, is it two Raid 1 mirrors,
or a single Raid 5 array?

 

The customer is  providing their own hardware and setup a single Raid 5
array.

 

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_shee
t0900aecd80587390.html

 

Jason Aarons

Consultant

 

Dimension Data North America

E-mail ::jason.aarons at us.didata.com

Mobile :: 904.338.3245 e-fax :: 904.339.9696

                      

http://www.dimensiondata.com <http://www.us.didata.com> 

Plan.Build.Support & Manage

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 

________________________________

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 jason.aarons at us.didata.com  Mon Aug 13 09:56:32 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 13 Aug 2007 09:56:32 -0400
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
	CenterExpress 5.0(1) (UCCX)
In-Reply-To: <BD89624EE83BD349BD31A08B453DE24401B03858@uogexb201.cfs.uoguelph.ca>
References: <C1FE15183DA37645BC0633BC604E44F00667F7A3@USNAEXCH.na.didata.local>
	<BD89624EE83BD349BD31A08B453DE24401B03858@uogexb201.cfs.uoguelph.ca>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00667F7EF@USNAEXCH.na.didata.local>

Is the UCCX 5.0(1) Cisco-provided Windows 2003 server just a boxed
retail copy of 2003 that looks like the retail edition or is it a
2003.4.4a eg Spherion build?

 

The customer wants to provide their own 2003 server media as they have a
Microsoft site license for 2003 server, and didn't order it from Cisco.

 

From: Rasim Duric [mailto:rduric at uoguelph.ca] 
Sent: Monday, August 13, 2007 9:50 AM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
CenterExpress 5.0(1) (UCCX)

 

Two RAID 1 mirrors.

 

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 Jason Aarons
(US)
Sent: Monday, August 13, 2007 9:45 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
CenterExpress 5.0(1) (UCCX)

 

This server MCS-7845-H2-CCX has 4 hard drives, is it two Raid 1 mirrors,
or a single Raid 5 array?

 

The customer is  providing their own hardware and setup a single Raid 5
array.

 

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_shee
t0900aecd80587390.html

 

Jason Aarons

Consultant

 

Dimension Data North America

E-mail ::jason.aarons at us.didata.com

Mobile :: 904.338.3245 e-fax :: 904.339.9696

                      

http://www.dimensiondata.com <http://www.us.didata.com> 

Plan.Build.Support & Manage

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 

________________________________

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
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 Richard.vanBerkel at telindus.nl  Mon Aug 13 10:05:18 2007
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Mon, 13 Aug 2007 16:05:18 +0200
Subject: [cisco-voip] Unity 5 Schema Additions
References: <5d093f9a0708130613i3789cd80rbb74c05ede71b430@mail.gmail.com>
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D886014F2A81@tnl1mx1.telindus.intra>

Hi Johnathan,

The schema attributes (Avdirmonex2k.ldf )are in this location
http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/release/notes/501
curelnotes.html




Groeten Richard,

Mailto:Richard.vanBerkel at Telindus.nl

  



 

 

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: maandag 13 augustus 2007 15:14
To: cisco voip list
Subject: [cisco-voip] Unity 5 Schema Additions
Importance: High

We are about to install a Unity 5 cluster for a customer. That
customer wants a detailed list of the Schema updates for Unity 5 and
why each are needed (IOW, what changes are made and why).

I can't find this on Cisco's web site for Unity 5, does such a document
exist?



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

From rduric at uoguelph.ca  Mon Aug 13 10:41:37 2007
From: rduric at uoguelph.ca (Rasim Duric)
Date: Mon, 13 Aug 2007 10:41:37 -0400
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
	CenterExpress 5.0(1) (UCCX)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00667F7EF@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00667F7A3@USNAEXCH.na.didata.local>
	<BD89624EE83BD349BD31A08B453DE24401B03858@uogexb201.cfs.uoguelph.ca>
	<C1FE15183DA37645BC0633BC604E44F00667F7EF@USNAEXCH.na.didata.local>
Message-ID: <BD89624EE83BD349BD31A08B453DE24401B0386B@uogexb201.cfs.uoguelph.ca>

It should be a prebuilt Cisco Version 2003.1.x OS.

 

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

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com] 
Sent: Monday, August 13, 2007 9:57 AM
To: Rasim Duric; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
CenterExpress 5.0(1) (UCCX)

 

Is the UCCX 5.0(1) Cisco-provided Windows 2003 server just a boxed
retail copy of 2003 that looks like the retail edition or is it a
2003.4.4a eg Spherion build?

 

The customer wants to provide their own 2003 server media as they have a
Microsoft site license for 2003 server, and didn't order it from Cisco.

 

From: Rasim Duric [mailto:rduric at uoguelph.ca] 
Sent: Monday, August 13, 2007 9:50 AM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
CenterExpress 5.0(1) (UCCX)

 

Two RAID 1 mirrors.

 

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 Jason Aarons
(US)
Sent: Monday, August 13, 2007 9:45 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
CenterExpress 5.0(1) (UCCX)

 

This server MCS-7845-H2-CCX has 4 hard drives, is it two Raid 1 mirrors,
or a single Raid 5 array?

 

The customer is  providing their own hardware and setup a single Raid 5
array.

 

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_shee
t0900aecd80587390.html

 

Jason Aarons

Consultant

 

Dimension Data North America

E-mail ::jason.aarons at us.didata.com

Mobile :: 904.338.3245 e-fax :: 904.339.9696

                      

http://www.dimensiondata.com <http://www.us.didata.com> 

Plan.Build.Support & Manage

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 

________________________________

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
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 is
strictly prohibited and may be unlawful. If you have received this
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From bob at smalltime.com  Mon Aug 13 10:55:18 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Mon, 13 Aug 2007 09:55:18 -0500
Subject: [cisco-voip] Networking Unity Express and Unity using VPIM
Message-ID: <46C070D6.2010306@smalltime.com>

If Unity 4.2(1) is running g.729 and UE 3.0.1 is g.711 only, then what 
happens when a message is forwarded from Unity to UE using VPIM?  Will 
the UE subscriber still be able to play the message using the TUI? 
Howabout VoiceView Express?

What about in the other direction?  UE -> Unity?

From svoll.voip at gmail.com  Mon Aug 13 11:45:41 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 13 Aug 2007 08:45:41 -0700
Subject: [cisco-voip] ipcc demo
Message-ID: <f84a38d30708130845q65c5cbbdta8ea4017f740e542@mail.gmail.com>

I'm hoping this is a quick question.

with CM you get some little demo type stuff.

with CM 5.x how does the IPCC work.  with CM 4.x it was co-resident but I'm
clueless on 5.x.

TIA

Scott
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From svoll.voip at gmail.com  Mon Aug 13 12:03:19 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 13 Aug 2007 09:03:19 -0700
Subject: [cisco-voip] CER w/ CM 5.1
Message-ID: <f84a38d30708130903l5d61a243p1ee3a6f9d63966fc@mail.gmail.com>

What version of CER do you have to have to work with CM 5.1?

Thanks

Scott
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From Matt.Slaga at us.didata.com  Mon Aug 13 12:16:28 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 13 Aug 2007 12:16:28 -0400
Subject: [cisco-voip] ipcc demo
In-Reply-To: <f84a38d30708130845q65c5cbbdta8ea4017f740e542@mail.gmail.com>
References: <f84a38d30708130845q65c5cbbdta8ea4017f740e542@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0066D87F9@USNAEXCH.na.didata.local>

I don't think you can run it co-resident, as IPCC 5.x runs win2003 and
Cm 5.x runs linux.  That was be an interesting combination....

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Monday, August 13, 2007 11:46 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ipcc demo

 

I'm hoping this is a quick question.

 

with CM you get some little demo type stuff.

 

with CM 5.x how does the IPCC work.  with CM 4.x it was co-resident but
I'm clueless on 5.x.  

 

TIA

 

Scott




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From wsisk at cisco.com  Mon Aug 13 12:16:51 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 13 Aug 2007 12:16:51 -0400
Subject: [cisco-voip] CER w/ CM 5.1
In-Reply-To: <f84a38d30708130903l5d61a243p1ee3a6f9d63966fc@mail.gmail.com>
References: <f84a38d30708130903l5d61a243p1ee3a6f9d63966fc@mail.gmail.com>
Message-ID: <46C083F3.9050509@cisco.com>

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From toddnh65 at gmail.com  Mon Aug 13 14:07:50 2007
From: toddnh65 at gmail.com (Todd Franklin)
Date: Mon, 13 Aug 2007 13:07:50 -0500
Subject: [cisco-voip] Recording Calls
Message-ID: <6101bda50708131107r72bd277fuad197301519c79f3@mail.gmail.com>

I have a need to record a few lines, at all times.
Has anyone had good experiences with any sort of recorder for the Cisco IP
phones (spec. 7940s and 7960s).
These are not POTS lines, so I can't place the recorder in front of the
gateway....the recorder would need to "sniff" the network for VOIP packets
and assemble them into recordings.

Thanks.

PS- I have been looking into CallReplay (call-replay.com) but am not sure
what else is out there.
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From deadpool.slayer.telecomm at gmail.com  Mon Aug 13 14:11:03 2007
From: deadpool.slayer.telecomm at gmail.com (Andrew F. Wolfe)
Date: Mon, 13 Aug 2007 13:11:03 -0500
Subject: [cisco-voip] CUCM 5.X, ODBC, and CDR
Message-ID: <913929690708131111w7e27e0d7k91d1f33b47b0a7a5@mail.gmail.com>

Hello all.  I am trying to wrap my brain around the DB for CDR and CUCM.
Since it is an Informix DB, I can't use the ODBC drivers included in
Windows.  Any suggestions on an Informix ODBC driver and any tips would be
greatly appreciated.  Please let me know.  Thanks.
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From Matt.Slaga at us.didata.com  Mon Aug 13 14:34:59 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 13 Aug 2007 14:34:59 -0400
Subject: [cisco-voip] CUCM 5.X, ODBC, and CDR
In-Reply-To: <913929690708131111w7e27e0d7k91d1f33b47b0a7a5@mail.gmail.com>
References: <913929690708131111w7e27e0d7k91d1f33b47b0a7a5@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0066D8AFB@USNAEXCH.na.didata.local>

No ODBC drivers for CallManager 5.x and up, CDR is all done via FTP flat
file push/pulls.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew F. Wolfe
Sent: Monday, August 13, 2007 2:11 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CUCM 5.X, ODBC, and CDR

 

Hello all.  I am trying to wrap my brain around the DB for CDR and CUCM.
Since it is an Informix DB, I can't use the ODBC drivers included in
Windows.  Any suggestions on an Informix ODBC driver and any tips would
be greatly appreciated.  Please let me know.  Thanks.  




-----------------------------------------
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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From pfabian at telesphere.com  Mon Aug 13 14:57:38 2007
From: pfabian at telesphere.com (Pete Fabian)
Date: Mon, 13 Aug 2007 11:57:38 -0700
Subject: [cisco-voip] Recording Calls
In-Reply-To: <6101bda50708131107r72bd277fuad197301519c79f3@mail.gmail.com>
Message-ID: <008401c7dddb$d195cbe0$34fea8c0@telesphereltd.com>

We have used the universal adapter from
http://www.usbcallrecord.com/universal_adapter.htm

 

It works ok, but you have to manually type in the phone number for each call

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Monday, August 13, 2007 11:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Recording Calls

 

I have a need to record a few lines, at all times.  
Has anyone had good experiences with any sort of recorder for the Cisco IP
phones (spec. 7940s and 7960s).
These are not POTS lines, so I can't place the recorder in front of the
gateway....the recorder would need to "sniff" the network for VOIP packets
and assemble them into recordings. 

Thanks.

PS- I have been looking into CallReplay (call-replay.com) but am not sure
what else is out there.



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From deadpool.slayer.telecomm at gmail.com  Mon Aug 13 15:07:21 2007
From: deadpool.slayer.telecomm at gmail.com (Andrew F. Wolfe)
Date: Mon, 13 Aug 2007 14:07:21 -0500
Subject: [cisco-voip] Recording Calls
In-Reply-To: <008401c7dddb$d195cbe0$34fea8c0@telesphereltd.com>
References: <6101bda50708131107r72bd277fuad197301519c79f3@mail.gmail.com>
	<008401c7dddb$d195cbe0$34fea8c0@telesphereltd.com>
Message-ID: <913929690708131207o71458226hae53933926c4770e@mail.gmail.com>

Telrex is another one that we have used.  Worked pretty well.

http://www.telrex.com/

If you are looking for a more high end system with reporting and such, check
out Witness.

http://www.witness.com

They have been in the call monitoring biz for a while.  They may be more
than what you are looking for  though.  Hope this helps.





On 8/13/07, Pete Fabian <pfabian at telesphere.com> wrote:
>
>  We have used the universal adapter from
> http://www.usbcallrecord.com/universal_adapter.htm
>
>
>
> It works ok, but you have to manually type in the phone number for each
> call
>
>
>
>
>   ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Todd Franklin
> *Sent:* Monday, August 13, 2007 11:08 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Recording Calls
>
>
>
> I have a need to record a few lines, at all times.
> Has anyone had good experiences with any sort of recorder for the Cisco IP
> phones (spec. 7940s and 7960s).
> These are not POTS lines, so I can't place the recorder in front of the
> gateway....the recorder would need to "sniff" the network for VOIP packets
> and assemble them into recordings.
>
> Thanks.
>
> PS- I have been looking into CallReplay (call-replay.com) but am not sure
> what else is out there.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From bcarter at sentinel.com  Mon Aug 13 15:41:19 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 13 Aug 2007 14:41:19 -0500
Subject: [cisco-voip] UCM 6.X Intercom
Message-ID: <6A0744639B86DD459433108DBF101D26018176AA@MAIL01.sentinel.com>

 
 
Can the UCM 6.X Intercom feature be used for paging to the phones?
Where 1 user could send a page to all IP Phones ?

From jason.aarons at us.didata.com  Mon Aug 13 15:51:18 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 13 Aug 2007 15:51:18 -0400
Subject: [cisco-voip] UCM 6.X Intercom
In-Reply-To: <6A0744639B86DD459433108DBF101D26018176AA@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26018176AA@MAIL01.sentinel.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0066D8D10@USNAEXCH.na.didata.local>

I understand it requires using a line on a phone, so if your 7941 has a
dn on both lines already you can't use Intercom unless you remove a
line. Can anyone vouch for this?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Monday, August 13, 2007 3:41 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] UCM 6.X Intercom

 
 
Can the UCM 6.X Intercom feature be used for paging to the phones?
Where 1 user could send a page to all IP Phones ?
_______________________________________________
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 bcarter at sentinel.com  Mon Aug 13 16:26:00 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 13 Aug 2007 15:26:00 -0500
Subject: [cisco-voip] UCM 6.X Intercom
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0066D8D10@USNAEXCH.na.didata.local>
References: <6A0744639B86DD459433108DBF101D26018176AA@MAIL01.sentinel.com>
	<C1FE15183DA37645BC0633BC604E44F0066D8D10@USNAEXCH.na.didata.local>
Message-ID: <6A0744639B86DD459433108DBF101D26018176BD@MAIL01.sentinel.com>

That is correct.  An Intercom line occupies 1 dedication button on a
phone. 

-----Original Message-----
From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com] 
Sent: Monday, August 13, 2007 2:51 PM
To: Carter, Bill; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] UCM 6.X Intercom

I understand it requires using a line on a phone, so if your 7941 has a
dn on both lines already you can't use Intercom unless you remove a
line. Can anyone vouch for this?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Monday, August 13, 2007 3:41 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] UCM 6.X Intercom

 
 
Can the UCM 6.X Intercom feature be used for paging to the phones?
Where 1 user could send a page to all IP Phones ?
_______________________________________________
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 josmon at rigozsaurus.com  Mon Aug 13 16:41:27 2007
From: josmon at rigozsaurus.com (John Osmon)
Date: Mon, 13 Aug 2007 14:41:27 -0600
Subject: [cisco-voip] Recording Calls
In-Reply-To: <6101bda50708131107r72bd277fuad197301519c79f3@mail.gmail.com>
References: <6101bda50708131107r72bd277fuad197301519c79f3@mail.gmail.com>
Message-ID: <20070813204127.GA25342@jeeves.rigozsaurus.com>

On Mon, Aug 13, 2007 at 01:07:50PM -0500, Todd Franklin wrote:
> I have a need to record a few lines, at all times.
> Has anyone had good experiences with any sort of recorder for the Cisco IP
> phones (spec. 7940s and 7960s).
> These are not POTS lines, so I can't place the recorder in front of the
> gateway....the recorder would need to "sniff" the network for VOIP packets
> and assemble them into recordings.

If you're a 'roll your own' type of shop:
  - run Asterisk and use it as your outbound/inbound gateway.  Let it
    open a second connection to the actual destination, and record
  - base something on pcapsipdump

>From a commercial standpoint, I've seen good results with:
  - http://www.telrex.com/
  - http://www.edigin.com/

From svoll.voip at gmail.com  Mon Aug 13 17:45:32 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 13 Aug 2007 14:45:32 -0700
Subject: [cisco-voip] Bulk import services -- CM 5.1
Message-ID: <f84a38d30708131445r44a79354v81389e0201d6b3bf@mail.gmail.com>

is there some kind of a bug with bulk import of phones where services don't
import when the templete has the services?

Scott
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From wsisk at cisco.com  Mon Aug 13 17:47:06 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 13 Aug 2007 17:47:06 -0400
Subject: [cisco-voip] Recording Calls
In-Reply-To: <6101bda50708131107r72bd277fuad197301519c79f3@mail.gmail.com>
References: <6101bda50708131107r72bd277fuad197301519c79f3@mail.gmail.com>
Message-ID: <46C0D15A.3020009@cisco.com>

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From wsisk at cisco.com  Mon Aug 13 17:47:30 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 13 Aug 2007 17:47:30 -0400
Subject: [cisco-voip] Bulk import services -- CM 5.1
In-Reply-To: <f84a38d30708131445r44a79354v81389e0201d6b3bf@mail.gmail.com>
References: <f84a38d30708131445r44a79354v81389e0201d6b3bf@mail.gmail.com>
Message-ID: <46C0D172.7050708@cisco.com>

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From egoppel at gmail.com  Mon Aug 13 17:50:37 2007
From: egoppel at gmail.com (Erik Goppel)
Date: Mon, 13 Aug 2007 23:50:37 +0200
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
	CenterExpress 5.0(1) (UCCX)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00667F7EF@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00667F7A3@USNAEXCH.na.didata.local>
	<BD89624EE83BD349BD31A08B453DE24401B03858@uogexb201.cfs.uoguelph.ca>
	<C1FE15183DA37645BC0633BC604E44F00667F7EF@USNAEXCH.na.didata.local>
Message-ID: <47fb51250708131450r69256501xf7f7c6342ac8cf22@mail.gmail.com>

Jason,

It is a cisco OS (yeah i know it is 2003 with some modifications)
If you try to install UCCX 5 on a normal 2003 server, it will fail, you will
have to emulate the spherion build.
But remember, your UCCX solution is only supported on the Cisco specified
OS, and that is the MCS OS.
Even if the customer had talked to bill gates themselves, and he came over
with a 2003 cd and key himself, cisco will not support it, as it is no
MCS-OS.
A good explanation to you customer is, that the OS needed for UCCX, is based
on win2k3 (about 99.99%), and some service modifications and addons have
been arranged in order to ensure the UCCX service can be delivered in a
supported manner, the cost for the os can be explained, by the 2003 part
that cisco has to pay to microsoft, the other part is the support (think SLA
and TAC support against Microsoft support) Cisco will provide on a Microsoft
OS, and the supported Cisco products that are running on it.

Greetings,

Erik Goppel
Dimension Data Netherlands




2007/8/13, Jason Aarons (US) <jason.aarons at us.didata.com>:
>
>  Is the UCCX 5.0(1) Cisco-provided Windows 2003 server just a boxed retail
> copy of 2003 that looks like the retail edition or is it a 2003.4.4a eg
> Spherion build?
>
>
>
> The customer wants to provide their own 2003 server media as they have a
> Microsoft site license for 2003 server, and didn't order it from Cisco.
>
>
>
> *From:* Rasim Duric [mailto:rduric at uoguelph.ca]
> *Sent:* Monday, August 13, 2007 9:50 AM
> *To:* Jason Aarons (US); cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
> CenterExpress 5.0(1) (UCCX)
>
>
>
> Two RAID 1 mirrors.
>
>
>
> 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 *Jason Aarons (US)
> *Sent:* Monday, August 13, 2007 9:45 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
> CenterExpress 5.0(1) (UCCX)
>
>
>
> This server MCS-7845-H2-CCX has 4 hard drives, is it two Raid 1 mirrors,
> or a single Raid 5 array?
>
>
>
> The customer is  providing their own hardware and setup a single Raid 5
> array.
>
>
>
>
> http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_sheet0900aecd80587390.html
>
>
>
> *Jason Aarons*
>
> Consultant
>
> [image: cid:image001.gif at 01C7407C.14EB3630]
>
> Dimension Data North America
>
> E-mail* *::jason.aarons at us.didata.com
>
> Mobile :: 904.338.3245 e-fax* *:: 904.339.9696
>
>
>
> http://www.dimensiondata.com <http://www.us.didata.com/>
>
> Plan.Build.Support & Manage
>
>
>
> For urgent issues notify your Project Manager, for 24x7 support contact
> the Dimension Data NOC at 800-974-6584.
>
>
>  ------------------------------
>
> *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
> 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 Darren.Smith2 at didata.com.au  Mon Aug 13 18:40:29 2007
From: Darren.Smith2 at didata.com.au (Darren Smith)
Date: Tue, 14 Aug 2007 08:40:29 +1000
Subject: [cisco-voip] UCM 6.X Intercom
Message-ID: <9BF888E063AF4D4590836EE9F01A61B0016EA7D0@AUNDDVSEXC01.au.didata.local>

The intercom feature can only work in a one to one manner, not one to
many.

If you require one to many functionality, i suggest looking at the
Berbee paging product.

 

 

Kind Regards

Darren Smith
Solutions Architect - Connectivity
Dimension Data
~~~~~~~~~~~~~~~~~~~~~~~~~
Level 1, 139 Coronation Drive,Milton
Brisbane, 4064
Direct: 07 3292 1999
Mobile: 0417 491055
Fax:  07 3292 1300
E-Mail: darren.smith2 at didata.com.au <mailto:darren.smith2 at didata.com.au>

For more information about Dimension Data, please go to
www.dimensiondata.com <http://www.dimensiondata.com/> 

 


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 - NOTICE FROM DIMENSION DATA AUSTRALIA
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From jason.aarons at us.didata.com  Mon Aug 13 20:49:05 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 13 Aug 2007 20:49:05 -0400
Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact
	CenterExpress 5.0(1) (UCCX)
References: <C1FE15183DA37645BC0633BC604E44F00667F7A3@USNAEXCH.na.didata.local><BD89624EE83BD349BD31A08B453DE24401B03858@uogexb201.cfs.uoguelph.ca><C1FE15183DA37645BC0633BC604E44F00667F7EF@USNAEXCH.na.didata.local>
	<47fb51250708131450r69256501xf7f7c6342ac8cf22@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E8C5@USNAEXCH.na.didata.local>

The issue may be CCX-7845H-50-HAS and it's availablity. Cisco status tool shows the ship date is November, probably due to new product hold limitations. Cisco approved UCCX 5.0(1) and gave the customer the media, except they forgot to provide/order the CCX-7845H-50-HAS. We'll see what the Cisco SE/AM can do in getting the cisco-provided 2003 server media.

________________________________

From: Erik Goppel [mailto:egoppel at gmail.com]
Sent: Mon 8/13/2007 5:50 PM
To: Jason Aarons (US)
Cc: Rasim Duric; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact CenterExpress 5.0(1) (UCCX)


Jason,
 
It is a cisco OS (yeah i know it is 2003 with some modifications)
If you try to install UCCX 5 on a normal 2003 server, it will fail, you will have to emulate the spherion build.
But remember, your UCCX solution is only supported on the Cisco specified OS, and that is the MCS OS.
Even if the customer had talked to bill gates themselves, and he came over with a 2003 cd and key himself, cisco will not support it, as it is no MCS-OS.
A good explanation to you customer is, that the OS needed for UCCX, is based on win2k3 (about 99.99%), and some service modifications and addons have been arranged in order to ensure the UCCX service can be delivered in a supported manner, the cost for the os can be explained, by the 2003 part that cisco has to pay to microsoft, the other part is the support (think SLA and TAC support against Microsoft support) Cisco will provide on a Microsoft OS, and the supported Cisco products that are running on it. 
 
Greetings,
 
Erik Goppel
Dimension Data Netherlands
 


 
2007/8/13, Jason Aarons (US) <jason.aarons at us.didata.com>: 

	Is the UCCX 5.0(1) Cisco-provided Windows 2003 server just a boxed retail copy of 2003 that looks like the retail edition or is it a 2003.4.4a eg Spherion build?

	 

	The customer wants to provide their own 2003 server media as they have a Microsoft site license for 2003 server, and didn't order it from Cisco.

	 

	From: Rasim Duric [mailto:rduric at uoguelph.ca ] 
	Sent: Monday, August 13, 2007 9:50 AM
	To: Jason Aarons (US); cisco-voip at puck.nether.net 
	Subject: RE: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact CenterExpress 5.0(1) (UCCX)

	 

	Two RAID 1 mirrors.

	 

	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 Jason Aarons (US)
	Sent: Monday, August 13, 2007 9:45 AM
	To: cisco-voip at puck.nether.net 
	Subject: [cisco-voip] Drive layout MCS-7845-H2-CCX1 and Contact CenterExpress 5.0(1) (UCCX)

	 

	This server MCS-7845-H2-CCX has 4 hard drives, is it two Raid 1 mirrors, or a single Raid 5 array?

	 

	The customer is  providing their own hardware and setup a single Raid 5 array.

	 

	http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_sheet0900aecd80587390.html 

	 

	Jason Aarons

	Consultant

	 cid:image001.gif at 01C7407C.14EB3630<https://webmail.us.didata.com/us/jason..aarons at us.didata.com/Drafts/image001.gif> 

	Dimension Data North America

	E-mail :: jason.aarons at us.didata.com

	Mobile :: 904.338.3245 e-fax :: 904.339.9696

	                      

	http://www.dimensiondata.com <http://www.us.didata.com/> 

	Plan.Build.Support & Manage

	 

	For urgent issues notify your Project Manager, for 24x7 support contact the Dimension Data NOC at 800-974-6584.

	 

	
________________________________


	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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	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 pat-cv at wcyv.com  Mon Aug 13 21:06:44 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Mon, 13 Aug 2007 21:06:44 -0400
Subject: [cisco-voip] Networking Unity Express and Unity using VPIM
In-Reply-To: <46C070D6.2010306@smalltime.com>
References: <46C070D6.2010306@smalltime.com>
Message-ID: <46C10024.8040604@wcyv.com>

I'm not sure exactly what happens if you send a 729 encoded message to 
CUE, but, you should be able to get around this by specifying the 
outgoing codec on the VPIM delivery location that you created in Unity.

Going the other way, Unity should be able to transcode the message on 
the fly without a problem. Hope this helps.

-------- Original Message  --------
From: Robert Kulagowski <bob at smalltime.com>
To: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Networking Unity Express and Unity using VPIM
Date: 8/13/2007 10:55 AM

> If Unity 4.2(1) is running g.729 and UE 3.0.1 is g.711 only, then what 
> happens when a message is forwarded from Unity to UE using VPIM?  Will 
> the UE subscriber still be able to play the message using the TUI? 
> Howabout VoiceView Express?
> 
> What about in the other direction?  UE -> Unity?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From nws at fredf.org  Mon Aug 13 23:30:17 2007
From: nws at fredf.org (Kent Roberts)
Date: Mon, 13 Aug 2007 21:30:17 -0600
Subject: [cisco-voip] Recording Calls
In-Reply-To: <mailman.10776.1187062022.1529.cisco-voip@puck.nether.net>
References: <mailman.10776.1187062022.1529.cisco-voip@puck.nether.net>
Message-ID: <3e6f56310708132030n59907552nb91441c7318d803e@mail.gmail.com>

>
> I been working with oreka, its nice and simple for plain recording.  I
> have it watching 2 ds3's on a 3.4gig xeon.  works well for free!   esp, if
> its a remote site that needs monitoring...
>
>
> http://oreka.sourceforge.net
>
>
>
> On 8/13/07, Todd Franklin <toddnh65 at gmail.com> wrote:
> >
> > I have a need to record a few lines, at all times.
> > Has anyone had good experiences with any sort of recorder for the Cisco
> > IP phones (spec. 7940s and 7960s).
> > These are not POTS lines, so I can't place the recorder in front of the
> > gateway....the recorder would need to "sniff" the network for VOIP packets
> > and assemble them into recordings.
> >
> > Thanks.
> >
> > PS- I have been looking into CallReplay (call-replay.com) but am not
> > sure what else is out there.
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
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From Nick.Kassel at Charles-Stanley.co.uk  Tue Aug 14 03:06:46 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 14 Aug 2007 08:06:46 +0100
Subject: [cisco-voip] Recieved Calls List Digits Lose Prefix
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC0145209A@CSEXCH01.charles-stanley.co.uk>

I think we were on 123-11 T3, I would recommend upgrading to 12.4 11 T2,
we have been using this with no problems at all.

-----Original Message-----
From: mika at loukola.com [mailto:mika at loukola.com] 
Sent: 13 August 2007 18:33
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: RE: Recieved Calls List Digits Lose Prefix

Hi,

I have a 2811 with c2800nm-spservicesk9-mz.123-11.T10.bin

Mika

> Hi Mika
>
> I can't remember the exactly but I think it may have been an issue the
> IOS version, what IOS are you using?
>
>
> -----Original Message-----
> From: mika at loukola.com [mailto:mika at loukola.com]
> Sent: 12 August 2007 22:44
> To: Nick Kassel
> Subject: Re: Recieved Calls List Digits Lose Prefix
>
> Hi Nick,
>
> I saw the email thread regarding the issue with gateway
translation-rule
> is being dropped once the call is answered.
>   http://www.threebit.net/mail-archive/cisco-voip/msg04775.html
>
> I have the same program with my CallManager setup.
>
> Did you find a solution to this proglem?
> (I didn't see the solution as part of the thread)
>
> Kind regards,
>
> Mika Loukola
>
>
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From Nick.Kassel at Charles-Stanley.co.uk  Tue Aug 14 03:17:49 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 14 Aug 2007 08:17:49 +0100
Subject: [cisco-voip] UCM 6.X Intercom
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC0145209B@CSEXCH01.charles-stanley.co.uk>

The intercom feature cannot be used for sending to more that one user it
is a point to point intercom.  

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: 13 August 2007 20:41
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] UCM 6.X Intercom

 
 
Can the UCM 6.X Intercom feature be used for paging to the phones?
Where 1 user could send a page to all IP Phones ?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
********************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
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You are advised that urgent, time sensitive and confidential communications should not be sent by e-mail. You 
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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  

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This footnote also confirms that this email message has been swept by McAfee VirusScan and SurfControl Email 
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From gokhanciscottl at yahoo.com  Tue Aug 14 03:41:17 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Tue, 14 Aug 2007 00:41:17 -0700 (PDT)
Subject: [cisco-voip] IPCC  license for the scenario
Message-ID: <673237.31703.qm@web39604.mail.mud.yahoo.com>

hello

on ccm 4.1.3  which kind of IPCC license comes .  standart or enhanced or premium?

the customer has an application which keeps the guest database and they want the calls will come to the agent and the database will search based on CallerID.  And for example CallerID is 555xxxx  and then from the database the guest who owns this number will be found and the data related to this guest will come to the screen
for this kind of application IPCC license  come with CCM is enough or should i upgrade it?

also what is the steps for enabling ipcc , i will register PAK coe and then install on CCM with IPCC cd, thats all?



thanks


       
____________________________________________________________________________________
Choose the right car based on your needs.  Check out Yahoo! Autos new Car Finder tool.
http://autos.yahoo.com/carfinder/
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From zohaibshabir at gmail.com  Tue Aug 14 06:13:27 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Tue, 14 Aug 2007 15:13:27 +0500
Subject: [cisco-voip] IPCC license for the scenario
In-Reply-To: <673237.31703.qm@web39604.mail.mud.yahoo.com>
References: <673237.31703.qm@web39604.mail.mud.yahoo.com>
Message-ID: <1ff12970708140313u30e9ea53h2e2395e063b9929a@mail.gmail.com>

Hello,
CCM 4.1.3 comes with standard version of IPCC express, the functionality you
require need an IPCC express enhanced version,you need to upgrade your
license to IPCC enhance version.

For registration just register IPCC express  serial number that will give
you the key which you will require in installation.


On 8/14/07, gokhan senol <gokhanciscottl at yahoo.com> wrote:
>
> hello
>
>
>
> on ccm 4.1.3  which kind of IPCC license comes .  standart or enhanced or
> premium?
>
>
>
> the customer has an application which keeps the guest database and they
> want the calls will come to the agent and the database will search based on
> CallerID.  And for example CallerID is 555xxxx  and then from the database
> the guest who owns this number will be found and the data related to this
> guest will come to the screen
>
> for this kind of application IPCC license  come with CCM is enough or
> should i upgrade it?
>
>
>
> also what is the steps for enabling ipcc , i will register PAK coe and
> then install on CCM with IPCC cd, thats all?
>
>
>
>
>
>
>
> thanks
>
>
>
>
>
>
>
>
>
> ------------------------------
> Choose the right car based on your needs. Check out Yahoo! Autos new Car
> Finder tool.<http://us.rd.yahoo.com/evt=48518/*http://autos.yahoo.com/carfinder/;_ylc=X3oDMTE3NWsyMDd2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDY2FyLWZpbmRlcg--+>
>
> _______________________________________________
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>



-- 
Regards,
Zohaib Shabir
Network Engineer(Voice)
DWP Group, TECH Division
Ph:+92-302-8232689
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From Johnny.Crothers at vanco.co.uk  Tue Aug 14 06:29:54 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Tue, 14 Aug 2007 11:29:54 +0100
Subject: [cisco-voip]  Alerting Name
Message-ID: <A56BD242744BF540B15D1672E0E3E875019E7189@LON-INT-MSGVS01.internal.vancocorp.com>

Hi,

 

We are using IPCC to provide queuing for receptionists, although they
are complaining about one thing.

 

When they are receiving an external call, they are unable to see what
the calling number is, all they see is the CTI port name.

 

Is there anyway of having the calling number displayed on the phone?

 

Cheers,

 

Johnny

 



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From aelnagar at ACT-EG.COM  Tue Aug 14 07:00:23 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 14 Aug 2007 14:00:23 +0300
Subject: [cisco-voip] Unity without Exchange
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED0115F3CE@mail.ACT-EG.COM>

Dear all;

 

I want to install unity server for a customer who has no Exchange for
domino server or any active directory and so on, can I install the unity
to run in this case withour installing Exchange server?? I will run it
for voice mail only not unified communication. Do I need to install
Exchange server on the unity box or what??; sorry I know it is an ABC
question but it is my first unity deployment.

 

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From ErasmuE4 at telkom.co.za  Tue Aug 14 07:07:37 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Tue, 14 Aug 2007 13:07:37 +0200
Subject: [cisco-voip] Unity without Exchange
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED0115F3CE@mail.ACT-EG.COM>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01984BB6@CNTRRA20-XCS01.telkom.co.za>


Yes - Unity is a bit of a mess as far as I am concerned when it comes
to the effort it takes just to get basic voicemail. In situations like
this you are better of installing Unity connection - does the same and
is a single application.

 

If you have to use unity - yes - you can install it as on its own
Exchange Island and in its own little domain not integrated with
anything else - But completely without exchange you can not as far as
I know because it uses some of the exchange elements. Not a major
problem - you don't really ahev to configure exchange full like you
would do in an exchage email deployment

 

I have a doc with summary steps explaining how to do it - will mail to
you - need to go and search for it first

 

 

Erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Tuesday, August 14, 2007 1:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity without Exchange

 

Dear all;

 

I want to install unity server for a customer who has no Exchange for
domino server or any active directory and so on, can I install the
unity to run in this case withour installing Exchange server?? I will
run it for voice mail only not unified communication. Do I need to
install Exchange server on the unity box or what??; sorry I know it is
an ABC question but it is my first unity deployment.

 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From pezhman_lali at yahoo.com  Tue Aug 14 06:36:57 2007
From: pezhman_lali at yahoo.com (Pezhman Lali)
Date: Tue, 14 Aug 2007 03:36:57 -0700 (PDT)
Subject: [cisco-voip] changing caller id
Message-ID: <367971.51208.qm@web52307.mail.re2.yahoo.com>

Hi dears
I have a cisco2600+nm-hdv+pri line(E1).
is any way to change callerid of outgoing calls to the
pstn.
best
Mani


       
____________________________________________________________________________________
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 zohaibshabir at gmail.com  Tue Aug 14 07:44:00 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Tue, 14 Aug 2007 16:44:00 +0500
Subject: [cisco-voip] changing caller id
In-Reply-To: <367971.51208.qm@web52307.mail.re2.yahoo.com>
References: <367971.51208.qm@web52307.mail.re2.yahoo.com>
Message-ID: <1ff12970708140444t3a5c4bke271b05fff43c83d@mail.gmail.com>

Yes
you can always change the callerid of calls if you have the PRI line for the
pstn. I dont know you config, but i am sure you if you share the config with
me, i will be able to help you out.

On 8/14/07, Pezhman Lali <pezhman_lali at yahoo.com> wrote:
>
> Hi dears
> I have a cisco2600+nm-hdv+pri line(E1).
> is any way to change callerid of outgoing calls to the
> pstn.
> best
> Mani
>
>
>
>
> ____________________________________________________________________________________
> 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
> _______________________________________________
> 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 aelnagar at ACT-EG.COM  Tue Aug 14 08:01:18 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 14 Aug 2007 15:01:18 +0300
Subject: [cisco-voip] Unity without Exchange
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED0115F4BF@mail.ACT-EG.COM>

One more question.

Does cisco supply the Exchange CDs with the Unity CDs ??

 

________________________________

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za] 
Sent: Tuesday, August 14, 2007 2:08 PM
To: Ahmed Elnagar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity without Exchange

 

Yes - Unity is a bit of a mess as far as I am concerned when it comes to
the effort it takes just to get basic voicemail. In situations like this
you are better of installing Unity connection - does the same and is a
single application.

 

If you have to use unity - yes - you can install it as on its own
Exchange Island and in its own little domain not integrated with
anything else - But completely without exchange you can not as far as I
know because it uses some of the exchange elements. Not a major problem
- you don't really ahev to configure exchange full like you would do in
an exchage email deployment

 

I have a doc with summary steps explaining how to do it - will mail to
you - need to go and search for it first

 

 

Erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Tuesday, August 14, 2007 1:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity without Exchange

 

Dear all;

 

I want to install unity server for a customer who has no Exchange for
domino server or any active directory and so on, can I install the unity
to run in this case withour installing Exchange server?? I will run it
for voice mail only not unified communication. Do I need to install
Exchange server on the unity box or what??; sorry I know it is an ABC
question but it is my first unity deployment.

 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	
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From jason.aarons at us.didata.com  Tue Aug 14 08:12:52 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 14 Aug 2007 08:12:52 -0400
Subject: [cisco-voip] Unity without Exchange
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED0115F4BF@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED0115F4BF@mail.ACT-EG.COM>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0066D918F@USNAEXCH.na.didata.local>

Yes Unity Voicemail Only comes with a Unity Message Store CD (Cisco Run
Time Edition  Exchange 2000)

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Tuesday, August 14, 2007 8:01 AM
To: Erik Erasmus (E); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity without Exchange

 

One more question.

Does cisco supply the Exchange CDs with the Unity CDs ??

 

________________________________

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za] 
Sent: Tuesday, August 14, 2007 2:08 PM
To: Ahmed Elnagar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity without Exchange

 

Yes - Unity is a bit of a mess as far as I am concerned when it comes to
the effort it takes just to get basic voicemail. In situations like this
you are better of installing Unity connection - does the same and is a
single application.

 

If you have to use unity - yes - you can install it as on its own
Exchange Island and in its own little domain not integrated with
anything else - But completely without exchange you can not as far as I
know because it uses some of the exchange elements. Not a major problem
- you don't really ahev to configure exchange full like you would do in
an exchage email deployment

 

I have a doc with summary steps explaining how to do it - will mail to
you - need to go and search for it first

 

 

Erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Tuesday, August 14, 2007 1:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity without Exchange

 

Dear all;

 

I want to install unity server for a customer who has no Exchange for
domino server or any active directory and so on, can I install the unity
to run in this case withour installing Exchange server?? I will run it
for voice mail only not unified communication. Do I need to install
Exchange server on the unity box or what??; sorry I know it is an ABC
question but it is my first unity deployment.

 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

 




-----------------------------------------
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 lelio at uoguelph.ca  Tue Aug 14 08:39:34 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 14 Aug 2007 08:39:34 -0400
Subject: [cisco-voip] Unity without Exchange
References: <88C8D019DA547C4BABECB9A806ED22ED0115F3CE@mail.ACT-EG.COM>
Message-ID: <008f01c7de70$2bc03a30$6601a8c0@pdp11>

I believe Cisco Unity connection is an all-in-one box for voicemail. No exchange required.

http://cisco.com/en/US/products/ps6509/index.html

  ----- Original Message ----- 
  From: Ahmed Elnagar 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, August 14, 2007 7:00 AM
  Subject: [cisco-voip] Unity without Exchange


  Dear all;

   

  I want to install unity server for a customer who has no Exchange for domino server or any active directory and so on, can I install the unity to run in this case withour installing Exchange server?? I will run it for voice mail only not unified communication. Do I need to install Exchange server on the unity box or what??; sorry I know it is an ABC question but it is my first unity deployment.

   



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From c3voip at nc.rr.com  Tue Aug 14 09:32:40 2007
From: c3voip at nc.rr.com (c3voip)
Date: Tue, 14 Aug 2007 09:32:40 -0400
Subject: [cisco-voip] phone provisioning with inventory database
Message-ID: <000001c7de77$95fcddc0$c1f69940$@rr.com>

I am looking for an application that can help my operations staff provision
phones and maintain a phone inventory with more descriptive fields than can
be stored in CCM. 

 

If there is no such app, then maybe just a phone inventory database.

 

Any ideas?

 

-C

 

 

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From Matt.Slaga at us.didata.com  Tue Aug 14 09:50:47 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 14 Aug 2007 09:50:47 -0400
Subject: [cisco-voip] phone provisioning with inventory database
In-Reply-To: <000001c7de77$95fcddc0$c1f69940$@rr.com>
References: <000001c7de77$95fcddc0$c1f69940$@rr.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0066D93CB@USNAEXCH.na.didata.local>

Try the Cisco Voice Provisioning Tool.  Not sure if it does inventory,
but it does allow a single interface and multiple search options.

 

http://www.cisco.com/en/US/partner/products/ps6524/products_data_sheet09
00aecd80313abd.html

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of c3voip
Sent: Tuesday, August 14, 2007 9:33 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] phone provisioning with inventory database

 

I am looking for an application that can help my operations staff
provision phones and maintain a phone inventory with more descriptive
fields than can be stored in CCM. 

 

If there is no such app, then maybe just a phone inventory database.

 

Any ideas?

 

-C

 

 




-----------------------------------------
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 matt at saskin.net  Tue Aug 14 10:01:33 2007
From: matt at saskin.net (Matthew Saskin)
Date: Tue, 14 Aug 2007 10:01:33 -0400
Subject: [cisco-voip] phone provisioning with inventory database
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0066D93CB@USNAEXCH.na.didata.local>
References: <000001c7de77$95fcddc0$c1f69940$@rr.com>
	<C1FE15183DA37645BC0633BC604E44F0066D93CB@USNAEXCH.na.didata.local>
Message-ID: <46C1B5BD.1010607@saskin.net>

One thing to remember is that VPT doesn't work if you're active 
directory integrated.  Looks like a cool product otherwise.

That said, I've yet to see anyone using it :)

-matt

Matt Slaga (US) wrote:
> Try the Cisco Voice Provisioning Tool.  Not sure if it does inventory, 
> but it does allow a single interface and multiple search options.
> 
>  
> 
> http://www.cisco.com/en/US/partner/products/ps6524/products_data_sheet0900aecd80313abd.html
> 
>  
> 
>  
> 
>  
> 
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *c3voip
> *Sent:* Tuesday, August 14, 2007 9:33 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] phone provisioning with inventory database
> 
>  
> 
> I am looking for an application that can help my operations staff 
> provision phones and maintain a phone inventory with more descriptive 
> fields than can be stored in CCM.
> 
>  
> 
> If there is no such app, then maybe just a phone inventory database.
> 
>  
> 
> Any ideas?
> 
>  
> 
> -C
> 
>  
> 
>  
> 
> ------------------------------------------------------------------------
> 
> * 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 at saskin.net  Tue Aug 14 10:03:30 2007
From: matt at saskin.net (Matthew Saskin)
Date: Tue, 14 Aug 2007 10:03:30 -0400
Subject: [cisco-voip] phone provisioning with inventory database
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0066D93CB@USNAEXCH.na.didata.local>
References: <000001c7de77$95fcddc0$c1f69940$@rr.com>
	<C1FE15183DA37645BC0633BC604E44F0066D93CB@USNAEXCH.na.didata.local>
Message-ID: <46C1B632.3050907@saskin.net>

Also - depending on your programming ability (or ability to snag a 
programmer to assist) it would be possible to write your own app that 
stores additional info in your own DB and tie it in w/ CM via the AXL 
interface.

Or just keep a separate database updated.  Possibly do an export of the 
CCM03xx database daily to an auxiliary machine and use the existing 
data/primary keys to tie in your own tables.

-matt

Matt Slaga (US) wrote:
> Try the Cisco Voice Provisioning Tool.  Not sure if it does inventory, 
> but it does allow a single interface and multiple search options.
> 
>  
> 
> http://www.cisco.com/en/US/partner/products/ps6524/products_data_sheet0900aecd80313abd.html
> 
>  
> 
>  
> 
>  
> 
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *c3voip
> *Sent:* Tuesday, August 14, 2007 9:33 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] phone provisioning with inventory database
> 
>  
> 
> I am looking for an application that can help my operations staff 
> provision phones and maintain a phone inventory with more descriptive 
> fields than can be stored in CCM.
> 
>  
> 
> If there is no such app, then maybe just a phone inventory database.
> 
>  
> 
> Any ideas?
> 
>  
> 
> -C
> 
>  
> 
>  
> 
> ------------------------------------------------------------------------
> 
> * 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 nmenoni at yahoo.com  Tue Aug 14 09:08:56 2007
From: nmenoni at yahoo.com (=?iso-8859-1?q?Nicol=E1s=20Menoni?=)
Date: Tue, 14 Aug 2007 08:08:56 -0500 (CDT)
Subject: [cisco-voip]  Codes for calls
In-Reply-To: <3e6f56310708132030n59907552nb91441c7318d803e@mail.gmail.com>
Message-ID: <181189.79049.qm@web50608.mail.re2.yahoo.com>

Hello. I'm working in a big company with two CCMs
4.2(3) in cluster. I want implement something for can
registering each call for each user when the users
call to the PSTN. One user can call for the PSTN from
any IP telephone, but they could enter a code in the
phone or something like this. In total there are 1.000
users. I don't know how do this, if someone has any
idea it's wellcome, thanks,

Nicol?s.

--- Kent Roberts <nws at fredf.org> escribi?:

> >
> > I been working with oreka, its nice and simple for
> plain recording.  I
> > have it watching 2 ds3's on a 3.4gig xeon.  works
> well for free!   esp, if
> > its a remote site that needs monitoring...
> >
> >
> > http://oreka.sourceforge.net
> >
> >
> >
> > On 8/13/07, Todd Franklin <toddnh65 at gmail.com>
> wrote:
> > >
> > > I have a need to record a few lines, at all
> times.
> > > Has anyone had good experiences with any sort of
> recorder for the Cisco
> > > IP phones (spec. 7940s and 7960s).
> > > These are not POTS lines, so I can't place the
> recorder in front of the
> > > gateway....the recorder would need to "sniff"
> the network for VOIP packets
> > > and assemble them into recordings.
> > >
> > > Thanks.
> > >
> > > PS- I have been looking into CallReplay
> (call-replay.com) but am not
> > > sure what else is out there.
> > >
> > >
> > >
> > > _______________________________________________
> > > 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


__________________________________________________
Correo Yahoo!
Espacio para todos tus mensajes, antivirus y antispam ?gratis! 
Reg?strate ya - http://correo.espanol.yahoo.com/ 

From c3voip at nc.rr.com  Tue Aug 14 10:11:00 2007
From: c3voip at nc.rr.com (c3voip)
Date: Tue, 14 Aug 2007 10:11:00 -0400
Subject: [cisco-voip] phone provisioning with inventory database
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0066D93CB@USNAEXCH.na.didata.local>
References: <000001c7de77$95fcddc0$c1f69940$@rr.com>
	<C1FE15183DA37645BC0633BC604E44F0066D93CB@USNAEXCH.na.didata.local>
Message-ID: <000e01c7de7c$f157c2c0$d4074840$@rr.com>

This link works better for most of us:

http://www.cisco.com/en/US/customer/products/ps6524/products_data_sheet0900a
ecd80313abd.html

 

but the inventory part is the most important part.  I need to know more
about my phones than we can input into CCM.  Information like Serial Number,
User name, Department, Building and office/cube, Location, Jack number,
Extension(s), Charge number, etc.

 

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Tuesday, August 14, 2007 9:51 AM
To: c3voip; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] phone provisioning with inventory database

 

Try the Cisco Voice Provisioning Tool.  Not sure if it does inventory, but
it does allow a single interface and multiple search options.

 

http://www.cisco.com/en/US/partner/products/ps6524/products_data_sheet0900ae
cd80313abd.html

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of c3voip
Sent: Tuesday, August 14, 2007 9:33 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] phone provisioning with inventory database

 

I am looking for an application that can help my operations staff provision
phones and maintain a phone inventory with more descriptive fields than can
be stored in CCM. 

 

If there is no such app, then maybe just a phone inventory database.

 

Any ideas?

 

-C

 

 

  _____  

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 matt at saskin.net  Tue Aug 14 10:23:29 2007
From: matt at saskin.net (Matthew Saskin)
Date: Tue, 14 Aug 2007 10:23:29 -0400
Subject: [cisco-voip] Codes for calls
In-Reply-To: <181189.79049.qm@web50608.mail.re2.yahoo.com>
References: <181189.79049.qm@web50608.mail.re2.yahoo.com>
Message-ID: <46C1BAE1.5040905@saskin.net>

For this you probably want to use Forced Authorization Codes (FAC's)

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803f3af7.html#wp15243

-matt

Nicol?s Menoni wrote:
> Hello. I'm working in a big company with two CCMs
> 4.2(3) in cluster. I want implement something for can
> registering each call for each user when the users
> call to the PSTN. One user can call for the PSTN from
> any IP telephone, but they could enter a code in the
> phone or something like this. In total there are 1.000
> users. I don't know how do this, if someone has any
> idea it's wellcome, thanks,
> 
> Nicol?s.
> 


From bob at smalltime.com  Tue Aug 14 10:40:33 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Tue, 14 Aug 2007 09:40:33 -0500
Subject: [cisco-voip] Networking Unity Express and Unity using VPIM
In-Reply-To: <46C10024.8040604@wcyv.com>
References: <46C070D6.2010306@smalltime.com> <46C10024.8040604@wcyv.com>
Message-ID: <46C1BEE1.9080808@smalltime.com>

Pat Hayes wrote:
> I'm not sure exactly what happens if you send a 729 encoded message to 
> CUE, but, you should be able to get around this by specifying the 
> outgoing codec on the VPIM delivery location that you created in Unity.
> 
> Going the other way, Unity should be able to transcode the message on 
> the fly without a problem. Hope this helps.

Will a VM that's forwarded from UE to Unity via VPIM have the 
appropriate headers applied so that it's marked as a VM in Outlook and 
can be played with Viewmail?  Or is it just an email with a .wav file 
attachment that has to be played with Media Player?



From pezhman_lali at yahoo.com  Tue Aug 14 09:43:56 2007
From: pezhman_lali at yahoo.com (Pezhman Lali)
Date: Tue, 14 Aug 2007 06:43:56 -0700 (PDT)
Subject: [cisco-voip] changing caller id
In-Reply-To: <1ff12970708140444t3a5c4bke271b05fff43c83d@mail.gmail.com>
Message-ID: <254032.59415.qm@web52305.mail.re2.yahoo.com>

dear 
thanks for your reply.
the config is:

!
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname test
!
enable password 111111
!
username admin password 0 111111
voice-card 1
!
aaa new-model
!
!
aaa authentication login default local
aaa authentication login vty local
aaa authentication login h323 group radius
aaa authentication login con none
aaa authorization exec default local 
aaa authorization exec vty local 
aaa authorization exec h323 group radius 
aaa accounting connection h323 start-stop group radius
aaa session-id common
ip subnet-zero
!
!
no ip domain lookup
ip name-server 4.2.2.4
!
isdn switch-type primary-net5
!
!
voice rtp send-recv
!
voice class codec 10
 codec preference 1 g723r53
 codec preference 2 g723r63
 codec preference 3 g723ar53
 codec preference 4 g723ar63
 codec preference 6 g729br8
!
!
!
!
!
!
voice class custom-cptone test
 dualtone busy
  frequency 425 425
  cadence 500 500
 dualtone ringback
  frequency 425 425
  cadence 1000 4000
 dualtone reorder
  frequency 425 425
  cadence 1000 1000
 dualtone out-of-service
  frequency 950
  cadence 330 330
 dualtone number-unobtainable
  frequency 425
  cadence 240 240
 dualtone disconnect
  frequency 425 425
  cadence 250 250
!
!
!
!
voice translation-rule 1
 rule 1 /^2/ /2197121112/
 rule 2 /^1/ /2197121111/
 rule 3 /^3/ /2197121113/
 rule 4 /^4/ /2197121114/
 rule 5 /^5/ /2197121115/
 rule 6 /^6/ /2197121116/
 rule 7 /^7/ /2197121117/
 rule 8 /^8/ /2197121118/
 rule 9 /^9/ /2197121119/
 rule 10 /^0/ /2197121110/
!
voice translation-rule 2
 rule 1 /^25221/ //
 rule 2 /^252/ /0/
!
!
voice translation-profile profile1
 translate called 1
!
voice translation-profile profile2
 translate called 2
!
!
mta receive maximum-recipients 0
!
controller E1 1/0
 framing NO-CRC4 
 clock source line primary
 line-termination 75-ohm
 pri-group timeslots 1-31
!
controller E1 1/1
 shutdown
 framing NO-CRC4 
 line-termination 75-ohm
!
translation-rule 1
!
translation-rule 2
!
!
!
!
interface FastEthernet0/0
 ip address 85.1.1.2 255.255.255.0
 duplex auto
 speed auto
!
interface Serial1/0:15
 no ip address
 no logging event link-status
 isdn switch-type primary-net5
 isdn incoming-voice voice
 no cdp enable
!
ip default-gateway 85.1.1.1
ip classless
ip route 0.0.0.0 0.0.0.0 85.1.1.1
no ip http server
ip pim bidir-enable
!
!
access-list 102 deny   tcp any any eq 4444
access-list 102 deny   tcp any any eq 593
access-list 102 deny   tcp any any eq 445
access-list 102 deny   tcp any any eq 139
access-list 102 deny   tcp any any eq 135
access-list 102 deny   udp any any eq tftp
access-list 102 deny   udp any any eq netbios-dgm
access-list 102 deny   udp any any eq 135
access-list 102 permit tcp any any eq telnet
access-list 102 permit ip any any
access-list 102 permit tcp any any
access-list 102 permit udp any any
!

radius-server host 85.1.1.3 auth-port 1812 acct-port
1813 key test
radius-server authorization permit missing
Service-Type
radius-server vsa send accounting
radius-server vsa send authentication
call rsvp-sync
!
call application voice cidial flash:cidial.tcl
call application voice cidial language 0 en
call application voice cidial set-location fa 0 flash:
!
voice-port 1/0:15
!
!
mgcp profile default
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 900 voip
 session protocol sipv2
 session target dns:did.test.com
 dtmf-relay rtp-nte
!
dial-peer voice 100 pots
 translation-profile outgoing profile2
 destination-pattern 252T
 clid restrict
 port 1/0:15
!
dial-peer voice 1112 pots
!
dial-peer voice 1111 pots
!
dial-peer voice 110 pots
 incoming called-number T
 direct-inward-dial
!
sip-ua 
 sip-server ipv4:217.10.79.219
!
!
line con 0
line aux 0
line vty 0 4
 password 11111
!
!
end

--- zohaib shabir <zohaibshabir at gmail.com> wrote:

> Yes
> you can always change the callerid of calls if you
> have the PRI line for the
> pstn. I dont know you config, but i am sure you if
> you share the config with
> me, i will be able to help you out.
> 
> On 8/14/07, Pezhman Lali <pezhman_lali at yahoo.com>
> wrote:
> >
> > Hi dears
> > I have a cisco2600+nm-hdv+pri line(E1).
> > is any way to change callerid of outgoing calls to
> the
> > pstn.
> > best
> > Mani
> >
> >
> >
> >
> >
>
____________________________________________________________________________________
> > 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
> > _______________________________________________
> > 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
> 



       
____________________________________________________________________________________Ready for the edge of your seat? 
Check out tonight's top picks on Yahoo! TV. 
http://tv.yahoo.com/

From joe.j.keller at gmail.com  Tue Aug 14 10:48:19 2007
From: joe.j.keller at gmail.com (Joe Keller)
Date: Tue, 14 Aug 2007 10:48:19 -0400
Subject: [cisco-voip] New Unity install 5.0 or 4.x and why?
Message-ID: <189b5daa0708140748t3d3f573ejc0231d4f9c8b0066@mail.gmail.com>

New installation, with message store on Unity. No restrictions. Install 4.x,
most probably 4.2 or 5.0. With migration of CCM 4.1.3 to CCM 5.1.
Any opinions? And why?
Is 5.0 ready for primetime?

Thanks
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From bms314 at gmail.com  Tue Aug 14 11:21:43 2007
From: bms314 at gmail.com (Brian)
Date: Tue, 14 Aug 2007 10:21:43 -0500
Subject: [cisco-voip] caller name with H.323
Message-ID: <7aaaf7f0708140821w6350f910j77274a86a643d76f@mail.gmail.com>

I see H.323 name display is supported as of 12.4(11)XW.  Does anyone know if
it is also supported on some level of the T train?  I'm running 12.4(15)T1
and CUCM 5.1(2), but I don't see the display-ie or supp-service commands on
the gateway.

Thanks,
Brian
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From jonvoip at gmail.com  Tue Aug 14 15:04:55 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 14 Aug 2007 14:04:55 -0500
Subject: [cisco-voip] ipcc demo
In-Reply-To: <f84a38d30708130845q65c5cbbdta8ea4017f740e542@mail.gmail.com>
References: <f84a38d30708130845q65c5cbbdta8ea4017f740e542@mail.gmail.com>
Message-ID: <5d093f9a0708141204k1c3a4aean44a80563ee2e54e0@mail.gmail.com>

IPCC MUST be on its own server with a CCM5.X install.

It comes with a  2-port demo license (for TAPS)



Jonathan

On 8/13/07, Scott Voll <svoll.voip at gmail.com> wrote:
> I'm hoping this is a quick question.
>
> with CM you get some little demo type stuff.
>
> with CM 5.x how does the IPCC work.  with CM 4.x it was co-resident but I'm
> clueless on 5.x.
>
> TIA
>
> Scott
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From nmenoni at yahoo.com  Tue Aug 14 15:07:13 2007
From: nmenoni at yahoo.com (=?iso-8859-1?q?Nicol=E1s=20Menoni?=)
Date: Tue, 14 Aug 2007 14:07:13 -0500 (CDT)
Subject: [cisco-voip] Codes for calls
In-Reply-To: <46C1BAE1.5040905@saskin.net>
Message-ID: <979604.59004.qm@web50610.mail.re2.yahoo.com>

It's just I was looking for, I will read more about
this, thank you very much Matthew, 

Nicol?s (from Uruguay, South America)

--- Matthew Saskin <matt at saskin.net> escribi?:

> For this you probably want to use Forced
> Authorization Codes (FAC's)
> 
>
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803f3af7.html#wp15243
> 
> -matt
> 
> Nicol?s Menoni wrote:
> > Hello. I'm working in a big company with two CCMs
> > 4.2(3) in cluster. I want implement something for
> can
> > registering each call for each user when the users
> > call to the PSTN. One user can call for the PSTN
> from
> > any IP telephone, but they could enter a code in
> the
> > phone or something like this. In total there are
> 1.000
> > users. I don't know how do this, if someone has
> any
> > idea it's wellcome, thanks,
> > 
> > Nicol?s.
> > 
> 
> 


__________________________________________________
Correo Yahoo!
Espacio para todos tus mensajes, antivirus y antispam ?gratis! 
Reg?strate ya - http://correo.espanol.yahoo.com/ 

From John.Cauthen at AGG.com  Tue Aug 14 16:22:00 2007
From: John.Cauthen at AGG.com (Cauthen, John A.)
Date: Tue, 14 Aug 2007 16:22:00 -0400
Subject: [cisco-voip] Static on 7960 Series Phone
Message-ID: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>

We have an issue with one of our 7960 series phones continually to have
static / crackling while the phone is in use.  It does not happen on
every phone call but does happen usually every day.  We have contacted
TAC support and have gone through the following troubleshooting but
cannot pin point the issue.

 

1. Phone has been replaced twice

2. Taken a non reported static phone and placed in her office, static
gets reported

3. Taken her phone out and had another user use it and no static
reported

4. We have run a new cable from her phone to a port outside her office
that has no static and she still get's static

5. Office has been checked for interfering devices (none found)

6. Static reported by her as being so bad she cannot hear the person she
is talking to all the way down to just crackling (like cell phone
reception), but the person on the other end does not hear static.

7. Have applied the latest and greatest phone load to user's phone

 

If anyone has any suggestions or ideas it would be very appreciated.

 

Thanks,

John


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 Aug 14 16:35:39 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 14 Aug 2007 16:35:39 -0400
Subject: [cisco-voip] Static on 7960 Series Phone
In-Reply-To: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>
References: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00671AD43@USNAEXCH.na.didata.local>

I would replace her phone at least two more times. Tends to be between
mic and handset.  Have you tried using a headset (Plantronics, etc) to
duplicate the problem?  Sure it's not a bad channel on a PRI? Or bad
PVDM2 ? Happens every call every time and she is the only one reporting
it?  Is your PSTN interface analog or digital? Open ticket with PSTN to
bring out a T-Berd and listen for static on calls? If she transfers call
to another dn does the static follow transfer?

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cauthen, John
A.
Sent: Tuesday, August 14, 2007 4:22 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Static on 7960 Series Phone

 

We have an issue with one of our 7960 series phones continually to have
static / crackling while the phone is in use.  It does not happen on
every phone call but does happen usually every day.  We have contacted
TAC support and have gone through the following troubleshooting but
cannot pin point the issue.

 

1. Phone has been replaced twice

2. Taken a non reported static phone and placed in her office, static
gets reported

3. Taken her phone out and had another user use it and no static
reported

4. We have run a new cable from her phone to a port outside her office
that has no static and she still get's static

5. Office has been checked for interfering devices (none found)

6. Static reported by her as being so bad she cannot hear the person she
is talking to all the way down to just crackling (like cell phone
reception), but the person on the other end does not hear static.

7. Have applied the latest and greatest phone load to user's phone

 

If anyone has any suggestions or ideas it would be very appreciated.

 

Thanks,

John

________________________________

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 adman at adman.net  Tue Aug 14 16:43:06 2007
From: adman at adman.net (Adam Blomfield)
Date: Tue, 14 Aug 2007 15:43:06 -0500
Subject: [cisco-voip] IPCC license for the scenario
In-Reply-To: <1ff12970708140313u30e9ea53h2e2395e063b9929a@mail.gmail.com>
References: <673237.31703.qm@web39604.mail.mud.yahoo.com>
	<1ff12970708140313u30e9ea53h2e2395e063b9929a@mail.gmail.com>
Message-ID: <c6f32ce40708141343o4eb91fb0m141c8319ce226c0c@mail.gmail.com>

Actually to do database lookups you will need the Premium version of UCCX
(IPCC). New orders for Communications Manager (CallManager) come with UCCX
Enhanced now instead of the Standard version that used to be bundled.

-Adam

On 8/14/07, zohaib shabir <zohaibshabir at gmail.com> wrote:
>
> Hello,
> CCM 4.1.3 comes with standard version of IPCC express, the functionality
> you require need an IPCC express enhanced version,you need to upgrade your
> license to IPCC enhance version.
>
> For registration just register IPCC express  serial number that will give
> you the key which you will require in installation.
>
>
> On 8/14/07, gokhan senol <gokhanciscottl at yahoo.com> wrote:
>
> > hello
> >
> >
> >
> > on ccm 4.1.3  which kind of IPCC license comes .  standart or enhanced
> > or premium?
> >
> >
> >
> > the customer has an application which keeps the guest database and they
> > want the calls will come to the agent and the database will search based on
> > CallerID.  And for example CallerID is 555xxxx  and then from the database
> > the guest who owns this number will be found and the data related to this
> > guest will come to the screen
> >
> > for this kind of application IPCC license  come with CCM is enough or
> > should i upgrade it?
> >
> >
> >
> > also what is the steps for enabling ipcc , i will register PAK coe and
> > then install on CCM with IPCC cd, thats all?
> >
> >
> >
> >
> >
> >
> >
> > thanks
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > ------------------------------
> > Choose the right car based on your needs. Check out Yahoo! Autos new Car
> > Finder tool.<http://us.rd.yahoo.com/evt=48518/*http://autos.yahoo.com/carfinder/;_ylc=X3oDMTE3NWsyMDd2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDY2FyLWZpbmRlcg--+>
> >
> > _______________________________________________
> > 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
>
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From svoll.voip at gmail.com  Tue Aug 14 17:21:29 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 14 Aug 2007 14:21:29 -0700
Subject: [cisco-voip] Translation error
Message-ID: <f84a38d30708141421qef7d726vc4a215e02c054b82@mail.gmail.com>

I'm trying to delete a Translation pattern and I'm getting the following
error.  Any idea how to fix it so I can delete it?

Error No. 1576:
CDBLException Dump: [Delete failed because record is in use by a related
table]

this is CM 4.1.3es100

Thanks

Scott
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From svoll.voip at gmail.com  Tue Aug 14 17:23:42 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 14 Aug 2007 14:23:42 -0700
Subject: [cisco-voip] Phone Proxy
Message-ID: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>

Hey--

we just got our Phone Proxy setup and it works pretty slick.

I think it only took about 4 hours to unpackage, configure, and get the
first Phone up and working.

I do however hate Cisco docs as it states you need a null modem.  so I look
high and low and finally find one to find out that in all reality you really
don't need one.  Got to love that.

Scott
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From lmarasci at cisco.com  Tue Aug 14 17:25:21 2007
From: lmarasci at cisco.com (Louis Marascio (lmarasci))
Date: Tue, 14 Aug 2007 14:25:21 -0700
Subject: [cisco-voip] Phone Proxy
In-Reply-To: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>
Message-ID: <6EE7932CC71C8D49A20C0E639BCD2165027644A5@xmb-sjc-21c.amer.cisco.com>

Scott, 
 
What doc does it talk about a null modem? I'll try to get it fixed.
Sorry for the trouble.
 
Louis

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Tuesday, August 14, 2007 4:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Phone Proxy


Hey--
 
we just got our Phone Proxy setup and it works pretty slick.
 
I think it only took about 4 hours to unpackage, configure, and get the
first Phone up and working.
 
I do however hate Cisco docs as it states you need a null modem.  so I
look high and low and finally find one to find out that in all reality
you really don't need one.  Got to love that.
 
Scott
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From jonvoip at gmail.com  Tue Aug 14 17:32:38 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 14 Aug 2007 16:32:38 -0500
Subject: [cisco-voip] CCM 5.1, unable to pull logs via RTMT..
Message-ID: <5d093f9a0708141432v166b5edcnb6c29ef1b14d7bf5@mail.gmail.com>

I have a CCM 5.1 cluster, I can pull logs via RTMT from the publisher
fine, but when I try from the subscriber, I get the following:

"An Error Occurred while Processing Request"

I am able to connect via RTMT and see performance stats (Virtual
Memory, CPU, registered phones, etc...) I just can't view log files.

Initially I couldn't even get into RTMT, so I restarted Cisco
Servicability and I was able to get in... Any ideas what could be
wrong here?



Jonathan

From jason.aarons at us.didata.com  Tue Aug 14 17:34:01 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 14 Aug 2007 17:34:01 -0400
Subject: [cisco-voip] Phone Proxy and IPC
In-Reply-To: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>
References: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00671AE5D@USNAEXCH.na.didata.local>

I wish Cisco IP Communicator had a profile option at startup, allowing
you to change your TFTP server easily.

 

 I have to change my tftp servers for each customer, for our phone
proxy, and for when I'm at our office. I have to keep a txt listing of
all the different tftp servers at sites, etc.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Voll
Sent: Tuesday, August 14, 2007 5:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Phone Proxy

 

Hey--

 

we just got our Phone Proxy setup and it works pretty slick.

 

I think it only took about 4 hours to unpackage, configure, and get the
first Phone up and working.

 

I do however hate Cisco docs as it states you need a null modem.  so I
look high and low and finally find one to find out that in all reality
you really don't need one.  Got to love that.

 

Scott




-----------------------------------------
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 adman at adman.net  Tue Aug 14 17:44:52 2007
From: adman at adman.net (Adam Blomfield)
Date: Tue, 14 Aug 2007 16:44:52 -0500
Subject: [cisco-voip] Phone Proxy and IPC
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00671AE5D@USNAEXCH.na.didata.local>
References: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F00671AE5D@USNAEXCH.na.didata.local>
Message-ID: <c6f32ce40708141444x299bfd74ncc31aa4cf97a13b2@mail.gmail.com>

I second the motion!

On 8/14/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  I wish Cisco IP Communicator had a profile option at startup, allowing
> you to change your TFTP server easily.
>
>
>
>  I have to change my tftp servers for each customer, for our phone proxy,
> and for when I'm at our office. I have to keep a txt listing of all the
> different tftp servers at sites, etc.
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Scott Voll
> *Sent:* Tuesday, August 14, 2007 5:24 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Phone Proxy
>
>
>
> Hey--
>
>
>
> we just got our Phone Proxy setup and it works pretty slick.
>
>
>
> I think it only took about 4 hours to unpackage, configure, and get the
> first Phone up and working.
>
>
>
> I do however hate Cisco docs as it states you need a null modem.  so I
> look high and low and finally find one to find out that in all reality you
> really don't need one.  Got to love that.
>
>
>
> Scott
>
> ------------------------------
>
> * 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 rsingleton at morsco.com  Tue Aug 14 17:46:18 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Tue, 14 Aug 2007 16:46:18 -0500
Subject: [cisco-voip] Static on 7960 Series Phone
In-Reply-To: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>
References: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>
Message-ID: <1187127978.1812.55.camel@robert.morsco.com>

On Tue, 2007-08-14 at 16:22 -0400, Cauthen, John A. wrote:
> We have an issue with one of our 7960 series phones continually to
> have static / crackling while the phone is in use.  It does not happen
> on every phone call but does happen usually every day.  We have
> contacted TAC support and have gone through the following
> troubleshooting but cannot pin point the issue.


> 1. Phone has been replaced twice

Same switch port

> 2. Taken a non reported static phone and placed in her office, static
> gets reported

Same switch port

> 3. Taken her phone out and had another user use it and no static
> reported

Different switch port

> 4. We have run a new cable from her phone to a port outside her office
> that has no static and she still get's static

Oops, different switch port, but has static.

> 5. Office has been checked for interfering devices (none found)

So, the office has nothing but furniture and the phone in it? How about
a PC monitor with connected to an outlet with a bad ground or otehr
wiring fault? How about the adjacent offices? Cabling that doesn't
necessarily terminate in that office but passes through the walls?

> 6. Static reported by her as being so bad she cannot hear the person
> she is talking to all the way down to just crackling (like cell phone
> reception), but the person on the other end does not hear static.

Have you tried the call statistics during the static time to see if
there is jitter or dropped packets? Sounds like something that is
specific to her calls; is the static only on outbound calls or both? If
only outbound, maybe her Device Pool has her in a region that can only
use a high bandwidth codec, but insufficient bandwidth is available?
Until the different switchport with the same issue came up above, I
thought that perhaps her switchport was limiting bandwidth or something
like that.

> 7. Have applied the latest and greatest phone load to user?s phone

Generally a good practice anyway...

Is the phone powered by the switch or a separate power supply? If a
power supply, maybe it has a noisy connection.

I'm tapped out for now...

Robert


From cisco at adman.net  Tue Aug 14 17:49:55 2007
From: cisco at adman.net (Adam)
Date: Tue, 14 Aug 2007 16:49:55 -0500
Subject: [cisco-voip] 3rd Party Import Tool
Message-ID: <c6f32ce40708141449j567a3d32oa0cf1e790c862548@mail.gmail.com>

I know a while back I ran across a third party tool that let you import more
than just the options in BAT... stuff like CSS and route patterns. I just
tried Googling for it, but can't remember the name of it and didn't have
much luck finding anything. Does anyone know the utility I'm talking about,
or am I just crazy and imagining things?

Thanks,
Adam
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From lmarasci at cisco.com  Tue Aug 14 17:51:46 2007
From: lmarasci at cisco.com (Louis Marascio (lmarasci))
Date: Tue, 14 Aug 2007 14:51:46 -0700
Subject: [cisco-voip] Phone Proxy and IPC
In-Reply-To: <c6f32ce40708141444x299bfd74ncc31aa4cf97a13b2@mail.gmail.com>
Message-ID: <6EE7932CC71C8D49A20C0E639BCD2165027644D5@xmb-sjc-21c.amer.cisco.com>

IP Communicator stores their TFTP information in the registry:
HK[CU,LM]\SOFTWARE\Cisco Systems, Inc.\Communicator. There are two keys,
TFTPServer1 and TFTPServer2. I suspect you could write a script to
change it automatically and launch IP Communicator for you. 
 
*cough*
I am not on the IPC team and this is obviously not supported, yada yada
yaaaadddddaaaaa....
*cough*
 
Louis

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Adam Blomfield
Sent: Tuesday, August 14, 2007 4:45 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Proxy and IPC


I second the motion!


On 8/14/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote: 

	I wish Cisco IP Communicator had a profile option at startup,
allowing you to change your TFTP server easily.

	 

	 I have to change my tftp servers for each customer, for our
phone proxy, and for when I'm at our office. I have to keep a txt
listing of all the different tftp servers at sites, etc.

	 

	From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Scott Voll
	Sent: Tuesday, August 14, 2007 5:24 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Phone Proxy

	 

	Hey--

	 

	we just got our Phone Proxy setup and it works pretty slick.

	 

	I think it only took about 4 hours to unpackage, configure, and
get the first Phone up and working.

	 

	I do however hate Cisco docs as it states you need a null modem.
so I look high and low and finally find one to find out that in all
reality you really don't need one.  Got to love that.

	 

	Scott

	

________________________________

	

	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 svoll.voip at gmail.com  Tue Aug 14 18:04:06 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 14 Aug 2007 15:04:06 -0700
Subject: [cisco-voip] Phone Proxy
In-Reply-To: <6EE7932CC71C8D49A20C0E639BCD2165027644A5@xmb-sjc-21c.amer.cisco.com>
References: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>
	<6EE7932CC71C8D49A20C0E639BCD2165027644A5@xmb-sjc-21c.amer.cisco.com>
Message-ID: <f84a38d30708141504q3928e913g69a7f41bcc5e7023@mail.gmail.com>

"Cisco Unified PhoneProxy installation and quick start guide" release 1.0(1)

PDF page 19 under procedure step 1.

Thanks

Scott


On 8/14/07, Louis Marascio (lmarasci) <lmarasci at cisco.com> wrote:
>
>  Scott,
>
> What doc does it talk about a null modem? I'll try to get it fixed. Sorry
> for the trouble.
>
> Louis
>
>  ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Scott Voll
> *Sent:* Tuesday, August 14, 2007 4:24 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Phone Proxy
>
>
>  Hey--
>
> we just got our Phone Proxy setup and it works pretty slick.
>
> I think it only took about 4 hours to unpackage, configure, and get the
> first Phone up and working.
>
> I do however hate Cisco docs as it states you need a null modem.  so I
> look high and low and finally find one to find out that in all reality you
> really don't need one.  Got to love that.
>
> Scott
>
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From kyle.hansen at hci.utah.edu  Tue Aug 14 17:54:03 2007
From: kyle.hansen at hci.utah.edu (Kyle Hansen (HCH))
Date: Tue, 14 Aug 2007 15:54:03 -0600
Subject: [cisco-voip] Unity Phone trees phone number masking
Message-ID: <916324E1C885C54989449104CB7225B60140FEB3@EMAIL1.hci.utah.edu>

Does anyone know if you can "mask" the numbers of a phone tree in Unity?
Example:  Caller calls 12345 gets the phone tree with different options
to press 1 or 2 or 3 or 4 Caller presses 2 (or any other option) but, it
shows the number that option 2 is transferring the caller to.  I hope
this makes sense.  Any help would be wonderful!!!

 

Thank you,

Kyle Hansen

Computing and Technology Group (C.A.T.G.)

Customer Support Lead/Project Manager

Huntsman Cancer Hospital

1950 Circle of Hope - Salt Lake City, UT  84112.  801.585.0330

If you need computer support please call the HELPDESK @ 585-0330 or

Submit a request for support at:
www.hci.utah.edu/internal/computer_support.html

 

 

 

 

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From kyle.hansen at hci.utah.edu  Tue Aug 14 17:50:38 2007
From: kyle.hansen at hci.utah.edu (Kyle Hansen (HCH))
Date: Tue, 14 Aug 2007 15:50:38 -0600
Subject: [cisco-voip] Configuration Utility 7920 & Windows Vista
Message-ID: <916324E1C885C54989449104CB7225B60140FEB2@EMAIL1.hci.utah.edu>

We are upgrading some of our systems to Windows Vista but, found out
that The Cisco 7920 Wireless phone Configuration utility does not work.
Has anyone found a fix?  I could not find anything on Cisco's website.

 

Thank you,

Kyle Hansen

Computing and Technology Group (C.A.T.G.)

Customer Support Lead/Project Manager

Huntsman Cancer Hospital

1950 Circle of Hope - Salt Lake City, UT  84112.  801.585.0330

If you need computer support please call the HELPDESK @ 585-0330 or

Submit a request for support at:
www.hci.utah.edu/internal/computer_support.html

 

 

 

 

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From Matt.Slaga at us.didata.com  Tue Aug 14 19:25:36 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 14 Aug 2007 19:25:36 -0400
Subject: [cisco-voip] Static on 7960 Series Phone
In-Reply-To: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>
References: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00671AEE9@USNAEXCH.na.didata.local>

Have the user wear a static wrist strap.  Some people just generate more
static electricity than others.

 

But really, make sure the user does not have his cellular phone lying
next to the phone itself.  Especially Nextel phones, they can reboot
some of the older models (it's amazing what transmitting at 400mhz can
do to a 400mhz processor).

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cauthen, John
A.
Sent: Tuesday, August 14, 2007 4:22 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Static on 7960 Series Phone

 

We have an issue with one of our 7960 series phones continually to have
static / crackling while the phone is in use.  It does not happen on
every phone call but does happen usually every day.  We have contacted
TAC support and have gone through the following troubleshooting but
cannot pin point the issue.

 

1. Phone has been replaced twice

2. Taken a non reported static phone and placed in her office, static
gets reported

3. Taken her phone out and had another user use it and no static
reported

4. We have run a new cable from her phone to a port outside her office
that has no static and she still get's static

5. Office has been checked for interfering devices (none found)

6. Static reported by her as being so bad she cannot hear the person she
is talking to all the way down to just crackling (like cell phone
reception), but the person on the other end does not hear static.

7. Have applied the latest and greatest phone load to user's phone

 

If anyone has any suggestions or ideas it would be very appreciated.

 

Thanks,

John

________________________________

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 BurwellJ at firstcharter.com  Tue Aug 14 16:29:26 2007
From: BurwellJ at firstcharter.com (Jason Burwell)
Date: Tue, 14 Aug 2007 16:29:26 -0400
Subject: [cisco-voip] Static on 7960 Series Phone
In-Reply-To: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>
Message-ID: <697161D0107BAD40AFE9A86B06509AC90D6F518E@FCEXCLUSTER.FirstCharter.com>

It may be an issue with the route the cable is run- like if it goes over
a fluorescent light or something. Also, have you swapped out switch
ports and patch cords on both ends?

 

Jason

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cauthen, John
A.
Sent: Tuesday, August 14, 2007 4:22 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Static on 7960 Series Phone

 

We have an issue with one of our 7960 series phones continually to have
static / crackling while the phone is in use.  It does not happen on
every phone call but does happen usually every day.  We have contacted
TAC support and have gone through the following troubleshooting but
cannot pin point the issue.

 

1. Phone has been replaced twice

2. Taken a non reported static phone and placed in her office, static
gets reported

3. Taken her phone out and had another user use it and no static
reported

4. We have run a new cable from her phone to a port outside her office
that has no static and she still get's static

5. Office has been checked for interfering devices (none found)

6. Static reported by her as being so bad she cannot hear the person she
is talking to all the way down to just crackling (like cell phone
reception), but the person on the other end does not hear static.

7. Have applied the latest and greatest phone load to user's phone

 

If anyone has any suggestions or ideas it would be very appreciated.

 

Thanks,

John

________________________________

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 jsteinberg at gmail.com  Tue Aug 14 20:56:45 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 14 Aug 2007 20:56:45 -0400
Subject: [cisco-voip] Static on 7960 Series Phone
In-Reply-To: <697161D0107BAD40AFE9A86B06509AC90D6F518E@FCEXCLUSTER.FirstCharter.com>
References: <4E23F66505C2E14CB4A1C1E5316F326D0B3AF8@ATLEX01.agg.local>
	<697161D0107BAD40AFE9A86B06509AC90D6F518E@FCEXCLUSTER.FirstCharter.com>
Message-ID: <f1b65f880708141756w21464e20r6ca4d15d34a788aa@mail.gmail.com>

Swap her hearing aid?

On 8/14/07, Jason Burwell <BurwellJ at firstcharter.com> wrote:
>
>  It may be an issue with the route the cable is run- like if it goes over
> a fluorescent light or something. Also, have you swapped out switch ports
> and patch cords on both ends?
>
>
>
> Jason
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Cauthen, John A.
> *Sent:* Tuesday, August 14, 2007 4:22 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Static on 7960 Series Phone
>
>
>
> We have an issue with one of our 7960 series phones continually to have
> static / crackling while the phone is in use.  It does not happen on every
> phone call but does happen usually every day.  We have contacted TAC support
> and have gone through the following troubleshooting but cannot pin point the
> issue.
>
>
>
> 1. Phone has been replaced twice
>
> 2. Taken a non reported static phone and placed in her office, static gets
> reported
>
> 3. Taken her phone out and had another user use it and no static reported
>
> 4. We have run a new cable from her phone to a port outside her office
> that has no static and she still get's static
>
> 5. Office has been checked for interfering devices (none found)
>
> 6. Static reported by her as being so bad she cannot hear the person she
> is talking to all the way down to just crackling (like cell phone
> reception), but the person on the other end does not hear static.
>
> 7. Have applied the latest and greatest phone load to user's phone
>
>
>
> If anyone has any suggestions or ideas it would be very appreciated.
>
>
>
> Thanks,
>
> John
>  ------------------------------
>
> *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 tednugent69 at yahoo.com  Tue Aug 14 22:19:36 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Tue, 14 Aug 2007 19:19:36 -0700 (PDT)
Subject: [cisco-voip] CM6 CallManager service will not stay running
Message-ID: <559959.67447.qm@web36114.mail.mud.yahoo.com>

Hey Guys
I've got a FRESH install of CM 6.01 which I've
attempted on both VMWare and a legitimate MCS and both
are reacting the same way. After the install is
complete I go to activate the "default" services and
the CM service runs for about 30 seconds and bombs all
other services remain running fine. I've read that
this is expected behavior on an upgrade however this
is a fresh install from the ground up so it SHOULD use
the "starter license"?? What am I missing? Any one
else run into this?

>From 6.01 Release Notes:
Starter Licenses
Cisco Unified Communications Manager comes
preinstalled with starter licenses, so a file upload
is not required. Starter licenses have no expiration
date and are available in limited quantities. Be aware
that starter licenses are only available for fresh
installations; they are not available for upgrades or
migrations from previous releases. Starter licenses
support only one Cisco Unified Communications Manager
node and up to 50 device license units.


Thanks
Ted


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

From adman at adman.net  Tue Aug 14 23:42:27 2007
From: adman at adman.net (Adam Blomfield)
Date: Tue, 14 Aug 2007 22:42:27 -0500
Subject: [cisco-voip] Phone Proxy and IPC
In-Reply-To: <6EE7932CC71C8D49A20C0E639BCD2165027644D5@xmb-sjc-21c.amer.cisco.com>
References: <c6f32ce40708141444x299bfd74ncc31aa4cf97a13b2@mail.gmail.com>
	<6EE7932CC71C8D49A20C0E639BCD2165027644D5@xmb-sjc-21c.amer.cisco.com>
Message-ID: <c6f32ce40708142042t598f4cfer824553f397e1a1eb@mail.gmail.com>

That sounded like too much of a good idea, so I had to go ahead and write
something to it. If anyone else is interested I have made it available on my
website at http://www.adman.net/cisco under IP Communicator Launcher. I
think it should work just fine, but as always, use at your own risk!

-Adam

On 8/14/07, Louis Marascio (lmarasci) <lmarasci at cisco.com> wrote:
>
>  IP Communicator stores their TFTP information in the registry:
> HK[CU,LM]\SOFTWARE\Cisco Systems, Inc.\Communicator. There are two keys,
> TFTPServer1 and TFTPServer2. I suspect you could write a script to change it
> automatically and launch IP Communicator for you.
>
> *cough*
> I am not on the IPC team and this is obviously not supported, yada yada
> yaaaadddddaaaaa....
> *cough*
>
> Louis
>
>  ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Adam Blomfield
> *Sent:* Tuesday, August 14, 2007 4:45 PM
> *To:* Jason Aarons (US)
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Phone Proxy and IPC
>
> I second the motion!
>
> On 8/14/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> >
> >  I wish Cisco IP Communicator had a profile option at startup, allowing
> > you to change your TFTP server easily.
> >
> >
> >
> >  I have to change my tftp servers for each customer, for our phone
> > proxy, and for when I'm at our office. I have to keep a txt listing of all
> > the different tftp servers at sites, etc.
> >
> >
> >
> > *From:* cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net]
> > *On Behalf Of *Scott Voll
> > *Sent:* Tuesday, August 14, 2007 5:24 PM
> > *To:* cisco-voip at puck.nether.net
> > * Subject:* [cisco-voip] Phone Proxy
> >
> >
> >
> > Hey--
> >
> >
> >
> > we just got our Phone Proxy setup and it works pretty slick.
> >
> >
> >
> > I think it only took about 4 hours to unpackage, configure, and get the
> > first Phone up and working.
> >
> >
> >
> > I do however hate Cisco docs as it states you need a null modem.  so I
> > look high and low and finally find one to find out that in all reality you
> > really don't need one.  Got to love that.
> >
> >
> >
> > Scott
> >
> > ------------------------------
> >
> > *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 Matthew.Hall at inxi.com  Wed Aug 15 00:08:09 2007
From: Matthew.Hall at inxi.com (Matthew Hall)
Date: Tue, 14 Aug 2007 23:08:09 -0500
Subject: [cisco-voip]  IPCC Music On Hold with Option to VM
Message-ID: <50AC803C707E0E48883F0FC9E7A86E32084F09C7@lewinxex01.InetX.com>

If you really must do this, I think I know a way.

 

Do the following:

 

Place a tag "Start Hold Loop"

Place Play Prompt "thanks for holding" or whatever

Place a Get Digit String set for interruptible, Prompt wave is your hold
music (long as you want), Barge in True, Continue on prompt error True,
initial timout 0, retries 0. Filter allows any digit, Input length 1.  

            On successful do your call to VM

            On Timeout do a Goto Label "Start Hold Loop"

            On  Unsuccessful do a Goto Label "Start Hold Loop"

 

Should work fine.  The Play prompt that says "thanks for holding" is
just there to make it sound less awkward if the prompt ends and loops
around.

 

Make sure you have an on exception step setup in your script to end on
ContactInactiveException as always.

 

Hope this helps

 

Matt

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From jason.aarons at us.didata.com  Wed Aug 15 01:18:54 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 15 Aug 2007 01:18:54 -0400
Subject: [cisco-voip] OS 2003.1.2a upgrade and BAD_POOL_HEADER blue screens
	death
Message-ID: <C1FE15183DA37645BC0633BC604E44F00671AFD4@USNAEXCH.na.didata.local>

After upgrading OS 2003.1.1 to win-OS-Upgrade-K9.2003-1-2a.exe I'm now
getting Blue Screens of Death with BAD_POOL_HEADER bug check has a value
of 0x00000019. Services are not starting (eg www). Http://ip/appadmin
fails as the CRS Node service is terminating unexpectedly.  Happening on
two separate boxes.   Searches on Google indicate bad device driver.
Anyone else seen this? My hardware is DL-380 G5.  I'm lucky the
equipment is still in testing phase.

 

DL-380 G5

UCCX 5.0(1)

 

 




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From myciscovoip at gmail.com  Wed Aug 15 02:52:05 2007
From: myciscovoip at gmail.com (Patrick Fischer)
Date: Wed, 15 Aug 2007 08:52:05 +0200
Subject: [cisco-voip] changing caller id
In-Reply-To: <254032.59415.qm@web52305.mail.re2.yahoo.com>
References: <1ff12970708140444t3a5c4bke271b05fff43c83d@mail.gmail.com>
	<254032.59415.qm@web52305.mail.re2.yahoo.com>
Message-ID: <fb5f789e0708142352p5a4d5249w9b85737d9d9a8b69@mail.gmail.com>

Hi Mani,

simply add another voice translation rule containing the translations you
want to have and add "translate calling <id>" to your voice
translation-profile profile2. That's it.

Regards,
Patrick


2007/8/14, Pezhman Lali <pezhman_lali at yahoo.com>:
>
> dear
> thanks for your reply.
> the config is:
>
> !
> version 12.2
> service timestamps debug uptime
> service timestamps log uptime
> no service password-encryption
> !
> hostname test
> !
> enable password 111111
> !
> username admin password 0 111111
> voice-card 1
> !
> aaa new-model
> !
> !
> aaa authentication login default local
> aaa authentication login vty local
> aaa authentication login h323 group radius
> aaa authentication login con none
> aaa authorization exec default local
> aaa authorization exec vty local
> aaa authorization exec h323 group radius
> aaa accounting connection h323 start-stop group radius
> aaa session-id common
> ip subnet-zero
> !
> !
> no ip domain lookup
> ip name-server 4.2.2.4
> !
> isdn switch-type primary-net5
> !
> !
> voice rtp send-recv
> !
> voice class codec 10
> codec preference 1 g723r53
> codec preference 2 g723r63
> codec preference 3 g723ar53
> codec preference 4 g723ar63
> codec preference 6 g729br8
> !
> !
> !
> !
> !
> !
> voice class custom-cptone test
> dualtone busy
>   frequency 425 425
>   cadence 500 500
> dualtone ringback
>   frequency 425 425
>   cadence 1000 4000
> dualtone reorder
>   frequency 425 425
>   cadence 1000 1000
> dualtone out-of-service
>   frequency 950
>   cadence 330 330
> dualtone number-unobtainable
>   frequency 425
>   cadence 240 240
> dualtone disconnect
>   frequency 425 425
>   cadence 250 250
> !
> !
> !
> !
> voice translation-rule 1
> rule 1 /^2/ /2197121112/
> rule 2 /^1/ /2197121111/
> rule 3 /^3/ /2197121113/
> rule 4 /^4/ /2197121114/
> rule 5 /^5/ /2197121115/
> rule 6 /^6/ /2197121116/
> rule 7 /^7/ /2197121117/
> rule 8 /^8/ /2197121118/
> rule 9 /^9/ /2197121119/
> rule 10 /^0/ /2197121110/
> !
> voice translation-rule 2
> rule 1 /^25221/ //
> rule 2 /^252/ /0/
> !
> !
> voice translation-profile profile1
> translate called 1
> !
> voice translation-profile profile2
> translate called 2
> !
> !
> mta receive maximum-recipients 0
> !
> controller E1 1/0
> framing NO-CRC4
> clock source line primary
> line-termination 75-ohm
> pri-group timeslots 1-31
> !
> controller E1 1/1
> shutdown
> framing NO-CRC4
> line-termination 75-ohm
> !
> translation-rule 1
> !
> translation-rule 2
> !
> !
> !
> !
> interface FastEthernet0/0
> ip address 85.1.1.2 255.255.255.0
> duplex auto
> speed auto
> !
> interface Serial1/0:15
> no ip address
> no logging event link-status
> isdn switch-type primary-net5
> isdn incoming-voice voice
> no cdp enable
> !
> ip default-gateway 85.1.1.1
> ip classless
> ip route 0.0.0.0 0.0.0.0 85.1.1.1
> no ip http server
> ip pim bidir-enable
> !
> !
> access-list 102 deny   tcp any any eq 4444
> access-list 102 deny   tcp any any eq 593
> access-list 102 deny   tcp any any eq 445
> access-list 102 deny   tcp any any eq 139
> access-list 102 deny   tcp any any eq 135
> access-list 102 deny   udp any any eq tftp
> access-list 102 deny   udp any any eq netbios-dgm
> access-list 102 deny   udp any any eq 135
> access-list 102 permit tcp any any eq telnet
> access-list 102 permit ip any any
> access-list 102 permit tcp any any
> access-list 102 permit udp any any
> !
>
> radius-server host 85.1.1.3 auth-port 1812 acct-port
> 1813 key test
> radius-server authorization permit missing
> Service-Type
> radius-server vsa send accounting
> radius-server vsa send authentication
> call rsvp-sync
> !
> call application voice cidial flash:cidial.tcl
> call application voice cidial language 0 en
> call application voice cidial set-location fa 0 flash:
> !
> voice-port 1/0:15
> !
> !
> mgcp profile default
> !
> !
> !
> dial-peer cor custom
> !
> !
> !
> dial-peer voice 900 voip
> session protocol sipv2
> session target dns:did.test.com
> dtmf-relay rtp-nte
> !
> dial-peer voice 100 pots
> translation-profile outgoing profile2
> destination-pattern 252T
> clid restrict
> port 1/0:15
> !
> dial-peer voice 1112 pots
> !
> dial-peer voice 1111 pots
> !
> dial-peer voice 110 pots
> incoming called-number T
> direct-inward-dial
> !
> sip-ua
> sip-server ipv4:217.10.79.219
> !
> !
> line con 0
> line aux 0
> line vty 0 4
> password 11111
> !
> !
> end
>
> --- zohaib shabir <zohaibshabir at gmail.com> wrote:
>
> > Yes
> > you can always change the callerid of calls if you
> > have the PRI line for the
> > pstn. I dont know you config, but i am sure you if
> > you share the config with
> > me, i will be able to help you out.
> >
> > On 8/14/07, Pezhman Lali <pezhman_lali at yahoo.com>
> > wrote:
> > >
> > > Hi dears
> > > I have a cisco2600+nm-hdv+pri line(E1).
> > > is any way to change callerid of outgoing calls to
> > the
> > > pstn.
> > > best
> > > Mani
> > >
> > >
> > >
> > >
> > >
> >
>
> ____________________________________________________________________________________
> > > 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
> > > _______________________________________________
> > > 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
> >
>
>
>
>
> ____________________________________________________________________________________Ready
> for the edge of your seat?
> Check out tonight's top picks on Yahoo! TV.
> http://tv.yahoo.com/
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From cisco at cips.nl  Wed Aug 15 03:40:28 2007
From: cisco at cips.nl (CIPS)
Date: Wed, 15 Aug 2007 09:40:28 +0200
Subject: [cisco-voip] FW:  3rd Party Import Tool
Message-ID: <016501c7df0f$8d7b73d0$a8725b70$@nl>

With CCM 3 and 4 you can use basic SQL programming to manipulate the Dbase.
With CCM 5 and 6 you have to use an AXL SOAP tool. I came across a tool
called AXLBridge, think their site is www.nzice.com. It was a little bit
complicated to get it working and I don't know if the tool is still
available and if it supports CCM 6.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Adam
Sent: dinsdag 14 augustus 2007 23:50
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 3rd Party Import Tool

 

I know a while back I ran across a third party tool that let you import more
than just the options in BAT... stuff like CSS and route patterns. I just
tried Googling for it, but can't remember the name of it and didn't have
much luck finding anything. Does anyone know the utility I'm talking about,
or am I just crazy and imagining things? 

Thanks,
Adam

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From Simon_Jones at mondial-assistance.co.uk  Wed Aug 15 05:53:48 2007
From: Simon_Jones at mondial-assistance.co.uk (Simon Jones)
Date: Wed, 15 Aug 2007 10:53:48 +0100
Subject: [cisco-voip] MoH - Sound Quality
Message-ID: <3273B98003B44648B4F256E759A2F36202F8E950@CROPAPP01.mondial.internal>

Hi,

 

Has anyone experienced this issue before? I have been tasked with
updating our existing MoH (currently classical music). I place the audio
files to be converted into the relevant directory in CCM and the files
are converted with no errors.

 

When listening to the new audio source I have created it sounds fine
when I am put on hold from  an internal IP Phone, analogue/POTS line at
home. However when I call from a mobile/cell phone and put myself on
hold the sound quality is very poor - a lot of background noise/hiss. I
have tried call from mobile phones on alternate networks and they all
have the same quailty problem. 

 

CCM 4.1(3) ES110

Product : Cisco Catalyst 6000 E1 VoIP Gateway

Device Protocol: Digital Access PRI

 

I have also tried a variety of music styles and also experimented with
the original file being in MP3 or WAV format, but still get this same
issue.

 

Any thoughts as to what could be the cause or hopefully the solution to
this issue?

 

 

Many Thanks,

 

Simon



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From ralf.eckhardt at web.de  Wed Aug 15 06:07:49 2007
From: ralf.eckhardt at web.de (Ralf)
Date: Wed, 15 Aug 2007 12:07:49 +0200
Subject: [cisco-voip] Unity 4.2.1 - Domino 7 --- MWI Delivery doesnt work as
	expected
Message-ID: <f9uj9n$fsj$1@sea.gmane.org>

Hi All,
I need some help/ suggestions maintaining this customers environment.
If a new message arrives at unity, it is recorded and delivered to the 
domino message store. Notes client shows the new message and it is 
accessable. Also possible is using the phone to play and delete the message.
But, MWI is not showing if a message is arrived od deleted!
If I manually do a refresh at the subscribers Message page, MWI will 
indicate the right status on or off related to the real message status.
The Unity port is dialling the extension.
 >>>But its not working automatically as it should.<<< just with this 
manual refresh
When the new message is delivered, Unity is expected to monitor this and 
  dial out the users extension. This is not happening! I compared this 
process with our companys Unity and there it's working that way.
The Unity process expected to do this is the Domino Monitor->Notifier-> 
Notifier.
How can I verify the delivery process using and analysing Micro Traces? 
The traces are already enebled.
I think the technical aspect is still possible (because of the manual 
delivery request), but something on the Domino monitoring is still wrong.
How can I veryfy that in more details?

Any suggestions?

Kind regards Ralf



From patrick.diener at gmail.com  Wed Aug 15 07:16:04 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Wed, 15 Aug 2007 13:16:04 +0200
Subject: [cisco-voip] CM6 CallManager service will not stay running
In-Reply-To: <559959.67447.qm@web36114.mail.mud.yahoo.com>
References: <559959.67447.qm@web36114.mail.mud.yahoo.com>
Message-ID: <c0deb5b80708150416p3a8ea887l1c3b1a23238196e@mail.gmail.com>

I ran into this once... the installation was done by someone not very
experienced in setting up linux based ccms so he used some strange
settings like DHCP enabled (meaning CM gets its IP from a DHCP Server
instead of setting it statically), DNS enabled etc. after
reinstallation with the "correct" settings it was working just fine...

anything you specified during installation you didn't know what it does?!?

Regards Patrick

On 8/15/07, Ted Nugent <tednugent69 at yahoo.com> wrote:
> Hey Guys
> I've got a FRESH install of CM 6.01 which I've
> attempted on both VMWare and a legitimate MCS and both
> are reacting the same way. After the install is
> complete I go to activate the "default" services and
> the CM service runs for about 30 seconds and bombs all
> other services remain running fine. I've read that
> this is expected behavior on an upgrade however this
> is a fresh install from the ground up so it SHOULD use
> the "starter license"?? What am I missing? Any one
> else run into this?
>
> >From 6.01 Release Notes:
> Starter Licenses
> Cisco Unified Communications Manager comes
> preinstalled with starter licenses, so a file upload
> is not required. Starter licenses have no expiration
> date and are available in limited quantities. Be aware
> that starter licenses are only available for fresh
> installations; they are not available for upgrades or
> migrations from previous releases. Starter licenses
> support only one Cisco Unified Communications Manager
> node and up to 50 device license units.
>
>
> Thanks
> Ted
>
>
>
> ____________________________________________________________________________________
> 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 pwalenta at wi.rr.com  Wed Aug 15 07:29:29 2007
From: pwalenta at wi.rr.com (pwalenta at wi.rr.com)
Date: Wed, 15 Aug 2007 06:29:29 -0500
Subject: [cisco-voip] Unity 4.2.1 - Domino 7 --- MWI Delivery doesnt
 work as expected
In-Reply-To: <f9uj9n$fsj$1@sea.gmane.org>
References: <f9uj9n$fsj$1@sea.gmane.org>
Message-ID: <c4d590b693658.93658c4d590b6@rdc-kc.rr.com>

Unity and Notes have a long and bad history with MWI.  

There are at least a dozen bugs opened against this.  Your best bet is
going to be to call Cisco TAC and see what the latest fixes are for this.

----- Original Message -----
From: Ralf <ralf.eckhardt at web.de>
Date: Wednesday, August 15, 2007 5:20 am
Subject: [cisco-voip] Unity 4.2.1 - Domino 7 --- MWI Delivery doesnt
work as expected
To: cisco-voip at puck.nether.net

> Hi All,
> I need some help/ suggestions maintaining this customers environment.
> If a new message arrives at unity, it is recorded and delivered to 
> the 
> domino message store. Notes client shows the new message and it is 
> accessable. Also possible is using the phone to play and delete the 
> message.But, MWI is not showing if a message is arrived od deleted!
> If I manually do a refresh at the subscribers Message page, MWI 
> will 
> indicate the right status on or off related to the real message 
> status.The Unity port is dialling the extension.
> >>>But its not working automatically as it should.<<< just with 
> this 
> manual refresh
> When the new message is delivered, Unity is expected to monitor 
> this and 
>  dial out the users extension. This is not happening! I compared 
> this 
> process with our companys Unity and there it's working that way.
> The Unity process expected to do this is the Domino Monitor-
> >Notifier-> 
> Notifier.
> How can I verify the delivery process using and analysing Micro 
> Traces? 
> The traces are already enebled.
> I think the technical aspect is still possible (because of the 
> manual 
> delivery request), but something on the Domino monitoring is still 
> wrong.How can I veryfy that in more details?
> 
> Any suggestions?
> 
> Kind regards Ralf
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From tednugent69 at yahoo.com  Wed Aug 15 08:28:08 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 15 Aug 2007 05:28:08 -0700 (PDT)
Subject: [cisco-voip] CM6 CallManager service will not stay running
In-Reply-To: <c0deb5b80708150416p3a8ea887l1c3b1a23238196e@mail.gmail.com>
Message-ID: <22420.12145.qm@web36106.mail.mud.yahoo.com>

Hmm, Thanks for the input but I think everything there
should be fine unless it requires NTP which is
currently disabled... I've installed CM5.x many times
with the same settings and have never run into this
issues. 


--- Patrick Diener <patrick.diener at gmail.com> wrote:

> I ran into this once... the installation was done by
> someone not very
> experienced in setting up linux based ccms so he
> used some strange
> settings like DHCP enabled (meaning CM gets its IP
> from a DHCP Server
> instead of setting it statically), DNS enabled etc.
> after
> reinstallation with the "correct" settings it was
> working just fine...
> 
> anything you specified during installation you
> didn't know what it does?!?
> 
> Regards Patrick
> 
> On 8/15/07, Ted Nugent <tednugent69 at yahoo.com>
> wrote:
> > Hey Guys
> > I've got a FRESH install of CM 6.01 which I've
> > attempted on both VMWare and a legitimate MCS and
> both
> > are reacting the same way. After the install is
> > complete I go to activate the "default" services
> and
> > the CM service runs for about 30 seconds and bombs
> all
> > other services remain running fine. I've read that
> > this is expected behavior on an upgrade however
> this
> > is a fresh install from the ground up so it SHOULD
> use
> > the "starter license"?? What am I missing? Any one
> > else run into this?
> >
> > >From 6.01 Release Notes:
> > Starter Licenses
> > Cisco Unified Communications Manager comes
> > preinstalled with starter licenses, so a file
> upload
> > is not required. Starter licenses have no
> expiration
> > date and are available in limited quantities. Be
> aware
> > that starter licenses are only available for fresh
> > installations; they are not available for upgrades
> or
> > migrations from previous releases. Starter
> licenses
> > support only one Cisco Unified Communications
> Manager
> > node and up to 50 device license units.
> >
> >
> > Thanks
> > Ted
> >
> >
> >
> >
>
____________________________________________________________________________________
> > 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
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________
Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase.
http://farechase.yahoo.com/

From lmarasci at cisco.com  Wed Aug 15 09:41:04 2007
From: lmarasci at cisco.com (Louis Marascio (lmarasci))
Date: Wed, 15 Aug 2007 06:41:04 -0700
Subject: [cisco-voip] Phone Proxy
In-Reply-To: <f84a38d30708141504q3928e913g69a7f41bcc5e7023@mail.gmail.com>
Message-ID: <6EE7932CC71C8D49A20C0E639BCD216502764643@xmb-sjc-21c.amer.cisco.com>

Thanks Scott, I'll get it fixed. Sorry for the trouble.
 
Louis

________________________________

From: Scott Voll [mailto:svoll.voip at gmail.com] 
Sent: Tuesday, August 14, 2007 5:04 PM
To: Louis Marascio (lmarasci)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Proxy


"Cisco Unified PhoneProxy installation and quick start guide" release
1.0(1)
 
PDF page 19 under procedure step 1.
 
Thanks
 
Scott

 
On 8/14/07, Louis Marascio (lmarasci) <lmarasci at cisco.com> wrote: 

	Scott, 
	 
	What doc does it talk about a null modem? I'll try to get it
fixed. Sorry for the trouble.
	 
	Louis

________________________________

	From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Scott Voll
	Sent: Tuesday, August 14, 2007 4:24 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Phone Proxy
	
	 
	
	Hey--
	 
	we just got our Phone Proxy setup and it works pretty slick.
	 
	I think it only took about 4 hours to unpackage, configure, and
get the first Phone up and working.
	 
	I do however hate Cisco docs as it states you need a null modem.
so I look high and low and finally find one to find out that in all
reality you really don't need one.  Got to love that.
	 
	Scott


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From cisco.voip at verizon.net  Wed Aug 15 10:09:37 2007
From: cisco.voip at verizon.net (cisco.voip)
Date: Wed, 15 Aug 2007 10:09:37 -0400
Subject: [cisco-voip] Voice-Gateway MGCP pri usage log
References: <mailman.11033.1187185277.1529.cisco-voip@puck.nether.net>
Message-ID: <001001c7df45$ebcfa560$661b1b0a@cs.myharris.net>

Does anyone know the mib I need for my 3845 voice gw to allow my snmp
manager to capture usage of the pri channels in and out on my PRI.



From cbomba at s4nets.com  Wed Aug 15 10:37:32 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Wed, 15 Aug 2007 10:37:32 -0400 (EDT)
Subject: [cisco-voip] changing caller id
Message-ID: <45131.192.168.1.71.1187188652.webmail@192.168.1.71>

To me it looks like you want to strip the "252" going out and prefix a "0".

Test your voice translation rule with the command below

test voice translation-rule 2 252XXXX and see if it gives you 0XXXX.  That is what it should give you when looking at:

voice translation-rule 2
 rule 1 /^25221/ //
 rule 2 /^252/ /0/

if you use |  test voice translation-rule 2 252221XXXX then you should get XXXX.  Now this is outgoing on your POTS dial-peer which will have you sending only 5 digits out to the PSTN or might only send 4 out to the PSTN.  And since you have it as "called" then you are translating your called number not your CLID.

If you want to translate your "calling" number you have to change the follow:

voice translation-rule 2
 rule 1 /^25221/ //
 rule 2 /^252/ /0/

voice translation-profile profile2
 translate called 2       <----------------should be translate calling 2

This would translate your CLID out to the PSTN to 4 digits if your calling number is 252221 or it will replace the 252 at the beginning of the calling number to 0.

Chris

|||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||

Date: Tue, 14 Aug 2007 06:43:56 -0700 (PDT)
From: Pezhman Lali <pezhman_lali at yahoo.com>
Subject: Re: [cisco-voip] changing caller id
To: zohaib shabir <zohaibshabir at gmail.com>
Cc: cisco-voip at puck.nether.net
Message-ID: <254032.59415.qm at web52305.mail.re2.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1

dear 
thanks for your reply.
the config is:

!
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname test
!
enable password 111111
!
username admin password 0 111111
voice-card 1
!
aaa new-model
!
!
aaa authentication login default local
aaa authentication login vty local
aaa authentication login h323 group radius
aaa authentication login con none
aaa authorization exec default local 
aaa authorization exec vty local 
aaa authorization exec h323 group radius 
aaa accounting connection h323 start-stop group radius
aaa session-id common
ip subnet-zero
!
!
no ip domain lookup
ip name-server 4.2.2.4
!
isdn switch-type primary-net5
!
!
voice rtp send-recv
!
voice class codec 10
 codec preference 1 g723r53
 codec preference 2 g723r63
 codec preference 3 g723ar53
 codec preference 4 g723ar63
 codec preference 6 g729br8
!
!
!
!
!
!
voice class custom-cptone test
 dualtone busy
  frequency 425 425
  cadence 500 500
 dualtone ringback
  frequency 425 425
  cadence 1000 4000
 dualtone reorder
  frequency 425 425
  cadence 1000 1000
 dualtone out-of-service
  frequency 950
  cadence 330 330
 dualtone number-unobtainable
  frequency 425
  cadence 240 240
 dualtone disconnect
  frequency 425 425
  cadence 250 250
!
!
!
!
voice translation-rule 1
 rule 1 /^2/ /2197121112/
 rule 2 /^1/ /2197121111/
 rule 3 /^3/ /2197121113/
 rule 4 /^4/ /2197121114/
 rule 5 /^5/ /2197121115/
 rule 6 /^6/ /2197121116/
 rule 7 /^7/ /2197121117/
 rule 8 /^8/ /2197121118/
 rule 9 /^9/ /2197121119/
 rule 10 /^0/ /2197121110/
!
voice translation-rule 2
 rule 1 /^25221/ //
 rule 2 /^252/ /0/
!
!
voice translation-profile profile1
 translate called 1
!
voice translation-profile profile2
 translate called 2
!
!
mta receive maximum-recipients 0
!
controller E1 1/0
 framing NO-CRC4 
 clock source line primary
 line-termination 75-ohm
 pri-group timeslots 1-31
!
controller E1 1/1
 shutdown
 framing NO-CRC4 
 line-termination 75-ohm
!
translation-rule 1
!
translation-rule 2
!
!
!
!
interface FastEthernet0/0
 ip address 85.1.1.2 255.255.255.0
 duplex auto
 speed auto
!
interface Serial1/0:15
 no ip address
 no logging event link-status
 isdn switch-type primary-net5
 isdn incoming-voice voice
 no cdp enable
!
ip default-gateway 85.1.1.1
ip classless
ip route 0.0.0.0 0.0.0.0 85.1.1.1
no ip http server
ip pim bidir-enable
!
!
access-list 102 deny   tcp any any eq 4444
access-list 102 deny   tcp any any eq 593
access-list 102 deny   tcp any any eq 445
access-list 102 deny   tcp any any eq 139
access-list 102 deny   tcp any any eq 135
access-list 102 deny   udp any any eq tftp
access-list 102 deny   udp any any eq netbios-dgm
access-list 102 deny   udp any any eq 135
access-list 102 permit tcp any any eq telnet
access-list 102 permit ip any any
access-list 102 permit tcp any any
access-list 102 permit udp any any
!

radius-server host 85.1.1.3 auth-port 1812 acct-port
1813 key test
radius-server authorization permit missing
Service-Type
radius-server vsa send accounting
radius-server vsa send authentication
call rsvp-sync
!
call application voice cidial flash:cidial.tcl
call application voice cidial language 0 en
call application voice cidial set-location fa 0 flash:
!
voice-port 1/0:15
!
!
mgcp profile default
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 900 voip
 session protocol sipv2
 session target dns:did.test.com
 dtmf-relay rtp-nte
!
dial-peer voice 100 pots
 translation-profile outgoing profile2
 destination-pattern 252T
 clid restrict
 port 1/0:15
!
dial-peer voice 1112 pots
!
dial-peer voice 1111 pots
!
dial-peer voice 110 pots
 incoming called-number T
 direct-inward-dial
!
sip-ua 
 sip-server ipv4:217.10.79.219
!
!
line con 0
line aux 0
line vty 0 4
 password 11111
!
!
end

--- zohaib shabir <zohaibshabir at gmail.com> wrote:

> Yes
> you can always change the callerid of calls if you
> have the PRI line for the
> pstn. I dont know you config, but i am sure you if
> you share the config with
> me, i will be able to help you out.
> 
> On 8/14/07, Pezhman Lali <pezhman_lali at yahoo.com>
> wrote:
> >
> > Hi dears
> > I have a cisco2600+nm-hdv+pri line(E1).
> > is any way to change callerid of outgoing calls to
> the
> > pstn.
> > best
> > Mani

_


From wsisk at cisco.com  Wed Aug 15 10:43:46 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 15 Aug 2007 10:43:46 -0400
Subject: [cisco-voip] CM6 CallManager service will not stay running
In-Reply-To: <559959.67447.qm@web36114.mail.mud.yahoo.com>
References: <559959.67447.qm@web36114.mail.mud.yahoo.com>
Message-ID: <46C31122.9090206@cisco.com>

No alarm in the application syslog or CM traces when it shuts down? /wes

Ted Nugent wrote:
> Hey Guys
> I've got a FRESH install of CM 6.01 which I've
> attempted on both VMWare and a legitimate MCS and both
> are reacting the same way. After the install is
> complete I go to activate the "default" services and
> the CM service runs for about 30 seconds and bombs all
> other services remain running fine. I've read that
> this is expected behavior on an upgrade however this
> is a fresh install from the ground up so it SHOULD use
> the "starter license"?? What am I missing? Any one
> else run into this?
>
> >From 6.01 Release Notes:
> Starter Licenses
> Cisco Unified Communications Manager comes
> preinstalled with starter licenses, so a file upload
> is not required. Starter licenses have no expiration
> date and are available in limited quantities. Be aware
> that starter licenses are only available for fresh
> installations; they are not available for upgrades or
> migrations from previous releases. Starter licenses
> support only one Cisco Unified Communications Manager
> node and up to 50 device license units.
>
>
> Thanks
> Ted
>
>
>        
> ____________________________________________________________________________________
> 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 wsisk at cisco.com  Wed Aug 15 10:50:25 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 15 Aug 2007 10:50:25 -0400
Subject: [cisco-voip] OS 2003.1.2a upgrade and BAD_POOL_HEADER blue
 screens death
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00671AFD4@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00671AFD4@USNAEXCH.na.didata.local>
Message-ID: <46C312B1.2030002@cisco.com>

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From wsisk at cisco.com  Wed Aug 15 10:54:26 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 15 Aug 2007 10:54:26 -0400
Subject: [cisco-voip] CCM 5.1, unable to pull logs via RTMT..
In-Reply-To: <5d093f9a0708141432v166b5edcnb6c29ef1b14d7bf5@mail.gmail.com>
References: <5d093f9a0708141432v166b5edcnb6c29ef1b14d7bf5@mail.gmail.com>
Message-ID: <46C313A2.6020401@cisco.com>

RTMT bombs if it tries to touch a file and does not have permission.  
This was initially caused by CSCsh36576.  Open a TAC case so we can get 
in with remote account and correct file permissions.  RTMT limitation is 
documented in CSCsi33810 and partial workaround supplied in CSCsi80661 
whereby RTMT will continue collecting files even if error occurs on one 
file.

/Wes

Jonathan Charles wrote:
> I have a CCM 5.1 cluster, I can pull logs via RTMT from the publisher
> fine, but when I try from the subscriber, I get the following:
>
> "An Error Occurred while Processing Request"
>
> I am able to connect via RTMT and see performance stats (Virtual
> Memory, CPU, registered phones, etc...) I just can't view log files.
>
> Initially I couldn't even get into RTMT, so I restarted Cisco
> Servicability and I was able to get in... Any ideas what could be
> wrong here?
>
>
>
> 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 Aug 15 10:55:16 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 15 Aug 2007 10:55:16 -0400
Subject: [cisco-voip] Translation error
In-Reply-To: <f84a38d30708141421qef7d726vc4a215e02c054b82@mail.gmail.com>
References: <f84a38d30708141421qef7d726vc4a215e02c054b82@mail.gmail.com>
Message-ID: <46C313D4.2050909@cisco.com>

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From jason.aarons at us.didata.com  Wed Aug 15 11:01:31 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 15 Aug 2007 11:01:31 -0400
Subject: [cisco-voip] Followup > OS 2003.1.2a upgrade and BAD_POOL_HEADER
	blue screensdeath
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00671AFD4@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00671AFD4@USNAEXCH.na.didata.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00671B41F@USNAEXCH.na.didata.local>

Notes from my UCCX 5.0(1) deployment;

 

Issue 1;

At boot F9, HP BIOS, System Options, Power Regulator, HP Static High
Performance change to OS Controlled, this fixed the Blue Screen of
Death.

 

Issue 2;

OS Update changed the binding order, made the disabled Local Area
Connection 2 the first in the binding. This creates problems with UCCX
5.x. You have to re-run the Desktop Administrator and re-boot after
fixing the binding order.

 

Issue 3;

The OS update 2003.1.2a seems to have issues, java complains out of
memory in logs, Task Manager sometimes opens, Event viewer may not open,
etc. TAC is still re-searching the root cause, appears I may have to
re-build and keep at OS 2003.1.1 (yeah I should have broken the mirror,
but hey it was a new buildout).

 

Issue 4;

You have to use the Microsoft SQL Server 2000 for Customer Response
Solutions. You can NOT supply your own media (OS/SQL) for any CRS
component. You can't even use MS SQL SP4, you have to use all Cisco per
TAC. On boot you get a message the SQL version was not used from the
Cisco supplied discs if you try to supply your own SQL 2000.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Wednesday, August 15, 2007 1:19 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] OS 2003.1.2a upgrade and BAD_POOL_HEADER blue
screensdeath

 

After upgrading OS 2003.1.1 to win-OS-Upgrade-K9.2003-1-2a.exe I'm now
getting Blue Screens of Death with BAD_POOL_HEADER bug check has a value
of 0x00000019. Services are not starting (eg www). Http://ip/appadmin
fails as the CRS Node service is terminating unexpectedly.  Happening on
two separate boxes.   Searches on Google indicate bad device driver.
Anyone else seen this? My hardware is DL-380 G5.  I'm lucky the
equipment is still in testing phase.

 

DL-380 G5

UCCX 5.0(1)

 

 

________________________________

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From myciscovoip at gmail.com  Wed Aug 15 11:03:30 2007
From: myciscovoip at gmail.com (Patrick Fischer)
Date: Wed, 15 Aug 2007 17:03:30 +0200
Subject: [cisco-voip] changing caller id
In-Reply-To: <338c6fe20708150004h3c026523oe266ae4d7300d34d@mail.gmail.com>
References: <1ff12970708140444t3a5c4bke271b05fff43c83d@mail.gmail.com>
	<254032.59415.qm@web52305.mail.re2.yahoo.com>
	<fb5f789e0708142352p5a4d5249w9b85737d9d9a8b69@mail.gmail.com>
	<338c6fe20708150004h3c026523oe266ae4d7300d34d@mail.gmail.com>
Message-ID: <fb5f789e0708150803p35a77311pbd67711ee8f76382@mail.gmail.com>

Hi Shaan

It depends on your provider. Ususally you should use "clid restrict" in your
dial-peer config. What you can do using voice translation rules is to send
out nothing. (rule 1 /1234/ //) But in many cases the provider will then
show your main number.

Regards,
Patrick

2007/8/15, Shanawaz <shanawaz at gmail.com>:
>
> Patrick,
>
> Can I send an 'unknown' caller ID the same way?
>
> Thanks,
> Shaan
>
>
> On 8/15/07, Patrick Fischer <myciscovoip at gmail.com> wrote:
> >
> > Hi Mani,
> >
> > simply add another voice translation rule containing the translations
> > you want to have and add "translate calling <id>" to your voice
> > translation-profile profile2. That's it.
> >
> > Regards,
> > Patrick
> >
> >
> > 2007/8/14, Pezhman Lali <pezhman_lali at yahoo.com>:
> > >
> > > dear
> > > thanks for your reply.
> > > the config is:
> > >
> > > !
> > > version 12.2
> > > service timestamps debug uptime
> > > service timestamps log uptime
> > > no service password-encryption
> > > !
> > > hostname test
> > > !
> > > enable password 111111
> > > !
> > > username admin password 0 111111
> > > voice-card 1
> > > !
> > > aaa new-model
> > > !
> > > !
> > > aaa authentication login default local
> > > aaa authentication login vty local
> > > aaa authentication login h323 group radius
> > > aaa authentication login con none
> > > aaa authorization exec default local
> > > aaa authorization exec vty local
> > > aaa authorization exec h323 group radius
> > > aaa accounting connection h323 start-stop group radius
> > > aaa session-id common
> > > ip subnet-zero
> > > !
> > > !
> > > no ip domain lookup
> > > ip name-server 4.2.2.4
> > > !
> > > isdn switch-type primary-net5
> > > !
> > > !
> > > voice rtp send-recv
> > > !
> > > voice class codec 10
> > > codec preference 1 g723r53
> > > codec preference 2 g723r63
> > > codec preference 3 g723ar53
> > > codec preference 4 g723ar63
> > > codec preference 6 g729br8
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > voice class custom-cptone test
> > > dualtone busy
> > >   frequency 425 425
> > >   cadence 500 500
> > > dualtone ringback
> > >   frequency 425 425
> > >   cadence 1000 4000
> > > dualtone reorder
> > >   frequency 425 425
> > >   cadence 1000 1000
> > > dualtone out-of-service
> > >   frequency 950
> > >   cadence 330 330
> > > dualtone number-unobtainable
> > >   frequency 425
> > >   cadence 240 240
> > > dualtone disconnect
> > >   frequency 425 425
> > >   cadence 250 250
> > > !
> > > !
> > > !
> > > !
> > > voice translation-rule 1
> > > rule 1 /^2/ /2197121112/
> > > rule 2 /^1/ /2197121111/
> > > rule 3 /^3/ /2197121113/
> > > rule 4 /^4/ /2197121114/
> > > rule 5 /^5/ /2197121115/
> > > rule 6 /^6/ /2197121116/
> > > rule 7 /^7/ /2197121117/
> > > rule 8 /^8/ /2197121118/
> > > rule 9 /^9/ /2197121119/
> > > rule 10 /^0/ /2197121110/
> > > !
> > > voice translation-rule 2
> > > rule 1 /^25221/ //
> > > rule 2 /^252/ /0/
> > > !
> > > !
> > > voice translation-profile profile1
> > > translate called 1
> > > !
> > > voice translation-profile profile2
> > > translate called 2
> > > !
> > > !
> > > mta receive maximum-recipients 0
> > > !
> > > controller E1 1/0
> > > framing NO-CRC4
> > > clock source line primary
> > > line-termination 75-ohm
> > > pri-group timeslots 1-31
> > > !
> > > controller E1 1/1
> > > shutdown
> > > framing NO-CRC4
> > > line-termination 75-ohm
> > > !
> > > translation-rule 1
> > > !
> > > translation-rule 2
> > > !
> > > !
> > > !
> > > !
> > > interface FastEthernet0/0
> > > ip address 85.1.1.2 255.255.255.0
> > > duplex auto
> > > speed auto
> > > !
> > > interface Serial1/0:15
> > > no ip address
> > > no logging event link-status
> > > isdn switch-type primary-net5
> > > isdn incoming-voice voice
> > > no cdp enable
> > > !
> > > ip default-gateway 85.1.1.1
> > > ip classless
> > > ip route 0.0.0.0 0.0.0.0 85.1.1.1
> > > no ip http server
> > > ip pim bidir-enable
> > > !
> > > !
> > > access-list 102 deny   tcp any any eq 4444
> > > access-list 102 deny   tcp any any eq 593
> > > access-list 102 deny   tcp any any eq 445
> > > access-list 102 deny   tcp any any eq 139
> > > access-list 102 deny   tcp any any eq 135
> > > access-list 102 deny   udp any any eq tftp
> > > access-list 102 deny   udp any any eq netbios-dgm
> > > access-list 102 deny   udp any any eq 135
> > > access-list 102 permit tcp any any eq telnet
> > > access-list 102 permit ip any any
> > > access-list 102 permit tcp any any
> > > access-list 102 permit udp any any
> > > !
> > >
> > > radius-server host 85.1.1.3 auth-port 1812 acct-port
> > > 1813 key test
> > > radius-server authorization permit missing
> > > Service-Type
> > > radius-server vsa send accounting
> > > radius-server vsa send authentication
> > > call rsvp-sync
> > > !
> > > call application voice cidial flash:cidial.tcl
> > > call application voice cidial language 0 en
> > > call application voice cidial set-location fa 0 flash:
> > > !
> > > voice-port 1/0:15
> > > !
> > > !
> > > mgcp profile default
> > > !
> > > !
> > > !
> > > dial-peer cor custom
> > > !
> > > !
> > > !
> > > dial-peer voice 900 voip
> > > session protocol sipv2
> > > session target dns: did.test.com
> > > dtmf-relay rtp-nte
> > > !
> > > dial-peer voice 100 pots
> > > translation-profile outgoing profile2
> > > destination-pattern 252T
> > > clid restrict
> > > port 1/0:15
> > > !
> > > dial-peer voice 1112 pots
> > > !
> > > dial-peer voice 1111 pots
> > > !
> > > dial-peer voice 110 pots
> > > incoming called-number T
> > > direct-inward-dial
> > > !
> > > sip-ua
> > > sip-server ipv4: 217.10.79.219
> > > !
> > > !
> > > line con 0
> > > line aux 0
> > > line vty 0 4
> > > password 11111
> > > !
> > > !
> > > end
> > >
> > > --- zohaib shabir < zohaibshabir at gmail.com> wrote:
> > >
> > > > Yes
> > > > you can always change the callerid of calls if you
> > > > have the PRI line for the
> > > > pstn. I dont know you config, but i am sure you if
> > > > you share the config with
> > > > me, i will be able to help you out.
> > > >
> > > > On 8/14/07, Pezhman Lali < pezhman_lali at yahoo.com >
> > > > wrote:
> > > > >
> > > > > Hi dears
> > > > > I have a cisco2600+nm-hdv+pri line(E1).
> > > > > is any way to change callerid of outgoing calls to
> > > > the
> > > > > pstn.
> > > > > best
> > > > > Mani
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > >
> > >
> > > ____________________________________________________________________________________
> > > > > 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
> > > > > _______________________________________________
> > > > > 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
> > > >
> > >
> > >
> > >
> > >
> > > ____________________________________________________________________________________Ready
> > > for the edge of your seat?
> > > Check out tonight's top picks on Yahoo! TV.
> > > http://tv.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 tednugent69 at yahoo.com  Wed Aug 15 10:32:51 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 15 Aug 2007 07:32:51 -0700 (PDT)
Subject: [cisco-voip] Resolved: CM6 CallManager service will not stay running
In-Reply-To: <c0deb5b80708150416p3a8ea887l1c3b1a23238196e@mail.gmail.com>
Message-ID: <998828.80876.qm@web36115.mail.mud.yahoo.com>

Thanks Patrick you actually changed my train of
thought. I was convinced that it was some sort of
licensing issue however it was much simpler then that.
The Server name was still set to localhost I switched
it to the IP address, rebooted the CM and now the
service is running stable, that?s a new one on me but
I learn something new everyday I guess. Thank for the
redirect.

Ted


--- Patrick Diener <patrick.diener at gmail.com> wrote:

> I ran into this once... the installation was done by
> someone not very
> experienced in setting up linux based ccms so he
> used some strange
> settings like DHCP enabled (meaning CM gets its IP
> from a DHCP Server
> instead of setting it statically), DNS enabled etc.
> after
> reinstallation with the "correct" settings it was
> working just fine...
> 
> anything you specified during installation you
> didn't know what it does?!?
> 
> Regards Patrick
> 
> On 8/15/07, Ted Nugent <tednugent69 at yahoo.com>
> wrote:
> > Hey Guys
> > I've got a FRESH install of CM 6.01 which I've
> > attempted on both VMWare and a legitimate MCS and
> both
> > are reacting the same way. After the install is
> > complete I go to activate the "default" services
> and
> > the CM service runs for about 30 seconds and bombs
> all
> > other services remain running fine. I've read that
> > this is expected behavior on an upgrade however
> this
> > is a fresh install from the ground up so it SHOULD
> use
> > the "starter license"?? What am I missing? Any one
> > else run into this?
> >
> > >From 6.01 Release Notes:
> > Starter Licenses
> > Cisco Unified Communications Manager comes
> > preinstalled with starter licenses, so a file
> upload
> > is not required. Starter licenses have no
> expiration
> > date and are available in limited quantities. Be
> aware
> > that starter licenses are only available for fresh
> > installations; they are not available for upgrades
> or
> > migrations from previous releases. Starter
> licenses
> > support only one Cisco Unified Communications
> Manager
> > node and up to 50 device license units.
> >
> >
> > Thanks
> > Ted
> >
> >
> >
> >
>
____________________________________________________________________________________
> > 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
> >
> _______________________________________________
> 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 BrDuncan at jackhenry.com  Wed Aug 15 11:52:23 2007
From: BrDuncan at jackhenry.com (Brian Duncan)
Date: Wed, 15 Aug 2007 10:52:23 -0500
Subject: [cisco-voip] Attendant Console CCM 5.1 Call Control Failure
Message-ID: <2B3745BC9D31514381A6FCC5E72D937301F7A849@SMOMAIL.jhacorp.com>

Hi All,
 
I have been attempting to set up Attendant Console with CCM 5.1 and I keep getting "Initialization of Call Control Failed.  Retrying..."  I have configured the AC user and the ACDeviceAuthenticationUser user as Application Users. The phones have been associated to both users and both are in the Standard CTI Allow Control of All Devices and Standard CTI Allow Call Park Monitoring groups.  Both phones in question have Allow Control of Device from CTI on both the phone and the line.  
 
Does anyone know how to bring up call control?
 
Thanks,
Brian Duncan
Jack Henry & Associates
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
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From wsisk at cisco.com  Wed Aug 15 12:07:08 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 15 Aug 2007 12:07:08 -0400
Subject: [cisco-voip] Attendant Console CCM 5.1 Call Control Failure
In-Reply-To: <2B3745BC9D31514381A6FCC5E72D937301F7A849@SMOMAIL.jhacorp.com>
References: <2B3745BC9D31514381A6FCC5E72D937301F7A849@SMOMAIL.jhacorp.com>
Message-ID: <46C324AC.9030600@cisco.com>

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From wsisk at cisco.com  Wed Aug 15 12:31:52 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 15 Aug 2007 12:31:52 -0400
Subject: [cisco-voip] Attendant Console CCM 5.1 Call Control Failure
In-Reply-To: <2B3745BC9D31514381A6FCC5E72D937301F7A849@SMOMAIL.jhacorp.com>
References: <2B3745BC9D31514381A6FCC5E72D937301F7A849@SMOMAIL.jhacorp.com>
Message-ID: <46C32A78.60000@cisco.com>

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From jonvoip at gmail.com  Wed Aug 15 14:15:15 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 15 Aug 2007 13:15:15 -0500
Subject: [cisco-voip] CCM 5.1, unable to pull logs via RTMT..
In-Reply-To: <46C313A2.6020401@cisco.com>
References: <5d093f9a0708141432v166b5edcnb6c29ef1b14d7bf5@mail.gmail.com>
	<46C313A2.6020401@cisco.com>
Message-ID: <5d093f9a0708151115y4691aacak8e842dfffa46d492@mail.gmail.com>

Thanks, I will open a case later today


Jonathan

On 8/15/07, Wes Sisk <wsisk at cisco.com> wrote:
> RTMT bombs if it tries to touch a file and does not have permission.
> This was initially caused by CSCsh36576.  Open a TAC case so we can get
> in with remote account and correct file permissions.  RTMT limitation is
> documented in CSCsi33810 and partial workaround supplied in CSCsi80661
> whereby RTMT will continue collecting files even if error occurs on one
> file.
>
> /Wes
>
> Jonathan Charles wrote:
> > I have a CCM 5.1 cluster, I can pull logs via RTMT from the publisher
> > fine, but when I try from the subscriber, I get the following:
> >
> > "An Error Occurred while Processing Request"
> >
> > I am able to connect via RTMT and see performance stats (Virtual
> > Memory, CPU, registered phones, etc...) I just can't view log files.
> >
> > Initially I couldn't even get into RTMT, so I restarted Cisco
> > Servicability and I was able to get in... Any ideas what could be
> > wrong here?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > 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 Aug 15 15:55:46 2007
From: deadpool.slayer.telecomm at gmail.com (Andrew F. Wolfe)
Date: Wed, 15 Aug 2007 14:55:46 -0500
Subject: [cisco-voip] CUCM 5.X, ODBC, and CDR
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0066D8AFB@USNAEXCH.na.didata.local>
References: <913929690708131111w7e27e0d7k91d1f33b47b0a7a5@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0066D8AFB@USNAEXCH.na.didata.local>
Message-ID: <913929690708151255i7184e86al43b99932cc34bcad@mail.gmail.com>

Sorry, it took a bit to read this.  I understand what you are saying and
with some more research I found what I was looking for in the way of CDR.
As for no ODBC drivers for CUCM 5, I find that very hard to believe since
they include the AXL SQL toolkit and it runs from my understanding IBM
Informix SQL.  I could be wrong there.  Any DB folks out there who have
decided to delve into this realm?

On 8/13/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
>
>  No ODBC drivers for CallManager 5.x and up, CDR is all done via FTP flat
> file push/pulls.
>
>
>
>
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Andrew F. Wolfe
> *Sent:* Monday, August 13, 2007 2:11 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] CUCM 5.X, ODBC, and CDR
>
>
>
> Hello all.  I am trying to wrap my brain around the DB for CDR and CUCM.
> Since it is an Informix DB, I can't use the ODBC drivers included in
> Windows.  Any suggestions on an Informix ODBC driver and any tips would be
> greatly appreciated.  Please let me know.  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 wsisk at cisco.com  Wed Aug 15 16:05:52 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 15 Aug 2007 16:05:52 -0400
Subject: [cisco-voip] CUCM 5.X, ODBC, and CDR
In-Reply-To: <913929690708151255i7184e86al43b99932cc34bcad@mail.gmail.com>
References: <913929690708131111w7e27e0d7k91d1f33b47b0a7a5@mail.gmail.com>	<C1FE15183DA37645BC0633BC604E44F0066D8AFB@USNAEXCH.na.didata.local>
	<913929690708151255i7184e86al43b99932cc34bcad@mail.gmail.com>
Message-ID: <46C35CA0.6050405@cisco.com>

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From svoll.voip at gmail.com  Wed Aug 15 16:44:07 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 15 Aug 2007 13:44:07 -0700
Subject: [cisco-voip] asterisk for Voicemail and CM 5.1
Message-ID: <f84a38d30708151344i3f2f7659gba1e8c78d110b7cb@mail.gmail.com>

has anyone used asterisk for Voicemail with CM 5.1.2?

is it pretty easy to setup?  are there any big problems?  does MWI work?

I've never played with Asterisk so sorry for the bone head questions.
Current VM / PBX is having issues when using CM 5.1 so I'm looking for
options.

are there any cost involved? other then a server?

Thanks

Scott
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From jeff at ocjtech.us  Wed Aug 15 17:13:32 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Wed, 15 Aug 2007 16:13:32 -0500
Subject: [cisco-voip] asterisk for Voicemail and CM 5.1
In-Reply-To: <f84a38d30708151344i3f2f7659gba1e8c78d110b7cb@mail.gmail.com>
References: <f84a38d30708151344i3f2f7659gba1e8c78d110b7cb@mail.gmail.com>
Message-ID: <1187212412.4066.32.camel@lt21223.campus.dmacc.edu>

On Wed, 2007-08-15 at 13:44 -0700, Scott Voll wrote:
> has anyone used asterisk for Voicemail with CM 5.1.2?

Unfortunately not seriously.

> is it pretty easy to setup?

It can take a bit to figure out if you're completely unfamiliar with it.

>   are there any big problems?
>   does MWI work?  

It works for SIP phones that talk directly to Asterisk.  MWI isn't going
to work for phones attached to your CallManager.  But since Asterisk is
open-source it *could* work - it'll just take a little coding/scripting.
 
> I've never played with Asterisk so sorry for the bone head questions.
> Current VM / PBX is having issues when using CM 5.1 so I'm looking for
> options.
>  
> are there any cost involved? other then a server?

Asterisk is GPL software and runs best on Linux (although ISTR that
there are some crazies out there that have ported it to BSD and
Windows).  I'd recommend using one of the "enterprise" Linuxes like
RHEL, CentOS or SLES (I'm sure that there are Debian "enterprise" Linux
variants but I'm unfamiliar with them).  CentOS is a free derivative of
RHEL.  RHEL and SLES are going to cost you some money (but come with
correspondingly better support).

Jeff


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From thomas.hartley at gmail.com  Wed Aug 15 21:11:48 2007
From: thomas.hartley at gmail.com (Thomas Hartley)
Date: Thu, 16 Aug 2007 11:11:48 +1000
Subject: [cisco-voip] asterisk for Voicemail and CM 5.1
In-Reply-To: <f84a38d30708151344i3f2f7659gba1e8c78d110b7cb@mail.gmail.com>
References: <f84a38d30708151344i3f2f7659gba1e8c78d110b7cb@mail.gmail.com>
Message-ID: <db324e8c0708151811r20800844j6cd85b5ac096ba0c@mail.gmail.com>

I'll try again, to the list this time (new to Gmail with mailing lists !!)

Works quite well using over a SIP trunk.

We have used this from CCM 4.1 and upgraded all the way to 5.1 with no
changes.

MWI works flawlessly for all all phones.

The details I used are here:

http://www.hurdman.net/mirror/voip-info/wiki/view/Asterisk+Cisco+CallManager+Voicemail+Integration.html

Images weren't working when I checked but it should give you an idea.


TH



On 8/16/07, Scott Voll <svoll.voip at gmail.com> wrote:
>
> has anyone used asterisk for Voicemail with CM 5.1.2?
>
> is it pretty easy to setup?  are there any big problems?  does MWI work?
>
> I've never played with Asterisk so sorry for the bone head questions.
> Current VM / PBX is having issues when using CM 5.1 so I'm looking for
> options.
>
> are there any cost involved? other then a server?
>
> 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 cshaffer at gmail.com  Thu Aug 16 06:38:32 2007
From: cshaffer at gmail.com (Curt Shaffer)
Date: Thu, 16 Aug 2007 06:38:32 -0400
Subject: [cisco-voip] UC500
Message-ID: <003c01c7dff1$981a20b0$c84e6210$@com>

I can't find a user limit for this device. I see mention of 16 users but it
doesn't say limited to 16 users. Anyone have insight on this? Also is anyone
aware of redundancy options for this?

 

Thanks!

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From akoul at dataways.gr  Thu Aug 16 06:46:30 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 16 Aug 2007 13:46:30 +0300
Subject: [cisco-voip] UC500
In-Reply-To: <003c01c7dff1$981a20b0$c84e6210$@com>
References: <003c01c7dff1$981a20b0$c84e6210$@com>
Message-ID: <46C42B06.1020805@dataways.gr>

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From Matt.Slaga at us.didata.com  Thu Aug 16 08:45:27 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 16 Aug 2007 08:45:27 -0400
Subject: [cisco-voip] UC500
In-Reply-To: <46C42B06.1020805@dataways.gr>
References: <003c01c7dff1$981a20b0$c84e6210$@com>
	<46C42B06.1020805@dataways.gr>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00675829A@USNAEXCH.na.didata.local>

Cisco has stated 16 phones for this device.  I am not sure if that is based on LAN port count, from what I understand it is only capable of 8 ports internally and expansion to one CAT500.  I'm sure you could patch it into another switch, but I'm not sure if Cisco limited the IOS to not allow more than 16 devices.

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony Kouloglou
Sent: Thursday, August 16, 2007 6:47 AM
To: Curt Shaffer
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] UC500

 

There is no obvious user limit i guess. Besides, it is a CCME after all.
Any limitations apply to the hardware capability of the platform just like CCME. There is no license schema as in CiscoCallmanager 5.X and above.

BTW,
have you managed to make it work by not using CLI?!! :-) :-( 

Chhers,
Anthony

Curt Shaffer wrote: 

I can?t find a user limit for this device. I see mention of 16 users but it doesn?t say limited to 16 users. Anyone have insight on this? Also is anyone aware of redundancy options for this?

 

Thanks!

 


________________________________



 
_______________________________________________
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
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From jonvoip at gmail.com  Thu Aug 16 08:59:38 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 16 Aug 2007 07:59:38 -0500
Subject: [cisco-voip] Resolved: CM6 CallManager service will not stay
	running
In-Reply-To: <998828.80876.qm@web36115.mail.mud.yahoo.com>
References: <c0deb5b80708150416p3a8ea887l1c3b1a23238196e@mail.gmail.com>
	<998828.80876.qm@web36115.mail.mud.yahoo.com>
Message-ID: <5d093f9a0708160559s50a9056fk94c080e027fdfbb@mail.gmail.com>

Was this an upgrade of a 5 box, or a new install?

Did you get an upgrade license?


Jonathan

On 8/15/07, Ted Nugent <tednugent69 at yahoo.com> wrote:
> Thanks Patrick you actually changed my train of
> thought. I was convinced that it was some sort of
> licensing issue however it was much simpler then that.
> The Server name was still set to localhost I switched
> it to the IP address, rebooted the CM and now the
> service is running stable, that's a new one on me but
> I learn something new everyday I guess. Thank for the
> redirect.
>
> Ted
>
>
> --- Patrick Diener <patrick.diener at gmail.com> wrote:
>
> > I ran into this once... the installation was done by
> > someone not very
> > experienced in setting up linux based ccms so he
> > used some strange
> > settings like DHCP enabled (meaning CM gets its IP
> > from a DHCP Server
> > instead of setting it statically), DNS enabled etc.
> > after
> > reinstallation with the "correct" settings it was
> > working just fine...
> >
> > anything you specified during installation you
> > didn't know what it does?!?
> >
> > Regards Patrick
> >
> > On 8/15/07, Ted Nugent <tednugent69 at yahoo.com>
> > wrote:
> > > Hey Guys
> > > I've got a FRESH install of CM 6.01 which I've
> > > attempted on both VMWare and a legitimate MCS and
> > both
> > > are reacting the same way. After the install is
> > > complete I go to activate the "default" services
> > and
> > > the CM service runs for about 30 seconds and bombs
> > all
> > > other services remain running fine. I've read that
> > > this is expected behavior on an upgrade however
> > this
> > > is a fresh install from the ground up so it SHOULD
> > use
> > > the "starter license"?? What am I missing? Any one
> > > else run into this?
> > >
> > > >From 6.01 Release Notes:
> > > Starter Licenses
> > > Cisco Unified Communications Manager comes
> > > preinstalled with starter licenses, so a file
> > upload
> > > is not required. Starter licenses have no
> > expiration
> > > date and are available in limited quantities. Be
> > aware
> > > that starter licenses are only available for fresh
> > > installations; they are not available for upgrades
> > or
> > > migrations from previous releases. Starter
> > licenses
> > > support only one Cisco Unified Communications
> > Manager
> > > node and up to 50 device license units.
> > >
> > >
> > > Thanks
> > > Ted
> > >
> > >
> > >
> > >
> >
> ____________________________________________________________________________________
> > > 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
> > >
> > _______________________________________________
> > 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/
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From AHolloway at enventis.com  Thu Aug 16 08:39:23 2007
From: AHolloway at enventis.com (Anthony Holloway (Enventis))
Date: Thu, 16 Aug 2007 07:39:23 -0500
Subject: [cisco-voip] caller name with H.323
In-Reply-To: <7aaaf7f0708140821w6350f910j77274a86a643d76f@mail.gmail.com>
References: <7aaaf7f0708140821w6350f910j77274a86a643d76f@mail.gmail.com>
Message-ID: <C0A504750F3BBE4FADE4ED6D672BA1790EEBFED7@mail.hickorytech.local>

I had this same question with Cisco TAC, and here is their response.
(Note that we're talking about having a PRI with switch type NI2)

 

            The "isdn supp-service name calling" command does enable
receipt of the calling name in the facility ie,

but as of CME 3.3 we could only deliver that to a SIP call leg.  CME 4.0
introduced the ability to deliver

that calling name to CME phones.Therefore the caller name will work with
CME 4.0 for sccp CME phones.

 

These are the IOS releases compatible with CME 4.0 and up.

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps4625/prod_insta
llation_guide09186a00805acf50.html

 

4.2        12.4(11)XW2

 

4.1        12.4(15)T 5, 7

            

4.1        12.4(11)XJ

 

4.0(2)    12.4(11)T 5

 

4.0(3)    12.4(4)XC4

 

4.0(0)    12.4(9)T 6

 

4.0(1)    12.4(4)XC

 

Anthony

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian
Sent: Tuesday, August 14, 2007 10:22 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] caller name with H.323

 

I see H.323 name display is supported as of 12.4(11)XW.  Does anyone
know if it is also supported on some level of the T train?  I'm running
12.4(15)T1 and CUCM 5.1(2), but I don't see the display-ie or
supp-service commands on the gateway. 

Thanks,
Brian



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From ealeatherman at gmail.com  Thu Aug 16 09:19:23 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 16 Aug 2007 09:19:23 -0400
Subject: [cisco-voip] H323 gateway question/problem
Message-ID: <94a1afde0708160619s3e8d28c6h9758c43afa47aff4@mail.gmail.com>

Hi Folks,

I'm pretty rusty on my H.323 gateways and dial peers etc, running into a
problem with a configuration in my lab that i'm hoping to use this Fall for
a project - was hoping someone might have an idea.

Goal here is to have a router with CME setup for "SRST Mode" and a number of
POTS lines in from pstn. Phones normally registered to callmanager (version
4.2 in my lab). During normal operation, if any of those lines ring, it
should just connect to call manager and ring a shared line appearance on X
number of phones at the location (probably about 8). During SRST operation,
essentially I want to do the same thing - using overlays etc.

I have the CME up and seems to work OK. The problem I have is during normal
operation. Any 1 line can ring in and it gets sent to call manager and the
correct line rings on all phones. Any other line that rings in after that
appears to get sent to call manager, the line rings once, and then the call
drops. It seems to me that perhaps something in the feature negotiation
phase is failing, since the call rings out once before failing. I dont have
any CAC setup, and its all using G.711ulaw. I don't know enough about H323
to know where to look next though - nothing in the debugs I looked at jumped
out at me as an error.

I did watch the debug for dial-peer matching and it looks to be matching my
dial-peer out to CCM before it matches the ephone-dn dial peers

router is basically setup like this:

voice-port 0/0/0
 connection plar 37111
voice-port 0/0/1
 connection plar 37111
voice-port 0/0/2
 connection plar 37111
!
dial-peer voice 100 voip
 destination-pattern 37111$
 session target ipv4:10.182.140.10  ! Call Manager 4.2
 no vad
 codec g711ulaw
!

I found that if I dont put the end of string marker in my destination
pattern, it likes to match the ephone-dn's before it matches the callmanager
dial-peer.

Thanks!
-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From AHolloway at enventis.com  Thu Aug 16 09:35:56 2007
From: AHolloway at enventis.com (Anthony Holloway (Enventis))
Date: Thu, 16 Aug 2007 08:35:56 -0500
Subject: [cisco-voip] caller name with H.323
In-Reply-To: <C0A504750F3BBE4FADE4ED6D672BA1790EEBFED7@mail.hickorytech.local>
References: <7aaaf7f0708140821w6350f910j77274a86a643d76f@mail.gmail.com>
	<C0A504750F3BBE4FADE4ED6D672BA1790EEBFED7@mail.hickorytech.local>
Message-ID: <C0A504750F3BBE4FADE4ED6D672BA1790EEBFF68@mail.hickorytech.local>

Just realized he wasn't talking CME.  Disregard.

 

Anthony

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
Holloway (Enventis)
Sent: Thursday, August 16, 2007 7:39 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] caller name with H.323

 

I had this same question with Cisco TAC, and here is their response.
(Note that we're talking about having a PRI with switch type NI2)

 

            The "isdn supp-service name calling" command does enable
receipt of the calling name in the facility ie,

but as of CME 3.3 we could only deliver that to a SIP call leg.  CME 4.0
introduced the ability to deliver

that calling name to CME phones.Therefore the caller name will work with
CME 4.0 for sccp CME phones.

 

These are the IOS releases compatible with CME 4.0 and up.

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps4625/prod_insta
llation_guide09186a00805acf50.html

 

4.2        12.4(11)XW2

 

4.1        12.4(15)T 5, 7

            

4.1        12.4(11)XJ

 

4.0(2)    12.4(11)T 5

 

4.0(3)    12.4(4)XC4

 

4.0(0)    12.4(9)T 6

 

4.0(1)    12.4(4)XC

 

Anthony

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian
Sent: Tuesday, August 14, 2007 10:22 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] caller name with H.323

 

I see H.323 name display is supported as of 12.4(11)XW.  Does anyone
know if it is also supported on some level of the T train?  I'm running
12.4(15)T1 and CUCM 5.1(2), but I don't see the display-ie or
supp-service commands on the gateway. 

Thanks,
Brian

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From bcarter at sentinel.com  Thu Aug 16 09:52:37 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 16 Aug 2007 08:52:37 -0500
Subject: [cisco-voip] CUCM 5.X, ODBC, and CDR
In-Reply-To: <46C35CA0.6050405@cisco.com>
References: <913929690708131111w7e27e0d7k91d1f33b47b0a7a5@mail.gmail.com>	<C1FE15183DA37645BC0633BC604E44F0066D8AFB@USNAEXCH.na.didata.local><913929690708151255i7184e86al43b99932cc34bcad@mail.gmail.com>
	<46C35CA0.6050405@cisco.com>
Message-ID: <6A0744639B86DD459433108DBF101D2601817945@MAIL01.sentinel.com>

Now we just need a AXL interface to use.  Without buying one.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Wednesday, August 15, 2007 3:06 PM
To: Andrew F. Wolfe
Cc: cisco-voip at puck.nether.net; Matt Slaga (US)
Subject: Re: [cisco-voip] CUCM 5.X, ODBC, and CDR


no odbc interface anymore.  database access is specifically blocked by
firewall.

can use AXL interface or CM can push flat text files to your (s)ftp
server.

/Wes

Andrew F. Wolfe wrote: 

	Sorry, it took a bit to read this.  I understand what you are
saying and with some more research I found what I was looking for in the
way of CDR.  As for no ODBC drivers for CUCM 5, I find that very hard to
believe since they include the AXL SQL toolkit and it runs from my
understanding IBM Informix SQL.  I could be wrong there.  Any DB folks
out there who have decided to delve into this realm?
	
	
	On 8/13/07, Matt Slaga (US) <Matt.Slaga at us.didata.com > wrote: 

		No ODBC drivers for CallManager 5.x and up, CDR is all
done via FTP flat file push/pulls.

		 

		 

		 

		From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Andrew F.
Wolfe
		Sent: Monday, August 13, 2007 2:11 PM
		To: cisco-voip at puck.nether.net
		Subject: [cisco-voip] CUCM 5.X, ODBC, and CDR

		 

		Hello all.  I am trying to wrap my brain around the DB
for CDR and CUCM.  Since it is an Informix DB, I can't use the ODBC
drivers included in Windows.  Any suggestions on an Informix ODBC driver
and any tips would be greatly appreciated.  Please let me know.  Thanks.


________________________________

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	_______________________________________________
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From joe.j.keller at gmail.com  Thu Aug 16 10:07:56 2007
From: joe.j.keller at gmail.com (Joe Keller)
Date: Thu, 16 Aug 2007 10:07:56 -0400
Subject: [cisco-voip] CCM 5.1 RAM feedback
Message-ID: <189b5daa0708160707m7dd64f5dsd9b6dcfa84028dfa@mail.gmail.com>

Any suggestions on minimum for 2000 users on Pub & Sub? 3GB on both OK? With
RAM so cheap, I was looking at 5GB on Pub, and maybe Subs.
Any noticeable performance improvement with more RAM?

Thanks
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From jason.aarons at us.didata.com  Thu Aug 16 10:15:29 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 16 Aug 2007 10:15:29 -0400
Subject: [cisco-voip] CCM 5.1 RAM feedback
In-Reply-To: <189b5daa0708160707m7dd64f5dsd9b6dcfa84028dfa@mail.gmail.com>
References: <189b5daa0708160707m7dd64f5dsd9b6dcfa84028dfa@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00675845E@USNAEXCH.na.didata.local>

32-bit address space is limited to 4GB addressable, do a Google search
for more info on 32-bit limitations.  This is one reason Exchange 2007
is 64-bit and requires 2003 Server 64 Bit.

 

You should be fine with 4GB, you can use RTMT to monitor hardware
resources.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Keller
Sent: Thursday, August 16, 2007 10:08 AM
To: Cisco-VoIP mailing list
Subject: [cisco-voip] CCM 5.1 RAM feedback

 

Any suggestions on minimum for 2000 users on Pub & Sub? 3GB on both OK?
With RAM so cheap, I was looking at 5GB on Pub, and maybe Subs.
Any noticeable performance improvement with more RAM?

Thanks




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From Simon_Jones at mondial-assistance.co.uk  Thu Aug 16 10:31:21 2007
From: Simon_Jones at mondial-assistance.co.uk (Simon Jones)
Date: Thu, 16 Aug 2007 15:31:21 +0100
Subject: [cisco-voip] MoH - Sound Quality
References: <3273B98003B44648B4F256E759A2F36202F8E950@CROPAPP01.mondial.internal> 
	<3e6f56310708151806kd585b86k85d9253cd3c40b82@mail.gmail.com>
Message-ID: <3273B98003B44648B4F256E759A2F36202F8EBB1@CROPAPP01.mondial.internal>

Hi Kent,

 

Thanks for the advice is there a specific program you use to put them in
the G729 format?

 

Many Thanks,

 

Simon

 

From: kent at fredf.org [mailto:kent at fredf.org] On Behalf Of Kent Roberts
Sent: 16 August 2007 02:06
To: Simon Jones
Subject: Re: [cisco-voip] MoH - Sound Quality

 

 

YUP.  make sure your origional files are recorded in G729  format,  Not
converted in any way shape or form.     Then drop them in the directory,
and let CCM do its work.  They should sound much better.

 



 

On 8/15/07, Simon Jones <Simon_Jones at mondial-assistance.co.uk> wrote: 

Hi,

 

Has anyone experienced this issue before? I have been tasked with
updating our existing MoH (currently classical music). I place the audio
files to be converted into the relevant directory in CCM and the files
are converted with no errors. 

 

When listening to the new audio source I have created it sounds fine
when I am put on hold from  an internal IP Phone, analogue/POTS line at
home. However when I call from a mobile/cell phone and put myself on
hold the sound quality is very poor - a lot of background noise/hiss. I
have tried call from mobile phones on alternate networks and they all
have the same quailty problem. 

 

CCM 4.1(3) ES110

Product : Cisco Catalyst 6000 E1 VoIP Gateway

Device Protocol: Digital Access PRI

 

I have also tried a variety of music styles and also experimented with
the original file being in MP3 or WAV format, but still get this same
issue.

 

Any thoughts as to what could be the cause or hopefully the solution to
this issue?

 

 

Many Thanks,

 

Simon



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Please note that we reserve the right to monitor and read any emails sent and received by the Company under the Telecommunications (Lawful Business Practice) (Interception of Communications) Regulations 2000.

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From Johnny.Crothers at vanco.co.uk  Thu Aug 16 11:18:52 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Thu, 16 Aug 2007 16:18:52 +0100
Subject: [cisco-voip]  IPCC Script - Ringback
Message-ID: <A56BD242744BF540B15D1672E0E3E87501A9648B@LON-INT-MSGVS01.internal.vancocorp.com>

Hi,

 

Is there a way of getting ringback instead of MoH when a caller is being
connecting to an agent? (Select Resource Step)

 

Any help would be great!

 

Cheers,

 

Johnny



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From msmith at laurinburg.org  Thu Aug 16 12:25:08 2007
From: msmith at laurinburg.org (Marcus Smith)
Date: Thu, 16 Aug 2007 11:25:08 -0500
Subject: [cisco-voip] masking incoming phone numbers
Message-ID: <000f01c7e022$030aae50$2900a8c0@laurinburg.org>

Good Morning All,

Our Police Dpt is looking at setting-up a Crime Tip Hotline, and here is the catcher, They cannot have access to the caller's number.   Is it possible to block the caller's number who places the tip call.  I am using CCM ver 4.2

Thank You,

Marcus Smith
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From patrick.diener at gmail.com  Thu Aug 16 11:27:15 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Thu, 16 Aug 2007 17:27:15 +0200
Subject: [cisco-voip] IPCC Script - Ringback
In-Reply-To: <A56BD242744BF540B15D1672E0E3E87501A9648B@LON-INT-MSGVS01.internal.vancocorp.com>
References: <A56BD242744BF540B15D1672E0E3E87501A9648B@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <c0deb5b80708160827l2bde7ac4ra16c321e877eb667@mail.gmail.com>

creating a music file containing the ringback tone will work...

Regards
Patrick

On 8/16/07, Johnny Crothers (c) <Johnny.Crothers at vanco.co.uk> wrote:
>
>
>
>
> Hi,
>
>
>
> Is there a way of getting ringback instead of MoH when a caller is being
> connecting to an agent? (Select Resource Step)
>
>
>
> Any help would be great!
>
>
>
> Cheers,
>
>
>
> Johnny
>
> **********************************************************************
>  Any opinions expressed in the email are those of the individual and not
> necessarily the company. This email and any files transmitted with it are
> confidential and solely for the use of the intended recipient. If you are
> not the intended recipient or the person responsible for delivering it to
> the intended recipient, be advised that you have received this email in
> error and that any dissemination, distribution, copying or use is strictly
> prohibited.
>
>  If you have received this email in error, or if you are concerned with the
> content of this email please e-mail to: e-security.support at vanco.info
>
>  The contents of an attachment to this e-mail may contain software viruses
> which could damage your own computer system. While the sender has taken
> every reasonable precaution to minimise this risk, we cannot accept
> liability for any damage which you sustain as a result of software viruses.
> You should carry out your own virus checks before opening any attachments to
> this e-mail.
>
>  Vanco UK Ltd Registered in England No: 2296733 Registered Office: John
> Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX
>
>  Please consider the environment before printing this e-mail
> **********************************************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From BRiley at jackhenry.com  Thu Aug 16 11:27:11 2007
From: BRiley at jackhenry.com (Bill Riley III)
Date: Thu, 16 Aug 2007 10:27:11 -0500
Subject: [cisco-voip] IPCC Script - Ringback
In-Reply-To: <A56BD242744BF540B15D1672E0E3E87501A9648B@LON-INT-MSGVS01.internal.vancocorp.com>
References: <A56BD242744BF540B15D1672E0E3E87501A9648B@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <2B3745BC9D31514381A6FCC5E72D9373021231BD@SMOMAIL.jhacorp.com>

Find a MOH file that plays a ringing noise and set it as the network MOH
for the CTI ports. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johnny Crothers
(c)
Sent: Thursday, August 16, 2007 10:19 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Script - Ringback

 

Hi,

 

Is there a way of getting ringback instead of MoH when a caller is being
connecting to an agent? (Select Resource Step)

 

Any help would be great!

 

Cheers,

 

Johnny



**********************************************************************
Any opinions expressed in the email are those of the individual and not
necessarily the company. This email and any files transmitted with it
are confidential and solely for the use of the intended recipient. If
you are not the intended recipient or the person responsible for
delivering it to the intended recipient, be advised that you have
received this email in error and that any dissemination, distribution,
copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with
the content of this email please e-mail to:
e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software
viruses which could damage your own computer system. While the sender
has taken every reasonable precaution to minimise this risk, we cannot
accept liability for any damage which you sustain as a result of
software viruses. You should carry out your own virus checks before
opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John
Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
**********************************************************************

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 
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From Johnny.Crothers at vanco.co.uk  Thu Aug 16 11:29:00 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Thu, 16 Aug 2007 16:29:00 +0100
Subject: [cisco-voip] IPCC Script - Ringback
In-Reply-To: <2B3745BC9D31514381A6FCC5E72D9373021231BD@SMOMAIL.jhacorp.com>
Message-ID: <A56BD242744BF540B15D1672E0E3E87501A96498@LON-INT-MSGVS01.internal.vancocorp.com>

Nice... Thanks :-) I'll have to setup another common profile etc.. but
should work fine, thanks!

 

Johnny Crothers

IPT Specialist

________________________________

From: Bill Riley III [mailto:BRiley at jackhenry.com] 
Sent: 16 August 2007 16:27
To: Johnny Crothers (c); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC Script - Ringback

 

Find a MOH file that plays a ringing noise and set it as the network MOH
for the CTI ports. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johnny Crothers
(c)
Sent: Thursday, August 16, 2007 10:19 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Script - Ringback

 

Hi,

 

Is there a way of getting ringback instead of MoH when a caller is being
connecting to an agent? (Select Resource Step)

 

Any help would be great!

 

Cheers,

 

Johnny



**********************************************************************
Any opinions expressed in the email are those of the individual and not
necessarily the company. This email and any files transmitted with it
are confidential and solely for the use of the intended recipient. If
you are not the intended recipient or the person responsible for
delivering it to the intended recipient, be advised that you have
received this email in error and that any dissemination, distribution,
copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with
the content of this email please e-mail to:
e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software
viruses which could damage your own computer system. While the sender
has taken every reasonable precaution to minimise this risk, we cannot
accept liability for any damage which you sustain as a result of
software viruses. You should carry out your own virus checks before
opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John
Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
**********************************************************************

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.
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From peder at networkoblivion.com  Thu Aug 16 12:21:58 2007
From: peder at networkoblivion.com (Peder @ NetworkOblivion)
Date: Thu, 16 Aug 2007 11:21:58 -0500
Subject: [cisco-voip] Factory reset pre UAL 7900
Message-ID: <46C479A6.5060600@networkoblivion.com>

Can anybody tell me how to factory reset a 7940 that is pre UAL?  The # 
key during reboot doesn't work and there is a password on there, so I 
can't do it that way.  I've googled it several ways and the only thing I 
can find is using the # key (which as I said doesn't work)...

Peder


From omar.parihuana at gmail.com  Thu Aug 16 12:27:15 2007
From: omar.parihuana at gmail.com (omar parihuana)
Date: Thu, 16 Aug 2007 11:27:15 -0500
Subject: [cisco-voip] Unregistered IP Phones
Message-ID: <834c50110708160927g54bbeb90p8d95faaaacebf406@mail.gmail.com>

Hi List,

Since two weeks I have issues with my  IP Phones, after check CCM 4.0 I saw
a lot of errors by device unregistered, and my users are complain because
the calls in course are dropped. I don't know why the IP Phones
unregistered, pls any suggestions... thanks.

My CCM logs are:


Event Type: Error

Event Source: Cisco CallManager

Event Category: None

Event ID: 3

Date: 16/08/2007

Time: 10:25:04 a.m.

User: N/A

Computer: PECMA01

Description:

Error: DeviceUnregistered - Device unregistered.

Device name.: SEP0013605818CA

Device IP address.: 10.20.0.132

Device type. [Optional]: 30007

Device description [Optional].: Patricia Verastegui 2237

Reason Code [Optional].: 8

App ID: Cisco CallManager

Cluster ID: PECMA01-Cluster

Node ID: 10.20.0.16

Explanation: A device that has previously registered with Cisco CallManager

has unregistered. This event may be issued as part of normal unregistration

event or due to some other reason such as loss of keepalives.

Recommended Action: No action is required if unregistration of this device

was expected..





Event Type: Error

Event Source: Cisco CallManager

Event Category: None

Event ID: 3

Date: 16/08/2007

Time: 10:24:52 a.m.

User: N/A

Computer: PECMA01

Description:

Error: DeviceUnregistered - Device unregistered.

Device name.: SEP00131A41940B

Device IP address.: 10.20.1.114

Device type. [Optional]: 30007

Device description [Optional].: Stephanie Kilimajer 2215

Reason Code [Optional].: 8

App ID: Cisco CallManager

Cluster ID: PECMA01-Cluster

Node ID: 10.20.0.16

Explanation: A device that has previously registered with Cisco CallManager

has unregistered. This event may be issued as part of normal unregistration

event or due to some other reason such as loss of keepalives.

Recommended Action: No action is required if unregistration of this device

was expected..





Event Type: Error

Event Source: Cisco CallManager

Event Category: None

Event ID: 3

Date: 16/08/2007

Time: 10:24:34 a.m.

User: N/A

Computer: PECMA01

Description:

Error: DeviceTransientConnection - Transient connection attempt.

Connecting Port: 0

Device name [Optional].:

Device IP address.: 10.20.20.45

Device type. [Optional]: 255

Reason Code [Optional].: 6

App ID: Cisco CallManager

Cluster ID: PECMA01-Cluster

Node ID: 10.20.0.16

Explanation: A connection was established and immediately dropped before

completing registration. Incomplete registration may indicate a device is

rehoming in the middle of registration. The alarm could also indicate a

device misconfiguration, database error, or an illegal/unknown device

trying to attempt a connection.

Recommended Action: No action is required if this event was issued as a

result of a normal device rehome..







Event Type: Error

Event Source: Cisco CallManager

Event Category: None

Event ID: 3

Date: 16/08/2007

Time: 10:24:29 a.m.

User: N/A

Computer: PECMA01

Description:

Error: DeviceUnregistered - Device unregistered.

Device name.: SEP001360581428

Device IP address.: 10.20.1.164

Device type. [Optional]: 30007

Device description [Optional].: Ursula Quevedo 1897

Reason Code [Optional].: 9

App ID: Cisco CallManager

Cluster ID: PECMA01-Cluster

Node ID: 10.20.0.16

Explanation: A device that has previously registered with Cisco CallManager

has unregistered. This event may be issued as part of normal unregistration

event or due to some other reason such as loss of keepalives.

Recommended Action: No action is required if unregistration of this device

was expected..







Event Type: Error

Event Source: Cisco CallManager

Event Category: None

Event ID: 3

Date: 16/08/2007

Time: 10:23:54 a.m.

User: N/A

Computer: PECMA01

Description:

Error: DeviceTransientConnection - Transient connection attempt.

Connecting Port: 2000

Device name [Optional].: SEP0012D9167518

Device IP address.: 10.20.1.228

Device type. [Optional]: 30007

Reason Code [Optional].: 1

App ID: Cisco CallManager

Cluster ID: PECMA01-Cluster

Node ID: 10.20.0.16

Explanation: A connection was established and immediately dropped before

completing registration. Incomplete registration may indicate a device is

rehoming in the middle of registration. The alarm could also indicate a

device misconfiguration, database error, or an illegal/unknown device

trying to attempt a connection.

Recommended Action: No action is required if this event was issued as a

result of a normal device rehome..






-- 
Omar E.P.T
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From ealeatherman at gmail.com  Thu Aug 16 12:34:48 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 16 Aug 2007 12:34:48 -0400
Subject: [cisco-voip] Nice page on CCO
Message-ID: <94a1afde0708160934s5a7d0973xe669f5e066a4b9f3@mail.gmail.com>

Figured I'd share this in case there were people like me that didnt know
about it where it might be useful:

Cisco IOS Voice Configuration Library
http://www.cisco.com/en/US/customer/products/ps6441/prod_configuration_guide09186a0080565f8a.html

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Thu Aug 16 12:38:33 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 16 Aug 2007 12:38:33 -0400
Subject: [cisco-voip] Unregistered IP Phones
In-Reply-To: <834c50110708160927g54bbeb90p8d95faaaacebf406@mail.gmail.com>
References: <834c50110708160927g54bbeb90p8d95faaaacebf406@mail.gmail.com>
Message-ID: <94a1afde0708160938q409cef72rc1f1ccf52f952576@mail.gmail.com>

I would guess its something on your network causing SCCP packets to get
dropped or delayed. Phone tries to register over to another server etc at
that point and puts messages in your event viewer. If the user is on a call
while the phone is trying to re-register and they try to use some feature
like hold or transfer, it will drop the call since there is no server to
facilitate the action.

Example, we had a PIX firewall messing with the sccp traffic at one site,
phone reset/re-registered constantly until we figured what was causing it.

On 8/16/07, omar parihuana <omar.parihuana at gmail.com> wrote:
>
> Hi List,
>
> Since two weeks I have issues with my  IP Phones, after check CCM 4.0 I
> saw a lot of errors by device unregistered, and my users are complain
> because the calls in course are dropped. I don't know why the IP Phones
> unregistered, pls any suggestions... thanks.
>
> My CCM logs are:
>
>
> Event Type: Error
>
> Event Source: Cisco CallManager
>
> Event Category: None
>
> Event ID: 3
>
> Date: 16/08/2007
>
> Time: 10:25:04 a.m.
>
> User: N/A
>
> Computer: PECMA01
>
> Description:
>
> Error: DeviceUnregistered - Device unregistered.
>
> Device name.: SEP0013605818CA
>
> Device IP address.: 10.20.0.132
>
> Device type. [Optional]: 30007
>
> Device description [Optional].: Patricia Verastegui 2237
>
> Reason Code [Optional].: 8
>
> App ID: Cisco CallManager
>
> Cluster ID: PECMA01-Cluster
>
> Node ID: 10.20.0.16
>
> Explanation: A device that has previously registered with Cisco
> CallManager
>
> has unregistered. This event may be issued as part of normal
> unregistration
>
> event or due to some other reason such as loss of keepalives.
>
> Recommended Action: No action is required if unregistration of this device
>
> was expected..
>
>
>
>
>
> Event Type: Error
>
> Event Source: Cisco CallManager
>
> Event Category: None
>
> Event ID: 3
>
> Date: 16/08/2007
>
> Time: 10:24:52 a.m.
>
> User: N/A
>
> Computer: PECMA01
>
> Description:
>
> Error: DeviceUnregistered - Device unregistered.
>
> Device name.: SEP00131A41940B
>
> Device IP address.: 10.20.1.114
>
> Device type. [Optional]: 30007
>
> Device description [Optional].: Stephanie Kilimajer 2215
>
> Reason Code [Optional].: 8
>
> App ID: Cisco CallManager
>
> Cluster ID: PECMA01-Cluster
>
> Node ID: 10.20.0.16
>
> Explanation: A device that has previously registered with Cisco
> CallManager
>
> has unregistered. This event may be issued as part of normal
> unregistration
>
> event or due to some other reason such as loss of keepalives.
>
> Recommended Action: No action is required if unregistration of this device
>
> was expected..
>
>
>
>
>
> Event Type: Error
>
> Event Source: Cisco CallManager
>
> Event Category: None
>
> Event ID: 3
>
> Date: 16/08/2007
>
> Time: 10:24:34 a.m.
>
> User: N/A
>
> Computer: PECMA01
>
> Description:
>
> Error: DeviceTransientConnection - Transient connection attempt.
>
> Connecting Port: 0
>
> Device name [Optional].:
>
> Device IP address.: 10.20.20.45
>
> Device type. [Optional]: 255
>
> Reason Code [Optional].: 6
>
> App ID: Cisco CallManager
>
> Cluster ID: PECMA01-Cluster
>
> Node ID: 10.20.0.16
>
> Explanation: A connection was established and immediately dropped before
>
> completing registration. Incomplete registration may indicate a device is
>
> rehoming in the middle of registration. The alarm could also indicate a
>
> device misconfiguration, database error, or an illegal/unknown device
>
> trying to attempt a connection.
>
> Recommended Action: No action is required if this event was issued as a
>
> result of a normal device rehome..
>
>
>
>
>
>
>
> Event Type: Error
>
> Event Source: Cisco CallManager
>
> Event Category: None
>
> Event ID: 3
>
> Date: 16/08/2007
>
> Time: 10:24:29 a.m.
>
> User: N/A
>
> Computer: PECMA01
>
> Description:
>
> Error: DeviceUnregistered - Device unregistered.
>
> Device name.: SEP001360581428
>
> Device IP address.: 10.20.1.164
>
> Device type. [Optional]: 30007
>
> Device description [Optional].: Ursula Quevedo 1897
>
> Reason Code [Optional].: 9
>
> App ID: Cisco CallManager
>
> Cluster ID: PECMA01-Cluster
>
> Node ID: 10.20.0.16
>
> Explanation: A device that has previously registered with Cisco
> CallManager
>
> has unregistered. This event may be issued as part of normal
> unregistration
>
> event or due to some other reason such as loss of keepalives.
>
> Recommended Action: No action is required if unregistration of this device
>
> was expected..
>
>
>
>
>
>
>
> Event Type: Error
>
> Event Source: Cisco CallManager
>
> Event Category: None
>
> Event ID: 3
>
> Date: 16/08/2007
>
> Time: 10:23:54 a.m.
>
> User: N/A
>
> Computer: PECMA01
>
> Description:
>
> Error: DeviceTransientConnection - Transient connection attempt.
>
> Connecting Port: 2000
>
> Device name [Optional].: SEP0012D9167518
>
> Device IP address.: 10.20.1.228
>
> Device type. [Optional]: 30007
>
> Reason Code [Optional].: 1
>
> App ID: Cisco CallManager
>
> Cluster ID: PECMA01-Cluster
>
> Node ID: 10.20.0.16
>
> Explanation: A connection was established and immediately dropped before
>
> completing registration. Incomplete registration may indicate a device is
>
> rehoming in the middle of registration. The alarm could also indicate a
>
> device misconfiguration, database error, or an illegal/unknown device
>
> trying to attempt a connection.
>
> Recommended Action: No action is required if this event was issued as a
>
> result of a normal device rehome..
>
>
>
>
>
>
> --
> Omar E.P.T
>
>
> _______________________________________________
> 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 Matt.Slaga at us.didata.com  Thu Aug 16 12:39:30 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 16 Aug 2007 12:39:30 -0400
Subject: [cisco-voip] Factory reset pre UAL 7900
In-Reply-To: <46C479A6.5060600@networkoblivion.com>
References: <46C479A6.5060600@networkoblivion.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0067587E9@USNAEXCH.na.didata.local>

Press Settings, **# (to unlock), press Network Settings, 33, then select
"Yes".

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Peder @
NetworkOblivion
Sent: Thursday, August 16, 2007 12:22 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Factory reset pre UAL 7900

Can anybody tell me how to factory reset a 7940 that is pre UAL?  The # 
key during reboot doesn't work and there is a password on there, so I 
can't do it that way.  I've googled it several ways and the only thing I

can find is using the # key (which as I said doesn't work)...

Peder

_______________________________________________
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 peder at networkoblivion.com  Thu Aug 16 13:25:31 2007
From: peder at networkoblivion.com (Peder @ NetworkOblivion)
Date: Thu, 16 Aug 2007 12:25:31 -0500
Subject: [cisco-voip] Factory reset pre UAL 7900
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0067587E9@USNAEXCH.na.didata.local>
References: <46C479A6.5060600@networkoblivion.com>
	<C1FE15183DA37645BC0633BC604E44F0067587E9@USNAEXCH.na.didata.local>
Message-ID: <46C4888B.2030703@networkoblivion.com>

I should have mentioned that it is already sip5.2, so **# doesn't work.

Matt Slaga (US) wrote:
> Press Settings, **# (to unlock), press Network Settings, 33, then select
> "Yes".
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Peder @
> NetworkOblivion
> Sent: Thursday, August 16, 2007 12:22 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Factory reset pre UAL 7900
> 
> Can anybody tell me how to factory reset a 7940 that is pre UAL?  The # 
> key during reboot doesn't work and there is a password on there, so I 
> can't do it that way.  I've googled it several ways and the only thing I
> 
> can find is using the # key (which as I said doesn't work)...
> 
> Peder
> 
> _______________________________________________
> 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 craig at staffin.org  Thu Aug 16 13:27:40 2007
From: craig at staffin.org (Craig Staffin)
Date: Thu, 16 Aug 2007 12:27:40 -0500
Subject: [cisco-voip] masking incoming phone numbers
In-Reply-To: <000f01c7e022$030aae50$2900a8c0@laurinburg.org>
References: <000f01c7e022$030aae50$2900a8c0@laurinburg.org>
Message-ID: <9b76d88c0708161027i1171dfb0v73d754328786e400@mail.gmail.com>

Marcus,

The easiest way is send the number through a translation pattern that strips
the calling party ID or puts something else in.

Craig

On 8/16/07, Marcus Smith <msmith at laurinburg.org> wrote:
>
>  Good Morning All,
>
> Our Police Dpt is looking at setting-up a Crime Tip Hotline, and here is
> the catcher, They cannot have access to the caller's number.   Is it
> possible to block the caller's number who places the tip call.  I am using
> CCM ver 4.2
>
> Thank You,
>
> Marcus Smith
>
> _______________________________________________
> 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 nmenoni at yahoo.com  Thu Aug 16 12:46:31 2007
From: nmenoni at yahoo.com (=?iso-8859-1?q?Nicol=E1s=20Menoni?=)
Date: Thu, 16 Aug 2007 11:46:31 -0500 (CDT)
Subject: [cisco-voip] Unregistered IP Phones
In-Reply-To: <834c50110708160927g54bbeb90p8d95faaaacebf406@mail.gmail.com>
Message-ID: <871149.79463.qm@web50607.mail.re2.yahoo.com>

I think it could be a problem in the NIC (network
interface card) of the CCM server. It could be a
mismatch in the speed or duplexing or something like
that, for this, the keepalive can't reach the
destination.

Ok, bye, Nicol?s.


--- omar parihuana <omar.parihuana at gmail.com>
escribi?:

> Hi List,
> 
> Since two weeks I have issues with my  IP Phones,
> after check CCM 4.0 I saw
> a lot of errors by device unregistered, and my users
> are complain because
> the calls in course are dropped. I don't know why
> the IP Phones
> unregistered, pls any suggestions... thanks.
> 
> My CCM logs are:
> 
> 
> Event Type: Error
> 
> Event Source: Cisco CallManager
> 
> Event Category: None
> 
> Event ID: 3
> 
> Date: 16/08/2007
> 
> Time: 10:25:04 a.m.
> 
> User: N/A
> 
> Computer: PECMA01
> 
> Description:
> 
> Error: DeviceUnregistered - Device unregistered.
> 
> Device name.: SEP0013605818CA
> 
> Device IP address.: 10.20.0.132
> 
> Device type. [Optional]: 30007
> 
> Device description [Optional].: Patricia Verastegui
> 2237
> 
> Reason Code [Optional].: 8
> 
> App ID: Cisco CallManager
> 
> Cluster ID: PECMA01-Cluster
> 
> Node ID: 10.20.0.16
> 
> Explanation: A device that has previously registered
> with Cisco CallManager
> 
> has unregistered. This event may be issued as part
> of normal unregistration
> 
> event or due to some other reason such as loss of
> keepalives.
> 
> Recommended Action: No action is required if
> unregistration of this device
> 
> was expected..
> 
> 
> 
> 
> 
> Event Type: Error
> 
> Event Source: Cisco CallManager
> 
> Event Category: None
> 
> Event ID: 3
> 
> Date: 16/08/2007
> 
> Time: 10:24:52 a.m.
> 
> User: N/A
> 
> Computer: PECMA01
> 
> Description:
> 
> Error: DeviceUnregistered - Device unregistered.
> 
> Device name.: SEP00131A41940B
> 
> Device IP address.: 10.20.1.114
> 
> Device type. [Optional]: 30007
> 
> Device description [Optional].: Stephanie Kilimajer
> 2215
> 
> Reason Code [Optional].: 8
> 
> App ID: Cisco CallManager
> 
> Cluster ID: PECMA01-Cluster
> 
> Node ID: 10.20.0.16
> 
> Explanation: A device that has previously registered
> with Cisco CallManager
> 
> has unregistered. This event may be issued as part
> of normal unregistration
> 
> event or due to some other reason such as loss of
> keepalives.
> 
> Recommended Action: No action is required if
> unregistration of this device
> 
> was expected..
> 
> 
> 
> 
> 
> Event Type: Error
> 
> Event Source: Cisco CallManager
> 
> Event Category: None
> 
> Event ID: 3
> 
> Date: 16/08/2007
> 
> Time: 10:24:34 a.m.
> 
> User: N/A
> 
> Computer: PECMA01
> 
> Description:
> 
> Error: DeviceTransientConnection - Transient
> connection attempt.
> 
> Connecting Port: 0
> 
> Device name [Optional].:
> 
> Device IP address.: 10.20.20.45
> 
> Device type. [Optional]: 255
> 
> Reason Code [Optional].: 6
> 
> App ID: Cisco CallManager
> 
> Cluster ID: PECMA01-Cluster
> 
> Node ID: 10.20.0.16
> 
> Explanation: A connection was established and
> immediately dropped before
> 
> completing registration. Incomplete registration may
> indicate a device is
> 
> rehoming in the middle of registration. The alarm
> could also indicate a
> 
> device misconfiguration, database error, or an
> illegal/unknown device
> 
> trying to attempt a connection.
> 
> Recommended Action: No action is required if this
> event was issued as a
> 
> result of a normal device rehome..
> 
> 
> 
> 
> 
> 
> 
> Event Type: Error
> 
> Event Source: Cisco CallManager
> 
> Event Category: None
> 
> Event ID: 3
> 
> Date: 16/08/2007
> 
> Time: 10:24:29 a.m.
> 
> User: N/A
> 
> Computer: PECMA01
> 
> 
=== message truncated ===>
_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


__________________________________________________
Correo Yahoo!
Espacio para todos tus mensajes, antivirus y antispam ?gratis! 
Reg?strate ya - http://correo.espanol.yahoo.com/ 

From tim.reimers at asheville.k12.nc.us  Thu Aug 16 13:30:25 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Thu, 16 Aug 2007 13:30:25 -0400
Subject: [cisco-voip] UNITY 4.01 - Changing the "recorded name" in a users
	voicemail box
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1AC23@webmail.asheville.k12.nc.us>

Hi -

 

I'm wondering what the menu options are for re-recording a users
'recorded name' in Unity 4.01 ???

 

We do not have Unity PCA set up, nor are users trained on it.

 

But I have several users that have gotten married and want to change the
recorded name in their Unity mailbox.

They can change their standard and alternate greetings, but cannot seem
to change their "recorded name"

 

I considered just blowing away their mailbox and creating a new one, but
some of them have voicemails they want to keep, some don't communicate
with me about having checked all voicemails, etc...

 

Is there a way the users can re-record their own "recorded name'  by
using the voice menus in Unity available via their phone ??

 

Thanks, Tim

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From jason.aarons at us.didata.com  Thu Aug 16 14:08:41 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 16 Aug 2007 14:08:41 -0400
Subject: [cisco-voip] CCM 5.1 RAM feedback
In-Reply-To: <189b5daa0708161048p6aceb8a6m1549f741b7c1e228@mail.gmail.com>
References: <189b5daa0708160707m7dd64f5dsd9b6dcfa84028dfa@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F00675845E@USNAEXCH.na.didata.local>
	<189b5daa0708161048p6aceb8a6m1549f741b7c1e228@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006795405@USNAEXCH.na.didata.local>

Is the hardware and RHE it's running on 64-bit (maybe seen from boot
console), would be a good question! I haven't seen a 64-bit app run on
32-bit hardware.

 

I hadn't heard of a 64-bit edition of CallManager, but I could be wrong.

 

From: Joe Keller [mailto:joe.j.keller at gmail.com] 
Sent: Thursday, August 16, 2007 1:49 PM
To: Jason Aarons (US)
Subject: Re: [cisco-voip] CCM 5.1 RAM feedback

 

So CCM 5.1 is 32-bit?

On 8/16/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:

32-bit address space is limited to 4GB addressable, do a Google search
for more info on 32-bit limitations.  This is one reason Exchange 2007
is 64-bit and requires 2003 Server 64 Bit.

 

You should be fine with 4GB, you can use RTMT to monitor hardware
resources.

 

From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Joe Keller
Sent: Thursday, August 16, 2007 10:08 AM
To: Cisco-VoIP mailing list
Subject: [cisco-voip] CCM 5.1 RAM feedback

 

Any suggestions on minimum for 2000 users on Pub & Sub? 3GB on both OK?
With RAM so cheap, I was looking at 5GB on Pub, and maybe Subs.
Any noticeable performance improvement with more RAM?

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 lelio at uoguelph.ca  Thu Aug 16 13:11:27 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 16 Aug 2007 13:11:27 -0400
Subject: [cisco-voip] masking incoming phone numbers
References: <000f01c7e022$030aae50$2900a8c0@laurinburg.org>
Message-ID: <003501c7e028$7bbe6f20$6601a8c0@pdp11>

Just setup a translation to their tip line that restricts the calling party name/number.

  ----- Original Message ----- 
  From: Marcus Smith 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, August 16, 2007 12:25 PM
  Subject: [cisco-voip] masking incoming phone numbers


  Good Morning All,

  Our Police Dpt is looking at setting-up a Crime Tip Hotline, and here is the catcher, They cannot have access to the caller's number.   Is it possible to block the caller's number who places the tip call.  I am using CCM ver 4.2

  Thank You,

  Marcus Smith


------------------------------------------------------------------------------


  _______________________________________________
  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  Thu Aug 16 14:22:59 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu, 16 Aug 2007 13:22:59 -0500
Subject: [cisco-voip] UNITY 4.01 - Changing the "recorded name" in a
	usersvoicemail box
In-Reply-To: <050944EE48FF384F9C663F7048C47DA702E1AC23@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA702E1AC23@webmail.asheville.k12.nc.us>
Message-ID: <002a01c7e032$8c2af2c0$0232a8c0@walentawk1>

When a user calls into voicemail option 4 -> option 3 -> option 2
 
http://www.cisco.com/application/pdf/en/us/guest/products/ps4608/c1626/ccmig
ration_09186a00803fac65.pdf

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Thursday, August 16, 2007 12:30 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] UNITY 4.01 - Changing the "recorded name" in a
usersvoicemail box



Hi -

 

I'm wondering what the menu options are for re-recording a users 'recorded
name' in Unity 4.01 ???

 

We do not have Unity PCA set up, nor are users trained on it.

 

But I have several users that have gotten married and want to change the
recorded name in their Unity mailbox.

They can change their standard and alternate greetings, but cannot seem to
change their "recorded name"

 

I considered just blowing away their mailbox and creating a new one, but
some of them have voicemails they want to keep, some don't communicate with
me about having checked all voicemails, etc.

 

Is there a way the users can re-record their own "recorded name'  by using
the voice menus in Unity available via their phone ??

 

Thanks, Tim

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From tim.reimers at asheville.k12.nc.us  Thu Aug 16 14:40:26 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Thu, 16 Aug 2007 14:40:26 -0400
Subject: [cisco-voip] UNITY 4.01 - Changing the "recorded name" in a
	usersvoicemail box
In-Reply-To: <002a01c7e032$8c2af2c0$0232a8c0@walentawk1>
References: <050944EE48FF384F9C663F7048C47DA702E1AC23@webmail.asheville.k12.nc.us>
	<002a01c7e032$8c2af2c0$0232a8c0@walentawk1>
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1AC29@webmail.asheville.k12.nc.us>

Thanks -

 

Don't know why I didn't trip over that...

 

________________________________

From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: Thursday, August 16, 2007 2:23 PM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] UNITY 4.01 - Changing the "recorded name" in a
usersvoicemail box

 

When a user calls into voicemail option 4 -> option 3 -> option 2

 

http://www.cisco.com/application/pdf/en/us/guest/products/ps4608/c1626/c
cmigration_09186a00803fac65.pdf

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Thursday, August 16, 2007 12:30 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] UNITY 4.01 - Changing the "recorded name" in a
usersvoicemail box

Hi -

 

I'm wondering what the menu options are for re-recording a users
'recorded name' in Unity 4.01 ???

 

We do not have Unity PCA set up, nor are users trained on it.

 

But I have several users that have gotten married and want to change the
recorded name in their Unity mailbox.

They can change their standard and alternate greetings, but cannot seem
to change their "recorded name"

 

I considered just blowing away their mailbox and creating a new one, but
some of them have voicemails they want to keep, some don't communicate
with me about having checked all voicemails, etc...

 

Is there a way the users can re-record their own "recorded name'  by
using the voice menus in Unity available via their phone ??

 

Thanks, Tim

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From svoll.voip at gmail.com  Thu Aug 16 14:56:13 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 16 Aug 2007 11:56:13 -0700
Subject: [cisco-voip] CM 5.1.2 Qsig redirect issue
Message-ID: <f84a38d30708161156q703875bl50ea89eef06383ac@mail.gmail.com>

Guys

have you seen any QSig issues with CM 5.1.2 with Redirect not working?  any
bug id's for me to look at.  I see a bunch but nothing really matches.

CM 5.1.2, 3845 running 12.4.15T1

ip phone calls IP phone that is CFA to PBX phone

IP phones --- CM ---- VGW ---- PRI QSig ---- PBX ---- PBX Phones

debug ISDN Q931 shows the redirect but on the PBX they aren't getting it.

Any ideas?

Scott
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From patrick.diener at gmail.com  Thu Aug 16 15:58:26 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Thu, 16 Aug 2007 21:58:26 +0200
Subject: [cisco-voip] H323 gateway question/problem
In-Reply-To: <94a1afde0708160619s3e8d28c6h9758c43afa47aff4@mail.gmail.com>
References: <94a1afde0708160619s3e8d28c6h9758c43afa47aff4@mail.gmail.com>
Message-ID: <c0deb5b80708161258h1b4187eer34670e1228e89d43@mail.gmail.com>

try to find out which end is dropping (releasing) the call (Gateway or
CallManager) and for what reason (h225 q931 cause code)

use "debug h225 q931" on the gateway to get this info...

Regards
Patrick

On 8/16/07, Ed Leatherman <ealeatherman at gmail.com> wrote:
> Hi Folks,
>
> I'm pretty rusty on my H.323 gateways and dial peers etc, running into a
> problem with a configuration in my lab that i'm hoping to use this Fall for
> a project - was hoping someone might have an idea.
>
> Goal here is to have a router with CME setup for "SRST Mode" and a number of
> POTS lines in from pstn. Phones normally registered to callmanager (version
> 4.2 in my lab). During normal operation, if any of those lines ring, it
> should just connect to call manager and ring a shared line appearance on X
> number of phones at the location (probably about 8). During SRST operation,
> essentially I want to do the same thing - using overlays etc.
>
> I have the CME up and seems to work OK. The problem I have is during normal
> operation. Any 1 line can ring in and it gets sent to call manager and the
> correct line rings on all phones. Any other line that rings in after that
> appears to get sent to call manager, the line rings once, and then the call
> drops. It seems to me that perhaps something in the feature negotiation
> phase is failing, since the call rings out once before failing. I dont have
> any CAC setup, and its all using G.711ulaw. I don't know enough about H323
> to know where to look next though - nothing in the debugs I looked at jumped
> out at me as an error.
>
> I did watch the debug for dial-peer matching and it looks to be matching my
> dial-peer out to CCM before it matches the ephone-dn dial peers
>
> router is basically setup like this:
>
> voice-port 0/0/0
>  connection plar 37111
> voice-port 0/0/1
>  connection plar 37111
> voice-port 0/0/2
>  connection plar 37111
> !
> dial-peer voice 100 voip
>  destination-pattern 37111$
>  session target ipv4:10.182.140.10  ! Call Manager 4.2
>  no vad
>  codec g711ulaw
> !
>
> I found that if I dont put the end of string marker in my destination
> pattern, it likes to match the ephone-dn's before it matches the callmanager
> dial-peer.
>
> Thanks!
> --
> 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 tednugent69 at yahoo.com  Thu Aug 16 20:36:36 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 16 Aug 2007 17:36:36 -0700 (PDT)
Subject: [cisco-voip] Resolved: CM6 CallManager service will not stay
	running
In-Reply-To: <5d093f9a0708160559s50a9056fk94c080e027fdfbb@mail.gmail.com>
Message-ID: <676641.35480.qm@web36105.mail.mud.yahoo.com>

Fresh install and after further investigation the
problem WAS caused by an install of DHCP and No DNS
Client contrary to my original beliefs. Another Fresh
install with a Static IP and DNS Client resolved the
issue.



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

> Was this an upgrade of a 5 box, or a new install?
> 
> Did you get an upgrade license?
> 
> 
> Jonathan
> 
> On 8/15/07, Ted Nugent <tednugent69 at yahoo.com>
> wrote:
> > Thanks Patrick you actually changed my train of
> > thought. I was convinced that it was some sort of
> > licensing issue however it was much simpler then
> that.
> > The Server name was still set to localhost I
> switched
> > it to the IP address, rebooted the CM and now the
> > service is running stable, that's a new one on me
> but
> > I learn something new everyday I guess. Thank for
> the
> > redirect.
> >
> > Ted
> >
> >
> > --- Patrick Diener <patrick.diener at gmail.com>
> wrote:
> >
> > > I ran into this once... the installation was
> done by
> > > someone not very
> > > experienced in setting up linux based ccms so he
> > > used some strange
> > > settings like DHCP enabled (meaning CM gets its
> IP
> > > from a DHCP Server
> > > instead of setting it statically), DNS enabled
> etc.
> > > after
> > > reinstallation with the "correct" settings it
> was
> > > working just fine...
> > >
> > > anything you specified during installation you
> > > didn't know what it does?!?
> > >
> > > Regards Patrick
> > >
> > > On 8/15/07, Ted Nugent <tednugent69 at yahoo.com>
> > > wrote:
> > > > Hey Guys
> > > > I've got a FRESH install of CM 6.01 which I've
> > > > attempted on both VMWare and a legitimate MCS
> and
> > > both
> > > > are reacting the same way. After the install
> is
> > > > complete I go to activate the "default"
> services
> > > and
> > > > the CM service runs for about 30 seconds and
> bombs
> > > all
> > > > other services remain running fine. I've read
> that
> > > > this is expected behavior on an upgrade
> however
> > > this
> > > > is a fresh install from the ground up so it
> SHOULD
> > > use
> > > > the "starter license"?? What am I missing? Any
> one
> > > > else run into this?
> > > >
> > > > >From 6.01 Release Notes:
> > > > Starter Licenses
> > > > Cisco Unified Communications Manager comes
> > > > preinstalled with starter licenses, so a file
> > > upload
> > > > is not required. Starter licenses have no
> > > expiration
> > > > date and are available in limited quantities.
> Be
> > > aware
> > > > that starter licenses are only available for
> fresh
> > > > installations; they are not available for
> upgrades
> > > or
> > > > migrations from previous releases. Starter
> > > licenses
> > > > support only one Cisco Unified Communications
> > > Manager
> > > > node and up to 50 device license units.
> > > >
> > > >
> > > > Thanks
> > > > Ted
> > > >
> > > >
> > > >
> > > >
> > >
> >
>
____________________________________________________________________________________
> > > > 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
> > > >
> > > _______________________________________________
> > > 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/
> > _______________________________________________
> > 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  Thu Aug 16 22:52:44 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 16 Aug 2007 19:52:44 -0700 (PDT)
Subject: [cisco-voip] UC500
In-Reply-To: <003c01c7dff1$981a20b0$c84e6210$@com>
Message-ID: <105599.34811.qm@web57210.mail.re3.yahoo.com>

http://www.cisco.com/en/US/products/ps7293/index.html

Here it says it supports up to 50 users.

Paul

--- Curt Shaffer <cshaffer at gmail.com> wrote:

> I can't find a user limit for this device. I see
> mention of 16 users but it
> doesn't say limited to 16 users. Anyone have insight
> on this? Also is anyone
> aware of redundancy options for this?
> 
>  
> 
> Thanks!
> 
> > _______________________________________________
> 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  Thu Aug 16 23:03:23 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 16 Aug 2007 20:03:23 -0700 (PDT)
Subject: [cisco-voip] a comparison of CCME configuration methods
In-Reply-To: <676641.35480.qm@web36105.mail.mud.yahoo.com>
Message-ID: <421560.17861.qm@web57208.mail.re3.yahoo.com>

There are three ways of configuring CCME, four if you
count QCT. I have only used the manual method and the
semi-automatic method with using ephone auto-reg. I
wanted to hear anyone's take on the other methods, in
particular, QCT/fully-automated installation where you
walk through a "wizard."

I find the ephone auto-reg to be quite useful. I
wanted to try the "wizard"-based configuration also
but wanted to hear others opinions on it first.

Paul


       
____________________________________________________________________________________
Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.
http://smallbusiness.yahoo.com/webhosting 

From Nesen.Kisten at za.didata.com  Fri Aug 17 01:23:15 2007
From: Nesen.Kisten at za.didata.com (Nesen Kisten (ZA))
Date: Fri, 17 Aug 2007 07:23:15 +0200
Subject: [cisco-voip] PCM Codec
Message-ID: <DA507CEAA8B853469A9A74AD432B58F41A5519@ZABRYSVCL17A.af.didata.local>

Hi All,
 
I need to change the PCM codec for voice calls from G711ulaw to G711alaw, I have changed the PCM setting on the E1 mgcp port on CCM, however when I do a sh voice calls I still see calls codec as G711ulaw, I have also changed compounding codec on voice port to g711alaw with no luck.
 
Please help
 
Kind Regards
 
Nesen

This email and all contents are subject to the following disclaimer:

"http://www.dimensiondata.com/emaildisclaimer.htm"
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From Andrew.Riley at getronics.com  Fri Aug 17 02:06:35 2007
From: Andrew.Riley at getronics.com (Riley, Andrew)
Date: Fri, 17 Aug 2007 16:06:35 +1000
Subject: [cisco-voip] UC500
In-Reply-To: <105599.34811.qm@web57210.mail.re3.yahoo.com>
References: <003c01c7dff1$981a20b0$c84e6210$@com>
	<105599.34811.qm@web57210.mail.re3.yahoo.com>
Message-ID: <E4B3AF9D5AA27041A2C8A5B9BC5518D90CF63F78@excausy100.australia.unity>

Be careful with the UC500's

- NO Routing protocol support: i.e. even no RIP

I have confirmed that the UC520W and UC500 Series feature set supports
all of the voice, video, data, security, and wireless features with the
following exceptions:

For Voice - MGCP / CCM / SRST

For Data - WAN Protocols such as FR / ATM / X.25 / MPLS / Tag / LAPB

For Security - L2TP

For Routing - Routing Protocols

For Network Management - SNMP Proxy

Other - GLBP (load balancing)

Dynamic IP routing(BGP etc..), CCM/MGCP/SRST support have been removed
from these images. Static IP routing is still supported. All other data
support exists in this image.

The UC500 series code is based on IOS 12.4 Advanced Service feature set,
however some of the above features has been stripped off including
dynamic routing. There is a internal bug CSCsh45692 raised to have the
dynamic routing removed.



Also the configuration Application is based around North America. I
recommend just using the UC500 like CME / CUE, much easier.

 
 




-----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, 17 August 2007 12:53 PM
To: Curt Shaffer; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] UC500

http://www.cisco.com/en/US/products/ps7293/index.html

Here it says it supports up to 50 users.

Paul

--- Curt Shaffer <cshaffer at gmail.com> wrote:

> I can't find a user limit for this device. I see mention of 16 users
> but it doesn't say limited to 16 users. Anyone have insight on this?
> Also is anyone aware of redundancy options for this?
>
> 
>
> Thanks!
>
> > _______________________________________________
> 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
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From craig at staffin.org  Fri Aug 17 09:57:20 2007
From: craig at staffin.org (Craig Staffin)
Date: Fri, 17 Aug 2007 08:57:20 -0500
Subject: [cisco-voip] FN - 62850 - MCS 7825-H2 Running Cisco Unified
	Communications Manager 5.1 and 6.0 or Cisco Unified Presence
	6.0 - Sporadic Server Hang
Message-ID: <9b76d88c0708170657u3b3cf146x734c5463c08bfa47@mail.gmail.com>

Wes,

Can you update this doc to include 6.x references.  Right now for instance
it only refers to a workaround for 5.1 and nothing about 6.x but yet the
title and description of the problem refer to 6 as well.

On a side note has anyone seen this issue on a 7825??

Thanks
Craig
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From keli at carocomp.ro  Fri Aug 17 13:28:03 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Fri, 17 Aug 2007 20:28:03 +0300
Subject: [cisco-voip] DHCP not really working
Message-ID: <46C5DAA3.6060407@carocomp.ro>

Hi!

It's a bit off-topic, however I'm all clueless, so any help would be 
appreciated.

I have a CCME (2811, with 12.4(13r)T), with 7911G phones installed. Two 
VLANs, one for data, one for voice.

The router should be responsible for DHCP for the phones, but it does 
not seem to work - phones already in the network have their IP adress, 
keep their address through a reset, but new phones added to the network 
can't get an IP.

However, more people have access to the router and it is very possible 
it might have been misconfigured.

For starters, I noticed a

no service dhcp

in the running config, so I removed that. The dhcp debug messages began 
to appear, but it seems like the router still can't find "itself":

Aug 17 14:13:41.160: DHCPD: DHCPREQUEST received from client 
0100.1bd4.60b5.e2.
Aug 17 14:13:41.160: DHCPD: Finding a relay for client 0100.1bd4.60b5.e2 
on interface FastEthernet0/1.2.
Aug 17 14:13:41.880: DHCPD: DHCPREQUEST received from client 
0100.1bd4.609c.cf.
Aug 17 14:13:41.880: DHCPD: Finding a relay for client 0100.1bd4.609c.cf 
on interface FastEthernet0/1.2.

Here are relevant parts of the configuration:

no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.42.1 192.168.42.2
!
ip dhcp pool ITS
   network 192.168.42.0 255.255.255.0
   option 150 ip 192.168.42.1
   default-router 192.168.42.1
!
!
no ip domain lookup

......................

any ideas? useful show commands?
most sh ip dhcp ... commands show nothing, except for

Router#sh ip dhcp server statistics
Memory usage         23512
Address pools        1
Database agents      0
Automatic bindings   0
Manual bindings      0
Expired bindings     0
Malformed messages   0
Secure arp entries   0

Message              Received
BOOTREQUEST          0
DHCPDISCOVER         0
DHCPREQUEST          1912
DHCPDECLINE          0
DHCPRELEASE          0
DHCPINFORM           0

Message              Sent
BOOTREPLY            0
DHCPOFFER            0
DHCPACK              0
DHCPNAK              0

Router#sh ip dhcp pool ITS         

Pool ITS :
 Utilization mark (high/low)    : 100 / 0
 Subnet size (first/next)       : 0 / 0
 Total addresses                : 254
 Leased addresses               : 0
 Pending event                  : none
 1 subnet is currently in the pool :
 Current index        IP address range                    Leased addresses
 192.168.42.1         192.168.42.1     - 192.168.42.254    0

Thanks,
  Zoltan

From wsisk at cisco.com  Fri Aug 17 10:40:04 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 17 Aug 2007 10:40:04 -0400
Subject: [cisco-voip] FN - 62850 - MCS 7825-H2 Running Cisco
 Unified	Communications Manager 5.1 and 6.0 or Cisco Unified Presence	6.0 -
 Sporadic Server Hang
In-Reply-To: <9b76d88c0708170657u3b3cf146x734c5463c08bfa47@mail.gmail.com>
References: <9b76d88c0708170657u3b3cf146x734c5463c08bfa47@mail.gmail.com>
Message-ID: <46C5B344.5060402@cisco.com>

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From pmowry at getgds.com  Fri Aug 17 10:11:36 2007
From: pmowry at getgds.com (Patrick Mowry)
Date: Fri, 17 Aug 2007 09:11:36 -0500
Subject: [cisco-voip]  masking incoming phone numbers
In-Reply-To: <mailman.11262.1187294310.1529.cisco-voip@puck.nether.net>
References: <mailman.11262.1187294310.1529.cisco-voip@puck.nether.net>
Message-ID: <4C792A099259634C83F97067671B37FE0318106D@GDSEXC.gds.pri>

I assume this would be with an H323 gateway so the caller ID never makes
it to CallManager.  If you use MGCP and a translation pattern in
CallManager, I would expect the callerID to still show up in the CDR
records.

  ----- Original Message ----- 
From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Subject: Re: [cisco-voip] masking incoming phone numbers
To: "Marcus Smith" <msmith at laurinburg.org>,
	<cisco-voip at puck.nether.net>
Message-ID: <003501c7e028$7bbe6f20$6601a8c0 at pdp11>
Content-Type: text/plain; charset="iso-8859-1"

Just setup a translation to their tip line that restricts the calling
party name/number.

  ----- Original Message ----- 
  From: Marcus Smith 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, August 16, 2007 12:25 PM
  Subject: [cisco-voip] masking incoming phone numbers


  Good Morning All,

  Our Police Dpt is looking at setting-up a Crime Tip Hotline, and here
is the catcher, They cannot have access to the caller's number.   Is it
possible to block the caller's number who places the tip call.  I am
using CCM ver 4.2

  Thank You,

  Marcus Smith


From svoll.voip at gmail.com  Fri Aug 17 11:17:06 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 17 Aug 2007 08:17:06 -0700
Subject: [cisco-voip] CM 5.1.2b download
Message-ID: <f84a38d30708170817x2fa5abeay19874668547baee@mail.gmail.com>

is anyone else having issues downloading and installing 5.1.2b?

I download it and it downloads as a .gz.gz but it should be gz.sgn.  or am I
missing a step.  I'm still really new to 5.x

Scott
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From patrick.diener at gmail.com  Fri Aug 17 11:17:36 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Fri, 17 Aug 2007 17:17:36 +0200
Subject: [cisco-voip] DHCP not really working
In-Reply-To: <46C5DAA3.6060407@carocomp.ro>
References: <46C5DAA3.6060407@carocomp.ro>
Message-ID: <c0deb5b80708170817k23ce8b6kf816b85c3a0bef2@mail.gmail.com>

do the IP Phones get the correct Voice VLAN from the switch? Somehow
it looks like the DHCP request is received on an subinterface with an
IP Address not in the 192.168.42.0/24 range...

Regards
Patrick

On 8/17/07, Kelemen Zoltan <keli at carocomp.ro> wrote:
> Hi!
>
> It's a bit off-topic, however I'm all clueless, so any help would be
> appreciated.
>
> I have a CCME (2811, with 12.4(13r)T), with 7911G phones installed. Two
> VLANs, one for data, one for voice.
>
> The router should be responsible for DHCP for the phones, but it does
> not seem to work - phones already in the network have their IP adress,
> keep their address through a reset, but new phones added to the network
> can't get an IP.
>
> However, more people have access to the router and it is very possible
> it might have been misconfigured.
>
> For starters, I noticed a
>
> no service dhcp
>
> in the running config, so I removed that. The dhcp debug messages began
> to appear, but it seems like the router still can't find "itself":
>
> Aug 17 14:13:41.160: DHCPD: DHCPREQUEST received from client
> 0100.1bd4.60b5.e2.
> Aug 17 14:13:41.160: DHCPD: Finding a relay for client 0100.1bd4.60b5.e2
> on interface FastEthernet0/1.2.
> Aug 17 14:13:41.880: DHCPD: DHCPREQUEST received from client
> 0100.1bd4.609c.cf.
> Aug 17 14:13:41.880: DHCPD: Finding a relay for client 0100.1bd4.609c.cf
> on interface FastEthernet0/1.2.
>
> Here are relevant parts of the configuration:
>
> no ip dhcp use vrf connected
> ip dhcp excluded-address 192.168.42.1 192.168.42.2
> !
> ip dhcp pool ITS
>    network 192.168.42.0 255.255.255.0
>    option 150 ip 192.168.42.1
>    default-router 192.168.42.1
> !
> !
> no ip domain lookup
>
> ......................
>
> any ideas? useful show commands?
> most sh ip dhcp ... commands show nothing, except for
>
> Router#sh ip dhcp server statistics
> Memory usage         23512
> Address pools        1
> Database agents      0
> Automatic bindings   0
> Manual bindings      0
> Expired bindings     0
> Malformed messages   0
> Secure arp entries   0
>
> Message              Received
> BOOTREQUEST          0
> DHCPDISCOVER         0
> DHCPREQUEST          1912
> DHCPDECLINE          0
> DHCPRELEASE          0
> DHCPINFORM           0
>
> Message              Sent
> BOOTREPLY            0
> DHCPOFFER            0
> DHCPACK              0
> DHCPNAK              0
>
> Router#sh ip dhcp pool ITS
>
> Pool ITS :
>  Utilization mark (high/low)    : 100 / 0
>  Subnet size (first/next)       : 0 / 0
>  Total addresses                : 254
>  Leased addresses               : 0
>  Pending event                  : none
>  1 subnet is currently in the pool :
>  Current index        IP address range                    Leased addresses
>  192.168.42.1         192.168.42.1     - 192.168.42.254    0
>
> Thanks,
>   Zoltan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From AHolloway at enventis.com  Fri Aug 17 11:18:47 2007
From: AHolloway at enventis.com (Anthony Holloway (Enventis))
Date: Fri, 17 Aug 2007 10:18:47 -0500
Subject: [cisco-voip] DHCP not really working
In-Reply-To: <46C5DAA3.6060407@carocomp.ro>
References: <46C5DAA3.6060407@carocomp.ro>
Message-ID: <C0A504750F3BBE4FADE4ED6D672BA1790F10D649@mail.hickorytech.local>

Can you show us the interface to the voice network (192.168.42.0/24)?

Anthony

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
Sent: Friday, August 17, 2007 12:28 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] DHCP not really working

Hi!

It's a bit off-topic, however I'm all clueless, so any help would be 
appreciated.

I have a CCME (2811, with 12.4(13r)T), with 7911G phones installed. Two 
VLANs, one for data, one for voice.

The router should be responsible for DHCP for the phones, but it does 
not seem to work - phones already in the network have their IP adress, 
keep their address through a reset, but new phones added to the network 
can't get an IP.

However, more people have access to the router and it is very possible 
it might have been misconfigured.

For starters, I noticed a

no service dhcp

in the running config, so I removed that. The dhcp debug messages began 
to appear, but it seems like the router still can't find "itself":

Aug 17 14:13:41.160: DHCPD: DHCPREQUEST received from client 
0100.1bd4.60b5.e2.
Aug 17 14:13:41.160: DHCPD: Finding a relay for client 0100.1bd4.60b5.e2

on interface FastEthernet0/1.2.
Aug 17 14:13:41.880: DHCPD: DHCPREQUEST received from client 
0100.1bd4.609c.cf.
Aug 17 14:13:41.880: DHCPD: Finding a relay for client 0100.1bd4.609c.cf

on interface FastEthernet0/1.2.

Here are relevant parts of the configuration:

no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.42.1 192.168.42.2
!
ip dhcp pool ITS
   network 192.168.42.0 255.255.255.0
   option 150 ip 192.168.42.1
   default-router 192.168.42.1
!
!
no ip domain lookup

......................

any ideas? useful show commands?
most sh ip dhcp ... commands show nothing, except for

Router#sh ip dhcp server statistics
Memory usage         23512
Address pools        1
Database agents      0
Automatic bindings   0
Manual bindings      0
Expired bindings     0
Malformed messages   0
Secure arp entries   0

Message              Received
BOOTREQUEST          0
DHCPDISCOVER         0
DHCPREQUEST          1912
DHCPDECLINE          0
DHCPRELEASE          0
DHCPINFORM           0

Message              Sent
BOOTREPLY            0
DHCPOFFER            0
DHCPACK              0
DHCPNAK              0

Router#sh ip dhcp pool ITS         

Pool ITS :
 Utilization mark (high/low)    : 100 / 0
 Subnet size (first/next)       : 0 / 0
 Total addresses                : 254
 Leased addresses               : 0
 Pending event                  : none
 1 subnet is currently in the pool :
 Current index        IP address range                    Leased
addresses
 192.168.42.1         192.168.42.1     - 192.168.42.254    0

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

From wsisk at cisco.com  Fri Aug 17 11:25:53 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 17 Aug 2007 11:25:53 -0400
Subject: [cisco-voip] CM 5.1.2b download
In-Reply-To: <f84a38d30708170817x2fa5abeay19874668547baee@mail.gmail.com>
References: <f84a38d30708170817x2fa5abeay19874668547baee@mail.gmail.com>
Message-ID: <46C5BE01.1020706@cisco.com>

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From craig at staffin.org  Fri Aug 17 11:29:50 2007
From: craig at staffin.org (Craig Staffin)
Date: Fri, 17 Aug 2007 10:29:50 -0500
Subject: [cisco-voip] FN - 62850 - MCS 7825-H2 Running Cisco Unified
	Communications Manager 5.1 and 6.0 or Cisco Unified Presence
	6.0 - Sporadic Server Hang
In-Reply-To: <46C5B344.5060402@cisco.com>
References: <9b76d88c0708170657u3b3cf146x734c5463c08bfa47@mail.gmail.com>
	<46C5B344.5060402@cisco.com>
Message-ID: <9b76d88c0708170829o38b06a8cybfaf17c1c8db9e67@mail.gmail.com>

Thanks Wes!

On 8/17/07, Wes Sisk <wsisk at cisco.com> wrote:
>
>  Hi Craig,
>
> 6.0 is fixed with 6.0(1a).  I will check on getting the doc updated.
>
> /Wes
>
> Craig Staffin wrote:
>
> Wes,
>
> Can you update this doc to include 6.x references.  Right now for instance
> it only refers to a workaround for 5.1 and nothing about 6.x but yet the
> title and description of the problem refer to 6 as well.
>
> On a side note has anyone seen this issue on a 7825??
>
> Thanks
> Craig
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.nethttps://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 svoll.voip at gmail.com  Fri Aug 17 12:10:03 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 17 Aug 2007 09:10:03 -0700
Subject: [cisco-voip] CM 5.1.2b download
In-Reply-To: <46C5BE01.1020706@cisco.com>
References: <f84a38d30708170817x2fa5abeay19874668547baee@mail.gmail.com>
	<46C5BE01.1020706@cisco.com>
Message-ID: <f84a38d30708170910s3d96a96an78d8b8c6d4c2141c@mail.gmail.com>

Thanks Wes

CSCsh20023 Some Browsers and WinZip

When the Cisco Unified CallManager upgrade patch file is downloaded from
cisco.com, some browsers may download files with the extension *tar.gz.sgn*as
*tar.gz.gz*.

Workaround

After the patch file is downloaded, rename it with the extension *.gz.sgn *(in
place of.gz.gz) and proceed with the install.







On 8/17/07, Wes Sisk <wsisk at cisco.com> wrote:
>
> check the release notes, it's your browser. /wes
>
> Scott Voll wrote:
>
>  is anyone else having issues downloading and installing 5.1.2b?
>
> I download it and it downloads as a .gz.gz but it should be gz.sgn.  or am
> I missing a step.  I'm still really new to 5.x
>
> Scott
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From omar.parihuana at gmail.com  Fri Aug 17 12:31:43 2007
From: omar.parihuana at gmail.com (omar parihuana)
Date: Fri, 17 Aug 2007 11:31:43 -0500
Subject: [cisco-voip] Unregistered IP Phones
In-Reply-To: <871149.79463.qm@web50607.mail.re2.yahoo.com>
References: <834c50110708160927g54bbeb90p8d95faaaacebf406@mail.gmail.com>
	<871149.79463.qm@web50607.mail.re2.yahoo.com>
Message-ID: <834c50110708170931n2dfc2c7fn18d6c8a703cb0b04@mail.gmail.com>

Hi,

Thanks for your response...

I re-checked the NIC and no errors, also the mismatch in the speed and
duplexing, all is ok!, neither Firewall between CCM and IP Phones, so I
don-know what's wrong? I checked the CCM resources, network resources, and
all seems ok. what other test can i realize?

Thanks again for your sugesstions...

Rgds.


On 8/16/07, Nicol?s Menoni <nmenoni at yahoo.com> wrote:
>
> I think it could be a problem in the NIC (network
> interface card) of the CCM server. It could be a
> mismatch in the speed or duplexing or something like
> that, for this, the keepalive can't reach the
> destination.
>
> Ok, bye, Nicol?s.
>
>
> --- omar parihuana <omar.parihuana at gmail.com>
> escribi?:
>
> > Hi List,
> >
> > Since two weeks I have issues with my  IP Phones,
> > after check CCM 4.0 I saw
> > a lot of errors by device unregistered, and my users
> > are complain because
> > the calls in course are dropped. I don't know why
> > the IP Phones
> > unregistered, pls any suggestions... thanks.
> >
> > My CCM logs are:
> >
> >
> > Event Type: Error
> >
> > Event Source: Cisco CallManager
> >
> > Event Category: None
> >
> > Event ID: 3
> >
> > Date: 16/08/2007
> >
> > Time: 10:25:04 a.m.
> >
> > User: N/A
> >
> > Computer: PECMA01
> >
> > Description:
> >
> > Error: DeviceUnregistered - Device unregistered.
> >
> > Device name.: SEP0013605818CA
> >
> > Device IP address.: 10.20.0.132
> >
> > Device type. [Optional]: 30007
> >
> > Device description [Optional].: Patricia Verastegui
> > 2237
> >
> > Reason Code [Optional].: 8
> >
> > App ID: Cisco CallManager
> >
> > Cluster ID: PECMA01-Cluster
> >
> > Node ID: 10.20.0.16
> >
> > Explanation: A device that has previously registered
> > with Cisco CallManager
> >
> > has unregistered. This event may be issued as part
> > of normal unregistration
> >
> > event or due to some other reason such as loss of
> > keepalives.
> >
> > Recommended Action: No action is required if
> > unregistration of this device
> >
> > was expected..
> >
> >
> >
> >
> >
> > Event Type: Error
> >
> > Event Source: Cisco CallManager
> >
> > Event Category: None
> >
> > Event ID: 3
> >
> > Date: 16/08/2007
> >
> > Time: 10:24:52 a.m.
> >
> > User: N/A
> >
> > Computer: PECMA01
> >
> > Description:
> >
> > Error: DeviceUnregistered - Device unregistered.
> >
> > Device name.: SEP00131A41940B
> >
> > Device IP address.: 10.20.1.114
> >
> > Device type. [Optional]: 30007
> >
> > Device description [Optional].: Stephanie Kilimajer
> > 2215
> >
> > Reason Code [Optional].: 8
> >
> > App ID: Cisco CallManager
> >
> > Cluster ID: PECMA01-Cluster
> >
> > Node ID: 10.20.0.16
> >
> > Explanation: A device that has previously registered
> > with Cisco CallManager
> >
> > has unregistered. This event may be issued as part
> > of normal unregistration
> >
> > event or due to some other reason such as loss of
> > keepalives.
> >
> > Recommended Action: No action is required if
> > unregistration of this device
> >
> > was expected..
> >
> >
> >
> >
> >
> > Event Type: Error
> >
> > Event Source: Cisco CallManager
> >
> > Event Category: None
> >
> > Event ID: 3
> >
> > Date: 16/08/2007
> >
> > Time: 10:24:34 a.m.
> >
> > User: N/A
> >
> > Computer: PECMA01
> >
> > Description:
> >
> > Error: DeviceTransientConnection - Transient
> > connection attempt.
> >
> > Connecting Port: 0
> >
> > Device name [Optional].:
> >
> > Device IP address.: 10.20.20.45
> >
> > Device type. [Optional]: 255
> >
> > Reason Code [Optional].: 6
> >
> > App ID: Cisco CallManager
> >
> > Cluster ID: PECMA01-Cluster
> >
> > Node ID: 10.20.0.16
> >
> > Explanation: A connection was established and
> > immediately dropped before
> >
> > completing registration. Incomplete registration may
> > indicate a device is
> >
> > rehoming in the middle of registration. The alarm
> > could also indicate a
> >
> > device misconfiguration, database error, or an
> > illegal/unknown device
> >
> > trying to attempt a connection.
> >
> > Recommended Action: No action is required if this
> > event was issued as a
> >
> > result of a normal device rehome..
> >
> >
> >
> >
> >
> >
> >
> > Event Type: Error
> >
> > Event Source: Cisco CallManager
> >
> > Event Category: None
> >
> > Event ID: 3
> >
> > Date: 16/08/2007
> >
> > Time: 10:24:29 a.m.
> >
> > User: N/A
> >
> > Computer: PECMA01
> >
> >
> === message truncated ===>
> _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> __________________________________________________
> Correo Yahoo!
> Espacio para todos tus mensajes, antivirus y antispam ?gratis!
> Reg?strate ya - http://correo.espanol.yahoo.com/
>



-- 
Omar E.P.T
-----------------
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From aizhar at ccbootcamp.com  Fri Aug 17 13:26:34 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Fri, 17 Aug 2007 10:26:34 -0700
Subject: [cisco-voip] Unregistered IP Phones
References: <834c50110708170931n2dfc2c7fn18d6c8a703cb0b04@mail.gmail.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C366356114C@nlimail.nlidomain>

Sounds like a networking problem to me, not necessarily directly connected to call manager.
 
Don't know your topology and services but a general network health check is always good thing to do, check things like broadcast per second, routers and switches cpu and memory load, spanning tree topologies, unstable links and such.
 
TAC case might not be a bad idea in scenarios like that. 
 
-------------------------------------------
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

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of omar parihuana
Sent: Fri 8/17/2007 9:31 AM
To: Nicol?s Menoni
Cc: cisco-voip
Subject: Re: [cisco-voip] Unregistered IP Phones [bcc][faked-from][html-rem]



Hi,

Thanks for your response...

I re-checked the NIC and no errors, also the mismatch in the speed and
duplexing, all is ok!, neither Firewall between CCM and IP Phones, so I
don-know what's wrong? I checked the CCM resources, network resources, and
all seems ok. what other test can i realize?

Thanks again for your sugesstions...

Rgds.


On 8/16/07, Nicol?s Menoni <nmenoni at yahoo.com> wrote:
>
> I think it could be a problem in the NIC (network
> interface card) of the CCM server. It could be a
> mismatch in the speed or duplexing or something like
> that, for this, the keepalive can't reach the
> destination.
>
> Ok, bye, Nicol?s.
>
>
> --- omar parihuana <omar.parihuana at gmail.com>
> escribi?:
>
> > Hi List,
> >
> > Since two weeks I have issues with my  IP Phones,
> > after check CCM 4.0 I saw
> > a lot of errors by device unregistered, and my users
> > are complain because
> > the calls in course are dropped. I don't know why
> > the IP Phones
> > unregistered, pls any suggestions... thanks.
> >
> > My CCM logs are:
> >
> >
> > Event Type: Error
> >
> > Event Source: Cisco CallManager
> >
> > Event Category: None
> >
> > Event ID: 3
> >
> > Date: 16/08/2007
> >
> > Time: 10:25:04 a.m.
> >
> > User: N/A
> >
> > Computer: PECMA01
> >
> > Description:
> >
> > Error: DeviceUnregistered - Device unregistered.
> >
> > Device name.: SEP0013605818CA
> >
> > Device IP address.: 10.20.0.132
> >
> > Device type. [Optional]: 30007
> >
> > Device description [Optional].: Patricia Verastegui
> > 2237
> >
> > Reason Code [Optional].: 8
> >
> > App ID: Cisco CallManager
> >
> > Cluster ID: PECMA01-Cluster
> >
> > Node ID: 10.20.0.16
> >
> > Explanation: A device that has previously registered
> > with Cisco CallManager
> >
> > has unregistered. This event may be issued as part
> > of normal unregistration
> >
> > event or due to some other reason such as loss of
> > keepalives.
> >
> > Recommended Action: No action is required if
> > unregistration of this device
> >
> > was expected..
> >
> >
> >
> >
> >
> > Event Type: Error
> >
> > Event Source: Cisco CallManager
> >
> > Event Category: None
> >
> > Event ID: 3
> >
> > Date: 16/08/2007
> >
> > Time: 10:24:52 a.m.
> >
> > User: N/A
> >
> > Computer: PECMA01
> >
> > Description:
> >
> > Error: DeviceUnregistered - Device unregistered.
> >
> > Device name.: SEP00131A41940B
> >
> > Device IP address.: 10.20.1.114
> >
> > Device type. [Optional]: 30007
> >
> > Device description [Optional].: Stephanie Kilimajer
> > 2215
> >
> > Reason Code [Optional].: 8
> >
> > App ID: Cisco CallManager
> >
> > Cluster ID: PECMA01-Cluster
> >
> > Node ID: 10.20.0.16
> >
> > Explanation: A device that has previously registered
> > with Cisco CallManager
> >
> > has unregistered. This event may be issued as part
> > of normal unregistration
> >
> > event or due to some other reason such as loss of
> > keepalives.
> >
> > Recommended Action: No action is required if
> > unregistration of this device
> >
> > was expected..
> >
> >
> >
> >
> >
> > Event Type: Error
> >
> > Event Source: Cisco CallManager
> >
> > Event Category: None
> >
> > Event ID: 3
> >
> > Date: 16/08/2007
> >
> > Time: 10:24:34 a.m.
> >
> > User: N/A
> >
> > Computer: PECMA01
> >
> > Description:
> >
> > Error: DeviceTransientConnection - Transient
> > connection attempt.
> >
> > Connecting Port: 0
> >
> > Device name [Optional].:
> >
> > Device IP address.: 10.20.20.45
> >
> > Device type. [Optional]: 255
> >
> > Reason Code [Optional].: 6
> >
> > App ID: Cisco CallManager
> >
> > Cluster ID: PECMA01-Cluster
> >
> > Node ID: 10.20.0.16
> >
> > Explanation: A connection was established and
> > immediately dropped before
> >
> > completing registration. Incomplete registration may
> > indicate a device is
> >
> > rehoming in the middle of registration. The alarm
> > could also indicate a
> >
> > device misconfiguration, database error, or an
> > illegal/unknown device
> >
> > trying to attempt a connection.
> >
> > Recommended Action: No action is required if this
> > event was issued as a
> >
> > result of a normal device rehome..
> >
> >
> >
> >
> >
> >
> >
> > Event Type: Error
> >
> > Event Source: Cisco CallManager
> >
> > Event Category: None
> >
> > Event ID: 3
> >
> > Date: 16/08/2007
> >
> > Time: 10:24:29 a.m.
> >
> > User: N/A
> >
> > Computer: PECMA01
> >
> >
> === message truncated ===>
> _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> __________________________________________________
> Correo Yahoo!
> Espacio para todos tus mensajes, antivirus y antispam ?gratis!
> Reg?strate ya - http://correo.espanol.yahoo.com/
>



--
Omar E.P.T
-----------------
_______________________________________________
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 Aug 17 13:40:27 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 17 Aug 2007 10:40:27 -0700 (PDT)
Subject: [cisco-voip] Unregistered IP Phones
In-Reply-To: <834c50110708170931n2dfc2c7fn18d6c8a703cb0b04@mail.gmail.com>
Message-ID: <506289.68052.qm@web57203.mail.re3.yahoo.com>

Why not show us your interface/subinterface
configurations?

--- omar parihuana <omar.parihuana at gmail.com> wrote:

> Hi,
> 
> Thanks for your response...
> 
> I re-checked the NIC and no errors, also the
> mismatch in the speed and
> duplexing, all is ok!, neither Firewall between CCM
> and IP Phones, so I
> don-know what's wrong? I checked the CCM resources,
> network resources, and
> all seems ok. what other test can i realize?
> 
> Thanks again for your sugesstions...
> 
> Rgds.
> 
> 
> On 8/16/07, Nicol?s Menoni <nmenoni at yahoo.com>
> wrote:
> >
> > I think it could be a problem in the NIC (network
> > interface card) of the CCM server. It could be a
> > mismatch in the speed or duplexing or something
> like
> > that, for this, the keepalive can't reach the
> > destination.
> >
> > Ok, bye, Nicol?s.
> >
> >
> > --- omar parihuana <omar.parihuana at gmail.com>
> > escribi?:
> >
> > > Hi List,
> > >
> > > Since two weeks I have issues with my  IP
> Phones,
> > > after check CCM 4.0 I saw
> > > a lot of errors by device unregistered, and my
> users
> > > are complain because
> > > the calls in course are dropped. I don't know
> why
> > > the IP Phones
> > > unregistered, pls any suggestions... thanks.
> > >
> > > My CCM logs are:
> > >
> > >
> > > Event Type: Error
> > >
> > > Event Source: Cisco CallManager
> > >
> > > Event Category: None
> > >
> > > Event ID: 3
> > >
> > > Date: 16/08/2007
> > >
> > > Time: 10:25:04 a.m.
> > >
> > > User: N/A
> > >
> > > Computer: PECMA01
> > >
> > > Description:
> > >
> > > Error: DeviceUnregistered - Device unregistered.
> > >
> > > Device name.: SEP0013605818CA
> > >
> > > Device IP address.: 10.20.0.132
> > >
> > > Device type. [Optional]: 30007
> > >
> > > Device description [Optional].: Patricia
> Verastegui
> > > 2237
> > >
> > > Reason Code [Optional].: 8
> > >
> > > App ID: Cisco CallManager
> > >
> > > Cluster ID: PECMA01-Cluster
> > >
> > > Node ID: 10.20.0.16
> > >
> > > Explanation: A device that has previously
> registered
> > > with Cisco CallManager
> > >
> > > has unregistered. This event may be issued as
> part
> > > of normal unregistration
> > >
> > > event or due to some other reason such as loss
> of
> > > keepalives.
> > >
> > > Recommended Action: No action is required if
> > > unregistration of this device
> > >
> > > was expected..
> > >
> > >
> > >
> > >
> > >
> > > Event Type: Error
> > >
> > > Event Source: Cisco CallManager
> > >
> > > Event Category: None
> > >
> > > Event ID: 3
> > >
> > > Date: 16/08/2007
> > >
> > > Time: 10:24:52 a.m.
> > >
> > > User: N/A
> > >
> > > Computer: PECMA01
> > >
> > > Description:
> > >
> > > Error: DeviceUnregistered - Device unregistered.
> > >
> > > Device name.: SEP00131A41940B
> > >
> > > Device IP address.: 10.20.1.114
> > >
> > > Device type. [Optional]: 30007
> > >
> > > Device description [Optional].: Stephanie
> Kilimajer
> > > 2215
> > >
> > > Reason Code [Optional].: 8
> > >
> > > App ID: Cisco CallManager
> > >
> > > Cluster ID: PECMA01-Cluster
> > >
> > > Node ID: 10.20.0.16
> > >
> > > Explanation: A device that has previously
> registered
> > > with Cisco CallManager
> > >
> > > has unregistered. This event may be issued as
> part
> > > of normal unregistration
> > >
> > > event or due to some other reason such as loss
> of
> > > keepalives.
> > >
> > > Recommended Action: No action is required if
> > > unregistration of this device
> > >
> > > was expected..
> > >
> > >
> > >
> > >
> > >
> > > Event Type: Error
> > >
> > > Event Source: Cisco CallManager
> > >
> > > Event Category: None
> > >
> > > Event ID: 3
> > >
> > > Date: 16/08/2007
> > >
> > > Time: 10:24:34 a.m.
> > >
> > > User: N/A
> > >
> > > Computer: PECMA01
> > >
> > > Description:
> > >
> > > Error: DeviceTransientConnection - Transient
> > > connection attempt.
> > >
> > > Connecting Port: 0
> > >
> > > Device name [Optional].:
> > >
> > > Device IP address.: 10.20.20.45
> > >
> > > Device type. [Optional]: 255
> > >
> > > Reason Code [Optional].: 6
> > >
> > > App ID: Cisco CallManager
> > >
> > > Cluster ID: PECMA01-Cluster
> > >
> > > Node ID: 10.20.0.16
> > >
> > > Explanation: A connection was established and
> > > immediately dropped before
> > >
> > > completing registration. Incomplete registration
> may
> > > indicate a device is
> > >
> > > rehoming in the middle of registration. The
> alarm
> > > could also indicate a
> > >
> > > device misconfiguration, database error, or an
> > > illegal/unknown device
> > >
> > > trying to attempt a connection.
> > >
> > > Recommended Action: No action is required if
> this
> > > event was issued as a
> > >
> > > result of a normal device rehome..
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > Event Type: Error
> > >
> > > Event Source: Cisco CallManager
> > >
> > > Event Category: None
> > >
> > > Event ID: 3
> > >
> > > Date: 16/08/2007
> > >
> > > Time: 10:24:29 a.m.
> > >
> > > User: N/A
> > >
> > > Computer: PECMA01
> > >
> > >
> > === message truncated ===>
> > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> > __________________________________________________
> > Correo Yahoo!
> > Espacio para todos tus mensajes, antivirus y
> antispam ?gratis!
> > Reg?strate ya - http://correo.espanol.yahoo.com/
> >
> 
> 
> 
> -- 
> Omar E.P.T
> -----------------
> > _______________________________________________
> 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.
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From CarlosOrtiz at bayviewfinancial.com  Fri Aug 17 18:24:44 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 17 Aug 2007 18:24:44 -0400
Subject: [cisco-voip] CME vs SRST
Message-ID: <OF2EAE33DE.D853A05F-ON8525733A.00791B1E-8525733A.007B1D57@bftg.com>

Can CME be used as a fallback from Call Mgr instead of SRST? I have never 
thought so......but a previous admin has a CME config on a router at a 
location that has the following excerpt:

ephone-hunt 1 sequential
 pilot 4700
 list 4739, 4740, 4743, 4744
 final 9XXXXXXXXXX
 preference 3 secondary 7
 timeout 16, 16, 16, 16
!
ephone-hunt 2 sequential
 pilot 4799
 list 4702, 4701
 final 9XXXXXXXXXX
 timeout 16, 0

The purpose of this config as explained by the business was when in "SRST" 
(I know this was CME config)  to route calls to a specific DID, hunt 
through lines and as a last resort dial outbound to a telco analog line 
with a mailbox to leave a message (important).

We did SRST testing for all of our locations locations about 2 months ago 
and this particular location did not function at all with this "CME" 
config.  I have since converted them to SRST, had an outage and their 
calls did work.

The business is claiming that this worked at one time during SRST but I am 
skeptical.   I am thinking that the previous admin told him it was 
configured but there was no real testing done.  Can anyone shed some light 
on this.  I need to know:

1) Can CME be used instead of SRST?
2) Can I accomplish what they want via SRST? (see red above - I think I 
can with the alias command but need to check)


  Below is the full CME config if needed.


 telephony-service
 srst mode auto-provision all
 srst dn line-mode dual
 max-ephones 100
 max-dn 200
 ip source-address 10.X.X.X port 2000
 max-conferences 8 gain -6
 web admin system name X password X
 dn-webedit 
 time-webedit 
 transfer-system full-consult
!
ephone-dn  1  dual-line
 number 4744
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  2  dual-line
 number 4743
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  3  dual-line
 number 4702
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  4  dual-line
 number 4745
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  5  dual-line
 number 4739
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  6  dual-line
 number 4742
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  7  dual-line
 number 4740
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  8  dual-line
 number 5539
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  9  dual-line
 number 4701
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  10  dual-line
 number 4710
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  11  dual-line
 number 1010
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  12  dual-line
 number 4703
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  13  dual-line
 number 1012
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  14  dual-line
 number 4704
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  15  dual-line
 number 4636
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  16  dual-line
 number 4705
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  17  dual-line
 number 4706
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  18  dual-line
 number 4707
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone-dn  19  dual-line
 number 7045
 description XXXXXXXXXX
 name XXXXXXXXXX
!
ephone  1
 mac-address 0016.4766.AB7C
 button  1:1
!
ephone  2
 mac-address 0015.F907.6BE1
!
ephone  3
 mac-address 000E.833C.AE08
 button  1:3
!
ephone  4
 mac-address 0011.5C0E.6108
 button  1:4
!
ephone  5
 mac-address 0015.F914.1515
!
ephone  6
 mac-address 0015.F914.0E19
 button  1:6
!
ephone  7
 mac-address 0015.F914.11FF
 button  1:7 2:15
!
ephone  8
 mac-address 0015.F914.0E4C
 button  1:9
!
ephone  9
 mac-address 00E0.7524.B312
 button  1:10
!
ephone  11
 mac-address 0015.FA0B.7202
 button  1:12
!
ephone  12
 mac-address 0017.9486.B302
 button  1:14
!
ephone  13
 mac-address 000F.2398.4524
 button  1:16
!
ephone  14
 mac-address 0012.0194.F685
 button  1:17
!
ephone  15
 mac-address 000F.2356.8144
 button  1:2
!
ephone  16
 mac-address 0014.A917.9FB7
 button  1:13 2:18
!
ephone-hunt 1 sequential
 pilot 4700
 list 4739, 4740, 4743, 4744
 final 9XXXXXXXXXX
 preference 3 secondary 7
 timeout 16, 16, 16, 16
!
ephone-hunt 2 sequential
 pilot 4799
 list 4702, 4701
 final 9XXXXXXXXXX
 timeout 16, 0

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From tednugent69 at yahoo.com  Fri Aug 17 19:04:54 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 17 Aug 2007 16:04:54 -0700 (PDT)
Subject: [cisco-voip] CME vs SRST
In-Reply-To: <OF2EAE33DE.D853A05F-ON8525733A.00791B1E-8525733A.007B1D57@bftg.com>
Message-ID: <584577.22199.qm@web36114.mail.mud.yahoo.com>

Yep, its pretty new

http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00807e0566.html


--- CarlosOrtiz at bayviewfinancial.com wrote:

> Can CME be used as a fallback from Call Mgr instead
> of SRST? I have never 
> thought so......but a previous admin has a CME
> config on a router at a 
> location that has the following excerpt:
> 
> ephone-hunt 1 sequential
>  pilot 4700
>  list 4739, 4740, 4743, 4744
>  final 9XXXXXXXXXX
>  preference 3 secondary 7
>  timeout 16, 16, 16, 16
> !
> ephone-hunt 2 sequential
>  pilot 4799
>  list 4702, 4701
>  final 9XXXXXXXXXX
>  timeout 16, 0
> 
> The purpose of this config as explained by the
> business was when in "SRST" 
> (I know this was CME config)  to route calls to a
> specific DID, hunt 
> through lines and as a last resort dial outbound to
> a telco analog line 
> with a mailbox to leave a message (important).
> 
> We did SRST testing for all of our locations
> locations about 2 months ago 
> and this particular location did not function at all
> with this "CME" 
> config.  I have since converted them to SRST, had an
> outage and their 
> calls did work.
> 
> The business is claiming that this worked at one
> time during SRST but I am 
> skeptical.   I am thinking that the previous admin
> told him it was 
> configured but there was no real testing done.  Can
> anyone shed some light 
> on this.  I need to know:
> 
> 1) Can CME be used instead of SRST?
> 2) Can I accomplish what they want via SRST? (see
> red above - I think I 
> can with the alias command but need to check)
> 
> 
>   Below is the full CME config if needed.
> 
> 
>  telephony-service
>  srst mode auto-provision all
>  srst dn line-mode dual
>  max-ephones 100
>  max-dn 200
>  ip source-address 10.X.X.X port 2000
>  max-conferences 8 gain -6
>  web admin system name X password X
>  dn-webedit 
>  time-webedit 
>  transfer-system full-consult
> !
> ephone-dn  1  dual-line
>  number 4744
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  2  dual-line
>  number 4743
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  3  dual-line
>  number 4702
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  4  dual-line
>  number 4745
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  5  dual-line
>  number 4739
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  6  dual-line
>  number 4742
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  7  dual-line
>  number 4740
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  8  dual-line
>  number 5539
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  9  dual-line
>  number 4701
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  10  dual-line
>  number 4710
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  11  dual-line
>  number 1010
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  12  dual-line
>  number 4703
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  13  dual-line
>  number 1012
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  14  dual-line
>  number 4704
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  15  dual-line
>  number 4636
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  16  dual-line
>  number 4705
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  17  dual-line
>  number 4706
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  18  dual-line
>  number 4707
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone-dn  19  dual-line
>  number 7045
>  description XXXXXXXXXX
>  name XXXXXXXXXX
> !
> ephone  1
>  mac-address 0016.4766.AB7C
>  button  1:1
> !
> ephone  2
>  mac-address 0015.F907.6BE1
> !
> ephone  3
>  mac-address 000E.833C.AE08
>  button  1:3
> !
> ephone  4
>  mac-address 0011.5C0E.6108
>  button  1:4
> !
> ephone  5
>  mac-address 0015.F914.1515
> !
> ephone  6
>  mac-address 0015.F914.0E19
>  button  1:6
> !
> ephone  7
>  mac-address 0015.F914.11FF
>  button  1:7 2:15
> !
> ephone  8
>  mac-address 0015.F914.0E4C
>  button  1:9
> !
> ephone  9
>  mac-address 00E0.7524.B312
>  button  1:10
> !
> ephone  11
>  mac-address 0015.FA0B.7202
>  button  1:12
> !
> ephone  12
>  mac-address 0017.9486.B302
>  button  1:14
> !
> 
=== message truncated ===>
_______________________________________________
> 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  Fri Aug 17 20:49:10 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Fri, 17 Aug 2007 19:49:10 -0500
Subject: [cisco-voip] FN - 62850 - MCS 7825-H2 Running Cisco Unified
	Communications Manager 5.1 and 6.0 or Cisco Unified Presence
	6.0 - Sporadic Server Hang
In-Reply-To: <9b76d88c0708170829o38b06a8cybfaf17c1c8db9e67@mail.gmail.com>
References: <9b76d88c0708170657u3b3cf146x734c5463c08bfa47@mail.gmail.com>
	<46C5B344.5060402@cisco.com>
	<9b76d88c0708170829o38b06a8cybfaf17c1c8db9e67@mail.gmail.com>
Message-ID: <f4445faf0708171749i2df5f76eq49dbd150bf9a6d5d@mail.gmail.com>

I've seen this on a MCS-7825H2 with CCM 5.1.1-3000-5 . The HP patch
appears to have fixed it up (patched it 3-4 weeks ago and so far no
hangs).

On 8/17/07, Craig Staffin <craig at staffin.org> wrote:
> Thanks Wes!
>
>
> On 8/17/07, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > Hi Craig,
> >
> > 6.0 is fixed with 6.0(1a).  I will check on getting the doc updated.
> >
> > /Wes
> >
> > Craig Staffin wrote:
> >
> > Wes,
> >
> > Can you update this doc to include 6.x references.  Right now for instance
> it only refers to a workaround for 5.1 and nothing about 6.x but yet the
> title and description of the problem refer to 6 as well.
> >
> > On a side note has anyone seen this issue on a 7825??
> >
> > Thanks
> > Craig
> >
> > ________________________________
>
> > _______________________________________________
> > 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
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From tednugent69 at yahoo.com  Fri Aug 17 22:52:52 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 17 Aug 2007 19:52:52 -0700 (PDT)
Subject: [cisco-voip] DHCP not really working
In-Reply-To: <46C5DAA3.6060407@carocomp.ro>
Message-ID: <69106.17706.qm@web36101.mail.mud.yahoo.com>

You don't happen to have a helper address on your 
interface FastEthernet0/1.2 interface do you?
Please provide a config for your interfaces and
telephone service. 
Ted

--- Kelemen Zoltan <keli at carocomp.ro> wrote:

> Hi!
> 
> It's a bit off-topic, however I'm all clueless, so
> any help would be 
> appreciated.
> 
> I have a CCME (2811, with 12.4(13r)T), with 7911G
> phones installed. Two 
> VLANs, one for data, one for voice.
> 
> The router should be responsible for DHCP for the
> phones, but it does 
> not seem to work - phones already in the network
> have their IP adress, 
> keep their address through a reset, but new phones
> added to the network 
> can't get an IP.
> 
> However, more people have access to the router and
> it is very possible 
> it might have been misconfigured.
> 
> For starters, I noticed a
> 
> no service dhcp
> 
> in the running config, so I removed that. The dhcp
> debug messages began 
> to appear, but it seems like the router still can't
> find "itself":
> 
> Aug 17 14:13:41.160: DHCPD: DHCPREQUEST received
> from client 
> 0100.1bd4.60b5.e2.
> Aug 17 14:13:41.160: DHCPD: Finding a relay for
> client 0100.1bd4.60b5.e2 
> on interface FastEthernet0/1.2.
> Aug 17 14:13:41.880: DHCPD: DHCPREQUEST received
> from client 
> 0100.1bd4.609c.cf.
> Aug 17 14:13:41.880: DHCPD: Finding a relay for
> client 0100.1bd4.609c.cf 
> on interface FastEthernet0/1.2.
> 
> Here are relevant parts of the configuration:
> 
> no ip dhcp use vrf connected
> ip dhcp excluded-address 192.168.42.1 192.168.42.2
> !
> ip dhcp pool ITS
>    network 192.168.42.0 255.255.255.0
>    option 150 ip 192.168.42.1
>    default-router 192.168.42.1
> !
> !
> no ip domain lookup
> 
> ......................
> 
> any ideas? useful show commands?
> most sh ip dhcp ... commands show nothing, except
> for
> 
> Router#sh ip dhcp server statistics
> Memory usage         23512
> Address pools        1
> Database agents      0
> Automatic bindings   0
> Manual bindings      0
> Expired bindings     0
> Malformed messages   0
> Secure arp entries   0
> 
> Message              Received
> BOOTREQUEST          0
> DHCPDISCOVER         0
> DHCPREQUEST          1912
> DHCPDECLINE          0
> DHCPRELEASE          0
> DHCPINFORM           0
> 
> Message              Sent
> BOOTREPLY            0
> DHCPOFFER            0
> DHCPACK              0
> DHCPNAK              0
> 
> Router#sh ip dhcp pool ITS         
> 
> Pool ITS :
>  Utilization mark (high/low)    : 100 / 0
>  Subnet size (first/next)       : 0 / 0
>  Total addresses                : 254
>  Leased addresses               : 0
>  Pending event                  : none
>  1 subnet is currently in the pool :
>  Current index        IP address range              
>      Leased addresses
>  192.168.42.1         192.168.42.1     -
> 192.168.42.254    0
> 
> 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 zohaibshabir at gmail.com  Sat Aug 18 03:12:49 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Sat, 18 Aug 2007 12:12:49 +0500
Subject: [cisco-voip] a comparison of CCME configuration methods
In-Reply-To: <421560.17861.qm@web57208.mail.re3.yahoo.com>
References: <676641.35480.qm@web36105.mail.mud.yahoo.com>
	<421560.17861.qm@web57208.mail.re3.yahoo.com>
Message-ID: <1ff12970708180012o55d7531epc41e5d5ada5269ce@mail.gmail.com>

Actually i had the experience of working both with QCT and through web
interface. I really like the QCT method because it saves a lot of time, and
it works perfectly on almost all routers.

On 8/17/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> There are three ways of configuring CCME, four if you
> count QCT. I have only used the manual method and the
> semi-automatic method with using ephone auto-reg. I
> wanted to hear anyone's take on the other methods, in
> particular, QCT/fully-automated installation where you
> walk through a "wizard."
>
> I find the ephone auto-reg to be quite useful. I
> wanted to try the "wizard"-based configuration also
> but wanted to hear others opinions on it first.
>
> Paul
>
>
>
>
> ____________________________________________________________________________________
> 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
>



-- 
Regards,
Zohaib Shabir
Network Engineer(Voice)
DWP Group, TECH Division
Ph:+92-302-8232689
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From keli at carocomp.ro  Sat Aug 18 03:52:32 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Sat, 18 Aug 2007 10:52:32 +0300
Subject: [cisco-voip] DHCP not really working
In-Reply-To: <69106.17706.qm@web36101.mail.mud.yahoo.com>
References: <69106.17706.qm@web36101.mail.mud.yahoo.com>
Message-ID: <20070818105232.mi7vjxjjdcc4csgo@mail.carocomp.ro>

I don't have a helper address there. I tried, pointing it to it's own  
IP, but obviously it just got worse :)

Anyway, the phones have the Administrative VLAN set manually, since  
they are on a Dlink PoE switch, however, they were working ok before  
... and seemingly the phones ARE trying to get an IP on the proper  
interface.

Here are some more configs: 0/1.2 is the voice VLAN

interface FastEthernet0/1
  no ip address
  duplex auto
  speed auto
!
interface FastEthernet0/1.1
  encapsulation dot1Q 1 native
  ip address 192.168.181.230 255.255.255.0
  h323-gateway voip interface
  h323-gateway voip bind srcaddr 192.168.181.230
!
interface FastEthernet0/1.2
  encapsulation dot1Q 42
  ip address 192.168.42.1 255.255.255.0
!

!
telephony-service
  load 7910 P00403020214
  load 7960-7940 P00307020200
  load 7911 SCCP11.8-0-4SR3AS
  max-ephones 42
  max-dn 42
  ip source-address 192.168.42.1 port 2000
  auto assign 1 to 30
  timeouts interdigit 5
  system message KESZ Srl
  url directories http://1.2.3.4/fw/apps/Speedy/xml/directory.asp
  url authentication http://1.2.3.4/fw/authenticate.asp
  time-zone 26
  time-format 24
  voicemail 899
  max-conferences 8 gain -6
  moh 14_kosheen_-_your_life.wav
  multicast moh 239.23.4.10 port 2000
  web admin system name *** password ***
  dn-webedit
  time-webedit
  transfer-system full-consult
  log password cisco
  secondary-dialtone 9
  xmltest
  create cnf-files version-stamp 7960 Aug 17 2007 17:32:58
!

thanks,
   Zoltan

Quoting Ted Nugent <tednugent69 at yahoo.com>:

> You don't happen to have a helper address on your
> interface FastEthernet0/1.2 interface do you?
> Please provide a config for your interfaces and
> telephone service.
> Ted
>
> --- Kelemen Zoltan <keli at carocomp.ro> wrote:
>
>> Hi!
>>
>> It's a bit off-topic, however I'm all clueless, so
>> any help would be
>> appreciated.
>>
>> I have a CCME (2811, with 12.4(13r)T), with 7911G
>> phones installed. Two
>> VLANs, one for data, one for voice.
>>
>> The router should be responsible for DHCP for the
>> phones, but it does
>> not seem to work - phones already in the network
>> have their IP adress,
>> keep their address through a reset, but new phones
>> added to the network
>> can't get an IP.
>>
>> However, more people have access to the router and
>> it is very possible
>> it might have been misconfigured.
>>
>> For starters, I noticed a
>>
>> no service dhcp
>>
>> in the running config, so I removed that. The dhcp
>> debug messages began
>> to appear, but it seems like the router still can't
>> find "itself":
>>
>> Aug 17 14:13:41.160: DHCPD: DHCPREQUEST received
>> from client
>> 0100.1bd4.60b5.e2.
>> Aug 17 14:13:41.160: DHCPD: Finding a relay for
>> client 0100.1bd4.60b5.e2
>> on interface FastEthernet0/1.2.
>> Aug 17 14:13:41.880: DHCPD: DHCPREQUEST received
>> from client
>> 0100.1bd4.609c.cf.
>> Aug 17 14:13:41.880: DHCPD: Finding a relay for
>> client 0100.1bd4.609c.cf
>> on interface FastEthernet0/1.2.
>>
>> Here are relevant parts of the configuration:
>>
>> no ip dhcp use vrf connected
>> ip dhcp excluded-address 192.168.42.1 192.168.42.2
>> !
>> ip dhcp pool ITS
>>    network 192.168.42.0 255.255.255.0
>>    option 150 ip 192.168.42.1
>>    default-router 192.168.42.1
>> !
>> !
>> no ip domain lookup
>>
>> ......................
>>
>> any ideas? useful show commands?
>> most sh ip dhcp ... commands show nothing, except
>> for
>>
>> Router#sh ip dhcp server statistics
>> Memory usage         23512
>> Address pools        1
>> Database agents      0
>> Automatic bindings   0
>> Manual bindings      0
>> Expired bindings     0
>> Malformed messages   0
>> Secure arp entries   0
>>
>> Message              Received
>> BOOTREQUEST          0
>> DHCPDISCOVER         0
>> DHCPREQUEST          1912
>> DHCPDECLINE          0
>> DHCPRELEASE          0
>> DHCPINFORM           0
>>
>> Message              Sent
>> BOOTREPLY            0
>> DHCPOFFER            0
>> DHCPACK              0
>> DHCPNAK              0
>>
>> Router#sh ip dhcp pool ITS
>>
>> Pool ITS :
>>  Utilization mark (high/low)    : 100 / 0
>>  Subnet size (first/next)       : 0 / 0
>>  Total addresses                : 254
>>  Leased addresses               : 0
>>  Pending event                  : none
>>  1 subnet is currently in the pool :
>>  Current index        IP address range
>>      Leased addresses
>>  192.168.42.1         192.168.42.1     -
>> 192.168.42.254    0
>>
>> Thanks,
>>   Zoltan
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
>
>
> ____________________________________________________________________________________
> Yahoo! oneSearch: Finally, mobile search
> that gives answers, not web links.
> http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC
>



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.


From tednugent69 at yahoo.com  Sat Aug 18 08:38:45 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sat, 18 Aug 2007 05:38:45 -0700 (PDT)
Subject: [cisco-voip] DHCP not really working
In-Reply-To: <20070818105232.mi7vjxjjdcc4csgo@mail.carocomp.ro>
Message-ID: <495994.63784.qm@web36103.mail.mud.yahoo.com>

For testing why not add another network and DHCP scope
to the unused interface of your router plug that into
an access port (single vlan,not trunk) on the switch
on the same vlan as one of the phones (also plugged
into an access port), clear the config on the phone
and see if it will pull an ip on a flat network.


--- keli at carocomp.ro wrote:

> I don't have a helper address there. I tried,
> pointing it to it's own  
> IP, but obviously it just got worse :)
> 
> Anyway, the phones have the Administrative VLAN set
> manually, since  
> they are on a Dlink PoE switch, however, they were
> working ok before  
> ... and seemingly the phones ARE trying to get an IP
> on the proper  
> interface.
> 
> Here are some more configs: 0/1.2 is the voice VLAN
> 
> interface FastEthernet0/1
>   no ip address
>   duplex auto
>   speed auto
> !
> interface FastEthernet0/1.1
>   encapsulation dot1Q 1 native
>   ip address 192.168.181.230 255.255.255.0
>   h323-gateway voip interface
>   h323-gateway voip bind srcaddr 192.168.181.230
> !
> interface FastEthernet0/1.2
>   encapsulation dot1Q 42
>   ip address 192.168.42.1 255.255.255.0
> !
> 
> !
> telephony-service
>   load 7910 P00403020214
>   load 7960-7940 P00307020200
>   load 7911 SCCP11.8-0-4SR3AS
>   max-ephones 42
>   max-dn 42
>   ip source-address 192.168.42.1 port 2000
>   auto assign 1 to 30
>   timeouts interdigit 5
>   system message KESZ Srl
>   url directories
> http://1.2.3.4/fw/apps/Speedy/xml/directory.asp
>   url authentication
> http://1.2.3.4/fw/authenticate.asp
>   time-zone 26
>   time-format 24
>   voicemail 899
>   max-conferences 8 gain -6
>   moh 14_kosheen_-_your_life.wav
>   multicast moh 239.23.4.10 port 2000
>   web admin system name *** password ***
>   dn-webedit
>   time-webedit
>   transfer-system full-consult
>   log password cisco
>   secondary-dialtone 9
>   xmltest
>   create cnf-files version-stamp 7960 Aug 17 2007
> 17:32:58
> !
> 
> thanks,
>    Zoltan
> 
> Quoting Ted Nugent <tednugent69 at yahoo.com>:
> 
> > You don't happen to have a helper address on your
> > interface FastEthernet0/1.2 interface do you?
> > Please provide a config for your interfaces and
> > telephone service.
> > Ted
> >
> > --- Kelemen Zoltan <keli at carocomp.ro> wrote:
> >
> >> Hi!
> >>
> >> It's a bit off-topic, however I'm all clueless,
> so
> >> any help would be
> >> appreciated.
> >>
> >> I have a CCME (2811, with 12.4(13r)T), with 7911G
> >> phones installed. Two
> >> VLANs, one for data, one for voice.
> >>
> >> The router should be responsible for DHCP for the
> >> phones, but it does
> >> not seem to work - phones already in the network
> >> have their IP adress,
> >> keep their address through a reset, but new
> phones
> >> added to the network
> >> can't get an IP.
> >>
> >> However, more people have access to the router
> and
> >> it is very possible
> >> it might have been misconfigured.
> >>
> >> For starters, I noticed a
> >>
> >> no service dhcp
> >>
> >> in the running config, so I removed that. The
> dhcp
> >> debug messages began
> >> to appear, but it seems like the router still
> can't
> >> find "itself":
> >>
> >> Aug 17 14:13:41.160: DHCPD: DHCPREQUEST received
> >> from client
> >> 0100.1bd4.60b5.e2.
> >> Aug 17 14:13:41.160: DHCPD: Finding a relay for
> >> client 0100.1bd4.60b5.e2
> >> on interface FastEthernet0/1.2.
> >> Aug 17 14:13:41.880: DHCPD: DHCPREQUEST received
> >> from client
> >> 0100.1bd4.609c.cf.
> >> Aug 17 14:13:41.880: DHCPD: Finding a relay for
> >> client 0100.1bd4.609c.cf
> >> on interface FastEthernet0/1.2.
> >>
> >> Here are relevant parts of the configuration:
> >>
> >> no ip dhcp use vrf connected
> >> ip dhcp excluded-address 192.168.42.1
> 192.168.42.2
> >> !
> >> ip dhcp pool ITS
> >>    network 192.168.42.0 255.255.255.0
> >>    option 150 ip 192.168.42.1
> >>    default-router 192.168.42.1
> >> !
> >> !
> >> no ip domain lookup
> >>
> >> ......................
> >>
> >> any ideas? useful show commands?
> >> most sh ip dhcp ... commands show nothing, except
> >> for
> >>
> >> Router#sh ip dhcp server statistics
> >> Memory usage         23512
> >> Address pools        1
> >> Database agents      0
> >> Automatic bindings   0
> >> Manual bindings      0
> >> Expired bindings     0
> >> Malformed messages   0
> >> Secure arp entries   0
> >>
> >> Message              Received
> >> BOOTREQUEST          0
> >> DHCPDISCOVER         0
> >> DHCPREQUEST          1912
> >> DHCPDECLINE          0
> >> DHCPRELEASE          0
> >> DHCPINFORM           0
> >>
> >> Message              Sent
> >> BOOTREPLY            0
> >> DHCPOFFER            0
> >> DHCPACK              0
> >> DHCPNAK              0
> >>
> >> Router#sh ip dhcp pool ITS
> >>
> >> Pool ITS :
> >>  Utilization mark (high/low)    : 100 / 0
> >>  Subnet size (first/next)       : 0 / 0
> >>  Total addresses                : 254
> >>  Leased addresses               : 0
> >>  Pending event                  : none
> >>  1 subnet is currently in the pool :
> >>  Current index        IP address range
> >>      Leased addresses
> >>  192.168.42.1         192.168.42.1     -
> >> 192.168.42.254    0
> >>
> >> Thanks,
> >>   Zoltan
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> >
> >
> >
>
____________________________________________________________________________________
> > Yahoo! oneSearch: Finally, mobile search
> > that gives answers, not web links.
> >
>
http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC
> >
> 
=== message truncated ===



      ____________________________________________________________________________________
Shape Yahoo! in your own image.  Join our Network Research Panel today!   http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 



From joe.j.keller at gmail.com  Sat Aug 18 19:24:59 2007
From: joe.j.keller at gmail.com (Joe Keller)
Date: Sat, 18 Aug 2007 19:24:59 -0400
Subject: [cisco-voip] CCM 5.1 RAM feedback
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006795405@USNAEXCH.na.didata.local>
References: <189b5daa0708160707m7dd64f5dsd9b6dcfa84028dfa@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F00675845E@USNAEXCH.na.didata.local>
	<189b5daa0708161048p6aceb8a6m1549f741b7c1e228@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F006795405@USNAEXCH.na.didata.local>
Message-ID: <189b5daa0708181624qdee0ba0l8e2ecff43dbc7b25@mail.gmail.com>

If CCM 5.1 OS is 64-bit, the extra RAM could be used by other
processes/apps, even if CCM 5.1 app could only use a maximum of 4GB. Can
anyone comment if the base OS of CCM 5.1 is 32-bit or 64-bit?


On 8/16/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  Is the hardware and RHE it's running on 64-bit (maybe seen from boot
> console), would be a good question! I haven't seen a 64-bit app run on
> 32-bit hardware.
>
>
>
> I hadn't heard of a 64-bit edition of CallManager, but I could be wrong.
>
>
>
> *From:* Joe Keller [mailto:joe.j.keller at gmail.com]
> *Sent:* Thursday, August 16, 2007 1:49 PM
> *To:* Jason Aarons (US)
> *Subject:* Re: [cisco-voip] CCM 5.1 RAM feedback
>
>
>
> So CCM 5.1 is 32-bit?
>
> On 8/16/07, *Jason Aarons (US)* <jason.aarons at us.didata.com> wrote:
>
> 32-bit address space is limited to 4GB addressable, do a Google search for
> more info on 32-bit limitations.  This is one reason Exchange 2007 is 64-bit
> and requires 2003 Server 64 Bit.
>
>
>
> You should be fine with 4GB, you can use RTMT to monitor hardware
> resources.
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net]
> *On Behalf Of *Joe Keller
> *Sent:* Thursday, August 16, 2007 10:08 AM
> *To:* Cisco-VoIP mailing list
> *Subject:* [cisco-voip] CCM 5.1 RAM feedback
>
>
>
> Any suggestions on minimum for 2000 users on Pub & Sub? 3GB on both OK?
> With RAM so cheap, I was looking at 5GB on Pub, and maybe Subs.
> Any noticeable performance improvement with more RAM?
>
> 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. *
>
>
>
> ------------------------------
>
> * 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 mmilanovic at ctiusa.com  Sat Aug 18 20:41:44 2007
From: mmilanovic at ctiusa.com (Mladen Milanovic)
Date: Sat, 18 Aug 2007 20:41:44 -0400
Subject: [cisco-voip] CME vs SRST
In-Reply-To: <OF2EAE33DE.D853A05F-ON8525733A.00791B1E-8525733A.007B1D57@bftg.com>
References: <OF2EAE33DE.D853A05F-ON8525733A.00791B1E-8525733A.007B1D57@bftg.com>
Message-ID: <4761C60AD1E74A4D8606CAF48582265AC1F13B@EXCHANGE.ctiusa.com>

Carlos,
 
you may use CCME instead of SRST in case of CallManager failure. This
aproach provide you more features than SRST, like hunt groups, basic
ACD, and if configured properly, entire CallManager failure can be
totally invisible for the enduser.
 
Defenetly, my recomendation is to go to the CCME SRST Fallback support
because this can provide more features than SRST. You may find more info
on:
 
 
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configura
tion_guide_chapter09186a00807e0566.html
 
 
 
Mladen

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Friday, August 17, 2007 6:25 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME vs SRST



Can CME be used as a fallback from Call Mgr instead of SRST? I have
never thought so......but a previous admin has a CME config on a router
at a location that has the following excerpt: 

ephone-hunt 1 sequential
pilot 4700
list 4739, 4740, 4743, 4744
final 9XXXXXXXXXX
preference 3 secondary 7
timeout 16, 16, 16, 16
!
ephone-hunt 2 sequential
pilot 4799
list 4702, 4701
final 9XXXXXXXXXX
timeout 16, 0

The purpose of this config as explained by the business was when in
"SRST" (I know this was CME config)  to route calls to a specific DID,
hunt through lines and as a last resort dial outbound to a telco analog
line with a mailbox to leave a message (important). 

We did SRST testing for all of our locations locations about 2 months
ago and this particular location did not function at all with this "CME"
config.  I have since converted them to SRST, had an outage and their
calls did work. 

The business is claiming that this worked at one time during SRST but I
am skeptical.   I am thinking that the previous admin told him it was
configured but there was no real testing done.  Can anyone shed some
light on this.  I need to know: 

1) Can CME be used instead of SRST? 
2) Can I accomplish what they want via SRST? (see red above - I think I
can with the alias command but need to check) 


  Below is the full CME config if needed. 


 telephony-service
srst mode auto-provision all
srst dn line-mode dual
max-ephones 100
max-dn 200
ip source-address 10.X.X.X port 2000
max-conferences 8 gain -6
web admin system name X password X
dn-webedit 
time-webedit 
transfer-system full-consult
!
ephone-dn  1  dual-line
number 4744
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  2  dual-line
number 4743
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  3  dual-line
number 4702
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  4  dual-line
number 4745
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  5  dual-line
number 4739
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  6  dual-line
number 4742
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  7  dual-line
number 4740
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  8  dual-line
number 5539
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  9  dual-line
number 4701
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  10  dual-line
number 4710
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  11  dual-line
number 1010
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  12  dual-line
number 4703
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  13  dual-line
number 1012
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  14  dual-line
number 4704
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  15  dual-line
number 4636
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  16  dual-line
number 4705
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  17  dual-line
number 4706
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  18  dual-line
number 4707
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone-dn  19  dual-line
number 7045
description XXXXXXXXXX
name XXXXXXXXXX
!
ephone  1
mac-address 0016.4766.AB7C
button  1:1
!
ephone  2
mac-address 0015.F907.6BE1
!
ephone  3
mac-address 000E.833C.AE08
button  1:3
!
ephone  4
mac-address 0011.5C0E.6108
button  1:4
!
ephone  5
mac-address 0015.F914.1515
!
ephone  6
mac-address 0015.F914.0E19
button  1:6
!
ephone  7
mac-address 0015.F914.11FF
button  1:7 2:15
!
ephone  8
mac-address 0015.F914.0E4C
button  1:9
!
ephone  9
mac-address 00E0.7524.B312
button  1:10
!
ephone  11
mac-address 0015.FA0B.7202
button  1:12
!
ephone  12
mac-address 0017.9486.B302
button  1:14
!
ephone  13
mac-address 000F.2398.4524
button  1:16
!
ephone  14
mac-address 0012.0194.F685
button  1:17
!
ephone  15
mac-address 000F.2356.8144
button  1:2
!
ephone  16
mac-address 0014.A917.9FB7
button  1:13 2:18
!
ephone-hunt 1 sequential
pilot 4700
list 4739, 4740, 4743, 4744
final 9XXXXXXXXXX
preference 3 secondary 7
timeout 16, 16, 16, 16
!
ephone-hunt 2 sequential
pilot 4799
list 4702, 4701
final 9XXXXXXXXXX
timeout 16, 0


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From sidumorjens at gmail.com  Sun Aug 19 11:46:33 2007
From: sidumorjens at gmail.com (Joe Smith)
Date: Sun, 19 Aug 2007 17:46:33 +0200
Subject: [cisco-voip] Display CallerInfo based on PSTN A-number
In-Reply-To: <8b68deba0708190837g5cbc4fe4ya90bb466b616ebcd@mail.gmail.com>
References: <8b68deba0708190837g5cbc4fe4ya90bb466b616ebcd@mail.gmail.com>
Message-ID: <8b68deba0708190846v3af0752atf4f05df056d623f8@mail.gmail.com>

Hello Everyone,

Im looking for a solution where caller information such as Name and Company
is displayed on Cisco 79x0/79x1 phone displays based on the incoming PSTN
A-number. The caller info could be retreived from a text file or ideally
from a LDAP directory such as CCM or AD.

The CallerInfo.jsp script in combination with PhotoDirectory.jsp from Cisco
IP Phone Services SDK seems to be out of the question in this scenario as
thay are designed for calls made between internal DN's. Perhaps someone has
found a way to tweak callerinfo.jsp for external PSTN calls?

A free tool called CallTracer from MyVoIPsource.com seems to be the obvious
choice in a CallManager 4.x environment. Im looking for a solution that
works with CCM 5.x and it seems like nobody has CallTracer working with CCM
5.x version to date..

A company called 4IPT has a product called 'WhoIsCalling' that does the job
but im hoping to find a non-commercial solution ;)

Any sugestions are most welcome!
//Stargazer
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From jgrace at adcapnet.com  Sun Aug 19 13:13:35 2007
From: jgrace at adcapnet.com (James Grace)
Date: Sun, 19 Aug 2007 13:13:35 -0400
Subject: [cisco-voip] Best way to covert skinny ipphone to SIP
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C736@mail.adcapnetsvc.local>

?
Im in the process of configuring CM 6.0 with SIP.  We are currently running as skinny. Want to know will the ipphone automatically converter over to SIP when we turn on the new CM with 6 SIP.  Im Just trying to see if I have to touch every phone or not,  since they are already deployed.  The phones are 7940-60
James Grace 
 
Systems Engineer
jgrace at 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 Aug 2007 13:13:29 -0400
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From patrick.diener at gmail.com  Sun Aug 19 13:20:50 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Sun, 19 Aug 2007 19:20:50 +0200
Subject: [cisco-voip] Best way to covert skinny ipphone to SIP
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C736@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C736@mail.adcapnetsvc.local>
Message-ID: <c0deb5b80708191020m2cfe3aaaqb2dbb7ff217a3b03@mail.gmail.com>

theres a BAT Task that can do this conversation for you. Unfortunately
there is no automatic way back to SCCP...

Regards
Patrick

On 8/19/07, James Grace <jgrace at adcapnet.com> wrote:
> ?
>
>
> Im in the process of configuring CM 6.0 with SIP.  We are currently running
> as skinny. Want to know will the ipphone automatically converter over to SIP
> when we turn on the new CM with 6 SIP.  Im Just trying to see if I have to
> touch every phone or not,  since they are already deployed.  The phones are
> 7940-60
>
>
> James Grace
>
> Systems Engineer
>
> jgrace at 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 Aug 2007
> 13:13:29 -0400
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jgrace at adcapnet.com  Sun Aug 19 15:04:13 2007
From: jgrace at adcapnet.com (James Grace)
Date: Sun, 19 Aug 2007 15:04:13 -0400
Subject: [cisco-voip] Best way to convert MGCP to SIP
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C739@mail.adcapnetsvc.local>

?
Im in the process of configuring CM 6.0 with SIP.  We are currently running as MGCP. Want to know will the ipphone automatically converter over to SIP when we turn on the new CM with 6 SIP.  Im Just trying to see if I have to touch every phone or edit some files,  since they are already deployed.  The phones are 7940-60
James Grace 
 
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 Aug 2007 15:09:08 -0400
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From mh at markholloway.com  Sun Aug 19 15:34:27 2007
From: mh at markholloway.com (Mark Holloway)
Date: Sun, 19 Aug 2007 12:34:27 -0700
Subject: [cisco-voip] Best way to covert skinny ipphone to SIP
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C736@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C736@mail.adcapnetsvc.local>
Message-ID: <003f01c7e297$f4c579b0$de506d10$@com>

Occasionally I run into phones that need SIP 8.x but came with the factory
load 3.02 image.  For phones that are running SCCP v3.x or v4.x, upgrade
them to SCCP 5.x first, then you can upgrade them to the latest SIP version.
If you don't use Call Manager to do the upgrade you can create an
XMLDefault.cnf file for the SCCP 5.x upgrades on a TFTP server.  Then you
can either edit the XMLDefault.cnf file to reflect SIP firmware or create a
SIPDefault.cnf and OS79XX.txt file set (but you must rename or remove the
XMLDefault.cnf file).  Typically you would use the latter method if your
Cisco phones are being used with a third party IP PBX like Asterisk or a
feature server like Broadworks. 

 

 

 

 

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Sunday, August 19, 2007 10:14 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Best way to covert skinny ipphone to SIP

 

?

Im in the process of configuring CM 6.0 with SIP.  We are currently running
as skinny. Want to know will the ipphone automatically converter over to SIP
when we turn on the new CM with 6 SIP.  Im Just trying to see if I have to
touch every phone or not,  since they are already deployed.  The phones are
7940-60

James Grace 

 

Systems Engineer

jgrace at 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 Aug 2007
13:13:29 -0400
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From jsteinberg at gmail.com  Sun Aug 19 18:23:00 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Sun, 19 Aug 2007 18:23:00 -0400
Subject: [cisco-voip] Best way to convert MGCP to SIP
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C739@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C739@mail.adcapnetsvc.local>
Message-ID: <f1b65f880708191523j70ee6128y661cf4efeb8cee69@mail.gmail.com>

I'm just curious here, what is the factor driving you to convert these
phones to SIP?  Why not use SCCP? The 7940 & 7960 loose quite a bit of
features with the SIP load.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_chapter09186a008085f625.html#wp1048590

Justin

On 8/19/07, James Grace <jgrace at adcapnet.com> wrote:
>
> ?  Im in the process of configuring CM 6.0 with SIP.  We are currently
> running as MGCP. Want to know will the ipphone automatically converter over
> to SIP when we turn on the new CM with 6 SIP.  Im Just trying to see if I
> have to touch every phone or edit some files,  since they are already
> deployed.  The phones are 7940-60
>  James Grace
>
>  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,
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>
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From Nick.Kassel at Charles-Stanley.co.uk  Mon Aug 20 06:11:27 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 20 Aug 2007 11:11:27 +0100
Subject: [cisco-voip] Recieved Calls List Digits Lose Prefix
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC014520BB@CSEXCH01.charles-stanley.co.uk>

Excellent!!!

-----Original Message-----
From: mika at loukola.com [mailto:mika at loukola.com] 
Sent: 20 August 2007 11:25
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: RE: Recieved Calls List Digits Lose Prefix

Hi,

I worked right away after software upgrade.
Thanks a lot for the helpful hint !

Regards,

Mika

> I think we were on 123-11 T3, I would recommend upgrading to 12.4 11
T2,
> we have been using this with no problems at all.
>
> -----Original Message-----
> From: mika at loukola.com [mailto:mika at loukola.com]
> Sent: 13 August 2007 18:33
> To: Nick Kassel
> Cc: cisco-voip at puck.nether.net
> Subject: RE: Recieved Calls List Digits Lose Prefix
>
> Hi,
>
> I have a 2811 with c2800nm-spservicesk9-mz.123-11.T10.bin
>
> Mika
>
>> Hi Mika
>>
>> I can't remember the exactly but I think it may have been an issue
the
>> IOS version, what IOS are you using?
>>
>>
>> -----Original Message-----
>> From: mika at loukola.com [mailto:mika at loukola.com]
>> Sent: 12 August 2007 22:44
>> To: Nick Kassel
>> Subject: Re: Recieved Calls List Digits Lose Prefix
>>
>> Hi Nick,
>>
>> I saw the email thread regarding the issue with gateway
> translation-rule
>> is being dropped once the call is answered.
>>   http://www.threebit.net/mail-archive/cisco-voip/msg04775.html
>>
>> I have the same program with my CallManager setup.
>>
>> Did you find a solution to this proglem?
>> (I didn't see the solution as part of the thread)
>>
>> Kind regards,
>>
>> Mika Loukola
>>
>>
>> ********************SAVE PAPER - THINK BEFORE YOU
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********************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 
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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.
*******************************************************************************************************


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


From ErasmuE4 at telkom.co.za  Mon Aug 20 06:21:53 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 20 Aug 2007 12:21:53 +0200
Subject: [cisco-voip] ccm 4-2 remote branch attendant problem
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01984EEC@CNTRRA20-XCS01.telkom.co.za>


Hi

\

I have a central call manager on a big campus and one remote branch.
The central site had serious issues with calls being stuck in the
cisco attendant console Queue. Cisco eventually gave up and told us to
rather install ARC. I am now trying to convince the customer to fork
out the bucks for it. 

 

Now one of the branches are playing up - also using the cisco
attendant console software combined with 7960 phones. I verified the
Qos on our wan MPLS links and looks fine - no drops in the Realtime
class where I put the g729 voice traffic as well as the signaling
traffic etc.  But the remote attendants complain about their consoles
freezing up. Has anyone - any good suggestions on an easy way to
monitor the attendant signaling with CCM. I looked at the Qos already
and also tested basic IP connectivity when they complain plus looked
at some of the performance counters in prefmon fro call manager
attendant and things look normal.

 

 

 

erik 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From aman.chugh at gmail.com  Mon Aug 20 06:53:05 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Mon, 20 Aug 2007 16:23:05 +0530
Subject: [cisco-voip] ccm 4-2 remote branch attendant problem
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F01984EEC@CNTRRA20-XCS01.telkom.co.za>
References: <472E195E692EE74BAE4E7B2880A78C1F01984EEC@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <d4c697ad0708200353n6b0f4867sa7d8bd4c60993e24@mail.gmail.com>

Eric,

I have seen issues with these before , this might help, check whether the
remote attendant has screensaver enabled on her computer where she has AC
installed. AC Client hangs when screensaver kicks in.

Thanks

Aman





On 8/20/07, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote:
>
>  Hi
>
> \
>
> I have a central call manager on a big campus and one remote branch. The
> central site had serious issues with calls being stuck in the cisco
> attendant console Queue. Cisco eventually gave up and told us to rather
> install ARC. I am now trying to convince the customer to fork out the bucks
> for it.
>
>
>
> Now one of the branches are playing up ? also using the cisco attendant
> console software combined with 7960 phones. I verified the Qos on our wan
> MPLS links and looks fine ? no drops in the Realtime class where I put the
> g729 voice traffic as well as the signaling traffic etc.  But the remote
> attendants complain about their consoles freezing up. Has anyone ? any good
> suggestions on an easy way to monitor the attendant signaling with CCM. I
> looked at the Qos already and also tested basic IP connectivity when they
> complain plus looked at some of the performance counters in prefmon fro call
> manager attendant and things look normal.
>
>
>
>
>
>
>
> erik
>   ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> This e-mail and its contents are subject to the Telkom SA Limited
> e-mail legal notice available at
> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>
> _______________________________________________
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From Richard.vanBerkel at telindus.nl  Mon Aug 20 07:03:34 2007
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Mon, 20 Aug 2007 13:03:34 +0200
Subject: [cisco-voip] Unity versus Microsoft Exchange 2007
References: <472E195E692EE74BAE4E7B2880A78C1F01984EEC@CNTRRA20-XCS01.telkom.co.za>
	<d4c697ad0708200353n6b0f4867sa7d8bd4c60993e24@mail.gmail.com>
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D886014F2C31@tnl1mx1.telindus.intra>

 

Hi,

 

Does anybody know if it it possible to set up Unified Messaging  with
Unified Communication Manager 6.X and Exchange 2007?

 

If so what version have you used to do this?

 

Groeten Richard,

Mailto:Richard.vanBerkel at Telindus.nl
<mailto:Richard.vanBerkel at Telindus.nl> 

  

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From keli at carocomp.ro  Mon Aug 20 14:29:32 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Mon, 20 Aug 2007 21:29:32 +0300
Subject: [cisco-voip] DHCP not really working [solved]
In-Reply-To: <495994.63784.qm@web36103.mail.mud.yahoo.com>
References: <495994.63784.qm@web36103.mail.mud.yahoo.com>
Message-ID: <46C9DD8C.3050209@carocomp.ro>

Hi,

Well it turns out, the DHCP server in Cisco did restart after the "no 
service dhcp" line was removed from the config, but the 7911G phones act 
rather irritatingly stupid as DHCP clients. They do not adhere to lease 
times, but only try to refresh their IP addresses in the DHCP server 
indefinitely.

Whatever a phone has received, it will not "let go" unless you 
explicitly set so in the phone's network menu.

The debugs I was seeing were related probably to the fact, that phones 
were trying to "refresh" the leased IP, but the router did not know 
about such a lease (since the service was stopped, it's database was 
empty). However none of the phones did a complete DHCP release/renew, 
even though their adresses have expired much longer that 24hs before.

The newly added phone (as it was a remote location, I didn't do it, but 
instructed somebody else) took an IP from the data VLAN before setting 
it's administrative VLAN ID. After the latter was set, the phone got 
blocked in a limbo, being in the voice VLAN, but looking for IPs in the 
data VLAN. And this phone didn't "let go" neither of it's IP, nor it's 
DHCP server, even after a long weekend.

So, my (rather miserable) opinion of the 7906/7911 series phones still 
holds, unfortunately ...

regards,
  Zoltan

Ted Nugent wrote:
> For testing why not add another network and DHCP scope
> to the unused interface of your router plug that into
> an access port (single vlan,not trunk) on the switch
> on the same vlan as one of the phones (also plugged
> into an access port), clear the config on the phone
> and see if it will pull an ip on a flat network.
>
>
> --- keli at carocomp.ro wrote:
>
>   
>> I don't have a helper address there. I tried,
>> pointing it to it's own  
>> IP, but obviously it just got worse :)
>>
>> Anyway, the phones have the Administrative VLAN set
>> manually, since  
>> they are on a Dlink PoE switch, however, they were
>> working ok before  
>> ... and seemingly the phones ARE trying to get an IP
>> on the proper  
>> interface.
>>
>> Here are some more configs: 0/1.2 is the voice VLAN
>>
>> interface FastEthernet0/1
>>   no ip address
>>   duplex auto
>>   speed auto
>> !
>> interface FastEthernet0/1.1
>>   encapsulation dot1Q 1 native
>>   ip address 192.168.181.230 255.255.255.0
>>   h323-gateway voip interface
>>   h323-gateway voip bind srcaddr 192.168.181.230
>> !
>> interface FastEthernet0/1.2
>>   encapsulation dot1Q 42
>>   ip address 192.168.42.1 255.255.255.0
>> !
>>
>> !
>> telephony-service
>>   load 7910 P00403020214
>>   load 7960-7940 P00307020200
>>   load 7911 SCCP11.8-0-4SR3AS
>>   max-ephones 42
>>   max-dn 42
>>   ip source-address 192.168.42.1 port 2000
>>   auto assign 1 to 30
>>   timeouts interdigit 5
>>   system message KESZ Srl
>>   url directories
>> http://1.2.3.4/fw/apps/Speedy/xml/directory.asp
>>   url authentication
>> http://1.2.3.4/fw/authenticate.asp
>>   time-zone 26
>>   time-format 24
>>   voicemail 899
>>   max-conferences 8 gain -6
>>   moh 14_kosheen_-_your_life.wav
>>   multicast moh 239.23.4.10 port 2000
>>   web admin system name *** password ***
>>   dn-webedit
>>   time-webedit
>>   transfer-system full-consult
>>   log password cisco
>>   secondary-dialtone 9
>>   xmltest
>>   create cnf-files version-stamp 7960 Aug 17 2007
>> 17:32:58
>> !
>>
>> thanks,
>>    Zoltan
>>
>> Quoting Ted Nugent <tednugent69 at yahoo.com>:
>>
>>     
>>> You don't happen to have a helper address on your
>>> interface FastEthernet0/1.2 interface do you?
>>> Please provide a config for your interfaces and
>>> telephone service.
>>> Ted
>>>
>>> --- Kelemen Zoltan <keli at carocomp.ro> wrote:
>>>
>>>       
>>>> Hi!
>>>>
>>>> It's a bit off-topic, however I'm all clueless,
>>>>         
>> so
>>     
>>>> any help would be
>>>> appreciated.
>>>>
>>>> I have a CCME (2811, with 12.4(13r)T), with 7911G
>>>> phones installed. Two
>>>> VLANs, one for data, one for voice.
>>>>
>>>> The router should be responsible for DHCP for the
>>>> phones, but it does
>>>> not seem to work - phones already in the network
>>>> have their IP adress,
>>>> keep their address through a reset, but new
>>>>         
>> phones
>>     
>>>> added to the network
>>>> can't get an IP.
>>>>
>>>> However, more people have access to the router
>>>>         
>> and
>>     
>>>> it is very possible
>>>> it might have been misconfigured.
>>>>
>>>> For starters, I noticed a
>>>>
>>>> no service dhcp
>>>>
>>>> in the running config, so I removed that. The
>>>>         
>> dhcp
>>     
>>>> debug messages began
>>>> to appear, but it seems like the router still
>>>>         
>> can't
>>     
>>>> find "itself":
>>>>
>>>> Aug 17 14:13:41.160: DHCPD: DHCPREQUEST received
>>>> from client
>>>> 0100.1bd4.60b5.e2.
>>>> Aug 17 14:13:41.160: DHCPD: Finding a relay for
>>>> client 0100.1bd4.60b5.e2
>>>> on interface FastEthernet0/1.2.
>>>> Aug 17 14:13:41.880: DHCPD: DHCPREQUEST received
>>>> from client
>>>> 0100.1bd4.609c.cf.
>>>> Aug 17 14:13:41.880: DHCPD: Finding a relay for
>>>> client 0100.1bd4.609c.cf
>>>> on interface FastEthernet0/1.2.
>>>>
>>>> Here are relevant parts of the configuration:
>>>>
>>>> no ip dhcp use vrf connected
>>>> ip dhcp excluded-address 192.168.42.1
>>>>         
>> 192.168.42.2
>>     
>>>> !
>>>> ip dhcp pool ITS
>>>>    network 192.168.42.0 255.255.255.0
>>>>    option 150 ip 192.168.42.1
>>>>    default-router 192.168.42.1
>>>> !
>>>> !
>>>> no ip domain lookup
>>>>
>>>> ......................
>>>>
>>>> any ideas? useful show commands?
>>>> most sh ip dhcp ... commands show nothing, except
>>>> for
>>>>
>>>> Router#sh ip dhcp server statistics
>>>> Memory usage         23512
>>>> Address pools        1
>>>> Database agents      0
>>>> Automatic bindings   0
>>>> Manual bindings      0
>>>> Expired bindings     0
>>>> Malformed messages   0
>>>> Secure arp entries   0
>>>>
>>>> Message              Received
>>>> BOOTREQUEST          0
>>>> DHCPDISCOVER         0
>>>> DHCPREQUEST          1912
>>>> DHCPDECLINE          0
>>>> DHCPRELEASE          0
>>>> DHCPINFORM           0
>>>>
>>>> Message              Sent
>>>> BOOTREPLY            0
>>>> DHCPOFFER            0
>>>> DHCPACK              0
>>>> DHCPNAK              0
>>>>
>>>> Router#sh ip dhcp pool ITS
>>>>
>>>> Pool ITS :
>>>>  Utilization mark (high/low)    : 100 / 0
>>>>  Subnet size (first/next)       : 0 / 0
>>>>  Total addresses                : 254
>>>>  Leased addresses               : 0
>>>>  Pending event                  : none
>>>>  1 subnet is currently in the pool :
>>>>  Current index        IP address range
>>>>      Leased addresses
>>>>  192.168.42.1         192.168.42.1     -
>>>> 192.168.42.254    0
>>>>
>>>> Thanks,
>>>>   Zoltan
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>>
>>>>         
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>     
>>>
>>>
>>>
>>>       
> ____________________________________________________________________________________
>   
>>> Yahoo! oneSearch: Finally, mobile search
>>> that gives answers, not web links.
>>>
>>>       
> http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC
>   
> === message truncated ===
>
>
>
>       ____________________________________________________________________________________
> Shape Yahoo! in your own image.  Join our Network Research Panel today!   http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 
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>
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From ErasmuE4 at telkom.co.za  Mon Aug 20 07:05:51 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 20 Aug 2007 13:05:51 +0200
Subject: [cisco-voip] ccm 4-2 remote branch attendant problem
In-Reply-To: <d4c697ad0708200353n6b0f4867sa7d8bd4c60993e24@mail.gmail.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01984F01@CNTRRA20-XCS01.telkom.co.za>


Thanks - will verify

 

Sending someone to site to see exactly what is happening there

 

 

 

erik

 

________________________________

From: Aman Chugh [mailto:aman.chugh at gmail.com] 
Sent: Monday, August 20, 2007 12:53 PM
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm 4-2 remote branch attendant problem

 

Eric,

 

I have seen issues with these before , this might help, check whether
the remote attendant has screensaver enabled on her computer where she
has AC installed. AC Client hangs when screensaver kicks in.

 

Thanks

 

Aman 

 

 



 

On 8/20/07, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote: 

Hi

\

I have a central call manager on a big campus and one remote branch.
The central site had serious issues with calls being stuck in the
cisco attendant console Queue. Cisco eventually gave up and told us to
rather install ARC. I am now trying to convince the customer to fork
out the bucks for it. 

 

Now one of the branches are playing up - also using the cisco
attendant console software combined with 7960 phones. I verified the
Qos on our wan MPLS links and looks fine - no drops in the Realtime
class where I put the g729 voice traffic as well as the signaling
traffic etc.  But the remote attendants complain about their consoles
freezing up. Has anyone - any good suggestions on an easy way to
monitor the attendant signaling with CCM. I looked at the Qos already
and also tested basic IP connectivity when they complain plus looked
at some of the performance counters in prefmon fro call manager
attendant and things look normal. 

 

 

 

erik 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 


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

 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
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From ErasmuE4 at telkom.co.za  Mon Aug 20 07:08:02 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 20 Aug 2007 13:08:02 +0200
Subject: [cisco-voip] Unity versus Microsoft Exchange 2007
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D886014F2C31@tnl1mx1.telindus.intra>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01984F02@CNTRRA20-XCS01.telkom.co.za>


Hi Richard\

 

I am not a unified messaging guru but if I were you I would start
looking at what is possible with Unity version 5

 

In terms of the CCM6 integration with Microsoft also look at Unified
presence server which will enable you to integrate with Microsoft Live
Communications server etc.

 

 

 

erik

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Richard van
Berkel
Sent: Monday, August 20, 2007 1:04 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity versus Microsoft Exchange 2007

 

 

Hi,

 

Does anybody know if it it possible to set up Unified Messaging  with
Unified Communication Manager 6.X and Exchange 2007?

 

If so what version have you used to do this?

 

Groeten Richard,

Mailto:Richard.vanBerkel at Telindus.nl
<mailto:Richard.vanBerkel at Telindus.nl> 

  



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From Matt.Slaga at us.didata.com  Mon Aug 20 08:23:50 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 20 Aug 2007 08:23:50 -0400
Subject: [cisco-voip] CCM 5.1 RAM feedback
In-Reply-To: <189b5daa0708181624qdee0ba0l8e2ecff43dbc7b25@mail.gmail.com>
References: <189b5daa0708160707m7dd64f5dsd9b6dcfa84028dfa@mail.gmail.com><C1FE15183DA37645BC0633BC604E44F00675845E@USNAEXCH.na.didata.local><189b5daa0708161048p6aceb8a6m1549f741b7c1e228@mail.gmail.com><C1FE15183DA37645BC0633BC604E44F006795405@USNAEXCH.na.didata.local>
	<189b5daa0708181624qdee0ba0l8e2ecff43dbc7b25@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0067D09D9@USNAEXCH.na.didata.local>

It must be 32 bit, because the minimum supported hardware platforms are
32 bit processors.  

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Keller
Sent: Saturday, August 18, 2007 7:25 PM
To: Cisco-VoIP mailing list
Subject: Re: [cisco-voip] CCM 5.1 RAM feedback

 

If CCM 5.1 OS is 64-bit, the extra RAM could be used by other
processes/apps, even if CCM 5.1 app could only use a maximum of 4GB. Can
anyone comment if the base OS of CCM 5.1 is 32-bit or 64-bit?



On 8/16/07, Jason Aarons (US) < jason.aarons at us.didata.com> wrote:

Is the hardware and RHE it's running on 64-bit (maybe seen from boot
console), would be a good question! I haven't seen a 64-bit app run on
32-bit hardware.

 

I hadn't heard of a 64-bit edition of CallManager, but I could be wrong.

 

From: Joe Keller [mailto:joe.j.keller at gmail.com] 
Sent: Thursday, August 16, 2007 1:49 PM
To: Jason Aarons (US)
Subject: Re: [cisco-voip] CCM 5.1 RAM feedback

 

So CCM 5.1 is 32-bit?

On 8/16/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:

32-bit address space is limited to 4GB addressable, do a Google search
for more info on 32-bit limitations.  This is one reason Exchange 2007
is 64-bit and requires 2003 Server 64 Bit.

 

You should be fine with 4GB, you can use RTMT to monitor hardware
resources.

 

From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Joe Keller
Sent: Thursday, August 16, 2007 10:08 AM
To: Cisco-VoIP mailing list
Subject: [cisco-voip] CCM 5.1 RAM feedback

 

Any suggestions on minimum for 2000 users on Pub & Sub? 3GB on both OK?
With RAM so cheap, I was looking at 5GB on Pub, and maybe Subs.
Any noticeable performance improvement with more RAM?

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,
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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
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message and deleting it from your computer. Thank you. 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
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error and that any use or reproduction of this email or its contents is
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message and deleting it from your computer. Thank you. 

 




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From Matt.Slaga at us.didata.com  Mon Aug 20 08:25:21 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 20 Aug 2007 08:25:21 -0400
Subject: [cisco-voip] Unity versus Microsoft Exchange 2007
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D886014F2C31@tnl1mx1.telindus.intra>
References: <472E195E692EE74BAE4E7B2880A78C1F01984EEC@CNTRRA20-XCS01.telkom.co.za><d4c697ad0708200353n6b0f4867sa7d8bd4c60993e24@mail.gmail.com>
	<B42A9A0E2A14B943A3B7EA2FA7A0D886014F2C31@tnl1mx1.telindus.intra>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0067D09E3@USNAEXCH.na.didata.local>

Yes, it is possible and works well.  However, you will need a third
party item to allow MWI.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Richard van
Berkel
Sent: Monday, August 20, 2007 7:04 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity versus Microsoft Exchange 2007

 

 

Hi,

 

Does anybody know if it it possible to set up Unified Messaging  with
Unified Communication Manager 6.X and Exchange 2007?

 

If so what version have you used to do this?

 

Groeten Richard,

Mailto:Richard.vanBerkel at Telindus.nl
<mailto:Richard.vanBerkel at Telindus.nl> 

  




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From d4rkf1ber at gmail.com  Mon Aug 20 09:32:22 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Mon, 20 Aug 2007 08:32:22 -0500
Subject: [cisco-voip] SemiOT: Roll your own point to point PRI?
Message-ID: <a1b07ea60708200632q30770d24s894638336265766e@mail.gmail.com>

I need to create two point to point PRI's between two locations to tie some
legacy phone/pbx equipment together.

Currently there are some bellsouth PRI's tying the PBX to its remote shelf
at a remote location, however the location is going to be moving and we want
to get some setup to the new location that will leverage the network and
some cisco gear to do this rather then have to depend on bellsouth.

Its not going to be managed or part of CallManager in any way.   I suspect
one could accomplish this with a couple of routers and FXO/FXS cards or
something.  But I may be way off.

Any thoughts?  Not sure what to call doing this, so I am having a difficult
time searching to find anything to read or to look for examples.

Any suggestions or links would be appreciated.
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From aman.chugh at gmail.com  Mon Aug 20 07:13:55 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Mon, 20 Aug 2007 16:43:55 +0530
Subject: [cisco-voip] ccm 4-2 remote branch attendant problem
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F01984F01@CNTRRA20-XCS01.telkom.co.za>
References: <d4c697ad0708200353n6b0f4867sa7d8bd4c60993e24@mail.gmail.com>
	<472E195E692EE74BAE4E7B2880A78C1F01984F01@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <d4c697ad0708200413r1bf8736bxa81504e7410311d2@mail.gmail.com>

If going on site, try to capture the file acclient.txt from attendant
console pc to check for errors.

Aman


On 8/20/07, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote:
>
>  Thanks ? will verify
>
>
>
> Sending someone to site to see exactly what is happening there
>
>
>
>
>
>
>
> erik
>
>
>  ------------------------------
>
> *From:* Aman Chugh [mailto:aman.chugh at gmail.com]
> *Sent:* Monday, August 20, 2007 12:53 PM
> *To:* Erik Erasmus (E)
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] ccm 4-2 remote branch attendant problem
>
>
>
> Eric,
>
>
>
> I have seen issues with these before , this might help, check whether the
> remote attendant has screensaver enabled on her computer where she has AC
> installed. AC Client hangs when screensaver kicks in.
>
>
>
> Thanks
>
>
>
> Aman
>
>
>
>
>
>
>
>
>
> On 8/20/07, *Erik Erasmus** (E)* <ErasmuE4 at telkom.co.za> wrote:
>
> Hi
>
> \
>
> I have a central call manager on a big campus and one remote branch. The
> central site had serious issues with calls being stuck in the cisco
> attendant console Queue. Cisco eventually gave up and told us to rather
> install ARC. I am now trying to convince the customer to fork out the bucks
> for it.
>
>
>
> Now one of the branches are playing up ? also using the cisco attendant
> console software combined with 7960 phones. I verified the Qos on our wan
> MPLS links and looks fine ? no drops in the Realtime class where I put the
> g729 voice traffic as well as the signaling traffic etc.  But the remote
> attendants complain about their consoles freezing up. Has anyone ? any good
> suggestions on an easy way to monitor the attendant signaling with CCM. I
> looked at the Qos already and also tested basic IP connectivity when they
> complain plus looked at some of the performance counters in prefmon fro call
> manager attendant and things look normal.
>
>
>
>
>
>
>
> erik
>
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> This e-mail and its contents are subject to the Telkom SA Limited
> e-mail legal notice available at
> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>
>
> _______________________________________________
> 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  Mon Aug 20 11:10:47 2007
From: jgrace at adcapnet.com (James Grace)
Date: Mon, 20 Aug 2007 11:10:47 -0400
Subject: [cisco-voip] Issues with administering cm 6.0 with explore 7
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C73B@mail.adcapnetsvc.local>

?
I having Issues with administering cm 6.0 with explore 7.  I can get to the admin page but there is a delay with every page  go into there after.  This is only happening the CM 6.  I can surf all other internet pages just fine.
 
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 20 Aug 2007 11:10:00 -0400
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From lelio at uoguelph.ca  Mon Aug 20 13:02:48 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 20 Aug 2007 13:02:48 -0400
Subject: [cisco-voip] location bw setting/resync
Message-ID: <015901c7e34b$efc5d2c0$56196883@cfs.uoguelph.ca>

We are looking at getting a backup link for our voice calls at a remote site. This will help during link flaps and complete outage (wireless + storm = bad).

I'd like to configure the phones at the remote site with a CallManager location with unlimited bandwith to start and then modify the bandwidth to something lower when we have to. I tried this and it didn't look like any phones had to be restarted when I made the change or when I resync'ed the bandwidth.

Thoughts?

--------------------------------------------------------------------------------
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 erickbee at gmail.com  Mon Aug 20 13:03:50 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Mon, 20 Aug 2007 12:03:50 -0500
Subject: [cisco-voip] Disabling low bandwidth in IP Communicator
Message-ID: <f4445faf0708201003u147d7195w8ba59462586bb258@mail.gmail.com>

Is there a way to disable the optimize ofr low bandwidth option so no
one can change it?
Registry setting, something else?

Thanks, Erick

From jason.aarons at us.didata.com  Mon Aug 20 09:03:15 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 20 Aug 2007 09:03:15 -0400
Subject: [cisco-voip] CCMUser CallForwardAll Line2
Message-ID: <C1FE15183DA37645BC0633BC604E44F0067D0ACD@USNAEXCH.na.didata.local>

After I use CCMUser to CallForwardAll on Line2, should the 7961 show a
animated CallForwardAll icon flashing with a arrow?  It's not, and I
want to say in the past it did show a animated icon.

 

CallManager 4.1(3)SR4D

Phone Load 8.2(2)SR2




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From cpoper at hamilton.ca  Mon Aug 20 12:51:33 2007
From: cpoper at hamilton.ca (Poper, Cathy)
Date: Mon, 20 Aug 2007 12:51:33 -0400
Subject: [cisco-voip] CER 1.2(3a)
Message-ID: <46262932FF67F74CA6D537C31A696987198CAB5F@endeavour.InfoNetwork>

I've been attempting to enter an Onsite ERL profile so that the
notification rings to an external telephone number (like a cell-phone).
I'm getting an error when CER attempts to call the number. Is there a
restriction in CER that only allows to an internal extension??

If anyone has successfully enabled the alert to an outside telephone
number please let me know...we're running CER 1.2(3a)...with CCM 4.1(3)+
Thanks

Cathy Poper
Network Support Analyst
City of Hamilton-Information Technology Services
55 York Blvd, 6th flr
Hamilton. Ont. Canada
L8L 3K7
Tel: 905-546-2424 x5481
Fax: 905-546-2333
Email: cpoper at hamilton.ca
~~~~~~~~~~~~~~~~~~~~

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From wsisk at cisco.com  Mon Aug 20 14:17:04 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 20 Aug 2007 14:17:04 -0400
Subject: [cisco-voip] location bw setting/resync
In-Reply-To: <015901c7e34b$efc5d2c0$56196883@cfs.uoguelph.ca>
References: <015901c7e34b$efc5d2c0$56196883@cfs.uoguelph.ca>
Message-ID: <46C9DAA0.7030506@cisco.com>

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From lelio at uoguelph.ca  Mon Aug 20 14:50:13 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 20 Aug 2007 14:50:13 -0400
Subject: [cisco-voip] location bw setting/resync
References: <015901c7e34b$efc5d2c0$56196883@cfs.uoguelph.ca>
	<46C9DAA0.7030506@cisco.com>
Message-ID: <020f01c7e35a$f1178240$56196883@cfs.uoguelph.ca>

Thanks Wes.

Will a gateway configured for specific location that has exceeded it's allotment trigger failover to the next route group? For example, route list A has route group X with 2 gateways in location X and route group Y with 2 gateways in the default location. If I place a cal from the default location to a route pattern with route list A and the b/w allocation to location X is exceeded,  will it go to route group Y?


  ----- Original Message ----- 
  From: Wes Sisk 
  To: Lelio Fulgenzi 
  Cc: Cisco VoIPoE List 
  Sent: Monday, August 20, 2007 2:17 PM
  Subject: Re: [cisco-voip] location bw setting/resync


  it does not reset/restart phones.  it only resets bw counters internal to the cm process. /wes

  Lelio Fulgenzi wrote: 
    We are looking at getting a backup link for our voice calls at a remote site. This will help during link flaps and complete outage (wireless + storm = bad).

    I'd like to configure the phones at the remote site with a CallManager location with unlimited bandwith to start and then modify the bandwidth to something lower when we have to. I tried this and it didn't look like any phones had to be restarted when I made the change or when I resync'ed the bandwidth.

    Thoughts?

    --------------------------------------------------------------------------------
    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
----------------------------------------------------------------------------
_______________________________________________
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  Mon Aug 20 15:48:29 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 20 Aug 2007 15:48:29 -0400
Subject: [cisco-voip] CER 1.2(3a)
In-Reply-To: <46262932FF67F74CA6D537C31A696987198CAB5F@endeavour.InfoNetwork>
References: <46262932FF67F74CA6D537C31A696987198CAB5F@endeavour.InfoNetwork>
Message-ID: <94a1afde0708201248v59e71b51oda17037f1a009cf3@mail.gmail.com>

Hi Cathy,

I have not set this up for external numbers, but I don't see why it wouldn't
work. Couple things you might look at:

- Make sure it will work for internal numbers first
- Check the CTI ports you configured to call out from, make sure they have a
calling search space that can dial the external number.
- Make sure your contact number in CER is entered as you would dial it

What error are you getting?

Ed

On 8/20/07, Poper, Cathy <cpoper at hamilton.ca> wrote:
>
>  I've been attempting to enter an Onsite ERL profile so that the
> notification rings to an external telephone number (like a cell-phone). I'm
> getting an error when CER attempts to call the number. Is there a
> restriction in CER that only allows to an internal extension??
>
> If anyone has successfully enabled the alert to an outside telephone
> number please let me know?we're running CER 1.2(3a)?with CCM 4.1(3)+
>
> Thanks
>
> *Cathy Poper*
> *Network Support Analyst*
> *City of Hamilton-Information Technology Services*
> *55 York Blvd, 6th flr*
> *Hamilton. Ont. Canada*
> *L8L 3K7*
> *Tel: 905-546-2424 x5481*
> *Fax: 905-546-2333*
> *Email: cpoper at hamilton.ca*
> *~~~~~~~~~~~~~~~~~~~~*
>
> _______________________________________________
> 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 wsisk at cisco.com  Mon Aug 20 17:05:20 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 20 Aug 2007 17:05:20 -0400
Subject: [cisco-voip] location bw setting/resync
In-Reply-To: <020f01c7e35a$f1178240$56196883@cfs.uoguelph.ca>
References: <015901c7e34b$efc5d2c0$56196883@cfs.uoguelph.ca>
	<46C9DAA0.7030506@cisco.com>
	<020f01c7e35a$f1178240$56196883@cfs.uoguelph.ca>
Message-ID: <46CA0210.2010605@cisco.com>

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From lelio at uoguelph.ca  Mon Aug 20 18:00:26 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 20 Aug 2007 18:00:26 -0400
Subject: [cisco-voip] location bw setting/resync
References: <015901c7e34b$efc5d2c0$56196883@cfs.uoguelph.ca>
	<46C9DAA0.7030506@cisco.com>
	<020f01c7e35a$f1178240$56196883@cfs.uoguelph.ca>
	<46CA0210.2010605@cisco.com>
Message-ID: <001301c7e375$847fdf40$60056883@pdp11>

Thanks Wes. We have 4.1(3)sr4d so I believe we have the fix. When I look in the service parameters, we have "Stop Routing on Out of Bandwidth Flag" but it mentions intercluster trunks, specifically: "This parameter determines routing behavior for calls through intercluster trunks when not enough bandwidth exists."

What do you think? Documentation error? I'm also guessing as long as we don't use Unity failover, we don't really have to worry about this.



  ----- Original Message ----- 
  From: Wes Sisk 
  To: Lelio Fulgenzi 
  Cc: Cisco VoIPoE List 
  Sent: Monday, August 20, 2007 5:05 PM
  Subject: Re: [cisco-voip] location bw setting/resync


  Engineer's favorite answer - it Depends.

  CM service parameter
  "Stop Hunting on Out of Bandwidth Flag"

  If you have a version with the fix for CSCsc62073 you will have 2 params, one for hunt lists and one for route lists.

  /Wes

  Lelio Fulgenzi wrote: 
    Thanks Wes.

    Will a gateway configured for specific location that has exceeded it's allotment trigger failover to the next route group? For example, route list A has route group X with 2 gateways in location X and route group Y with 2 gateways in the default location. If I place a cal from the default location to a route pattern with route list A and the b/w allocation to location X is exceeded,  will it go to route group Y?


      ----- Original Message ----- 
      From: Wes Sisk 
      To: Lelio Fulgenzi 
      Cc: Cisco VoIPoE List 
      Sent: Monday, August 20, 2007 2:17 PM
      Subject: Re: [cisco-voip] location bw setting/resync


      it does not reset/restart phones.  it only resets bw counters internal to the cm process. /wes

      Lelio Fulgenzi wrote: 
        We are looking at getting a backup link for our voice calls at a remote site. This will help during link flaps and complete outage (wireless + storm = bad).

        I'd like to configure the phones at the remote site with a CallManager location with unlimited bandwith to start and then modify the bandwidth to something lower when we have to. I tried this and it didn't look like any phones had to be restarted when I made the change or when I resync'ed the bandwidth.

        Thoughts?

        --------------------------------------------------------------------------------
        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
------------------------------------------------------------------------
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From tettokalle at hotmail.com  Mon Aug 20 18:07:00 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Tue, 21 Aug 2007 00:07:00 +0200
Subject: [cisco-voip] CME vs SRST
In-Reply-To: <584577.22199.qm@web36114.mail.mud.yahoo.com>
References: <OF2EAE33DE.D853A05F-ON8525733A.00791B1E-8525733A.007B1D57@bftg.com>
	<584577.22199.qm@web36114.mail.mud.yahoo.com>
Message-ID: <fad3a2$81c$1@sea.gmane.org>

this is a feature since cme 4.0. and we are using this quite often.

the only problem is to have both configs synchronized, but the good 
thing is cme does not cost more than srst.

Ted Nugent schrieb:
> Yep, its pretty new
> 
> http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00807e0566.html
> 
> 
> --- CarlosOrtiz at bayviewfinancial.com wrote:
> 
>> Can CME be used as a fallback from Call Mgr instead
>> of SRST? I have never 
>> thought so......but a previous admin has a CME
>> config on a router at a 
>> location that has the following excerpt:
>>
>> ephone-hunt 1 sequential
>>  pilot 4700
>>  list 4739, 4740, 4743, 4744
>>  final 9XXXXXXXXXX
>>  preference 3 secondary 7
>>  timeout 16, 16, 16, 16
>> !
>> ephone-hunt 2 sequential
>>  pilot 4799
>>  list 4702, 4701
>>  final 9XXXXXXXXXX
>>  timeout 16, 0
>>
>> The purpose of this config as explained by the
>> business was when in "SRST" 
>> (I know this was CME config)  to route calls to a
>> specific DID, hunt 
>> through lines and as a last resort dial outbound to
>> a telco analog line 
>> with a mailbox to leave a message (important).
>>
>> We did SRST testing for all of our locations
>> locations about 2 months ago 
>> and this particular location did not function at all
>> with this "CME" 
>> config.  I have since converted them to SRST, had an
>> outage and their 
>> calls did work.
>>
>> The business is claiming that this worked at one
>> time during SRST but I am 
>> skeptical.   I am thinking that the previous admin
>> told him it was 
>> configured but there was no real testing done.  Can
>> anyone shed some light 
>> on this.  I need to know:
>>
>> 1) Can CME be used instead of SRST?
>> 2) Can I accomplish what they want via SRST? (see
>> red above - I think I 
>> can with the alias command but need to check)
>>
>>
>>   Below is the full CME config if needed.
>>
>>
>>  telephony-service
>>  srst mode auto-provision all
>>  srst dn line-mode dual
>>  max-ephones 100
>>  max-dn 200
>>  ip source-address 10.X.X.X port 2000
>>  max-conferences 8 gain -6
>>  web admin system name X password X
>>  dn-webedit 
>>  time-webedit 
>>  transfer-system full-consult
>> !
>> ephone-dn  1  dual-line
>>  number 4744
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  2  dual-line
>>  number 4743
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  3  dual-line
>>  number 4702
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  4  dual-line
>>  number 4745
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  5  dual-line
>>  number 4739
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  6  dual-line
>>  number 4742
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  7  dual-line
>>  number 4740
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  8  dual-line
>>  number 5539
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  9  dual-line
>>  number 4701
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  10  dual-line
>>  number 4710
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  11  dual-line
>>  number 1010
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  12  dual-line
>>  number 4703
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  13  dual-line
>>  number 1012
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  14  dual-line
>>  number 4704
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  15  dual-line
>>  number 4636
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  16  dual-line
>>  number 4705
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  17  dual-line
>>  number 4706
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  18  dual-line
>>  number 4707
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone-dn  19  dual-line
>>  number 7045
>>  description XXXXXXXXXX
>>  name XXXXXXXXXX
>> !
>> ephone  1
>>  mac-address 0016.4766.AB7C
>>  button  1:1
>> !
>> ephone  2
>>  mac-address 0015.F907.6BE1
>> !
>> ephone  3
>>  mac-address 000E.833C.AE08
>>  button  1:3
>> !
>> ephone  4
>>  mac-address 0011.5C0E.6108
>>  button  1:4
>> !
>> ephone  5
>>  mac-address 0015.F914.1515
>> !
>> ephone  6
>>  mac-address 0015.F914.0E19
>>  button  1:6
>> !
>> ephone  7
>>  mac-address 0015.F914.11FF
>>  button  1:7 2:15
>> !
>> ephone  8
>>  mac-address 0015.F914.0E4C
>>  button  1:9
>> !
>> ephone  9
>>  mac-address 00E0.7524.B312
>>  button  1:10
>> !
>> ephone  11
>>  mac-address 0015.FA0B.7202
>>  button  1:12
>> !
>> ephone  12
>>  mac-address 0017.9486.B302
>>  button  1:14
>> !
>>
> === message truncated ===>
> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
>        
> ____________________________________________________________________________________
> Sick sense of humor? Visit Yahoo! TV's 
> Comedy with an Edge to see what's on, when. 
> http://tv.yahoo.com/collections/222


From wsisk at cisco.com  Mon Aug 20 18:20:00 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 20 Aug 2007 18:20:00 -0400
Subject: [cisco-voip] location bw setting/resync
In-Reply-To: <001301c7e375$847fdf40$60056883@pdp11>
References: <015901c7e34b$efc5d2c0$56196883@cfs.uoguelph.ca>
	<46C9DAA0.7030506@cisco.com>
	<020f01c7e35a$f1178240$56196883@cfs.uoguelph.ca>
	<46CA0210.2010605@cisco.com> <001301c7e375$847fdf40$60056883@pdp11>
Message-ID: <46CA1390.7060100@cisco.com>

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From techguy at gmail.com  Mon Aug 20 18:25:33 2007
From: techguy at gmail.com (TechGuy)
Date: Mon, 20 Aug 2007 17:25:33 -0500
Subject: [cisco-voip] SemiOT: Roll your own point to point PRI?
In-Reply-To: <a1b07ea60708200632q30770d24s894638336265766e@mail.gmail.com>
References: <a1b07ea60708200632q30770d24s894638336265766e@mail.gmail.com>
Message-ID: <a1b07ea60708201525h5677a738h99076bee876fa37e@mail.gmail.com>

>
> I need to create two point to point PRI's between two locations to tie
> some legacy phone/pbx equipment together.
>
> Currently there are some bellsouth PRI's tying the PBX to its remote shelf
> at a remote location, however the location is going to be moving and we want
> to get something setup at the new location that will leverage the network
> and some cisco gear to do this rather then have to depend on bellsouth.
>
> Its not going to be managed or part of CallManager in any way.   I suspect
> one could accomplish this with a couple of routers and FXO/FXS cards or
> something.  But I may be way off.
>
> Any thoughts?  Not sure what to call doing this, so I am having a
> difficult time searching to find anything to read or to look for examples.
>
> Any suggestions or links would be appreciated.
>
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From lelio at uoguelph.ca  Mon Aug 20 18:13:27 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 20 Aug 2007 18:13:27 -0400
Subject: [cisco-voip] CME vs SRST
References: <OF2EAE33DE.D853A05F-ON8525733A.00791B1E-8525733A.007B1D57@bftg.com><584577.22199.qm@web36114.mail.mud.yahoo.com>
	<fad3a2$81c$1@sea.gmane.org>
Message-ID: <005101c7e377$5570aed0$60056883@pdp11>

don't forget the limitations in endpoints of CME vs SRST. SRST can scale 
quite well.

also, I haven't read much about it b/c of the scaling issue, but it sounded 
like you needed a bit more configuration as well. the data sheet said 
something about 'one time configuration' as a benefit for SRST.

----- Original Message ----- 
From: "Kalle Tetto" <tettokalle at hotmail.com>
To: <cisco-voip at puck.nether.net>
Sent: Monday, August 20, 2007 6:07 PM
Subject: Re: [cisco-voip] CME vs SRST


> this is a feature since cme 4.0. and we are using this quite often.
>
> the only problem is to have both configs synchronized, but the good
> thing is cme does not cost more than srst.
>
> Ted Nugent schrieb:
>> Yep, its pretty new
>>
>> http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00807e0566.html
>>
>>
>> --- CarlosOrtiz at bayviewfinancial.com wrote:
>>
>>> Can CME be used as a fallback from Call Mgr instead
>>> of SRST? I have never
>>> thought so......but a previous admin has a CME
>>> config on a router at a
>>> location that has the following excerpt:
>>>
>>> ephone-hunt 1 sequential
>>>  pilot 4700
>>>  list 4739, 4740, 4743, 4744
>>>  final 9XXXXXXXXXX
>>>  preference 3 secondary 7
>>>  timeout 16, 16, 16, 16
>>> !
>>> ephone-hunt 2 sequential
>>>  pilot 4799
>>>  list 4702, 4701
>>>  final 9XXXXXXXXXX
>>>  timeout 16, 0
>>>
>>> The purpose of this config as explained by the
>>> business was when in "SRST"
>>> (I know this was CME config)  to route calls to a
>>> specific DID, hunt
>>> through lines and as a last resort dial outbound to
>>> a telco analog line
>>> with a mailbox to leave a message (important).
>>>
>>> We did SRST testing for all of our locations
>>> locations about 2 months ago
>>> and this particular location did not function at all
>>> with this "CME"
>>> config.  I have since converted them to SRST, had an
>>> outage and their
>>> calls did work.
>>>
>>> The business is claiming that this worked at one
>>> time during SRST but I am
>>> skeptical.   I am thinking that the previous admin
>>> told him it was
>>> configured but there was no real testing done.  Can
>>> anyone shed some light
>>> on this.  I need to know:
>>>
>>> 1) Can CME be used instead of SRST?
>>> 2) Can I accomplish what they want via SRST? (see
>>> red above - I think I
>>> can with the alias command but need to check)
>>>
>>>
>>>   Below is the full CME config if needed.
>>>
>>>
>>>  telephony-service
>>>  srst mode auto-provision all
>>>  srst dn line-mode dual
>>>  max-ephones 100
>>>  max-dn 200
>>>  ip source-address 10.X.X.X port 2000
>>>  max-conferences 8 gain -6
>>>  web admin system name X password X
>>>  dn-webedit
>>>  time-webedit
>>>  transfer-system full-consult
>>> !
>>> ephone-dn  1  dual-line
>>>  number 4744
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  2  dual-line
>>>  number 4743
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  3  dual-line
>>>  number 4702
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  4  dual-line
>>>  number 4745
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  5  dual-line
>>>  number 4739
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  6  dual-line
>>>  number 4742
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  7  dual-line
>>>  number 4740
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  8  dual-line
>>>  number 5539
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  9  dual-line
>>>  number 4701
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  10  dual-line
>>>  number 4710
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  11  dual-line
>>>  number 1010
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  12  dual-line
>>>  number 4703
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  13  dual-line
>>>  number 1012
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  14  dual-line
>>>  number 4704
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  15  dual-line
>>>  number 4636
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  16  dual-line
>>>  number 4705
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  17  dual-line
>>>  number 4706
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  18  dual-line
>>>  number 4707
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  19  dual-line
>>>  number 7045
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone  1
>>>  mac-address 0016.4766.AB7C
>>>  button  1:1
>>> !
>>> ephone  2
>>>  mac-address 0015.F907.6BE1
>>> !
>>> ephone  3
>>>  mac-address 000E.833C.AE08
>>>  button  1:3
>>> !
>>> ephone  4
>>>  mac-address 0011.5C0E.6108
>>>  button  1:4
>>> !
>>> ephone  5
>>>  mac-address 0015.F914.1515
>>> !
>>> ephone  6
>>>  mac-address 0015.F914.0E19
>>>  button  1:6
>>> !
>>> ephone  7
>>>  mac-address 0015.F914.11FF
>>>  button  1:7 2:15
>>> !
>>> ephone  8
>>>  mac-address 0015.F914.0E4C
>>>  button  1:9
>>> !
>>> ephone  9
>>>  mac-address 00E0.7524.B312
>>>  button  1:10
>>> !
>>> ephone  11
>>>  mac-address 0015.FA0B.7202
>>>  button  1:12
>>> !
>>> ephone  12
>>>  mac-address 0017.9486.B302
>>>  button  1:14
>>> !
>>>
>> === message truncated ===>
>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>>
>> ____________________________________________________________________________________
>> Sick sense of humor? Visit Yahoo! TV's
>> Comedy with an Edge to see what's on, when.
>> http://tv.yahoo.com/collections/222
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From Timothy_Frazee at adp.com  Mon Aug 20 20:41:47 2007
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Mon, 20 Aug 2007 19:41:47 -0500
Subject: [cisco-voip] softkey template
In-Reply-To: <005101c7e377$5570aed0$60056883@pdp11>
References: <OF2EAE33DE.D853A05F-ON8525733A.00791B1E-8525733A.007B1D57@bftg.com><584577.22199.qm@web36114.mail.mud.yahoo.com><fad3a2$81c$1@sea.gmane.org>
	<005101c7e377$5570aed0$60056883@pdp11>
Message-ID: <723FA17C6AA4C4428CDF572DCAEE08D404E18FD0@EXCHANGEDS01.ds.ad.adp.com>

If I set a softkey template at the device pool level to "template1", and
at the device it is set to "<none>", what is the resulting template?

template1 or <none> ???

Does anyone know how to batch change phones from a setting of "<none>"
to "template1"? Just in case?

I don't see a softkey template setting for update phones.

Thanks.


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 jsteinberg at gmail.com  Mon Aug 20 21:12:48 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Mon, 20 Aug 2007 21:12:48 -0400
Subject: [cisco-voip] softkey template
In-Reply-To: <723FA17C6AA4C4428CDF572DCAEE08D404E18FD0@EXCHANGEDS01.ds.ad.adp.com>
References: <OF2EAE33DE.D853A05F-ON8525733A.00791B1E-8525733A.007B1D57@bftg.com>
	<584577.22199.qm@web36114.mail.mud.yahoo.com>
	<fad3a2$81c$1@sea.gmane.org> <005101c7e377$5570aed0$60056883@pdp11>
	<723FA17C6AA4C4428CDF572DCAEE08D404E18FD0@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <f1b65f880708201812l1d494dd7hea9c01e6e99d4668@mail.gmail.com>

template1



On 8/20/07, Frazee, Timothy <Timothy_Frazee at adp.com> wrote:
>
> If I set a softkey template at the device pool level to "template1", and
> at the device it is set to "<none>", what is the resulting template?
>
> template1 or <none> ???
>
> Does anyone know how to batch change phones from a setting of "<none>"
> to "template1"? Just in case?
>
> I don't see a softkey template setting for update phones.
>
> Thanks.
>
>
> 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 erickbee at gmail.com  Mon Aug 20 22:03:38 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Mon, 20 Aug 2007 21:03:38 -0500
Subject: [cisco-voip] CCM 5.11 MOH error -- Unmapped exception
Message-ID: <f4445faf0708201903n56762710mae78821445ad05cc@mail.gmail.com>

When trying to add a MOH file, the find and list comes up with this.

They had a MP3 file they converted to a WAV file then this happened
when they tried to upload it. The new file shows under MOH audio
sources but saids the file is not there and lists no properties for
the audio source.

Unmapped exception. The entity name must immediately follow the '&' in
the entity reference.

0 records found, 133 minutes available.

From tednugent69 at yahoo.com  Mon Aug 20 21:16:48 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Mon, 20 Aug 2007 18:16:48 -0700 (PDT)
Subject: [cisco-voip] softkey template
In-Reply-To: <723FA17C6AA4C4428CDF572DCAEE08D404E18FD0@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <130289.61555.qm@web36110.mail.mud.yahoo.com>

The phone will use whatever is in the devicepool
unless its overriden by the phone device level
configuration. You can use BAT or CM5 bulk phone
update to query and bulk update multiple devices.


--- "Frazee, Timothy" <Timothy_Frazee at adp.com> wrote:

> If I set a softkey template at the device pool level
> to "template1", and
> at the device it is set to "<none>", what is the
> resulting template?
> 
> template1 or <none> ???
> 
> Does anyone know how to batch change phones from a
> setting of "<none>"
> to "template1"? Just in case?
> 
> I don't see a softkey template setting for update
> phones.
> 
> Thanks.
> 
> 
> 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 gokhanciscottl at yahoo.com  Tue Aug 21 02:10:23 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Mon, 20 Aug 2007 23:10:23 -0700 (PDT)
Subject: [cisco-voip] how to see calling number and called number on mgcp
	gateway
Message-ID: <660093.42967.qm@web39603.mail.mud.yahoo.com>

hi

on mgcp gateway  I wanna see calling number and called number on FXO ports and ISDN port
i try  debug voice ccapi inout and debug voip vtsp all but they show UNKNOWN number
how can i see calling and called numbers on mgcp gateway

thanks


       
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From Johnny.Crothers at vanco.co.uk  Tue Aug 21 05:02:25 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Tue, 21 Aug 2007 10:02:25 +0100
Subject: [cisco-voip] how to see calling number and called number on
 mgcpgateway
In-Reply-To: <660093.42967.qm@web39603.mail.mud.yahoo.com>
Message-ID: <A56BD242744BF540B15D1672E0E3E87501A96E12@LON-INT-MSGVS01.internal.vancocorp.com>

Try "debug isdn q931"

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of gokhan senol
Sent: 21 August 2007 07:10
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to see calling number and called number on
mgcpgateway

 

hi

 

on mgcp gateway  I wanna see calling number and called number on FXO
ports and ISDN port

i try  debug voice ccapi inout and debug voip vtsp all but they show
UNKNOWN number

how can i see calling and called numbers on mgcp gateway

 

thanks

 

________________________________

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From gokhanciscottl at yahoo.com  Tue Aug 21 04:38:41 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Tue, 21 Aug 2007 01:38:41 -0700 (PDT)
Subject: [cisco-voip] path code violation error
Message-ID: <588232.58199.qm@web39609.mail.mud.yahoo.com>

Hi
I have trouble on my pri line. The line  comes from SDH but the distance between the line termination point and the router is 200 meters and  i put 2 alcatel modems between them.
As you can see on the output i am getting PATH CODE VIOLATION error. 
I talked telecom an they sai everything is ok on their config. do you have any idea about the error below.
This error  often causes stopping speeching on the line

thanks  a lot 


network-clock-participate wic 2 
network-clock-select 1 E1 0/2/0

controller E1 0/2/0
 pri-group timeslots 1-31 service mgcp

E1 0/2/0 is up.
  Applique type is Channelized E1 - balanced
  No alarms detected.
  alarm-trigger is not set
  Version info Firmware: 20060711, FPGA: 13, spm_count = 0
  Framing is CRC4, Line Code is HDB3, Clock Source is Line.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (691 seconds elapsed):
     0 Line Code Violations, 173 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
  Total Data (last 1 15 minute intervals):
     0 Line Code Violations, 234 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


       
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From pwalenta at wi.rr.com  Tue Aug 21 08:54:44 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 21 Aug 2007 07:54:44 -0500
Subject: [cisco-voip] path code violation error
In-Reply-To: <588232.58199.qm@web39609.mail.mud.yahoo.com>
References: <588232.58199.qm@web39609.mail.mud.yahoo.com>
Message-ID: <005c01c7e3f2$72c51d60$0232a8c0@walentawk1>

You might want to look at the "cablelength" command under your E1
controller.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of gokhan senol
Sent: Tuesday, August 21, 2007 3:39 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] path code violation error



Hi

I have trouble on my pri line. The line  comes from SDH but the distance
between the line termination point and the router is 200 meters and  i put 2
alcatel modems between them.

As you can see on the output i am getting PATH CODE VIOLATION error. 
I talked telecom an they sai everything is ok on their config. do you have
any idea about the error below.

This error  often causes stopping speeching on the line

 

thanks  a lot 

 

 

network-clock-participate wic 2 
network-clock-select 1 E1 0/2/0

 

controller E1 0/2/0
 pri-group timeslots 1-31 service mgcp

 

E1 0/2/0 is up.
  Applique type is Channelized E1 - balanced
  No alarms detected.
  alarm-trigger is not set
  Version info Firmware: 20060711, FPGA: 13, spm_count = 0
  Framing is CRC4, Line Code is HDB3, Clock Source is Line.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (691 seconds elapsed):
     0 Line Code Violations, 173 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
  Total Data (last 1 15 minute intervals):
     0 Line Code Violations, 234 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


  _____  

Be a better Globetrotter. Get
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From CarlosOrtiz at bayviewfinancial.com  Tue Aug 21 09:13:30 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 21 Aug 2007 09:13:30 -0400
Subject: [cisco-voip] CME vs SRST
In-Reply-To: <005101c7e377$5570aed0$60056883@pdp11>
Message-ID: <OFE1259B08.BD971730-ON8525733E.00487306-8525733E.0048A5E3@bftg.com>

Lelio -  But it could serve its purpose for smaller offices that need a 
specific functionality.

Kate - What do you mean keeping the configs synchronized?

Carlos



"Lelio Fulgenzi" <lelio at uoguelph.ca> 
Sent by: cisco-voip-bounces at puck.nether.net
08/20/2007 06:13 PM

To
<cisco-voip at puck.nether.net>, "Kalle Tetto" <tettokalle at hotmail.com>
cc

Subject
Re: [cisco-voip] CME vs SRST






don't forget the limitations in endpoints of CME vs SRST. SRST can scale 
quite well.

also, I haven't read much about it b/c of the scaling issue, but it 
sounded 
like you needed a bit more configuration as well. the data sheet said 
something about 'one time configuration' as a benefit for SRST.

----- Original Message ----- 
From: "Kalle Tetto" <tettokalle at hotmail.com>
To: <cisco-voip at puck.nether.net>
Sent: Monday, August 20, 2007 6:07 PM
Subject: Re: [cisco-voip] CME vs SRST


> this is a feature since cme 4.0. and we are using this quite often.
>
> the only problem is to have both configs synchronized, but the good
> thing is cme does not cost more than srst.
>
> Ted Nugent schrieb:
>> Yep, its pretty new
>>
>> 
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00807e0566.html

>>
>>
>> --- CarlosOrtiz at bayviewfinancial.com wrote:
>>
>>> Can CME be used as a fallback from Call Mgr instead
>>> of SRST? I have never
>>> thought so......but a previous admin has a CME
>>> config on a router at a
>>> location that has the following excerpt:
>>>
>>> ephone-hunt 1 sequential
>>>  pilot 4700
>>>  list 4739, 4740, 4743, 4744
>>>  final 9XXXXXXXXXX
>>>  preference 3 secondary 7
>>>  timeout 16, 16, 16, 16
>>> !
>>> ephone-hunt 2 sequential
>>>  pilot 4799
>>>  list 4702, 4701
>>>  final 9XXXXXXXXXX
>>>  timeout 16, 0
>>>
>>> The purpose of this config as explained by the
>>> business was when in "SRST"
>>> (I know this was CME config)  to route calls to a
>>> specific DID, hunt
>>> through lines and as a last resort dial outbound to
>>> a telco analog line
>>> with a mailbox to leave a message (important).
>>>
>>> We did SRST testing for all of our locations
>>> locations about 2 months ago
>>> and this particular location did not function at all
>>> with this "CME"
>>> config.  I have since converted them to SRST, had an
>>> outage and their
>>> calls did work.
>>>
>>> The business is claiming that this worked at one
>>> time during SRST but I am
>>> skeptical.   I am thinking that the previous admin
>>> told him it was
>>> configured but there was no real testing done.  Can
>>> anyone shed some light
>>> on this.  I need to know:
>>>
>>> 1) Can CME be used instead of SRST?
>>> 2) Can I accomplish what they want via SRST? (see
>>> red above - I think I
>>> can with the alias command but need to check)
>>>
>>>
>>>   Below is the full CME config if needed.
>>>
>>>
>>>  telephony-service
>>>  srst mode auto-provision all
>>>  srst dn line-mode dual
>>>  max-ephones 100
>>>  max-dn 200
>>>  ip source-address 10.X.X.X port 2000
>>>  max-conferences 8 gain -6
>>>  web admin system name X password X
>>>  dn-webedit
>>>  time-webedit
>>>  transfer-system full-consult
>>> !
>>> ephone-dn  1  dual-line
>>>  number 4744
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  2  dual-line
>>>  number 4743
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  3  dual-line
>>>  number 4702
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  4  dual-line
>>>  number 4745
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  5  dual-line
>>>  number 4739
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  6  dual-line
>>>  number 4742
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  7  dual-line
>>>  number 4740
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  8  dual-line
>>>  number 5539
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  9  dual-line
>>>  number 4701
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  10  dual-line
>>>  number 4710
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  11  dual-line
>>>  number 1010
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  12  dual-line
>>>  number 4703
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  13  dual-line
>>>  number 1012
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  14  dual-line
>>>  number 4704
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  15  dual-line
>>>  number 4636
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  16  dual-line
>>>  number 4705
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  17  dual-line
>>>  number 4706
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  18  dual-line
>>>  number 4707
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone-dn  19  dual-line
>>>  number 7045
>>>  description XXXXXXXXXX
>>>  name XXXXXXXXXX
>>> !
>>> ephone  1
>>>  mac-address 0016.4766.AB7C
>>>  button  1:1
>>> !
>>> ephone  2
>>>  mac-address 0015.F907.6BE1
>>> !
>>> ephone  3
>>>  mac-address 000E.833C.AE08
>>>  button  1:3
>>> !
>>> ephone  4
>>>  mac-address 0011.5C0E.6108
>>>  button  1:4
>>> !
>>> ephone  5
>>>  mac-address 0015.F914.1515
>>> !
>>> ephone  6
>>>  mac-address 0015.F914.0E19
>>>  button  1:6
>>> !
>>> ephone  7
>>>  mac-address 0015.F914.11FF
>>>  button  1:7 2:15
>>> !
>>> ephone  8
>>>  mac-address 0015.F914.0E4C
>>>  button  1:9
>>> !
>>> ephone  9
>>>  mac-address 00E0.7524.B312
>>>  button  1:10
>>> !
>>> ephone  11
>>>  mac-address 0015.FA0B.7202
>>>  button  1:12
>>> !
>>> ephone  12
>>>  mac-address 0017.9486.B302
>>>  button  1:14
>>> !
>>>
>> === message truncated ===>
>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>>
>> 
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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 ealeatherman at gmail.com  Tue Aug 21 09:33:37 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 21 Aug 2007 09:33:37 -0400
Subject: [cisco-voip] CME vs SRST
In-Reply-To: <OFE1259B08.BD971730-ON8525733E.00487306-8525733E.0048A5E3@bftg.com>
References: <005101c7e377$5570aed0$60056883@pdp11>
	<OFE1259B08.BD971730-ON8525733E.00487306-8525733E.0048A5E3@bftg.com>
Message-ID: <94a1afde0708210633n6cebe096s74950e546d1611fe@mail.gmail.com>

Carlos,

AFAIK..
To use the extra features in CME SRST would would have to have them
provisioned ahead of time in the router config.. ephones, ephone-dns, plus
your hunt groups etc. So if your goal was to keep the user experience the
same, you would need to keep the CME configuration up to date with any
changes you made on call manager.



On 8/21/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote:
>
>
> Lelio -  But it could serve its purpose for smaller offices that need a
> specific functionality.
>
> Kate - What do you mean keeping the configs synchronized?
>
> Carlos
>
>
>  *"Lelio Fulgenzi" <lelio at uoguelph.ca>*
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 08/20/200
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From erickbee at gmail.com  Tue Aug 21 09:53:35 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Tue, 21 Aug 2007 08:53:35 -0500
Subject: [cisco-voip] CCM 5.11 MOH error -- Unmapped exception
In-Reply-To: <f4445faf0708201903n56762710mae78821445ad05cc@mail.gmail.com>
References: <f4445faf0708201903n56762710mae78821445ad05cc@mail.gmail.com>
Message-ID: <f4445faf0708210653h3f59e55bw528dc65e148cf954@mail.gmail.com>

Nevermind, this is bug id  CSCsh57225

On 8/20/07, Erick Bergquist <erickbee at gmail.com> wrote:
> When trying to add a MOH file, the find and list comes up with this.
>
> They had a MP3 file they converted to a WAV file then this happened
> when they tried to upload it. The new file shows under MOH audio
> sources but saids the file is not there and lists no properties for
> the audio source.
>
> Unmapped exception. The entity name must immediately follow the '&' in
> the entity reference.
>
> 0 records found, 133 minutes available.
>

From KandasEE at telkom.co.za  Tue Aug 21 10:30:38 2007
From: KandasEE at telkom.co.za (Errol Kandasami (EE))
Date: Tue, 21 Aug 2007 16:30:38 +0200
Subject: [cisco-voip] CCME AA - calls stuck in queue
Message-ID: <420AB64AE2F18546BA1C1E289B9FA7D401247B91@CNTRRA20-XCS01.telkom.co.za>


Hi,

 

I have configured the AA feature on the CCME ver 4.0 on a Cisco 3845
platform. When I dial the pilot number for the AA application, the
welcome and options messages play. However when you select the option,
the call gets routed to the ephone-hunt group but the IP Phone does
not ring and displays a message" 1 call in queue" and I am unable to
retrieve the call. The ringer light flashes but there is no audible
ring and answering the call does not retrieve it from the call queue.

 

Can any please assist.

 

Regards, 
Errol Kandasami 
LAN/WAN Design and Systems Integration 
Telephone: 012- 6806332 
Cell: 082 4106480 
Fax: 012 - 6808222 
e-mail: kandasee at telkom.co.za 
The one who kneels before God can stand up to anything.

 



<html>
<body>
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<a href="http://www.telkom.co.za/TelkomEMailLegalNotice.PDF">http://www.telkom.co.za/TelkomEMailLegalNotice.PDF</a>
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From jmilner at dpsciences.com  Tue Aug 21 10:26:47 2007
From: jmilner at dpsciences.com (Milner, Jamey)
Date: Tue, 21 Aug 2007 10:26:47 -0400
Subject: [cisco-voip] CCMAdmin
References: <588232.58199.qm@web39609.mail.mud.yahoo.com>
Message-ID: <54EC1C8498CCE54AAE3128187EB01A16489AF9@ink.dpsciences.com>

?

I am getting a funky message trying to log into the CCMAdmin page.

 

"You could not be authenicated successfully. System encountered internal error"

 

The servers are integrated into the AD LDAP. I tried logging in as Local Admin and it does not allow me to login. One of the Admins is able to log into the CM's with his AD username but, it takes awhile for him to finally get logged in. Any ideas.

I took a mirror drive from the CM's and put it in a LAB but, now there in no AD and I cant get in at all. Any ideas

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From Johnny.Crothers at vanco.co.uk  Tue Aug 21 10:35:29 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Tue, 21 Aug 2007 15:35:29 +0100
Subject: [cisco-voip]  IPCC IP Phone Agents
Message-ID: <A56BD242744BF540B15D1672E0E3E87501B26AE0@LON-INT-MSGVS01.internal.vancocorp.com>

Hi All,
 
The customer we have implemented IPCC for is complaining that they used
to have an "arrow" next to one of their lines to indicate that there was
a call waiting for them, and they are asking me to provide them with a
visal aid to show them there is a call in queue without using the
"state" softkey of the IPCC Phone Agent service. (Some people dont like
change as we all know!)
 
Has anyone ever used a 3rd party that has provided something similair to
this?
 
Cheers,
 
Johnny


**********************************************************************
Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient.  If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with the content of this email please e-mail to: e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

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From Johnny.Crothers at vanco.co.uk  Tue Aug 21 10:38:41 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Tue, 21 Aug 2007 15:38:41 +0100
Subject: [cisco-voip] CCMAdmin
In-Reply-To: <54EC1C8498CCE54AAE3128187EB01A16489AF9@ink.dpsciences.com>
Message-ID: <A56BD242744BF540B15D1672E0E3E87501B26AE2@LON-INT-MSGVS01.internal.vancocorp.com>

Is MLA enabled? You could try using the ccmadministrator account, once
in you'll need to assign privalages to each user account that requires a
login.
 
The ccmadministrator account password should be the same as the local
administrator account.
 
________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Milner, Jamey
Sent: 21 August 2007 15:27
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCMAdmin


? 

I am getting a funky message trying to log into the CCMAdmin page.

 

"You could not be authenicated successfully. System encountered internal
error"

 

The servers are integrated into the AD LDAP. I tried logging in as Local
Admin and it does not allow me to login. One of the Admins is able to
log into the CM's with his AD username but, it takes awhile for him to
finally get logged in. Any ideas.

I took a mirror drive from the CM's and put it in a LAB but, now there
in no AD and I cant get in at all. Any ideas



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From CarlosOrtiz at bayviewfinancial.com  Tue Aug 21 10:50:25 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 21 Aug 2007 10:50:25 -0400
Subject: [cisco-voip] CME vs SRST
In-Reply-To: <94a1afde0708210633n6cebe096s74950e546d1611fe@mail.gmail.com>
Message-ID: <OF1250F8A3.F4BEDF06-ON8525733E.00515A5B-8525733E.00518575@bftg.com>

Actually the goal is to keep it simple and provide a couple of functions. 
I think I can do the hunting and forwarding to offisite analog mailbox 
with the alias commands in SRST.   Im going to try that first without 
complicating matters....... 





"Ed Leatherman" <ealeatherman at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
08/21/2007 09:33 AM

To
"CarlosOrtiz at bayviewfinancial.com" <CarlosOrtiz at bayviewfinancial.com>
cc
cisco-voip at puck.nether.net, Kalle Tetto <tettokalle at hotmail.com>
Subject
Re: [cisco-voip] CME vs SRST






Carlos,

AFAIK..
To use the extra features in CME SRST would would have to have them 
provisioned ahead of time in the router config.. ephones, ephone-dns, plus 
your hunt groups etc. So if your goal was to keep the user experience the 
same, you would need to keep the CME configuration up to date with any 
changes you made on call manager. 



On 8/21/07, CarlosOrtiz at bayviewfinancial.com < 
CarlosOrtiz at bayviewfinancial.com> wrote:

Lelio -  But it could serve its purpose for smaller offices that need a 
specific functionality. 

Kate - What do you mean keeping the configs synchronized? 

Carlos 


"Lelio Fulgenzi" <lelio at uoguelph.ca> 
Sent by: cisco-voip-bounces at puck.nether.net 
08/20/200




-- 
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 tim.reimers at asheville.k12.nc.us  Tue Aug 21 10:59:52 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue, 21 Aug 2007 10:59:52 -0400
Subject: [cisco-voip] question about AD and creating Unity accounts...
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1AC7A@webmail.asheville.k12.nc.us>

 

 

We use Unity 4.01 in it's own AD domain (a Messaging only deployment)
and we have our own AD domain on a different VLAN.

 

We are in the process of fully automating the creation of users on our
data domain via VBS scripting and potentially Microsoft's MIIS product
to integrate with a statewide student management system.

 

We'd like to consider including the creation and deletion of Unity
voicemail accounts and _possibly_ phone and Global directory user
accounts in this automation.

 

Does anyone know if Cisco supports MIIS or any other automated scripting
technologies?

 

I'm familiar with BAT for CCM and I think there's something similar in
Tools Depot for Unity ---

 

But I'm wondering if anyone has automated those tools to the "next
level" by doing something like auto-creating the import files they need
and then automatically launching the BAT/import tools on CCM and Unity
instead of doing the BAT manually?

 

What does Cisco think of automating this? Do they provide any sort of
framework to design this from, or does the TAC say "you're on your own
if you want to break your system"   ;-)

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From wsisk at cisco.com  Tue Aug 21 11:03:09 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 21 Aug 2007 11:03:09 -0400
Subject: [cisco-voip] CCMAdmin
In-Reply-To: <54EC1C8498CCE54AAE3128187EB01A16489AF9@ink.dpsciences.com>
References: <588232.58199.qm@web39609.mail.mud.yahoo.com>
	<54EC1C8498CCE54AAE3128187EB01A16489AF9@ink.dpsciences.com>
Message-ID: <46CAFEAD.7040209@cisco.com>

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From KandasEE at telkom.co.za  Tue Aug 21 11:39:47 2007
From: KandasEE at telkom.co.za (Errol Kandasami (EE))
Date: Tue, 21 Aug 2007 17:39:47 +0200
Subject: [cisco-voip] CCME AA - calls stuck in queue (Errol Kandasami (EE))
Message-ID: <420AB64AE2F18546BA1C1E289B9FA7D401247B92@CNTRRA20-XCS01.telkom.co.za>


Hi,

 

I have managed to resolve the problem. The Call queue script was
failing because the MoH file under the telephony-service was
incorrect. 

 

Thanks.

 

Regards, 
Errol Kandasami 
LAN/WAN Design and Systems Integration 
Telephone: 012- 6806332 
Cell: 082 4106480 
Fax: 012 - 6808222 
e-mail: kandasee at telkom.co.za 
The one who kneels before God can stand up to anything.

 



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<a href="http://www.telkom.co.za/TelkomEMailLegalNotice.PDF"> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF</a>
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From svoll.voip at gmail.com  Tue Aug 21 13:01:13 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 21 Aug 2007 10:01:13 -0700
Subject: [cisco-voip] CER 1.3 and CM 5.1
Message-ID: <f84a38d30708211001r47029951s2f83811adbcaf6c6@mail.gmail.com>

i'm looking at the 1.31 CER admin guide and it talks about installing with
5.0.  is there any difference in the install with CM 5.0 vs 5.1?  if so, is
there a guide or updated guide I should be looking at?

Thanks

Scott

installing CER 1.3.2 with CM 5.1.2b
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From pmowry at getgds.com  Tue Aug 21 13:28:12 2007
From: pmowry at getgds.com (Patrick Mowry)
Date: Tue, 21 Aug 2007 12:28:12 -0500
Subject: [cisco-voip] CCME/CUE: What is the proper way to CFD busy to CUE
	when the incoming call is from a SIP trunk?
In-Reply-To: <mailman.11709.1187562184.1529.cisco-voip@puck.nether.net>
References: <mailman.11709.1187562184.1529.cisco-voip@puck.nether.net>
Message-ID: <4C792A099259634C83F97067671B37FE031FDBC9@GDSEXC.gds.pri>

Hello and thanks for the help,

  I have an incoming SIP trunk to a CCME 4.1 system. Incoming G729 calls
work just fine and if I set one of the DID numbers as the trigger to the
auto-attendant in CUE, the transcoders kick in and the call is answered
just fine.  
  My problem is with call-forward busy calls. I get the INVITE for an
incoming call and then send Trying.  This is followed by a SIP/2.0 302
Moved Temporarily with Contact: <sip:2002 at 10.33.64.145>.  The provider
then responds with an ACK and the caller gets a busy signal.  How do I
prevent CCME from sending redirect messages or to at least send the
Public IP address of the CME instead of the private IP of the service
Engine?  I do not have a firewall feature set.

Thanks,

-Patrick

From svoll.voip at gmail.com  Tue Aug 21 13:35:06 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 21 Aug 2007 10:35:06 -0700
Subject: [cisco-voip] CER
Message-ID: <f84a38d30708211035g413ff334jd1bb4497d03beb2@mail.gmail.com>

for 911 RP do you just select one phone partition that all CSS have access
to or do you add a 911 RP to each and every phone partition?

IE. partitions
DO
ES
HS
MS

but all CSS include DO so just do it there or add a RP for 911 in each
Partition.

TIA

Scott
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From jsteinberg at gmail.com  Tue Aug 21 14:19:23 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 21 Aug 2007 14:19:23 -0400
Subject: [cisco-voip] CCME/CUE: What is the proper way to CFD busy to
	CUE when the incoming call is from a SIP trunk?
In-Reply-To: <4C792A099259634C83F97067671B37FE031FDBC9@GDSEXC.gds.pri>
References: <mailman.11709.1187562184.1529.cisco-voip@puck.nether.net>
	<4C792A099259634C83F97067671B37FE031FDBC9@GDSEXC.gds.pri>
Message-ID: <f1b65f880708211119x33f242abid81a2b09f0c33faa@mail.gmail.com>

you need a session border controller / IP IP GW / Unified Border Element



On 8/21/07, Patrick Mowry <pmowry at getgds.com> wrote:
>
> Hello and thanks for the help,
>
>   I have an incoming SIP trunk to a CCME 4.1 system. Incoming G729 calls
> work just fine and if I set one of the DID numbers as the trigger to the
> auto-attendant in CUE, the transcoders kick in and the call is answered
> just fine.
>   My problem is with call-forward busy calls. I get the INVITE for an
> incoming call and then send Trying.  This is followed by a SIP/2.0 302
> Moved Temporarily with Contact: <sip:2002 at 10.33.64.145>.  The provider
> then responds with an ACK and the caller gets a busy signal.  How do I
> prevent CCME from sending redirect messages or to at least send the
> Public IP address of the CME instead of the private IP of the service
> Engine?  I do not have a firewall feature set.
>
> Thanks,
>
> -Patrick
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From pmowry at getgds.com  Tue Aug 21 14:19:37 2007
From: pmowry at getgds.com (Patrick Mowry)
Date: Tue, 21 Aug 2007 13:19:37 -0500
Subject: [cisco-voip] CCME/CUE: What is the proper way to CFD busy to
	CUE when the incoming call is from a SIP trunk?
References: <mailman.11709.1187562184.1529.cisco-voip@puck.nether.net> 
Message-ID: <4C792A099259634C83F97067671B37FE031FDBDF@GDSEXC.gds.pri>

Found it:


voice service voip
 no supplementary-service sip moved-temporarily


Unless there is a better way...


-Patrick


-----Original Message-----
From: Patrick Mowry 
Sent: Tuesday, August 21, 2007 12:28 PM
To: 'cisco-voip at puck.nether.net'
Subject: CCME/CUE: What is the proper way to CFD busy to CUE when the
incoming call is from a SIP trunk?

Hello and thanks for the help,

  I have an incoming SIP trunk to a CCME 4.1 system. Incoming G729 calls
work just fine and if I set one of the DID numbers as the trigger to the
auto-attendant in CUE, the transcoders kick in and the call is answered
just fine.  
  My problem is with call-forward busy calls. I get the INVITE for an
incoming call and then send Trying.  This is followed by a SIP/2.0 302
Moved Temporarily with Contact: <sip:2002 at 10.33.64.145>.  The provider
then responds with an ACK and the caller gets a busy signal.  How do I
prevent CCME from sending redirect messages or to at least send the
Public IP address of the CME instead of the private IP of the service
Engine?  I do not have a firewall feature set.

Thanks,

-Patrick

From svoll.voip at gmail.com  Tue Aug 21 16:15:12 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 21 Aug 2007 13:15:12 -0700
Subject: [cisco-voip] BAT -- CM 4.1 vs 5.1
Message-ID: <f84a38d30708211315k50a33be5g4f425c395acb0fd7@mail.gmail.com>

Good Afternoon all--

in CM 4.1 with BAT you can go in select phones and change a field.... IE>
CSS or partition or device pool, etc.

in CM 5.1 it looks like you export and then reimport.  Am I missing
something?  it sure is easier with 4.1 if that's the case.

Let me know if I'm out to lunch.

Scott
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From mark.mills at baesystems.com  Tue Aug 21 20:22:03 2007
From: mark.mills at baesystems.com (MILLS, Mark)
Date: Wed, 22 Aug 2007 09:52:03 +0930
Subject: [cisco-voip] CM 6 Australian Dial Plan Installer Error
Message-ID: <54DD8B39132EFA4293A5C0565004E5460995C9C6@SBEMAIL1.au.baesystems.com>

Hi,

I am currently doing my first CM 6 implementation, and have stumbled
across a problem trying to install the Australian Numbering Plan.

I have uploaded the file dp-ffr.1-1-4.AS.cop.sgn to all the CallManagers
in the cluster, but when going to do the actual install via Call Routing
-> Dial Plan Installer and selecting the plan, I get the following
error:

Database connection failed An Error Occurred: HY000

The info that is listed for the dialplan in Dial Plan Installer is AUNP
- Australian Numbering Plan 1.1(4).

Any points in the right direction would be appreciated... Hopefully its
something simple I have overlooked!

Thanks,
   Mark


From tednugent69 at yahoo.com  Tue Aug 21 21:52:07 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Tue, 21 Aug 2007 18:52:07 -0700 (PDT)
Subject: [cisco-voip] BAT -- CM 4.1 vs 5.1
In-Reply-To: <f84a38d30708211315k50a33be5g4f425c395acb0fd7@mail.gmail.com>
Message-ID: <915335.35568.qm@web36111.mail.mud.yahoo.com>

Hmm.. Thats not the case actually its very similar to
4.1 BAT, as far as I can tell so far you can update
both device or line on a query and device using csv
just like 4.1. 
HTH
Ted



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

> Good Afternoon all--
> 
> in CM 4.1 with BAT you can go in select phones and
> change a field.... IE>
> CSS or partition or device pool, etc.
> 
> in CM 5.1 it looks like you export and then
> reimport.  Am I missing
> something?  it sure is easier with 4.1 if that's the
> case.
> 
> Let me know if I'm out to lunch.
> 
> Scott
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



      ____________________________________________________________________________________
Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/ 

From mark.mills at baesystems.com  Wed Aug 22 03:18:01 2007
From: mark.mills at baesystems.com (MILLS, Mark)
Date: Wed, 22 Aug 2007 16:48:01 +0930
Subject: [cisco-voip] voice recognition telephone system for quadriplegic
Message-ID: <54DD8B39132EFA4293A5C0565004E54609A06FD1@SBEMAIL1.au.baesystems.com>


Hi All,

We currently utilise a Cisco CallManager telephone system, and have a
quadriplegic person in our employment that we need to come up with a
telephone solution for.

He uses dragon naturally speaking for using his PC, has anyone been in a
similar situation and can suggest a softphone or similar solution that
is CCM compatibile that works well with Dragon, or has other voice
recognition ability?

Thanks,  
   Mark

From Jordi.Riba at eu.didata.com  Wed Aug 22 05:43:35 2007
From: Jordi.Riba at eu.didata.com (Jordi Riba (Europe))
Date: Wed, 22 Aug 2007 11:43:35 +0200
Subject: [cisco-voip] Unity 4.0.3 upgrade to 4.2
Message-ID: <FA233791F398B848B978464507739E8FB6B752@EUBEBRUSVEX1.eu.didata.local>

Hi,

 

We have a customer with a CCM 4.2 and a Unity 4.0.3 (7825) we need to
upgrade hardware to a (7835) and software to 4.2. I have already
received new hardware and software.

 

I would like to know the steps I should follow to move all information
(Users, message, etc...) from the old hardware (version 4.03) to the new
hardware (version 4.2). I haven't found any guide explaining the steps
to follow and I thought that probably some of you have already realized
this kind of migration.

 

Any information will be appreciated.

 

Regards,

 

Jordi!

 

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From cisco.voip at verizon.net  Wed Aug 22 09:07:05 2007
From: cisco.voip at verizon.net (cisco.voip)
Date: Wed, 22 Aug 2007 09:07:05 -0400
Subject: [cisco-voip] CCM 5.1 appliance license
References: <mailman.12081.1187775854.1529.cisco-voip@puck.nether.net>
Message-ID: <000601c7e4bf$6f145b40$661b1b0a@cs.myharris.net>

Can anyone point me to or send me a link on how to enable the license on my
new Cisco 5.1 appliance server.
It it asking for a license file, but all I have in the box are the license
cards with the serial numbers on it?

Where do I enter the license number or am I missing something in the box?

Thanks!



From Matt.Slaga at us.didata.com  Wed Aug 22 09:34:34 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 22 Aug 2007 09:34:34 -0400
Subject: [cisco-voip] CCM 5.1 appliance license
In-Reply-To: <000601c7e4bf$6f145b40$661b1b0a@cs.myharris.net>
References: <mailman.12081.1187775854.1529.cisco-voip@puck.nether.net>
	<000601c7e4bf$6f145b40$661b1b0a@cs.myharris.net>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00685388A@USNAEXCH.na.didata.local>

You should have a PAK key on the box the software came in.  You would
then use this to access the licensing web page on CCO, along with your
server's mac address.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of cisco.voip
Sent: Wednesday, August 22, 2007 9:07 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM 5.1 appliance license

Can anyone point me to or send me a link on how to enable the license on
my
new Cisco 5.1 appliance server.
It it asking for a license file, but all I have in the box are the
license
cards with the serial numbers on it?

Where do I enter the license number or am I missing something in the
box?

Thanks!


_______________________________________________
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
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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 LuisYrigoyen at bayviewfinancial.com  Wed Aug 22 09:43:47 2007
From: LuisYrigoyen at bayviewfinancial.com (LuisYrigoyen at bayviewfinancial.com)
Date: Wed, 22 Aug 2007 09:43:47 -0400
Subject: [cisco-voip] IPCC IP Phone Agents
In-Reply-To: <A56BD242744BF540B15D1672E0E3E87501B26AE0@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <OF2A8A3D64.1AFBEDAD-ON8525733F.004A5829-8525733F.004B6C11@bftg.com>

Johnny,
        I don't have an answer to your question but another question:
What is IPCC Phone Agent service? how do one get that? is it free? 
>From your email and a quick googling I can deduce that, is a phone service 
that allows agents to log in/out via the phones etc. but I'm interested on 
what else it can do.
We are currently running IPCC Express Enhanced 4.0(4).

I'm very new with IPCC and any info or documentation will be greatly 
appreciated.

thank you




Luis Yrigoyen

Network Operations
Bayview Financial, L.P.
4425 Ponce de Leon Blvd.
Coral Gables, FL 33146

T 305.817.5083
F 305.646.4744
E luisyrigoyen at bayviewfinancial.com




"Johnny Crothers \(c\)" <Johnny.Crothers at vanco.co.uk> 
Sent by: cisco-voip-bounces at puck.nether.net
08/21/2007 10:35 AM

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

Subject
[cisco-voip]  IPCC IP Phone Agents






Hi All,
 
The customer we have implemented IPCC for is complaining that they used to 
have an "arrow" next to one of their lines to indicate that there was a 
call waiting for them, and they are asking me to provide them with a visal 
aid to show them there is a call in queue without using the "state" 
softkey of the IPCC Phone Agent service. (Some people dont like change as 
we all know!)
 
Has anyone ever used a 3rd party that has provided something similair to 
this?
 
Cheers,
 
Johnny


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From Johnny.Crothers at vanco.co.uk  Wed Aug 22 10:07:55 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Wed, 22 Aug 2007 15:07:55 +0100
Subject: [cisco-voip] IPCC IP Phone Agents
In-Reply-To: <OF2A8A3D64.1AFBEDAD-ON8525733F.004A5829-8525733F.004B6C11@bftg.com>
Message-ID: <A56BD242744BF540B15D1672E0E3E87501B26E47@LON-INT-MSGVS01.internal.vancocorp.com>

Hi Luis,
 
Yes, IP Phone Agents are free, all you need to do is configure a service
(Like you do for extension mobility). 
 
URL you need to use is : 
 
http://<IPCC IP Address>:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp
<http://<IPCC IP
Address>:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp> 
 
Then subscribe the service to the phones or device profiles..

Johnny Crothers

IPT Specialist

 

________________________________

From: LuisYrigoyen at bayviewfinancial.com
[mailto:LuisYrigoyen at bayviewfinancial.com] 
Sent: 22 August 2007 14:44
To: Johnny Crothers (c)
Cc: cisco-voip at puck.nether.net; cisco-voip-bounces at puck.nether.net
Subject: Re: [cisco-voip] IPCC IP Phone Agents



Johnny, 
        I don't have an answer to your question but another question: 
What is IPCC Phone Agent service? how do one get that? is it free? 
>From your email and a quick googling I can deduce that, is a phone
service that allows agents to log in/out via the phones etc. but I'm
interested on what else it can do. 
We are currently running IPCC Express Enhanced 4.0(4). 

I'm very new with IPCC and any info or documentation will be greatly
appreciated. 

thank you 




Luis Yrigoyen

Network Operations
Bayview Financial, L.P.
4425 Ponce de Leon Blvd.
Coral Gables, FL 33146

T 305.817.5083
F 305.646.4744
E luisyrigoyen at bayviewfinancial.com




"Johnny Crothers \(c\)" <Johnny.Crothers at vanco.co.uk> 
Sent by: cisco-voip-bounces at puck.nether.net 

08/21/2007 10:35 AM 

To
<cisco-voip at puck.nether.net> 
cc
Subject
[cisco-voip]  IPCC IP Phone Agents

	




Hi All, 
  
The customer we have implemented IPCC for is complaining that they used
to have an "arrow" next to one of their lines to indicate that there was
a call waiting for them, and they are asking me to provide them with a
visal aid to show them there is a call in queue without using the
"state" softkey of the IPCC Phone Agent service. (Some people dont like
change as we all know!) 
  
Has anyone ever used a 3rd party that has provided something similair to
this? 
  
Cheers, 
  
Johnny 


**********************************************************************
Any opinions expressed in the email are those of the individual and not
necessarily the company. This email and any files transmitted with it
are confidential and solely for the use of the intended recipient. If
you are not the intended recipient or the person responsible for
delivering it to the intended recipient, be advised that you have
received this email in error and that any dissemination, distribution,
copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with
the content of this email please e-mail to:
e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software
viruses which could damage your own computer system. While the sender
has taken every reasonable precaution to minimise this risk, we cannot
accept liability for any damage which you sustain as a result of
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opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John
Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
**********************************************************************__
_____________________________________________
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From Matt.Slaga at us.didata.com  Wed Aug 22 10:11:57 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 22 Aug 2007 10:11:57 -0400
Subject: [cisco-voip] IPCC IP Phone Agents
In-Reply-To: <A56BD242744BF540B15D1672E0E3E87501B26E47@LON-INT-MSGVS01.internal.vancocorp.com>
References: <OF2A8A3D64.1AFBEDAD-ON8525733F.004A5829-8525733F.004B6C11@bftg.com>
	<A56BD242744BF540B15D1672E0E3E87501B26E47@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006853996@USNAEXCH.na.didata.local>

It's not free so to speak, but starting with UCCX 4.x each user license
comprises of a selection of CAD (agent desktop), supervisor or IPPA
(phone agent).  In previous versions, you had to purchase each
separately.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johnny Crothers
(c)
Sent: Wednesday, August 22, 2007 10:08 AM
To: LuisYrigoyen at bayviewfinancial.com
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC IP Phone Agents

 

Hi Luis,

 

Yes, IP Phone Agents are free, all you need to do is configure a service
(Like you do for extension mobility). 

 

URL you need to use is : 

 

http://<IPCC IP Address>:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp
<http://%3cIPCC%20IP%20Address%3e:6293/ipphone/jsp/sciphonexml/IPAgentIn
itial.jsp> 

 

Then subscribe the service to the phones or device profiles..

Johnny Crothers

IPT Specialist

 

 

________________________________

From: LuisYrigoyen at bayviewfinancial.com
[mailto:LuisYrigoyen at bayviewfinancial.com] 
Sent: 22 August 2007 14:44
To: Johnny Crothers (c)
Cc: cisco-voip at puck.nether.net; cisco-voip-bounces at puck.nether.net
Subject: Re: [cisco-voip] IPCC IP Phone Agents


Johnny, 
        I don't have an answer to your question but another question: 
What is IPCC Phone Agent service? how do one get that? is it free? 
From your email and a quick googling I can deduce that, is a phone
service that allows agents to log in/out via the phones etc. but I'm
interested on what else it can do. 
We are currently running IPCC Express Enhanced 4.0(4). 

I'm very new with IPCC and any info or documentation will be greatly
appreciated. 

thank you 




Luis Yrigoyen

Network Operations
Bayview Financial, L.P.
4425 Ponce de Leon Blvd.
Coral Gables, FL 33146

T 305.817.5083
F 305.646.4744
E luisyrigoyen at bayviewfinancial.com




"Johnny Crothers \(c\)" <Johnny.Crothers at vanco.co.uk> 
Sent by: cisco-voip-bounces at puck.nether.net 

08/21/2007 10:35 AM 

To

<cisco-voip at puck.nether.net> 

cc

	
Subject

[cisco-voip]  IPCC IP Phone Agents

 

		




Hi All, 
  
The customer we have implemented IPCC for is complaining that they used
to have an "arrow" next to one of their lines to indicate that there was
a call waiting for them, and they are asking me to provide them with a
visal aid to show them there is a call in queue without using the
"state" softkey of the IPCC Phone Agent service. (Some people dont like
change as we all know!) 
  
Has anyone ever used a 3rd party that has provided something similair to
this? 
  
Cheers, 
  
Johnny 


**********************************************************************
Any opinions expressed in the email are those of the individual and not
necessarily the company. This email and any files transmitted with it
are confidential and solely for the use of the intended recipient. If
you are not the intended recipient or the person responsible for
delivering it to the intended recipient, be advised that you have
received this email in error and that any dissemination, distribution,
copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with
the content of this email please e-mail to:
e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software
viruses which could damage your own computer system. While the sender
has taken every reasonable precaution to minimise this risk, we cannot
accept liability for any damage which you sustain as a result of
software viruses. You should carry out your own virus checks before
opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John
Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
**********************************************************************__
_____________________________________________
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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designated addressee(s) named above only.  If you are not the
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From ErasmuE4 at telkom.co.za  Wed Aug 22 11:05:27 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 22 Aug 2007 17:05:27 +0200
Subject: [cisco-voip] call manager - event viewer strange events
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F019850F9@CNTRRA20-XCS01.telkom.co.za>


HI 

 

I am currently struggling with what sounds like a one way voice issue
at a remote call manager branch using an h323 gateway and phones
registering to a central cluster at HQ. I looked at all the usual
stuff - clocking on gateway is fine etc etc. The problem is basically
the remote cisco attendant users acting as operators say that
sporadically they have a couple of minutes per day when they answer
pstn incoming calls but it sounds silent on the line and they cannot
hear the caller calling in - the phone shows the call as connected. 

 

 

I also found a bit of conflicting info on the cisco web site - cisco
say in some places try not to use MTP and not to tick the require
media termination on the gateway config page in CCM unless it is
absolutely necessary for example when using non cisco h323 devices
etc. However - in other docs where they talk about possible one way
voice / voice cut through issues the say tick the require Media
Termination if you have these issues. Anyway. I also started looking
at the event viewer on the pub and sub (active ccm) and noticed a lot
of application errors as shown below. One example is loads of the
unregister events at times when I know not that many phones will
actually unregister and the transient connection attempts. Has anyone
seen similar things on CCM 4-2 x 

 

Any advice appreciated. 

 

Error: DeviceUnregistered - Device unregistered.

  Device name.: SEP0019560A32B2

  Device IP address.: 172.1.3.21

  Device type. [Optional]: 30007

  Device description [Optional].: Auto 50142

  Reason Code [Optional].: 9

  App ID: Cisco CallManager

  Cluster ID: StandAloneCluster

  Node ID: 172.1.14.11

Explanation: A device that has previously registered with Cisco
CallManager has unregistered. This event may be issued as part of
normal unregistration event or due to some other reason such as loss
of keepalives.

Recommended Action: No action is required if unregistration of this
device was expected.. 

 

----------------------------------------------------------------------
--------------------------------------------------------------- 

 

Error: DeviceTransientConnection - Transient connection attempt.

  Connecting Port: 51471

  Device name [Optional].: 

  Device IP address.: 172.1.5.40

  Device type. [Optional]: 255

  Reason Code [Optional].: 6

  App ID: Cisco CallManager

  Cluster ID: StandAloneCluster

  Node ID: 172.1.14.10

Explanation: A connection was established and immediately dropped
before completing registration. Incomplete registration may indicate a
device is rehoming in the middle of registration. The alarm could also
indicate a device misconfiguration, database error, or an
illegal/unknown device trying to attempt a connection.

Recommended Action: No action is required if this event was issued as
a result of a normal device rehome.. 

 

----------------------------------------------------------------------
----------------------------------------------------------------------
-----  

 

Error: DeviceTransientConnection - Transient connection attempt.

  Connecting Port: 18715

  Device name [Optional].: 

  Device IP address.: 172.1.11.60

  Device type. [Optional]: 255

  Reason Code [Optional].: 6

  App ID: Cisco CallManager

  Cluster ID: StandAloneCluster

  Node ID: 172.1.14.10

Explanation: A connection was established and immediately dropped
before completing registration. Incomplete registration may indicate a
device is rehoming in the middle of registration. The alarm could also
indicate a device misconfiguration, database error, or an
illegal/unknown device trying to attempt a connection.

Recommended Action: No action is required if this event was issued as
a result of a normal device rehome.. 

 

 

 



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From svoll.voip at gmail.com  Wed Aug 22 11:34:31 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 22 Aug 2007 08:34:31 -0700
Subject: [cisco-voip] BAT -- CM 4.1 vs 5.1
In-Reply-To: <915335.35568.qm@web36111.mail.mud.yahoo.com>
References: <f84a38d30708211315k50a33be5g4f425c395acb0fd7@mail.gmail.com>
	<915335.35568.qm@web36111.mail.mud.yahoo.com>
Message-ID: <f84a38d30708220834p317a0f04rb2cabe462f7eaa62@mail.gmail.com>

Perfect.  That was what I was looking for.

Thanks
Scott


On 8/21/07, Ted Nugent <tednugent69 at yahoo.com> wrote:
>
> Hmm.. Thats not the case actually its very similar to
> 4.1 BAT, as far as I can tell so far you can update
> both device or line on a query and device using csv
> just like 4.1.
> HTH
> Ted
>
>
>
> --- Scott Voll <svoll.voip at gmail.com> wrote:
>
> > Good Afternoon all--
> >
> > in CM 4.1 with BAT you can go in select phones and
> > change a field.... IE>
> > CSS or partition or device pool, etc.
> >
> > in CM 5.1 it looks like you export and then
> > reimport.  Am I missing
> > something?  it sure is easier with 4.1 if that's the
> > case.
> >
> > Let me know if I'm out to lunch.
> >
> > Scott
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ____________________________________________________________________________________
> Park yourself in front of a world of choices in alternative vehicles.
> Visit the Yahoo! Auto Green Center.
> http://autos.yahoo.com/green_center/
>
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From svoll.voip at gmail.com  Wed Aug 22 12:54:43 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 22 Aug 2007 09:54:43 -0700
Subject: [cisco-voip] Exchange 2007 is cool
Message-ID: <f84a38d30708220954k2c0062bag6956648195bffe0a@mail.gmail.com>

we just got Exchange 2007 up on a couple test accounts and I'm inpressed.
The Voice portal to Exchange calandars / VM / Email / etc is nice.

I'm hoping to upgrade my current installation of Unity to 5 and see the
differences but believe it or not M$ has done a pretty good job, other then
having to buy a third party MWI package

I don't care for the "welcome to exchange" on a lot of the greetings but
other then that, how can you argue with the price if you already have
exchange.

Scott
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From lelio at uoguelph.ca  Wed Aug 22 10:20:17 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 22 Aug 2007 10:20:17 -0400
Subject: [cisco-voip] problem with phone getting DHCP address from voicevlan
Message-ID: <009601c7e4c7$902be890$56196883@cfs.uoguelph.ca>

We have a number of new phones with a problem whereby they get a DHCP address from the data VLAN not the voice VLAN. The only way to make it work is to remove the data VLAN, let the phone upgrade and then everything works after replacing the data VLAN even after erasing the phone configuration.

Here's the specific phone load information:

App Load ID: P003030IMFG2
Boot Load ID PC0303010200
Version: 3.1 (MF.G2)

Here's my switch config:

interface FastEthernet1/0/40
 switchport access vlan 1220                  *this is what I remove to make things work*
 switchport mode access
 switchport voice vlan 3104
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast
end

--------------------------------------------------------------------------------
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 paramds at gmail.com  Wed Aug 22 15:01:49 2007
From: paramds at gmail.com (param singh)
Date: Wed, 22 Aug 2007 14:01:49 -0500
Subject: [cisco-voip] Sanity check for changing passwords on the call
	manager 4.1.3
Message-ID: <867640cd0708221201i67019ca3kce48a2d734bd878f@mail.gmail.com>

Hi All,

I have to change the windows administrator password as well as the web
password for my customer's call manager.

Following are the steps I am going to follow. Please let me know if this
should work for me or I should do something different.

This is 4.1(3) call manager and 4.0(4) IPCC.

CHANGING PASSWORDS ON CALL MANAGER



   1. Windows Administrator Passwords on Call manager :





*Note: *The Administrator account serves as the default Windows NT
administration account. Cisco CallManager does not use this password.



Use any of these methods :



   - Choose *Start *> *Program Files *> *Administrative Tools *> *Computer
   Management *> *Users*, select *Administrator Account *and right click
   *Set Password*.



   - The Windows Administrator account password is a login password into
   the physical server. You can change the Windows Administrator password when
   you press *Ctrl+Alt+Delete *and click on *Change Password*.



It is necessary to reboot the cluster after you change the passwords in
order to remove any potential problems with Windows password caching.





   1. Changing CCMAdministrator Password on the call manager :



 Since MLA is enabled on the call manager.

 Disable MLA and re-enable MLA. Once you do this , it is going to ask for
the new password.  After this is done, restart the IIS on all the servers.


I guess this should do it. Also do I need to run "CCMPwdchanger" or this
should be enough. I don't have to change Directory Manager , CCMSysuser
accounts. Any Ideas.




Thanks
Param
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From ez4me2c3d at gmail.com  Wed Aug 22 15:25:43 2007
From: ez4me2c3d at gmail.com (Anthony H)
Date: Wed, 22 Aug 2007 14:25:43 -0500
Subject: [cisco-voip] Sanity check for changing passwords on the call
	manager 4.1.3
In-Reply-To: <867640cd0708221201i67019ca3kce48a2d734bd878f@mail.gmail.com>
References: <867640cd0708221201i67019ca3kce48a2d734bd878f@mail.gmail.com>
Message-ID: <49921b8d0708221225o121ea215ib27fedd9b67639f5@mail.gmail.com>

*Note: *The Administrator account serves as the default Windows NT
administration account. Cisco CallManager does not use this password.


What is that supposed to mean?  It says CCM does not use this password, then
proceeds to tell you how to change it?  Is this a typo?

Anthony


On 8/22/07, param singh <paramds at gmail.com> wrote:
>
> Hi All,
>
> I have to change the windows administrator password as well as the web
> password for my customer's call manager.
>
> Following are the steps I am going to follow. Please let me know if this
> should work for me or I should do something different.
>
> This is 4.1(3) call manager and 4.0(4) IPCC.
>
> CHANGING PASSWORDS ON CALL MANAGER
>
>
>
>    1. Windows Administrator Passwords on Call manager :
>
>
>
>
>
> *Note: *The Administrator account serves as the default Windows NT
> administration account. Cisco CallManager does not use this password.
>
>
>
> Use any of these methods :
>
>
>
>    - Choose *Start *> *Program Files *> *Administrative Tools *> *Computer
>    Management *> *Users*, select *Administrator Account *and right
>    click *Set Password*.
>
>
>
>    - The Windows Administrator account password is a login password
>    into the physical server. You can change the Windows Administrator password
>    when you press *Ctrl+Alt+Delete *and click on *Change Password*.
>
>
>
> It is necessary to reboot the cluster after you change the passwords in
> order to remove any potential problems with Windows password caching.
>
>
>
>
>
>    1. Changing CCMAdministrator Password on the call manage r :
>
>
>
>  Since MLA is enabled on the call manager.
>
>  Disable MLA and re-enable MLA. Once you do this , it is going to ask for
> the new password.  After this is done, restart the IIS on all the servers.
>
>
> I guess this should do it. Also do I need to run "CCMPwdchanger" or this
> should be enough. I don't have to change Directory Manager , CCMSysuser
> accounts. Any Ideas.
>
>
>
>
> Thanks
> Param
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Anthony
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From paramds at gmail.com  Wed Aug 22 15:30:57 2007
From: paramds at gmail.com (param singh)
Date: Wed, 22 Aug 2007 14:30:57 -0500
Subject: [cisco-voip] Sanity check for changing passwords on the call
	manager 4.1.3
In-Reply-To: <49921b8d0708221225o121ea215ib27fedd9b67639f5@mail.gmail.com>
References: <867640cd0708221201i67019ca3kce48a2d734bd878f@mail.gmail.com>
	<49921b8d0708221225o121ea215ib27fedd9b67639f5@mail.gmail.com>
Message-ID: <867640cd0708221230o18700c20k6cb31a6059c20eda@mail.gmail.com>

I guess that means that none of the Call manager services are effected by
changing the Window's Administration Account. It is just to log into the
Windows box. My customer wants to change that too..

On 8/22/07, Anthony H <ez4me2c3d at gmail.com> wrote:
>
> *Note: *The Administrator account serves as the default Windows NT
> administration account. Cisco CallManager does not use this password.
>
>
> What is that supposed to mean?  It says CCM does not use this password,
> then proceeds to tell you how to change it?  Is this a typo?
>
> Anthony
>
>
> On 8/22/07, param singh <paramds at gmail.com> wrote:
>
> > Hi All,
> >
> > I have to change the windows administrator password as well as the web
> > password for my customer's call manager.
> >
> > Following are the steps I am going to follow. Please let me know if this
> > should work for me or I should do something different.
> >
> > This is 4.1(3) call manager and 4.0(4) IPCC.
> >
> > CHANGING PASSWORDS ON CALL MANAGER
> >
> >
> >
> >    1. Windows Administrator Passwords on Call manager :
> >
> >
> >
> >
> >
> > *Note: *The Administrator account serves as the default Windows NT
> > administration account. Cisco CallManager does not use this password.
> >
> >
> >
> > Use any of these methods :
> >
> >
> >
> >    - Choose *Start *> *Program Files *> *Administrative Tools *> *Computer
> >    Management *> *Users*, select *Administrator Account *and right
> >    click *Set Password*.
> >
> >
> >
> >    - The Windows Administrator account password is a login password
> >    into the physical server. You can change the Windows Administrator password
> >    when you press *Ctrl+Alt+Delete *and click on *Change Password*.
> >
> >
> >
> > It is necessary to reboot the cluster after you change the passwords in
> > order to remove any potential problems with Windows password caching.
> >
> >
> >
> >
> >
> >    1. Changing CCMAdministrator Password on the call manage r :
> >
> >
> >
> >  Since MLA is enabled on the call manager.
> >
> >  Disable MLA and re-enable MLA. Once you do this , it is going to ask
> > for the new password.  After this is done, restart the IIS on all the
> > servers.
> >
> >
> > I guess this should do it. Also do I need to run "CCMPwdchanger" or this
> > should be enough. I don't have to change Directory Manager , CCMSysuser
> > accounts. Any Ideas.
> >
> >
> >
> >
> > Thanks
> > Param
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> --
> Anthony
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From pfabian at telesphere.com  Wed Aug 22 15:37:24 2007
From: pfabian at telesphere.com (Pete Fabian)
Date: Wed, 22 Aug 2007 12:37:24 -0700
Subject: [cisco-voip] problem with phone getting DHCP address from
	voicevlan
In-Reply-To: <009601c7e4c7$902be890$56196883@cfs.uoguelph.ca>
Message-ID: <006c01c7e4f3$deff5610$34fea8c0@telesphereltd.com>

Try,

interface FastEthernet1/0/40
 switchport access vlan 3104                  
 switchport mode trunk encapsulation dot1q

 switchport trunk native vlan 1220
 switchport voice vlan 3104
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast
end

 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, August 22, 2007 7:20 AM
To: Cisco VoIPoE List
Subject: [cisco-voip] problem with phone getting DHCP address from voicevlan

 

We have a number of new phones with a problem whereby they get a DHCP
address from the data VLAN not the voice VLAN. The only way to make it work
is to remove the data VLAN, let the phone upgrade and then everything works
after replacing the data VLAN even after erasing the phone configuration.

 

Here's the specific phone load information:

 

App Load ID: P003030IMFG2

Boot Load ID PC0303010200

Version: 3.1 (MF.G2)

 

Here's my switch config:

 

interface FastEthernet1/0/40
 switchport access vlan 1220                  *this is what I remove to make
things work*
 switchport mode access
 switchport voice vlan 3104
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast
end

 

----------------------------------------------------------------------------
----
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 lelio at uoguelph.ca  Wed Aug 22 15:51:23 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 22 Aug 2007 15:51:23 -0400
Subject: [cisco-voip] problem with phone getting DHCP address from
	voicevlan
References: <006c01c7e4f3$deff5610$34fea8c0@telesphereltd.com>
Message-ID: <007f01c7e4f5$d1aa7c40$56196883@cfs.uoguelph.ca>

I believe this only works with a phone attached. We want our ports to work even without a phone.
  ----- Original Message ----- 
  From: Pete Fabian 
  To: 'Lelio Fulgenzi' ; 'Cisco VoIPoE List' 
  Sent: Wednesday, August 22, 2007 3:37 PM
  Subject: RE: [cisco-voip] problem with phone getting DHCP address from voicevlan


  Try,

  interface FastEthernet1/0/40
   switchport access vlan 3104                  
   switchport mode trunk encapsulation dot1q

   switchport trunk native vlan 1220
   switchport voice vlan 3104
   srr-queue bandwidth share 10 10 60 20
   srr-queue bandwidth shape  10  0  0  0
   mls qos trust device cisco-phone
   mls qos trust cos
   auto qos voip cisco-phone
   spanning-tree portfast
  end

   

   

   


------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Wednesday, August 22, 2007 7:20 AM
  To: Cisco VoIPoE List
  Subject: [cisco-voip] problem with phone getting DHCP address from voicevlan

   

  We have a number of new phones with a problem whereby they get a DHCP address from the data VLAN not the voice VLAN. The only way to make it work is to remove the data VLAN, let the phone upgrade and then everything works after replacing the data VLAN even after erasing the phone configuration.

   

  Here's the specific phone load information:

   

  App Load ID: P003030IMFG2

  Boot Load ID PC0303010200

  Version: 3.1 (MF.G2)

   

  Here's my switch config:

   

  interface FastEthernet1/0/40
   switchport access vlan 1220                  *this is what I remove to make things work*
   switchport mode access
   switchport voice vlan 3104
   srr-queue bandwidth share 10 10 60 20
   srr-queue bandwidth shape  10  0  0  0
   mls qos trust device cisco-phone
   mls qos trust cos
   auto qos voip cisco-phone
   spanning-tree portfast
  end

   

  --------------------------------------------------------------------------------
  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 Chris.Ellington at inin.com  Wed Aug 22 15:59:14 2007
From: Chris.Ellington at inin.com (Ellington, Chris)
Date: Wed, 22 Aug 2007 15:59:14 -0400
Subject: [cisco-voip] problem with phone getting DHCP address from
	voicevlan
In-Reply-To: <mailman.12188.1187811061.1529.cisco-voip@puck.nether.net>
References: <mailman.12188.1187811061.1529.cisco-voip@puck.nether.net>
Message-ID: <8F9F1C6AA1D6834EACC379C8821533A605771DE2@i3mail1.i3domain.inin.com>

With your setup, you would initially tell the switch to use untagged
packets and vlan 1220 (I'm assuming this is your data vlan).  When you
drop that, my _guess_ is that the voice vlan takes over but I don't have
a switch and a phone handy to test.

The switch voice vlan command is only useful if you are using dot1q
trunks -or- if you are going to use dot1p for CoS purposes.  Here's a
good link on it
http://www.cisco.com/en/US/docs/switches/lan/catalyst3550/software/relea
se/12.1_12c_ea1/configuration/guide/swvoip.html

chris


Message: 2
Date: Wed, 22 Aug 2007 10:20:17 -0400
From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Subject: [cisco-voip] problem with phone getting DHCP address from
	voicevlan
To: "Cisco VoIPoE List" <cisco-voip at puck.nether.net>
Message-ID: <009601c7e4c7$902be890$56196883 at cfs.uoguelph.ca>
Content-Type: text/plain; charset="iso-8859-1"

We have a number of new phones with a problem whereby they get a DHCP
address from the data VLAN not the voice VLAN. The only way to make it
work is to remove the data VLAN, let the phone upgrade and then
everything works after replacing the data VLAN even after erasing the
phone configuration.

Here's the specific phone load information:

App Load ID: P003030IMFG2
Boot Load ID PC0303010200
Version: 3.1 (MF.G2)

Here's my switch config:

interface FastEthernet1/0/40
 switchport access vlan 1220                  *this is what I remove to
make things work*
 switchport mode access
 switchport voice vlan 3104
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast
end

------------------------------------------------------------------------
--------
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 Chris.Ellington at inin.com  Wed Aug 22 16:01:21 2007
From: Chris.Ellington at inin.com (Ellington, Chris)
Date: Wed, 22 Aug 2007 16:01:21 -0400
Subject: [cisco-voip] cisco-voip Digest, Vol 54, Issue 97
In-Reply-To: <mailman.12195.1187812288.1529.cisco-voip@puck.nether.net>
References: <mailman.12195.1187812288.1529.cisco-voip@puck.nether.net>
Message-ID: <8F9F1C6AA1D6834EACC379C8821533A605771DE3@i3mail1.i3domain.inin.com>

Change the access vlan to the data vlan, remove the switchport mode
trunk and it will work wonders; a non-dot1q client won't negotiate a
trunk and will get the native.  A phone will get a trunk and I think
that's what you want...

Chris


Christopher Ellington | VoIP/SIP Engineer 
phone & fax +1.317.715.8578 | chris.ellington at inin.com
CCIE #6814

Interactive Intelligence Inc.
Deliberately Innovative
www.inin.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: Wednesday, August 22, 2007 3:51 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 54, Issue 97

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: problem with phone getting DHCP address from	voicevlan
      (Pete Fabian)
   2. Re: problem with phone getting DHCP address from	voicevlan
      (Lelio Fulgenzi)


----------------------------------------------------------------------

Message: 1
Date: Wed, 22 Aug 2007 12:37:24 -0700
From: "Pete Fabian" <pfabian at telesphere.com>
Subject: Re: [cisco-voip] problem with phone getting DHCP address from
	voicevlan
To: "'Lelio Fulgenzi'" <lelio at uoguelph.ca>,	"'Cisco VoIPoE List'"
	<cisco-voip at puck.nether.net>
Message-ID: <006c01c7e4f3$deff5610$34fea8c0 at telesphereltd.com>
Content-Type: text/plain; charset="us-ascii"

Try,

interface FastEthernet1/0/40
 switchport access vlan 3104                  
 switchport mode trunk encapsulation dot1q

 switchport trunk native vlan 1220
 switchport voice vlan 3104
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast
end

 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, August 22, 2007 7:20 AM
To: Cisco VoIPoE List
Subject: [cisco-voip] problem with phone getting DHCP address from
voicevlan

 

We have a number of new phones with a problem whereby they get a DHCP
address from the data VLAN not the voice VLAN. The only way to make it
work
is to remove the data VLAN, let the phone upgrade and then everything
works
after replacing the data VLAN even after erasing the phone
configuration.

 

Here's the specific phone load information:

 

App Load ID: P003030IMFG2

Boot Load ID PC0303010200

Version: 3.1 (MF.G2)

 

Here's my switch config:

 

interface FastEthernet1/0/40
 switchport access vlan 1220                  *this is what I remove to
make
things work*
 switchport mode access
 switchport voice vlan 3104
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast
end

 

------------------------------------------------------------------------
----
----
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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------------------------------

Message: 2
Date: Wed, 22 Aug 2007 15:51:23 -0400
From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Subject: Re: [cisco-voip] problem with phone getting DHCP address from
	voicevlan
To: "Pete Fabian" <pfabian at telesphere.com>,	"'Cisco VoIPoE List'"
	<cisco-voip at puck.nether.net>
Message-ID: <007f01c7e4f5$d1aa7c40$56196883 at cfs.uoguelph.ca>
Content-Type: text/plain; charset="iso-8859-1"

I believe this only works with a phone attached. We want our ports to
work even without a phone.
  ----- Original Message ----- 
  From: Pete Fabian 
  To: 'Lelio Fulgenzi' ; 'Cisco VoIPoE List' 
  Sent: Wednesday, August 22, 2007 3:37 PM
  Subject: RE: [cisco-voip] problem with phone getting DHCP address from
voicevlan


  Try,

  interface FastEthernet1/0/40
   switchport access vlan 3104                  
   switchport mode trunk encapsulation dot1q

   switchport trunk native vlan 1220
   switchport voice vlan 3104
   srr-queue bandwidth share 10 10 60 20
   srr-queue bandwidth shape  10  0  0  0
   mls qos trust device cisco-phone
   mls qos trust cos
   auto qos voip cisco-phone
   spanning-tree portfast
  end

   

   

   


------------------------------------------------------------------------
------

  From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Wednesday, August 22, 2007 7:20 AM
  To: Cisco VoIPoE List
  Subject: [cisco-voip] problem with phone getting DHCP address from
voicevlan

   

  We have a number of new phones with a problem whereby they get a DHCP
address from the data VLAN not the voice VLAN. The only way to make it
work is to remove the data VLAN, let the phone upgrade and then
everything works after replacing the data VLAN even after erasing the
phone configuration.

   

  Here's the specific phone load information:

   

  App Load ID: P003030IMFG2

  Boot Load ID PC0303010200

  Version: 3.1 (MF.G2)

   

  Here's my switch config:

   

  interface FastEthernet1/0/40
   switchport access vlan 1220                  *this is what I remove
to make things work*
   switchport mode access
   switchport voice vlan 3104
   srr-queue bandwidth share 10 10 60 20
   srr-queue bandwidth shape  10  0  0  0
   mls qos trust device cisco-phone
   mls qos trust cos
   auto qos voip cisco-phone
   spanning-tree portfast
  end

   

 
------------------------------------------------------------------------
--------
  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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------------------------------

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

End of cisco-voip Digest, Vol 54, Issue 97
******************************************



From CarlosOrtiz at bayviewfinancial.com  Wed Aug 22 15:56:19 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 22 Aug 2007 15:56:19 -0400
Subject: [cisco-voip] 7961 long reset
Message-ID: <OFBD01A6BC.934C9557-ON8525733F.006D2869-8525733F.006D8704@bftg.com>

When updating lines on some new 7961 phones I am seeing a large delay (1 
minute) for the phone to reset.   Is this normal or am I possibly hitting 
a bug?

Load file is  SCCP41.8-2-2SR1S.
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From jason.aarons at us.didata.com  Wed Aug 22 16:07:37 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 22 Aug 2007 16:07:37 -0400
Subject: [cisco-voip] Sanity check for changing passwords on the
	callmanager 4.1.3
In-Reply-To: <867640cd0708221230o18700c20k6cb31a6059c20eda@mail.gmail.com>
References: <867640cd0708221201i67019ca3kce48a2d734bd878f@mail.gmail.com><49921b8d0708221225o121ea215ib27fedd9b67639f5@mail.gmail.com>
	<867640cd0708221230o18700c20k6cb31a6059c20eda@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0068541B2@USNAEXCH.na.didata.local>

I recall certain updates will require you to rename the account back to
administrator.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of param singh
Sent: Wednesday, August 22, 2007 3:31 PM
To: ez4me2c3d at gmail.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Sanity check for changing passwords on the
callmanager 4.1.3

 

I guess that means that none of the Call manager services are effected
by changing the Window's Administration Account. It is just to log into
the  Windows box. My customer wants to change that too.. 

On 8/22/07, Anthony H <ez4me2c3d at gmail.com> wrote:

Note: The Administrator account serves as the default Windows NT
administration account. Cisco CallManager does not use this password.


What is that supposed to mean?  It says CCM does not use this password,
then proceeds to tell you how to change it?  Is this a typo? 


Anthony



On 8/22/07, param singh < paramds at gmail.com <mailto:paramds at gmail.com> >
wrote:

	Hi All, 
	
	I have to change the windows administrator password as well as
the web password for my customer's call manager. 
	
	Following are the steps I am going to follow. Please let me know
if this should work for me or I should do something different. 
	
	This is 4.1(3) call manager and 4.0(4) IPCC. 

	CHANGING PASSWORDS ON CALL MANAGER

	 

	1.	Windows Administrator Passwords on Call manager : 

	 

	 

	Note: The Administrator account serves as the default Windows NT
administration account. Cisco CallManager does not use this password.

	 

	Use any of these methods : 

	 

	*	Choose Start > Program Files > Administrative Tools >
Computer Management > Users, select Administrator Account and right
click Set Password.

	 

	*	The Windows Administrator account password is a login
password into the physical server. You can change the Windows
Administrator password when you press Ctrl+Alt+Delete and click on
Change Password.

	 

	It is necessary to reboot the cluster after you change the
passwords in order to remove any potential problems with Windows
password caching.

	 

	 

	2.	Changing CCMAdministrator Password on the call manage r
: 

	 

	 Since MLA is enabled on the call manager. 

	 Disable MLA and re-enable MLA. Once you do this , it is going
to ask for the new password.  After this is done, restart the IIS on all
the servers.

	 

	 

	I guess this should do it. Also do I need to run "CCMPwdchanger"
or this should be enough. I don't have to change Directory Manager ,
CCMSysuser accounts. Any Ideas. 

	 

	
	
	Thanks 
	Param 
	
	
	

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




-- 
Anthony 

 




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From jason.aarons at us.didata.com  Wed Aug 22 16:07:42 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 22 Aug 2007 16:07:42 -0400
Subject: [cisco-voip] Sanity check for changing passwords on the
	callmanager 4.1.3
In-Reply-To: <867640cd0708221230o18700c20k6cb31a6059c20eda@mail.gmail.com>
References: <867640cd0708221201i67019ca3kce48a2d734bd878f@mail.gmail.com><49921b8d0708221225o121ea215ib27fedd9b67639f5@mail.gmail.com>
	<867640cd0708221230o18700c20k6cb31a6059c20eda@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0068541B5@USNAEXCH.na.didata.local>

Since the SAMAccountName is easily obtainable with common security tools
on the Internet, I would focus on password security versus renaming
accounts.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of param singh
Sent: Wednesday, August 22, 2007 3:31 PM
To: ez4me2c3d at gmail.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Sanity check for changing passwords on the
callmanager 4.1.3

 

I guess that means that none of the Call manager services are effected
by changing the Window's Administration Account. It is just to log into
the  Windows box. My customer wants to change that too.. 

On 8/22/07, Anthony H <ez4me2c3d at gmail.com> wrote:

Note: The Administrator account serves as the default Windows NT
administration account. Cisco CallManager does not use this password.


What is that supposed to mean?  It says CCM does not use this password,
then proceeds to tell you how to change it?  Is this a typo? 


Anthony



On 8/22/07, param singh < paramds at gmail.com <mailto:paramds at gmail.com> >
wrote:

	Hi All, 
	
	I have to change the windows administrator password as well as
the web password for my customer's call manager. 
	
	Following are the steps I am going to follow. Please let me know
if this should work for me or I should do something different. 
	
	This is 4.1(3) call manager and 4.0(4) IPCC. 

	CHANGING PASSWORDS ON CALL MANAGER

	 

	1.	Windows Administrator Passwords on Call manager : 

	 

	 

	Note: The Administrator account serves as the default Windows NT
administration account. Cisco CallManager does not use this password.

	 

	Use any of these methods : 

	 

	*	Choose Start > Program Files > Administrative Tools >
Computer Management > Users, select Administrator Account and right
click Set Password.

	 

	*	The Windows Administrator account password is a login
password into the physical server. You can change the Windows
Administrator password when you press Ctrl+Alt+Delete and click on
Change Password.

	 

	It is necessary to reboot the cluster after you change the
passwords in order to remove any potential problems with Windows
password caching.

	 

	 

	2.	Changing CCMAdministrator Password on the call manage r
: 

	 

	 Since MLA is enabled on the call manager. 

	 Disable MLA and re-enable MLA. Once you do this , it is going
to ask for the new password.  After this is done, restart the IIS on all
the servers.

	 

	 

	I guess this should do it. Also do I need to run "CCMPwdchanger"
or this should be enough. I don't have to change Directory Manager ,
CCMSysuser accounts. Any Ideas. 

	 

	
	
	Thanks 
	Param 
	
	
	

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




-- 
Anthony 

 




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From paramds at gmail.com  Wed Aug 22 16:26:49 2007
From: paramds at gmail.com (param singh)
Date: Wed, 22 Aug 2007 15:26:49 -0500
Subject: [cisco-voip] Sanity check for changing passwords on the
	callmanager 4.1.3
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0068541B2@USNAEXCH.na.didata.local>
References: <867640cd0708221201i67019ca3kce48a2d734bd878f@mail.gmail.com>
	<49921b8d0708221225o121ea215ib27fedd9b67639f5@mail.gmail.com>
	<867640cd0708221230o18700c20k6cb31a6059c20eda@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0068541B2@USNAEXCH.na.didata.local>
Message-ID: <867640cd0708221326w7ab7265cyca78260cdd0d5a08@mail.gmail.com>

Since somebody left their company, they just want to change the
ccmadministrator password and the login password to the windows. Yes, I
guess I have seen a post in this list when someone had a problem changing
the MLA passwords along with the windows login passwords( even if there is
no relation between them) . Just want to accomplish this task without any
problem. Please let me know if anybody had issues while doing these.

thanks
Param

On 8/22/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  I recall certain updates will require you to rename the account back to
> administrator.
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *param singh
> *Sent:* Wednesday, August 22, 2007 3:31 PM
> *To:* ez4me2c3d at gmail.com
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Sanity check for changing passwords on the
> callmanager 4.1.3
>
>
>
> I guess that means that none of the Call manager services are effected by
> changing the Window's Administration Account. It is just to log into the
> Windows box. My customer wants to change that too..
>
> On 8/22/07, *Anthony H* <ez4me2c3d at gmail.com> wrote:
>
> *Note: *The Administrator account serves as the default Windows NT
> administration account. Cisco CallManager does not use this password.
>
>
> What is that supposed to mean?  It says CCM does not use this password,
> then proceeds to tell you how to change it?  Is this a typo?
>
>
> Anthony
>
>   On 8/22/07, *param singh* < paramds at gmail.com> wrote:
>
>  Hi All,
>
> I have to change the windows administrator password as well as the web
> password for my customer's call manager.
>
> Following are the steps I am going to follow. Please let me know if this
> should work for me or I should do something different.
>
> This is 4.1(3) call manager and 4.0(4) IPCC.
>
> CHANGING PASSWORDS ON CALL MANAGER
>
>
>
>    1. *Windows Administrator Passwords on Call manager* :
>
>
>
>
>
> *Note: *The Administrator account serves as the default Windows NT
> administration account. Cisco CallManager does not use this password.
>
>
>
> Use any of these methods :
>
>
>
>    - Choose *Start *> *Program Files *> *Administrative Tools *> *Computer
>    Management *> *Users*, select *Administrator Account *and right
>    click *Set Password*.
>
>
>
>    - The Windows Administrator account password is a login password
>    into the physical server. You can change the Windows Administrator password
>    when you press *Ctrl+Alt+Delete *and click on *Change Password*.
>
>
>
> It is necessary to reboot the cluster after you change the passwords in
> order to remove any potential problems with Windows password caching.
>
>
>
>
>
>    1. *Changing CCMAdministrator Password on the call manage *r :
>
>
>
>  Since MLA is enabled on the call manager.
>
>  Disable MLA and re-enable MLA. Once you do this , it is going to ask for
> the new password.  After this is done, restart the IIS on all the servers.
>
>
>
>
>
> I guess this should do it. Also do I need to run "CCMPwdchanger" or this
> should be enough. I don't have to change Directory Manager , CCMSysuser
> accounts. Any Ideas.
>
>
>
>
>
> Thanks
> Param
>
>
>  _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> --
> Anthony
>
>
>
> ------------------------------
>
> * 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 jsteinberg at gmail.com  Wed Aug 22 16:38:50 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 22 Aug 2007 16:38:50 -0400
Subject: [cisco-voip] problem with phone getting DHCP address from
	voicevlan
In-Reply-To: <8F9F1C6AA1D6834EACC379C8821533A605771DE2@i3mail1.i3domain.inin.com>
References: <mailman.12188.1187811061.1529.cisco-voip@puck.nether.net>
	<8F9F1C6AA1D6834EACC379C8821533A605771DE2@i3mail1.i3domain.inin.com>
Message-ID: <f1b65f880708221338q7637a1b8i58305502ae226c3f@mail.gmail.com>

lelio.

make sure that whatever server is the DHCP server for your DATA vlan
does not have the command below applied to the switchport:

 switchport voice vlan 3104

If the data DHCP server has this command it may be incorrectly
responding to DHCP broadcasts.  Some NIC's in some servers don't work
properly with dot1q tags and may respond to the request.

Justin

On 8/22/07, Ellington, Chris <Chris.Ellington at inin.com> wrote:
> With your setup, you would initially tell the switch to use untagged
> packets and vlan 1220 (I'm assuming this is your data vlan).  When you
> drop that, my _guess_ is that the voice vlan takes over but I don't have
> a switch and a phone handy to test.
>
> The switch voice vlan command is only useful if you are using dot1q
> trunks -or- if you are going to use dot1p for CoS purposes.  Here's a
> good link on it
> http://www.cisco.com/en/US/docs/switches/lan/catalyst3550/software/relea
> se/12.1_12c_ea1/configuration/guide/swvoip.html
>
> chris
>
>
> Message: 2
> Date: Wed, 22 Aug 2007 10:20:17 -0400
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: [cisco-voip] problem with phone getting DHCP address from
>         voicevlan
> To: "Cisco VoIPoE List" <cisco-voip at puck.nether.net>
> Message-ID: <009601c7e4c7$902be890$56196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> We have a number of new phones with a problem whereby they get a DHCP
> address from the data VLAN not the voice VLAN. The only way to make it
> work is to remove the data VLAN, let the phone upgrade and then
> everything works after replacing the data VLAN even after erasing the
> phone configuration.
>
> Here's the specific phone load information:
>
> App Load ID: P003030IMFG2
> Boot Load ID PC0303010200
> Version: 3.1 (MF.G2)
>
> Here's my switch config:
>
> interface FastEthernet1/0/40
>  switchport access vlan 1220                  *this is what I remove to
> make things work*
>  switchport mode access
>  switchport voice vlan 3104
>  srr-queue bandwidth share 10 10 60 20
>  srr-queue bandwidth shape  10  0  0  0
>  mls qos trust device cisco-phone
>  mls qos trust cos
>  auto qos voip cisco-phone
>  spanning-tree portfast
> end
>
> ------------------------------------------------------------------------
> --------
> 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
> -------------- next part --------------
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> https://puck.nether.net/pipermail/cisco-voip/attachments/20070822/9e324c
> 25/attachment-0001.html
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From lelio at uoguelph.ca  Wed Aug 22 16:42:52 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 22 Aug 2007 16:42:52 -0400
Subject: [cisco-voip] problem with phone getting DHCP address
	fromvoicevlan
References: <mailman.12188.1187811061.1529.cisco-voip@puck.nether.net><8F9F1C6AA1D6834EACC379C8821533A605771DE2@i3mail1.i3domain.inin.com>
	<f1b65f880708221338q7637a1b8i58305502ae226c3f@mail.gmail.com>
Message-ID: <011501c7e4fd$02c37050$56196883@cfs.uoguelph.ca>

This was definately one of my thoughts, however, if this was the case, then 
removing the data VLAN on the port I was working wouldn't have fixed the 
problem I'm guessing.

----- Original Message ----- 
From: "Justin Steinberg" <jsteinberg at gmail.com>
To: <lelio at uiguelph.ca>
Cc: <cisco-voip at puck.nether.net>
Sent: Wednesday, August 22, 2007 4:38 PM
Subject: Re: [cisco-voip] problem with phone getting DHCP address 
fromvoicevlan


> lelio.
>
> make sure that whatever server is the DHCP server for your DATA vlan
> does not have the command below applied to the switchport:
>
> switchport voice vlan 3104
>
> If the data DHCP server has this command it may be incorrectly
> responding to DHCP broadcasts.  Some NIC's in some servers don't work
> properly with dot1q tags and may respond to the request.
>
> Justin
>
> On 8/22/07, Ellington, Chris <Chris.Ellington at inin.com> wrote:
>> With your setup, you would initially tell the switch to use untagged
>> packets and vlan 1220 (I'm assuming this is your data vlan).  When you
>> drop that, my _guess_ is that the voice vlan takes over but I don't have
>> a switch and a phone handy to test.
>>
>> The switch voice vlan command is only useful if you are using dot1q
>> trunks -or- if you are going to use dot1p for CoS purposes.  Here's a
>> good link on it
>> http://www.cisco.com/en/US/docs/switches/lan/catalyst3550/software/relea
>> se/12.1_12c_ea1/configuration/guide/swvoip.html
>>
>> chris
>>
>>
>> Message: 2
>> Date: Wed, 22 Aug 2007 10:20:17 -0400
>> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
>> Subject: [cisco-voip] problem with phone getting DHCP address from
>>         voicevlan
>> To: "Cisco VoIPoE List" <cisco-voip at puck.nether.net>
>> Message-ID: <009601c7e4c7$902be890$56196883 at cfs.uoguelph.ca>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> We have a number of new phones with a problem whereby they get a DHCP
>> address from the data VLAN not the voice VLAN. The only way to make it
>> work is to remove the data VLAN, let the phone upgrade and then
>> everything works after replacing the data VLAN even after erasing the
>> phone configuration.
>>
>> Here's the specific phone load information:
>>
>> App Load ID: P003030IMFG2
>> Boot Load ID PC0303010200
>> Version: 3.1 (MF.G2)
>>
>> Here's my switch config:
>>
>> interface FastEthernet1/0/40
>>  switchport access vlan 1220                  *this is what I remove to
>> make things work*
>>  switchport mode access
>>  switchport voice vlan 3104
>>  srr-queue bandwidth share 10 10 60 20
>>  srr-queue bandwidth shape  10  0  0  0
>>  mls qos trust device cisco-phone
>>  mls qos trust cos
>>  auto qos voip cisco-phone
>>  spanning-tree portfast
>> end
>>
>> ------------------------------------------------------------------------
>> --------
>> 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
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL:
>> https://puck.nether.net/pipermail/cisco-voip/attachments/20070822/9e324c
>> 25/attachment-0001.html
>>
>>
>> _______________________________________________
>> 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 bms314 at gmail.com  Wed Aug 22 17:17:00 2007
From: bms314 at gmail.com (Brian)
Date: Wed, 22 Aug 2007 16:17:00 -0500
Subject: [cisco-voip] No MWI - QSIG trunk between CCM and Avaya S8700 with
	Unity VM
Message-ID: <7aaaf7f0708221417i18b0cf3ftdc6c43f13c74df3f@mail.gmail.com>

We have an environment with CCM 5.1(2)b and an Avaya S8700 connected via
QSIG PRI's.  We would like to convert some of the Avaya sets to a Unity
voicemail box.  I have successfully been able to leave and retrieve
voicemail's on the Avaya phones that have a Unity VM.  However, I am unable
to get MWI to work.  In the Avaya world, *50XXXX turns on MWI and #50XXXX
turns it off.  I've created route patterns to point to the Avaya, but can
not get the MWI to light.

Does anyone have a similar environment where this is working?

Thanks,
Brian
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From jason.aarons at us.didata.com  Wed Aug 22 20:35:28 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 22 Aug 2007 20:35:28 -0400
Subject: [cisco-voip] Unity listening to message being left
Message-ID: <C1FE15183DA37645BC0633BC604E44F006854439@USNAEXCH.na.didata.local>

I understand in Unity 4.2 you can listen to someone leaving you a
voicemail, if you like what they are saying you can pickup the call?
Allows you to screen calls from pest salespeople...

 

I don't see where in the guides how to use this or configure Unity call
monitoring.

 

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_administr
ation_guide_book09186a008043ea39.html

 

 




-----------------------------------------
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 jsteinberg at gmail.com  Wed Aug 22 20:59:23 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 22 Aug 2007 20:59:23 -0400
Subject: [cisco-voip] Unity listening to message being left
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006854439@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F006854439@USNAEXCH.na.didata.local>
Message-ID: <f1b65f880708221759j2f67a447g7dafc60ed8656294@mail.gmail.com>

The feature is called message monitor and it was first look in 4.2.

Looks like they pulled the PDFs from ciscounitytools.com but you can
still download the videos and check them out.

http://www.ciscounitytools.com/TOI_Unity42.htm



On 8/22/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> I understand in Unity 4.2 you can listen to someone leaving you a voicemail,
> if you like what they are saying you can pickup the call? Allows you to
> screen calls from pest salespeople?
>
>
>
> I don't see where in the guides how to use this or configure Unity call
> monitoring.
>
>
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_administration_guide_book09186a008043ea39.html
>
>
>
>
>
>
> ________________________________
>
>
>
>
>  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 lelio at uoguelph.ca  Wed Aug 22 21:02:20 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 22 Aug 2007 21:02:20 -0400
Subject: [cisco-voip] Unity listening to message being left
References: <C1FE15183DA37645BC0633BC604E44F006854439@USNAEXCH.na.didata.local>
Message-ID: <002101c7e521$425d22f0$6601a8c0@pdp11>

I believe the "message monitor" feature you are looking for is not fully supported until v5 but it is available as a "first look" in v4.2.

Take a look here and do a CTRL-F for "message monitor" I believe it's the last one.


http://www.ciscounitytools.com/TOI_Unity42.htm


  ----- Original Message ----- 
  From: Jason Aarons (US) 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, August 22, 2007 8:35 PM
  Subject: [cisco-voip] Unity listening to message being left


  I understand in Unity 4.2 you can listen to someone leaving you a voicemail, if you like what they are saying you can pickup the call? Allows you to screen calls from pest salespeople.

   

  I don't see where in the guides how to use this or configure Unity call monitoring.

   

  http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_administration_guide_book09186a008043ea39.html

   

   



------------------------------------------------------------------------------


  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 lists at jerkys.org  Wed Aug 22 22:47:31 2007
From: lists at jerkys.org (Jerky)
Date: Wed, 22 Aug 2007 21:47:31 -0500
Subject: [cisco-voip] CCM 5.1 appliance license
Message-ID: <2F94AEFA-A971-4849-B2D4-8DF72C4AB863@jerkys.org>

I recently upgraded to 5.x so the URL is still fresh in my mind. The  
tool you are looking for can be found at:

https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet? 
FormId=806

As was mentioned before you'll need your Publisher's MAC address and  
obviously a valid CCO account.

jerky

 >You should have a PAK key on the box the software came in.  You >would
 >then use this to access the licensing web page on CCO, along >with  
your
 >server's mac address.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of cisco.voip
Sent: Wednesday, August 22, 2007 9:07 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM 5.1 appliance license

Can anyone point me to or send me a link on how to enable the license on
my
new Cisco 5.1 appliance server.
It it asking for a license file, but all I have in the box are the
license
cards with the serial numbers on it?

Where do I enter the license number or am I missing something in the
box?

Thanks!
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From MLinsemier at apcapital.com  Thu Aug 23 03:36:18 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Thu, 23 Aug 2007 03:36:18 -0400
Subject: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE after
	installing OS 4.5a SR2
Message-ID: <AB0A35F2486F81468971AA68558BE71906D9A4AD@smieln01winm01.micoanet.com>

All,

 

This evening while installing the latest OS service pack from Cisco (OS
4.5a SR2) and restarting the CCM server (publisher) I am now greeted
with a BSOD displaying Stop 0x0000007B INACCESABLE_BOOT_DEVICE.  This is
an IBM 7815-I1 server with a single SATA drive.  I didn't do anything
different then I normally do when I patch servers and have never had a
problem before this one.  I have tried booting using the last known good
configuration, safe mode with network, safe mode, and command prompt.  I
booted with a Windows 2000 Server CD and ran the recovery console and
proceeded to run a CHKDSK /P.  The first time through it got to 50%
before stopping and stating that "one or more unrecoverable errors have
been discovered".  The second time I ran CHKDSK it only got to 25%
before displaying the same message.  Is it conceivable that I lost the
drive on a reboot and patching the box was just a coincidence?

 

I have a recent BARS backup (8/22/07) so I am pretty confident that I
can restore the server (even though I have never actually done this
before).  Is it difficult to do so?  I am running CCM 4.1(3) with IPCC
3.5(1) co-resident.  Any links or experiences (or ideas to get back at
my data) would be appreciated.

 

Matt 


CONFIDENTIALITY STATEMENT
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From CarlosOrtiz at bayviewfinancial.com  Thu Aug 23 09:22:08 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 23 Aug 2007 09:22:08 -0400
Subject: [cisco-voip] 7961 long reset
In-Reply-To: <5BFD4B266FCBED44938AE3BB35ACD0AA02F3132E@UVEXVS01.bc.broward.cty>
Message-ID: <OF7F599DD6.5C310B3B-ON85257340.004848BD-85257340.00497023@bftg.com>

Just my phone.  I am updating the CSS of the line which causes the phone 
to reset.  I don't remember it taking this long even when I first 
installed the phone.  I just reset it and it took 1 minute to come back. 
Anyone seen this before?

 



"Socarras, Luis" <LSOCARRAS at broward.org> 
08/23/2007 08:53 AM

To
<CarlosOrtiz at bayviewfinancial.com>
cc

Subject
RE: [cisco-voip] 7961 long reset






Our load file is SCCP41.8-2-1S.  When you say a delay of about a minute, 
are you resetting one phone at a time or are you selecting a group from 
the results of a search on the Find and List Phone Page. 
 
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Wednesday, August 22, 2007 3:56 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7961 long reset
 

When updating lines on some new 7961 phones I am seeing a large delay (1 
minute) for the phone to reset.   Is this normal or am I possibly hitting 
a bug? 

Load file is  SCCP41.8-2-2SR1S.
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From Johnny.Crothers at vanco.co.uk  Thu Aug 23 09:54:15 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers (c))
Date: Thu, 23 Aug 2007 14:54:15 +0100
Subject: [cisco-voip] 7961 long reset
In-Reply-To: <OF7F599DD6.5C310B3B-ON85257340.004848BD-85257340.00497023@bftg.com>
Message-ID: <A56BD242744BF540B15D1672E0E3E87501B271AC@LON-INT-MSGVS01.internal.vancocorp.com>

Is it just your phone? Are all phones on the same switch?

 

Which step is taking the longest? 

 

It might be there is some sort of network problem, maybe delays in DHCP
or TFTP?

 

You might need to sniff the traffic and see what problems there are.

 

Johnny Crothers

IPT Specialist

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: 23 August 2007 14:22
To: Socarras, Luis
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7961 long reset

 


Just my phone.  I am updating the CSS of the line which causes the phone
to reset.  I don't remember it taking this long even when I first
installed the phone.  I just reset it and it took 1 minute to come back.
Anyone seen this before? 

  



"Socarras, Luis" <LSOCARRAS at broward.org> 

08/23/2007 08:53 AM 

To

<CarlosOrtiz at bayviewfinancial.com> 

cc

 

Subject

RE: [cisco-voip] 7961 long reset

 

 

 




Our load file is SCCP41.8-2-1S.  When you say a delay of about a minute,
are you resetting one phone at a time or are you selecting a group from
the results of a search on the Find and List Phone Page.   
  
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Wednesday, August 22, 2007 3:56 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7961 long reset 
  

When updating lines on some new 7961 phones I am seeing a large delay (1
minute) for the phone to reset.   Is this normal or am I possibly
hitting a bug? 

Load file is  SCCP41.8-2-2SR1S. 



**********************************************************************
Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient.  If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

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The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

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From WOCAMPO at broward.org  Thu Aug 23 10:00:23 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Thu, 23 Aug 2007 10:00:23 -0400
Subject: [cisco-voip] 7961 long reset
In-Reply-To: <A56BD242744BF540B15D1672E0E3E87501B271AC@LON-INT-MSGVS01.internal.vancocorp.com>
References: <OF7F599DD6.5C310B3B-ON85257340.004848BD-85257340.00497023@bftg.com>
	<A56BD242744BF540B15D1672E0E3E87501B271AC@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA4A9@UVEXVS01.bc.broward.cty>

Johnny is right,

 

It is simple to detect, 

In the 7961, check the network setting and check the IP address of the
DHCP server as well the TFTP servers IP address. Reset the phone and
check during the reset process which is taking the most time, the DHCP
or the TFP loading on the phone. From there you could start to
troubleshoot.

 

 

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

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johnny Crothers
(c)
Sent: Thursday, August 23, 2007 9:54 AM
To: CarlosOrtiz at bayviewfinancial.com; Socarras, Luis
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7961 long reset

 

Is it just your phone? Are all phones on the same switch?

 

Which step is taking the longest? 

 

It might be there is some sort of network problem, maybe delays in DHCP
or TFTP?

 

You might need to sniff the traffic and see what problems there are.

 

Johnny Crothers

IPT Specialist

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: 23 August 2007 14:22
To: Socarras, Luis
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7961 long reset

 


Just my phone.  I am updating the CSS of the line which causes the phone
to reset.  I don't remember it taking this long even when I first
installed the phone.  I just reset it and it took 1 minute to come back.
Anyone seen this before? 

  

"Socarras, Luis" <LSOCARRAS at broward.org> 

08/23/2007 08:53 AM 

To

<CarlosOrtiz at bayviewfinancial.com> 

cc

 

Subject

RE: [cisco-voip] 7961 long reset

 

 

 




Our load file is SCCP41.8-2-1S.  When you say a delay of about a minute,
are you resetting one phone at a time or are you selecting a group from
the results of a search on the Find and List Phone Page.   
  
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Wednesday, August 22, 2007 3:56 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7961 long reset 
  

When updating lines on some new 7961 phones I am seeing a large delay (1
minute) for the phone to reset.   Is this normal or am I possibly
hitting a bug? 

Load file is  SCCP41.8-2-2SR1S. 



**********************************************************************
Any opinions expressed in the email are those of the individual and not
necessarily the company. This email and any files transmitted with it
are confidential and solely for the use of the intended recipient. If
you are not the intended recipient or the person responsible for
delivering it to the intended recipient, be advised that you have
received this email in error and that any dissemination, distribution,
copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with
the content of this email please e-mail to:
e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software
viruses which could damage your own computer system. While the sender
has taken every reasonable precaution to minimise this risk, we cannot
accept liability for any damage which you sustain as a result of
software viruses. You should carry out your own virus checks before
opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John
Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
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From wsisk at cisco.com  Thu Aug 23 10:22:31 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 23 Aug 2007 10:22:31 -0400
Subject: [cisco-voip] No MWI - QSIG trunk between CCM and Avaya S8700
 with	Unity VM
In-Reply-To: <7aaaf7f0708221417i18b0cf3ftdc6c43f13c74df3f@mail.gmail.com>
References: <7aaaf7f0708221417i18b0cf3ftdc6c43f13c74df3f@mail.gmail.com>
Message-ID: <46CD9827.2080609@cisco.com>

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From cpennington at ncaa.org  Thu Aug 23 10:10:01 2007
From: cpennington at ncaa.org (Pennington, Craig)
Date: Thu, 23 Aug 2007 10:10:01 -0400
Subject: [cisco-voip] How to change the corp. directory to search multiple
	CCM clusters
Message-ID: <B0353BDEA8A5A14D8E3CA39AADCA46B606A00E9F@NCAA-MAIL.ncaa.org>

Has anyone figured out how to modify the corporate directory on the
phones to look up results in multiple (in our case 2) CCM clusters and
display all of the results?  We have just implemented a second cluster
at another site.  I don't want to mess with active directory
integration.  We tracked the code down and it appears to hit a .dll file
for the actual finding of results so there is no web page code to
modify.  Does anyone have any ideas?  I heard one suggestion to
implement a separate ldap server and have it sync with both clusters
then have the lookup go to it but I think that would still involve a
re-write of the corp. directory code.  Has anyone tried it?
 
Craig Pennington
Network Engineer
National Collegiate Athletic Association
Fax: 317-917-6888
cpennington at ncaa.org <mailto:cpennington at ncaa.org> 
www.ncaa.org <http://www.ncaa.org/> 

This email and any attachments may contain confidential and privileged information.  If you are not the intended recipient, please notify the sender immediately by return email, delete this message and destroy any copies.  Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.
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From wsisk at cisco.com  Thu Aug 23 11:19:59 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 23 Aug 2007 11:19:59 -0400
Subject: [cisco-voip] How to change the corp. directory to search
 multiple CCM clusters
In-Reply-To: <B0353BDEA8A5A14D8E3CA39AADCA46B606A00E9F@NCAA-MAIL.ncaa.org>
References: <B0353BDEA8A5A14D8E3CA39AADCA46B606A00E9F@NCAA-MAIL.ncaa.org>
Message-ID: <46CDA59F.4050105@cisco.com>

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From gnasses at tcfbank.com  Thu Aug 23 10:45:06 2007
From: gnasses at tcfbank.com (Nasses, Gus)
Date: Thu, 23 Aug 2007 09:45:06 -0500
Subject: [cisco-voip] Call redirection using alias command in event of WAN
	failure
Message-ID: <200708231521.l7NFLZNM003220@puck.nether.net>

Hello List, 

 

I am looking to setup a call redirection for inbound DID survivability in a
office with remote VoIP gateway using the alias command, and previous testing
did not seem to work. 

 

I have office A with a PRI full of DIDs terminating at a 2821 and WAN
connection to Office B (head site) where the call Managers live. Office A
phones register to CM at office B. Office C also has a WAN to Office B and
registers its phones there. Office C gets its DIDs from the PRI at Office A.
This all works fine, but we want to eliminate any busy tones if WAN links die
during office hours. If office C loses its WAN, then calls will go to unity
from the CM at Office B or can be redirected from there, which is fine. 

 

Problem is, If Office A loses its WAN, SRST kicks in and they can still
receive calls, but then the DIDs for Office C go to busy. We want to redirect
these DID calls to another extension at Office A until the WAN comes back up,
so the calls can be answered by another user.  Here is how we had it
configured (on the Office A 2821) and it did not seem to work. (1000 and 1002
are extensions at Office A, and 1008-1012 are extensions at Office C)

 

call-manager-fallback 

 timeouts interdigit 3

 ip source-address 172.21.1.1 port 2000 strict-match

 max-ephones 144

 max-dn 288

 default-destination 1000

 access-code pri 9 direct-inward-dial

 alias 1 1008 to 1002

 alias 1 1009 to 1000

 alias 1 1010 to 1000

 alias 1 1012 to 1002

 

 

Is there something else I am missing? Do I need to somehow tell the SRST
config something more about the inbound DIDs since it has lost connection to
CM? 

 

TIA, 

 

Gus Nasses

gnasses at yahoo.com

 

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From bms314 at gmail.com  Thu Aug 23 11:44:55 2007
From: bms314 at gmail.com (Brian)
Date: Thu, 23 Aug 2007 10:44:55 -0500
Subject: [cisco-voip] No MWI - QSIG trunk between CCM and Avaya S8700
	with Unity VM
In-Reply-To: <46CD9827.2080609@cisco.com>
References: <7aaaf7f0708221417i18b0cf3ftdc6c43f13c74df3f@mail.gmail.com>
	<46CD9827.2080609@cisco.com>
Message-ID: <7aaaf7f0708230844t47ce56f3k14f842e2de76c96e@mail.gmail.com>

In Unity, 71098 MWI on & 71099 MWI off.

I need to clarify our setup.  We have 2 CallManager clusters and an Avaya
s8700.

It goes Unity>CCM_51>Intercluster Trunk>CCM_41>QSIG>Avaya s8700.

How does this configuration impact the CSS's?

CCM_51 - Has MWI_CSS, with the CCM_51 device partitions and the partition
that the route patterns (using the intercluster trunk) are in.

CCM_41 - Has MWI_CSS, with the CCM_41 device partitions and the partition
that the route patterns (using the QSIG trunks) are in.

Shouldn't we be able to dial the Avaya's MWI on code (*50) followed by an
Avaya phone to light MWI, all from a phone on the CCM_41 cluster?

Thanks.
Brian

On 8/23/07, Wes Sisk <wsisk at cisco.com> wrote:
>
>  In unity what DN is configured for message waiting?
>
> QSIG MWI works like this with Unity/CM:
>
> Unity MWI port (high port usually) goes offhook and specifies calling
> party number = the DN to light MWI for and called party = MWI on number in
> CM.
> When the call reaches the MWI on number CM recognizes this as an MWI
> request and converts the SCCP call to a QSIG call with necessary APDUs.
> Effectively this alters the call so the calling device is the MWI on number
> and the called party number is the DN to light MWI for.
>
> At this point the MWI CSS must contain a route pattern that points to a
> QSIG enabled route list that points the gateway and trunk that provides the
> QSIG link between CM and PBX.
>
>
> Net result - are your DN's, CSS, and route patterns correct?
>
> /Wes
>
> Brian wrote:
>
> We have an environment with CCM 5.1(2)b and an Avaya S8700 connected via
> QSIG PRI's.  We would like to convert some of the Avaya sets to a Unity
> voicemail box.  I have successfully been able to leave and retrieve
> voicemail's on the Avaya phones that have a Unity VM.  However, I am unable
> to get MWI to work.  In the Avaya world, *50XXXX turns on MWI and #50XXXX
> turns it off.  I've created route patterns to point to the Avaya, but can
> not get the MWI to light.
>
> Does anyone have a similar environment where this is working?
>
> Thanks,
> Brian
>
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From svoll.voip at gmail.com  Thu Aug 23 11:49:01 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 23 Aug 2007 08:49:01 -0700
Subject: [cisco-voip] RTMT Media list exhausted
Message-ID: <f84a38d30708230849j79ec4e25q48bda9d8d0c9068c@mail.gmail.com>

I keep getting these alarts from RTMT


*At 08:32:46 on 08/23/2007 on cluster CMPUB-Cluster.*

*Number of MediaListExhausted events exceed 3 within 60 minutes.*

I think it coralates to calls transfering out of Unity.  Could that be the
cause? and if so, how do I fix it?

Thanks

Scott
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From wsisk at cisco.com  Thu Aug 23 11:52:10 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 23 Aug 2007 11:52:10 -0400
Subject: [cisco-voip] No MWI - QSIG trunk between CCM and Avaya S8700
 with Unity VM
In-Reply-To: <7aaaf7f0708230844t47ce56f3k14f842e2de76c96e@mail.gmail.com>
References: <7aaaf7f0708221417i18b0cf3ftdc6c43f13c74df3f@mail.gmail.com>	
	<46CD9827.2080609@cisco.com>
	<7aaaf7f0708230844t47ce56f3k14f842e2de76c96e@mail.gmail.com>
Message-ID: <46CDAD2A.2080209@cisco.com>

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From wsisk at cisco.com  Thu Aug 23 11:52:53 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 23 Aug 2007 11:52:53 -0400
Subject: [cisco-voip] RTMT Media list exhausted
In-Reply-To: <f84a38d30708230849j79ec4e25q48bda9d8d0c9068c@mail.gmail.com>
References: <f84a38d30708230849j79ec4e25q48bda9d8d0c9068c@mail.gmail.com>
Message-ID: <46CDAD55.5020107@cisco.com>

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From dwfielder at gmail.com  Thu Aug 23 12:22:40 2007
From: dwfielder at gmail.com (David Fielder)
Date: Thu, 23 Aug 2007 12:22:40 -0400
Subject: [cisco-voip] External Meetingplace Web Conferencing
Message-ID: <001401c7e5a1$d431fe50$673ea8c0@na.didata.local>

I have recently added an External MeetingPlace Web server for a client. I have configured a NAT to the outside for the Server but I am not using one from the DMZ to the inside. I am able to connect from the inside to the external web page but not from the outside. The web page resolves to the NATed address. Cisco TAC told me that NAT was not supported but the documentation says that NAT is not supported between the Audio server and MP components. However, I would think that I would be able to NAT to the outside. Any help would be much appreciated.


David Fielder
Sr. Network Engineer
Dimension Data NA

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From WOCAMPO at broward.org  Thu Aug 23 13:04:25 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Thu, 23 Aug 2007 13:04:25 -0400
Subject: [cisco-voip] RTMT Media list exhausted
In-Reply-To: <46CDAD55.5020107@cisco.com>
References: <f84a38d30708230849j79ec4e25q48bda9d8d0c9068c@mail.gmail.com>
	<46CDAD55.5020107@cisco.com>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA4AB@UVEXVS01.bc.broward.cty>

After you check the alarm details, go to the call manager serviceability
under alarm definitions type the alarm details that you obtained, it
will tell you the possibilities  cause of the alarm. Depending of the
media list exhausted  name, check the dependency records of it. See if
it is been used by many users. 

 

 

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

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Thursday, August 23, 2007 11:53 AM
To: Scott Voll
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RTMT Media list exhausted

 

look at the details of the alarm in RTMT, it will show you the list
name.  Work from there to see what devices use that list and what its'
purpose is.

/Wes

Scott Voll wrote: 

I keep getting these alarts from RTMT

 

At 08:32:46 on 08/23/2007 on cluster CMPUB-Cluster.

Number of MediaListExhausted events exceed 3 within 60 minutes.

I think it coralates to calls transfering out of Unity.  Could that be
the cause? and if so, how do I fix it?

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 d4rkf1ber at gmail.com  Thu Aug 23 15:03:16 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Thu, 23 Aug 2007 14:03:16 -0500
Subject: [cisco-voip] CallManager OS Confusion?
Message-ID: <a1b07ea60708231203ga8cd65ct7d18a947a55a378b@mail.gmail.com>

I have two callmanagers (publisher and subscriber) that are both running the
following OS:

2000.4.2sr16

And the CallManager version is 4.1(3)sr4d

I also have two IPCC Express Premium servers running OS:

2000.4.3aSR4

And IPCC version 4.0(5)_Build042

When I go to look for OS upgrades on the Cisco site I see version numbers
that are all over the place.

2000.4.4a sr10 (Aug. 21 2007)
2000.4.5a sr2 (Aug. 21 2007)
2003.1.1 sr7 (Aug. 21 2007)
2003.1.2a sr2 (Aug. 21 2007)


I recognize that two of them are for Windows Server 2000 and two of them are
for Windows Server 2003.

I assume if I want / have to stay with 2000 for now I would have to choose
from either:

 2000.4.4a sr10 (Aug. 21 2007)
2000.4.5a sr2 (Aug. 21 2007)

>From the version numbers you would think one is more up to date then the
other since .5 is higher then .4, but since the dates are the same it would
seem that they are both equal in terms of being "up to date".

Anyone want to shed some light on this?  I have to admit I am being lazy,
and honestly don't want to have to read through 4 different readme's to
understand the pro's and con's of them all.

I want to get things up to date and secure, but I am also trying to avoid
any sort of major upgrade right now.  I would love to have one consistent OS
across the four servers if possible but I suppose there are alot of other
factors that will come into play.

Anyone have any thoughts regarding this?  Becoming quite the pain to keep
track of the various versions, and now I have to bring in a couple of CER
servers too so I am trying to get a handle on all of this before I begin.

Thanks!
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From lists at jerkys.org  Wed Aug 22 22:47:31 2007
From: lists at jerkys.org (Jerky)
Date: Wed, 22 Aug 2007 21:47:31 -0500
Subject: [cisco-voip] CCM 5.1 appliance license
Message-ID: <2F94AEFA-A971-4849-B2D4-8DF72C4AB863@jerkys.org>

I recently upgraded to 5.x so the URL is still fresh in my mind. The  
tool you are looking for can be found at:

https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet? 
FormId=806

As was mentioned before you'll need your Publisher's MAC address and  
obviously a valid CCO account.

jerky

 >You should have a PAK key on the box the software came in.  You >would
 >then use this to access the licensing web page on CCO, along >with  
your
 >server's mac address.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of cisco.voip
Sent: Wednesday, August 22, 2007 9:07 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM 5.1 appliance license

Can anyone point me to or send me a link on how to enable the license on
my
new Cisco 5.1 appliance server.
It it asking for a license file, but all I have in the box are the
license
cards with the serial numbers on it?

Where do I enter the license number or am I missing something in the
box?

Thanks!
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From VLoschiavo at data-corporation.com  Thu Aug 23 14:48:12 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Thu, 23 Aug 2007 14:48:12 -0400
Subject: [cisco-voip] External Meetingplace Web Conferencing
In-Reply-To: <001401c7e5a1$d431fe50$673ea8c0@na.didata.local>
References: <001401c7e5a1$d431fe50$673ea8c0@na.didata.local>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF571356FE13@MAIL02.hq.data-corporation.com>

Meetingplace Express?

 

Make sure you are using both NIC's on the MCS server....with different
IPs.

 

The reason is:  the same TCP port is used on both IPs for different
services.

 

____________________________________________________________
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 David Fielder
Sent: Thursday, August 23, 2007 12:23 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] External Meetingplace Web Conferencing

 

I have recently added an External MeetingPlace Web server for a client.
I have configured a NAT to the outside for the Server but I am not using
one from the DMZ to the inside. I am able to connect from the inside to
the external web page but not from the outside. The web page resolves to
the NATed address. Cisco TAC told me that NAT was not supported but the
documentation says that NAT is not supported between the Audio server
and MP components. However, I would think that I would be able to NAT to
the outside. Any help would be much appreciated.

 

 

David Fielder

Sr. Network Engineer

Dimension Data NA

 

 

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From svoll.voip at gmail.com  Thu Aug 23 16:20:42 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 23 Aug 2007 13:20:42 -0700
Subject: [cisco-voip] CallManager OS Confusion?
In-Reply-To: <a1b07ea60708231203ga8cd65ct7d18a947a55a378b@mail.gmail.com>
References: <a1b07ea60708231203ga8cd65ct7d18a947a55a378b@mail.gmail.com>
Message-ID: <f84a38d30708231320r2556519aydeaf43820bdfc94f@mail.gmail.com>

4.2, 4.3, 4.4, etc are the train of OS (major upgrade)  the SR1, sr2, sr3b
is the minor upgrade.

you will need to read the readme to make sure all hardware is supported but
you should be able to use the newest windows 2000 major (major upgrade 2-3
reboots) and then install the minor on all your CM, IPCCx, CER, etc. (single
reboot) (YMMV)

win 2003 is just for CM 4.3 as that runs on win2k3.

Hope that clears things up.  (READ THE READMEs) ;-)

Scott


On 8/23/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
>
> I have two callmanagers (publisher and subscriber) that are both running
> the following OS:
>
> 2000.4.2sr16
>
> And the CallManager version is 4.1(3)sr4d
>
> I also have two IPCC Express Premium servers running OS:
>
> 2000.4.3aSR4
>
> And IPCC version 4.0(5)_Build042
>
> When I go to look for OS upgrades on the Cisco site I see version numbers
> that are all over the place.
>
> 2000.4.4a sr10 (Aug. 21 2007)
> 2000.4.5a sr2 (Aug. 21 2007)
> 2003.1.1 sr7 (Aug. 21 2007)
> 2003.1.2a sr2 (Aug. 21 2007)
>
>
> I recognize that two of them are for Windows Server 2000 and two of them
> are for Windows Server 2003.
>
> I assume if I want / have to stay with 2000 for now I would have to choose
> from either:
>
>  2000.4.4a sr10 (Aug. 21 2007)
> 2000.4.5a sr2 (Aug. 21 2007)
>
> From the version numbers you would think one is more up to date then the
> other since .5 is higher then .4, but since the dates are the same it would
> seem that they are both equal in terms of being "up to date".
>
> Anyone want to shed some light on this?  I have to admit I am being lazy,
> and honestly don't want to have to read through 4 different readme's to
> understand the pro's and con's of them all.
>
> I want to get things up to date and secure, but I am also trying to avoid
> any sort of major upgrade right now.  I would love to have one consistent OS
> across the four servers if possible but I suppose there are alot of other
> factors that will come into play.
>
> Anyone have any thoughts regarding this?  Becoming quite the pain to keep
> track of the various versions, and now I have to bring in a couple of CER
> servers too so I am trying to get a handle on all of this before I begin.
>
> Thanks!
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From malhatma at gmx.net  Thu Aug 23 16:22:12 2007
From: malhatma at gmx.net (malhatma)
Date: Thu, 23 Aug 2007 22:22:12 +0200
Subject: [cisco-voip] CME Shared Line with SIP Phone and SCCP Ephone
Message-ID: <46CDEC74.3010406@gmx.net>

Hi List,

I try to understand how to setup a shared line with a Cisco 7960 (reg. via SCCP) and a Nokia E60 Sip Phone.
Calling in general works fine. The setup is as follows:

voice register dn  2
  number 53
  name ne60
  no-reg
  label NE60

voice register pool  2
  id mac 0000.1111.2222
  number 1 dn 2
  username ne60 password xxxxx
  codec g711alaw

ephone-dn  1  dual-line
  number 53 no-reg primary
  label 53
  call-forward busy 41
  call-forward noan 41 timeout 17
  huntstop channel
  no huntstop

When I call the 53 from another phone, only the SCCP phone rings.
"Show dialplan number" reveals the problem:

test_gw# sh dialplan number 53 | i Peer|destination|Matched
VoiceEncapPeer20001
         tag = 20001, destination-pattern = `53$',
Matched: 53   Digits: 3
VoiceOverIpPeer40001
         tag = 40001, destination-pattern = `53',
Matched: 53   Digits: 2
test_gw#

The $ seems to be interpreted as an additional match.
Well, ok. You can configure the SIP dn number as follows:

voice register dn  2
  number 53$

"sh run" looks as pasted below:

test_gw#sh run | s voice register dn  2
voice register dn  2
  number 53$
  name ne60
  no-reg
  label NE60

So far so good. But "sh dialplan numer" still shows:

test_gw# sh dialplan number 53 | i Peer|destination|Matched
VoiceEncapPeer20001
         tag = 20001, destination-pattern = `53$',
Matched: 53   Digits: 3
VoiceOverIpPeer40001
         tag = 40001, destination-pattern = `53',
Matched: 53   Digits: 2
test_gw#

And, sure enough, only the SCCP phone rings. I?m slowly running out of ideas.
Hunt Groups are no alternative, cause there is no broadcast feature withe CME.

Any hints?

Regards
Andre



From dwfielder at gmail.com  Thu Aug 23 16:28:39 2007
From: dwfielder at gmail.com (David Fielder)
Date: Thu, 23 Aug 2007 16:28:39 -0400
Subject: [cisco-voip] External Meetingplace Web Conferencing
References: <001401c7e5a1$d431fe50$673ea8c0@na.didata.local>
	<CB4BC8C24EF6E149AED9FDB9B9AF571356FE13@MAIL02.hq.data-corporation.com>
Message-ID: <001001c7e5c4$36e163c0$673ea8c0@na.didata.local>

I'm not using MP express, does using both NICs still apply?
  ----- Original Message ----- 
  From: Vince Loschiavo 
  To: David Fielder ; cisco-voip at puck.nether.net 
  Sent: Thursday, August 23, 2007 2:48 PM
  Subject: RE: [cisco-voip] External Meetingplace Web Conferencing


  Meetingplace Express?

   

  Make sure you are using both NIC's on the MCS server..with different IPs.

   

  The reason is:  the same TCP port is used on both IPs for different services.

   

  ____________________________________________________________
  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 David Fielder
  Sent: Thursday, August 23, 2007 12:23 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] External Meetingplace Web Conferencing

   

  I have recently added an External MeetingPlace Web server for a client. I have configured a NAT to the outside for the Server but I am not using one from the DMZ to the inside. I am able to connect from the inside to the external web page but not from the outside. The web page resolves to the NATed address. Cisco TAC told me that NAT was not supported but the documentation says that NAT is not supported between the Audio server and MP components. However, I would think that I would be able to NAT to the outside. Any help would be much appreciated.

   

   

  David Fielder

  Sr. Network Engineer

  Dimension Data NA

   

   
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From jonvoip at gmail.com  Thu Aug 23 18:10:16 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 23 Aug 2007 17:10:16 -0500
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
Message-ID: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>

Has anyone done it successfully?

I have a LOT of customers on Unity 4.X running Windows 2000 who want
to upgrade to Unity 5 and Windows 2003. However, my first attempt was
a complete disaster (ended up having to install from scratch).

Here is the problem:

You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity is
installed (says so in the upgrade documentation).

So, you must do a DRT backup, install Windows 2003 clean, install your
old version of Unity 4.X and restore it. Then you can upgrade to 5

This is not even slightly as simple as it sounds and has a remarkably
high failure rate.

(we got an error that the hostname didn't match and the upgrade to unity failed)

TAC refused to support the upgrade (this was early July, it had come
off of NPH that day).

If anyone has, I am curious as to what tricks you did to make it work...



Jonathan

From aizhar at ccbootcamp.com  Thu Aug 23 18:18:31 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Thu, 23 Aug 2007 15:18:31 -0700
Subject: [cisco-voip] Call redirection using alias command in event of
	WANfailure
References: <200708231521.l7NFLZNM003220@puck.nether.net>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663561168@nlimail.nlidomain>

Hi,

Are you getting four digits from the PSTN at site A?

Perhaps there is no match to the 1008-1012 range and therefor the alias does not kick in.

-------------------------------------------
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 Nasses, Gus
Sent: Thu 8/23/2007 7:45 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call redirection using alias command in event of WANfailure [bcc][faked-from][html-rem]
 
Hello List, 

 

I am looking to setup a call redirection for inbound DID survivability in a
office with remote VoIP gateway using the alias command, and previous testing
did not seem to work. 

 

I have office A with a PRI full of DIDs terminating at a 2821 and WAN
connection to Office B (head site) where the call Managers live. Office A
phones register to CM at office B. Office C also has a WAN to Office B and
registers its phones there. Office C gets its DIDs from the PRI at Office A.
This all works fine, but we want to eliminate any busy tones if WAN links die
during office hours. If office C loses its WAN, then calls will go to unity
from the CM at Office B or can be redirected from there, which is fine. 

 

Problem is, If Office A loses its WAN, SRST kicks in and they can still
receive calls, but then the DIDs for Office C go to busy. We want to redirect
these DID calls to another extension at Office A until the WAN comes back up,
so the calls can be answered by another user.  Here is how we had it
configured (on the Office A 2821) and it did not seem to work. (1000 and 1002
are extensions at Office A, and 1008-1012 are extensions at Office C)

 

call-manager-fallback 

 timeouts interdigit 3

 ip source-address 172.21.1.1 port 2000 strict-match

 max-ephones 144

 max-dn 288

 default-destination 1000

 access-code pri 9 direct-inward-dial

 alias 1 1008 to 1002

 alias 1 1009 to 1000

 alias 1 1010 to 1000

 alias 1 1012 to 1002

 

 

Is there something else I am missing? Do I need to somehow tell the SRST
config something more about the inbound DIDs since it has lost connection to
CM? 

 

TIA, 

 

Gus Nasses

gnasses at 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 jason.aarons at us.didata.com  Thu Aug 23 18:53:41 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 23 Aug 2007 18:53:41 -0400
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006892486@USNAEXCH.na.didata.local>

Most of my Unity 4.2 are running 2003 Server they provided, I wonder if
that will present any issues -jason

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, August 23, 2007 6:10 PM
To: cisco voip list
Subject: [cisco-voip] Unity 5 upgrading from Unity 4

Has anyone done it successfully?

I have a LOT of customers on Unity 4.X running Windows 2000 who want
to upgrade to Unity 5 and Windows 2003. However, my first attempt was
a complete disaster (ended up having to install from scratch).

Here is the problem:

You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity is
installed (says so in the upgrade documentation).

So, you must do a DRT backup, install Windows 2003 clean, install your
old version of Unity 4.X and restore it. Then you can upgrade to 5

This is not even slightly as simple as it sounds and has a remarkably
high failure rate.

(we got an error that the hostname didn't match and the upgrade to unity
failed)

TAC refused to support the upgrade (this was early July, it had come
off of NPH that day).

If anyone has, I am curious as to what tricks you did to make it work...



Jonathan
_______________________________________________
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
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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 svoll.voip at gmail.com  Thu Aug 23 18:55:03 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 23 Aug 2007 15:55:03 -0700
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>
Message-ID: <f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com>

I'm planning on this upgrade October / Novemberish but have not started the
research on it yet.  If anyone has any comments I would like to know also.

I don't have the Windows issue as I'm already at windows 2k3 with Exchange
2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a little easier
then Jonathan is talking about.

Scott


On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Has anyone done it successfully?
>
> I have a LOT of customers on Unity 4.X running Windows 2000 who want
> to upgrade to Unity 5 and Windows 2003. However, my first attempt was
> a complete disaster (ended up having to install from scratch).
>
> Here is the problem:
>
> You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity is
> installed (says so in the upgrade documentation).
>
> So, you must do a DRT backup, install Windows 2003 clean, install your
> old version of Unity 4.X and restore it. Then you can upgrade to 5
>
> This is not even slightly as simple as it sounds and has a remarkably
> high failure rate.
>
> (we got an error that the hostname didn't match and the upgrade to unity
> failed)
>
> TAC refused to support the upgrade (this was early July, it had come
> off of NPH that day).
>
> If anyone has, I am curious as to what tricks you did to make it work...
>
>
>
> 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  Thu Aug 23 19:34:05 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 23 Aug 2007 18:34:05 -0500
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>
	<f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com>
Message-ID: <5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com>

The Windows issue is the big one I am going to be running into... most
of the people who want to upgrade is because they want Windows 2000
out of their network (that is why they went to CCM5.1, MPX, etc...)

A few customers have Unity 4.2(1) on Win2k3, which means the upgrade
should be easy, but if they are running Win2K, that means a complete
blow away and start over...

I did one back in July, and it was a nightmare, we ended up abandoning
the old system and recreating everything from scratch...


Jonathan

On 8/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> I'm planning on this upgrade October / Novemberish but have not started the
> research on it yet.  If anyone has any comments I would like to know also.
>
> I don't have the Windows issue as I'm already at windows 2k3 with Exchange
> 2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a little easier
> then Jonathan is talking about.
>
> Scott
>
>
> On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > Has anyone done it successfully?
> >
> > I have a LOT of customers on Unity 4.X running Windows 2000 who want
> > to upgrade to Unity 5 and Windows 2003. However, my first attempt was
> > a complete disaster (ended up having to install from scratch).
> >
> > Here is the problem:
> >
> > You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity is
> > installed (says so in the upgrade documentation).
> >
> > So, you must do a DRT backup, install Windows 2003 clean, install your
> > old version of Unity 4.X and restore it. Then you can upgrade to 5
> >
> > This is not even slightly as simple as it sounds and has a remarkably
> > high failure rate.
> >
> > (we got an error that the hostname didn't match and the upgrade to unity
> failed)
> >
> > TAC refused to support the upgrade (this was early July, it had come
> > off of NPH that day).
> >
> > If anyone has, I am curious as to what tricks you did to make it work...
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From keith.klevenski at rig.net  Thu Aug 23 19:52:01 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Thu, 23 Aug 2007 18:52:01 -0500
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com><f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com>
	<5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com>
Message-ID: <E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org>

This does not sound promising as I'm about to embark on the win2k/4.0x to win2k3/4.2 to 5.0 myself.  I was going to upgrade to to 4.2 first (since 4.03sr1 isn't supported on win2k3), dirt backup, install fresh win2k3 install, dirt restore, upgrade to 5.0.  This is going to be ugly...

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
Sent: Thu 8/23/2007 6:34 PM
To: Scott Voll
Cc: cisco voip list
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4



The Windows issue is the big one I am going to be running into... most
of the people who want to upgrade is because they want Windows 2000
out of their network (that is why they went to CCM5.1, MPX, etc...)

A few customers have Unity 4.2(1) on Win2k3, which means the upgrade
should be easy, but if they are running Win2K, that means a complete
blow away and start over...

I did one back in July, and it was a nightmare, we ended up abandoning
the old system and recreating everything from scratch...


Jonathan

On 8/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> I'm planning on this upgrade October / Novemberish but have not started the
> research on it yet.  If anyone has any comments I would like to know also.
>
> I don't have the Windows issue as I'm already at windows 2k3 with Exchange
> 2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a little easier
> then Jonathan is talking about.
>
> Scott
>
>
> On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > Has anyone done it successfully?
> >
> > I have a LOT of customers on Unity 4.X running Windows 2000 who want
> > to upgrade to Unity 5 and Windows 2003. However, my first attempt was
> > a complete disaster (ended up having to install from scratch).
> >
> > Here is the problem:
> >
> > You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity is
> > installed (says so in the upgrade documentation).
> >
> > So, you must do a DRT backup, install Windows 2003 clean, install your
> > old version of Unity 4.X and restore it. Then you can upgrade to 5
> >
> > This is not even slightly as simple as it sounds and has a remarkably
> > high failure rate.
> >
> > (we got an error that the hostname didn't match and the upgrade to unity
> failed)
> >
> > TAC refused to support the upgrade (this was early July, it had come
> > off of NPH that day).
> >
> > If anyone has, I am curious as to what tricks you did to make it work...
> >
> >
> >
> > 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 jonvoip at gmail.com  Thu Aug 23 23:18:12 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 23 Aug 2007 22:18:12 -0500
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>
	<f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com>
	<5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com>
	<E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org>
Message-ID: <5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com>

That is exactly what I did...

Then when I went to run the Unity 5 install, it failed saying there
was a hostname mismatch... I called TAC, they had no idea what to do
and said, 'this should work...' and then refused to support it.



Jonathan

On 8/23/07, Keith Klevenski <keith.klevenski at rig.net> wrote:
>
>
>
> This does not sound promising as I'm about to embark on the win2k/4.0x to
> win2k3/4.2 to 5.0 myself.  I was going to upgrade to to 4.2 first (since
> 4.03sr1 isn't supported on win2k3), dirt backup, install fresh win2k3
> install, dirt restore, upgrade to 5.0.  This is going to be ugly...
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net on behalf of
> Jonathan Charles
> Sent: Thu 8/23/2007 6:34 PM
> To: Scott Voll
> Cc: cisco voip list
> Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4
>
>
>
>
> The Windows issue is the big one I am going to be running into... most
> of the people who want to upgrade is because they want Windows 2000
> out of their network (that is why they went to CCM5.1, MPX, etc...)
>
> A few customers have Unity 4.2(1) on Win2k3, which means the upgrade
> should be easy, but if they are running Win2K, that means a complete
> blow away and start over...
>
> I did one back in July, and it was a nightmare, we ended up abandoning
> the old system and recreating everything from scratch...
>
>
> Jonathan
>
> On 8/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > I'm planning on this upgrade October / Novemberish but have not started
> the
> > research on it yet.  If anyone has any comments I would like to know also.
> >
> > I don't have the Windows issue as I'm already at windows 2k3 with Exchange
> > 2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a little
> easier
> > then Jonathan is talking about.
> >
> > Scott
> >
> >
> > On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > > Has anyone done it successfully?
> > >
> > > I have a LOT of customers on Unity 4.X running Windows 2000 who want
> > > to upgrade to Unity 5 and Windows 2003. However, my first attempt was
> > > a complete disaster (ended up having to install from scratch).
> > >
> > > Here is the problem:
> > >
> > > You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity is
> > > installed (says so in the upgrade documentation).
> > >
> > > So, you must do a DRT backup, install Windows 2003 clean, install your
> > > old version of Unity 4.X and restore it. Then you can upgrade to 5
> > >
> > > This is not even slightly as simple as it sounds and has a remarkably
> > > high failure rate.
> > >
> > > (we got an error that the hostname didn't match and the upgrade to unity
> > failed)
> > >
> > > TAC refused to support the upgrade (this was early July, it had come
> > > off of NPH that day).
> > >
> > > If anyone has, I am curious as to what tricks you did to make it work...
> > >
> > >
> > >
> > > 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  Fri Aug 24 00:23:27 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 23 Aug 2007 23:23:27 -0500
Subject: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE after
	installing OS 4.5a SR2
In-Reply-To: <AB0A35F2486F81468971AA68558BE71906D9A4AD@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE71906D9A4AD@smieln01winm01.micoanet.com>
Message-ID: <5d093f9a0708232123n6c6bfaeaye0a032f3b18058b1@mail.gmail.com>

If your backup is good (I would say check the logs, but you can't),
then the restore is easy.

A couple of quick things:

the IP address and hostname MUST be identical or the backup will fail.

Also, you must install the EXACT same CCM version (check your
subscriber), and the same OS version that the backup was done (this
may be optional, but why risk it...), and then you MUST have the exact
same version of BARS.

As long as all of that is the same, you should be fine...


Jonathan

On 8/23/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
>
> All,
>
>
>
> This evening while installing the latest OS service pack from Cisco (OS 4.5a
> SR2) and restarting the CCM server (publisher) I am now greeted with a BSOD
> displaying Stop 0x0000007B INACCESABLE_BOOT_DEVICE.  This is an IBM 7815-I1
> server with a single SATA drive.  I didn't do anything different then I
> normally do when I patch servers and have never had a problem before this
> one.  I have tried booting using the last known good configuration, safe
> mode with network, safe mode, and command prompt.  I booted with a Windows
> 2000 Server CD and ran the recovery console and proceeded to run a CHKDSK
> /P.  The first time through it got to 50% before stopping and stating that
> "one or more unrecoverable errors have been discovered".  The second time I
> ran CHKDSK it only got to 25% before displaying the same message.  Is it
> conceivable that I lost the drive on a reboot and patching the box was just
> a coincidence?
>
>
>
> I have a recent BARS backup (8/22/07) so I am pretty confident that I can
> restore the server (even though I have never actually done this before).  Is
> it difficult to do so?  I am running CCM 4.1(3) with IPCC 3.5(1)
> co-resident.  Any links or experiences (or ideas to get back at my data)
> would be appreciated.
>
>
>
> Matt
>
>  ________________________________
>
>
>
>
> 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 GRhodes at rbh.com  Fri Aug 24 11:00:35 2007
From: GRhodes at rbh.com (Rhodes, Geoff)
Date: Fri, 24 Aug 2007 11:00:35 -0400
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com><f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com><5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com><E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org>
	<5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com>
Message-ID: <6350516B6159444E9D646523EB4526F403EBA87B@apollo.rbh.local>

Maybe they don't want us going to Unity 5, instead they want to push us
into using Exchange 2007 Unified Messaging...  :)


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, August 23, 2007 11:18 PM
To: Keith Klevenski
Cc: cisco voip list
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4

That is exactly what I did...

Then when I went to run the Unity 5 install, it failed saying there was
a hostname mismatch... I called TAC, they had no idea what to do and
said, 'this should work...' and then refused to support it.



Jonathan

On 8/23/07, Keith Klevenski <keith.klevenski at rig.net> wrote:
>
>
>
> This does not sound promising as I'm about to embark on the win2k/4.0x

> to
> win2k3/4.2 to 5.0 myself.  I was going to upgrade to to 4.2 first 
> (since
> 4.03sr1 isn't supported on win2k3), dirt backup, install fresh win2k3 
> install, dirt restore, upgrade to 5.0.  This is going to be ugly...
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan 
> Charles
> Sent: Thu 8/23/2007 6:34 PM
> To: Scott Voll
> Cc: cisco voip list
> Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4
>
>
>
>
> The Windows issue is the big one I am going to be running into... most

> of the people who want to upgrade is because they want Windows 2000 
> out of their network (that is why they went to CCM5.1, MPX, etc...)
>
> A few customers have Unity 4.2(1) on Win2k3, which means the upgrade 
> should be easy, but if they are running Win2K, that means a complete 
> blow away and start over...
>
> I did one back in July, and it was a nightmare, we ended up abandoning

> the old system and recreating everything from scratch...
>
>
> Jonathan
>
> On 8/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > I'm planning on this upgrade October / Novemberish but have not 
> > started
> the
> > research on it yet.  If anyone has any comments I would like to know
also.
> >
> > I don't have the Windows issue as I'm already at windows 2k3 with 
> > Exchange
> > 2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a 
> > little
> easier
> > then Jonathan is talking about.
> >
> > Scott
> >
> >
> > On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > > Has anyone done it successfully?
> > >
> > > I have a LOT of customers on Unity 4.X running Windows 2000 who 
> > > want to upgrade to Unity 5 and Windows 2003. However, my first 
> > > attempt was a complete disaster (ended up having to install from
scratch).
> > >
> > > Here is the problem:
> > >
> > > You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity 
> > > is installed (says so in the upgrade documentation).
> > >
> > > So, you must do a DRT backup, install Windows 2003 clean, install 
> > > your old version of Unity 4.X and restore it. Then you can upgrade

> > > to 5
> > >
> > > This is not even slightly as simple as it sounds and has a 
> > > remarkably high failure rate.
> > >
> > > (we got an error that the hostname didn't match and the upgrade to

> > > unity
> > failed)
> > >
> > > TAC refused to support the upgrade (this was early July, it had 
> > > come off of NPH that day).
> > >
> > > If anyone has, I am curious as to what tricks you did to make it
work...
> > >
> > >
> > >
> > > 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 keith.klevenski at rig.net  Fri Aug 24 11:12:22 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 24 Aug 2007 10:12:22 -0500
Subject: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE
	afterinstalling OS 4.5a SR2
In-Reply-To: <5d093f9a0708232123n6c6bfaeaye0a032f3b18058b1@mail.gmail.com>
References: <AB0A35F2486F81468971AA68558BE71906D9A4AD@smieln01winm01.micoanet.com>
	<5d093f9a0708232123n6c6bfaeaye0a032f3b18058b1@mail.gmail.com>
Message-ID: <E0F48CE151B5D440B3649903D7221DA2381CA9@HOUEXH02.rignet.org>

Also make sure when installing CCM that you use the same local administrator, sa, private password phrase, and dc directory passwords as well as IP, computer name, workgroup, etc.  It really is pretty straightforward as Jonathan said.  I just recently did a hardware replacement for a couple of clusters.  I installed the exact same version of CCM (OS was different for me but worked fine), CCM service release, and BARS (along with all the passwords above being the same as your current system) on the new hardware.  Ran the restore from BARS, rebooted and voila, my old configuration was back and everything worked perfectly.

Cisco has done a good job with CCM restores from my experience.

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

"Extending the power of your knowledge"


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Thursday, August 23, 2007 11:23 PM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE afterinstalling OS 4.5a SR2

If your backup is good (I would say check the logs, but you can't),
then the restore is easy.

A couple of quick things:

the IP address and hostname MUST be identical or the backup will fail.

Also, you must install the EXACT same CCM version (check your
subscriber), and the same OS version that the backup was done (this
may be optional, but why risk it...), and then you MUST have the exact
same version of BARS.

As long as all of that is the same, you should be fine...


Jonathan

On 8/23/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
>
> All,
>
>
>
> This evening while installing the latest OS service pack from Cisco (OS 4.5a
> SR2) and restarting the CCM server (publisher) I am now greeted with a BSOD
> displaying Stop 0x0000007B INACCESABLE_BOOT_DEVICE.  This is an IBM 7815-I1
> server with a single SATA drive.  I didn't do anything different then I
> normally do when I patch servers and have never had a problem before this
> one.  I have tried booting using the last known good configuration, safe
> mode with network, safe mode, and command prompt.  I booted with a Windows
> 2000 Server CD and ran the recovery console and proceeded to run a CHKDSK
> /P.  The first time through it got to 50% before stopping and stating that
> "one or more unrecoverable errors have been discovered".  The second time I
> ran CHKDSK it only got to 25% before displaying the same message.  Is it
> conceivable that I lost the drive on a reboot and patching the box was just
> a coincidence?
>
>
>
> I have a recent BARS backup (8/22/07) so I am pretty confident that I can
> restore the server (even though I have never actually done this before).  Is
> it difficult to do so?  I am running CCM 4.1(3) with IPCC 3.5(1)
> co-resident.  Any links or experiences (or ideas to get back at my data)
> would be appreciated.
>
>
>
> Matt
>
>  ________________________________
>
>
>
>
> 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

From keith.klevenski at rig.net  Fri Aug 24 11:12:43 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 24 Aug 2007 10:12:43 -0500
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>
	<f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com>
	<5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com>
	<E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org>
	<5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com>
Message-ID: <E0F48CE151B5D440B3649903D7221DA2381CAA@HOUEXH02.rignet.org>

Nice.  :c|

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

"Extending the power of your knowledge"


-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Thursday, August 23, 2007 10:18 PM
To: Keith Klevenski
Cc: Scott Voll; cisco voip list
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4

That is exactly what I did...

Then when I went to run the Unity 5 install, it failed saying there
was a hostname mismatch... I called TAC, they had no idea what to do
and said, 'this should work...' and then refused to support it.



Jonathan

On 8/23/07, Keith Klevenski <keith.klevenski at rig.net> wrote:
>
>
>
> This does not sound promising as I'm about to embark on the win2k/4.0x to
> win2k3/4.2 to 5.0 myself.  I was going to upgrade to to 4.2 first (since
> 4.03sr1 isn't supported on win2k3), dirt backup, install fresh win2k3
> install, dirt restore, upgrade to 5.0.  This is going to be ugly...
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net on behalf of
> Jonathan Charles
> Sent: Thu 8/23/2007 6:34 PM
> To: Scott Voll
> Cc: cisco voip list
> Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4
>
>
>
>
> The Windows issue is the big one I am going to be running into... most
> of the people who want to upgrade is because they want Windows 2000
> out of their network (that is why they went to CCM5.1, MPX, etc...)
>
> A few customers have Unity 4.2(1) on Win2k3, which means the upgrade
> should be easy, but if they are running Win2K, that means a complete
> blow away and start over...
>
> I did one back in July, and it was a nightmare, we ended up abandoning
> the old system and recreating everything from scratch...
>
>
> Jonathan
>
> On 8/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > I'm planning on this upgrade October / Novemberish but have not started
> the
> > research on it yet.  If anyone has any comments I would like to know also.
> >
> > I don't have the Windows issue as I'm already at windows 2k3 with Exchange
> > 2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a little
> easier
> > then Jonathan is talking about.
> >
> > Scott
> >
> >
> > On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > > Has anyone done it successfully?
> > >
> > > I have a LOT of customers on Unity 4.X running Windows 2000 who want
> > > to upgrade to Unity 5 and Windows 2003. However, my first attempt was
> > > a complete disaster (ended up having to install from scratch).
> > >
> > > Here is the problem:
> > >
> > > You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity is
> > > installed (says so in the upgrade documentation).
> > >
> > > So, you must do a DRT backup, install Windows 2003 clean, install your
> > > old version of Unity 4.X and restore it. Then you can upgrade to 5
> > >
> > > This is not even slightly as simple as it sounds and has a remarkably
> > > high failure rate.
> > >
> > > (we got an error that the hostname didn't match and the upgrade to unity
> > failed)
> > >
> > > TAC refused to support the upgrade (this was early July, it had come
> > > off of NPH that day).
> > >
> > > If anyone has, I am curious as to what tricks you did to make it work...
> > >
> > >
> > >
> > > 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  Fri Aug 24 11:47:41 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 24 Aug 2007 11:47:41 -0400
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <6350516B6159444E9D646523EB4526F403EBA87B@apollo.rbh.local>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com><f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com><5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com><E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org><5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com>
	<6350516B6159444E9D646523EB4526F403EBA87B@apollo.rbh.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0068CD4B4@USNAEXCH.na.didata.local>

The Exchange 2007 upgrade has it's own caveats, 64-bit hardware, 64-bit
OS and migration of mailboxes....either way you need someone skilled!

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rhodes, Geoff
Sent: Friday, August 24, 2007 11:01 AM
To: cisco voip list
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4

Maybe they don't want us going to Unity 5, instead they want to push us
into using Exchange 2007 Unified Messaging...  :)


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, August 23, 2007 11:18 PM
To: Keith Klevenski
Cc: cisco voip list
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4

That is exactly what I did...

Then when I went to run the Unity 5 install, it failed saying there was
a hostname mismatch... I called TAC, they had no idea what to do and
said, 'this should work...' and then refused to support it.



Jonathan

On 8/23/07, Keith Klevenski <keith.klevenski at rig.net> wrote:
>
>
>
> This does not sound promising as I'm about to embark on the win2k/4.0x

> to
> win2k3/4.2 to 5.0 myself.  I was going to upgrade to to 4.2 first 
> (since
> 4.03sr1 isn't supported on win2k3), dirt backup, install fresh win2k3 
> install, dirt restore, upgrade to 5.0.  This is going to be ugly...
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan 
> Charles
> Sent: Thu 8/23/2007 6:34 PM
> To: Scott Voll
> Cc: cisco voip list
> Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4
>
>
>
>
> The Windows issue is the big one I am going to be running into... most

> of the people who want to upgrade is because they want Windows 2000 
> out of their network (that is why they went to CCM5.1, MPX, etc...)
>
> A few customers have Unity 4.2(1) on Win2k3, which means the upgrade 
> should be easy, but if they are running Win2K, that means a complete 
> blow away and start over...
>
> I did one back in July, and it was a nightmare, we ended up abandoning

> the old system and recreating everything from scratch...
>
>
> Jonathan
>
> On 8/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > I'm planning on this upgrade October / Novemberish but have not 
> > started
> the
> > research on it yet.  If anyone has any comments I would like to know
also.
> >
> > I don't have the Windows issue as I'm already at windows 2k3 with 
> > Exchange
> > 2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a 
> > little
> easier
> > then Jonathan is talking about.
> >
> > Scott
> >
> >
> > On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > > Has anyone done it successfully?
> > >
> > > I have a LOT of customers on Unity 4.X running Windows 2000 who 
> > > want to upgrade to Unity 5 and Windows 2003. However, my first 
> > > attempt was a complete disaster (ended up having to install from
scratch).
> > >
> > > Here is the problem:
> > >
> > > You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity 
> > > is installed (says so in the upgrade documentation).
> > >
> > > So, you must do a DRT backup, install Windows 2003 clean, install 
> > > your old version of Unity 4.X and restore it. Then you can upgrade

> > > to 5
> > >
> > > This is not even slightly as simple as it sounds and has a 
> > > remarkably high failure rate.
> > >
> > > (we got an error that the hostname didn't match and the upgrade to

> > > unity
> > failed)
> > >
> > > TAC refused to support the upgrade (this was early July, it had 
> > > come off of NPH that day).
> > >
> > > If anyone has, I am curious as to what tricks you did to make it
work...
> > >
> > >
> > >
> > > 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
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From d4rkf1ber at gmail.com  Fri Aug 24 11:49:59 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Fri, 24 Aug 2007 10:49:59 -0500
Subject: [cisco-voip] IP Communicator 2.1(1) Crashes when ending call
Message-ID: <a1b07ea60708240849sf41354dy2428b115ba2340b7@mail.gmail.com>

I seem to be running into a problem with IP Communicator.  It launches and
seems to work fine, I can take a call and talk but then when you end the
call it continues to show the counter as though the call is there.
Eventually it starts to get sluggish and hang, non responsive, then you see
the CM is down message on IP Communicator display.

Anyone seen this?  I read the docs and bugs but none seem to point to an
issue when ending a call.

Just thought I would ask.
-------------- next part --------------
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From joe.j.keller at gmail.com  Fri Aug 24 11:54:19 2007
From: joe.j.keller at gmail.com (Joe Keller)
Date: Fri, 24 Aug 2007 11:54:19 -0400
Subject: [cisco-voip] BARS fails due to SQL Server, after OS 4.4aSR9 update
Message-ID: <189b5daa0708240854l22fe2abbn9e85bdc3c4da5de7@mail.gmail.com>

CCM 4.1(3)
BARS 4.0.12
OS was 2000.4.3aSR9 updated to 2000.4.4aSR9. Update log showed no
errors. BARS fails since update.

Bars status window shows:
 [02:00:19 AM]  Dumping Ccm0302 database on x.x.x.x
 *** [02:00:24 AM]  Could not verify db backup.
 [02:00:24 AM]  Staging database dump from x.x.x.x to staging directory.
 *** [02:00:24 AM]  Failed to stage the database Ccm0302.
 *** [02:00:24 AM]  Could not backup database Ccm0302 successfully.
 [02:00:24 AM]  Staging Locales folder C:\Program
Files\Cisco\CallManager\DBInstallCSV\Products
 [02:00:26 AM] Backing up URL Services Page.
 *** [02:00:26 AM]  Failed to copy URL Services page. Couldn't find
IIS Root Directory for CCMCIP in x.x.x.x
 [02:00:26 AM]  Skipping CDR database backup on x.x.x.x.
 [
BARS trace log shows:
[02:00:24 AM]   Could not verify database backup successfully:
-2147218303    [Mic-
                rosoft][ODBC SQL Server Driver][SQL Server]Cannot open backup d-
                evice 'stiFileDump'. Device error or device off-line. See the S-
                QL Server error log for more details.
[Microsoft][ODBC SQL Serv-
                er Driver][SQL Server]VERIFY DATABASE is terminating abnormally.
                .
[02:00:24 AM]   Could not verify db backup.
[02:00:24 AM]   Moving dump to staging dir
[02:00:24 AM]   Failed to stage the database Ccm0302.
[02:00:24 AM]   Error in modBackup DB: 450    Wrong number of
arguments or invalid-
                 property assignment
[02:00:24 AM]   Could not backup database Ccm0302 successfully.
[02:00:24 AM]   Staging Locales folder C:\Program Files\Cisco\CallManager\DBIns-
                tallCSV\Products
[02:00:24 AM]   Locales Folder Created
[02:00:24 AM]   In function modStageDir
[02:00:26 AM]   Successfully backed up locales folder.
[02:00:26 AM]   The CCM database, DBNAME from registry CCM0302 and the  Connect-
                ion String registry value for DBConnection0 is CCM0302 .
[02:00:26 AM]   Inside function modGetLocalComputerName
[02:00:26 AM]   Local Computer name is CCMPUB
[02:00:26 AM]   Exiting function modGetLocalComputerName
[02:00:26 AM]   Initializing Sockets
[02:00:26 AM]   Inside function modGetIpAddress
[02:00:26 AM]   IPAddress = x.x.x.x
[02:00:26 AM]   Closing Sockets
[02:00:26 AM]   Exiting function modGetIpAddress
[02:00:26 AM]   Succesfully wrote the Publisher details to the catalog.ini file.
[02:00:26 AM]   Inside Function modExecuteQuery
[02:00:26 AM]   modExecuteQuery=http://x.x.x.x/CCMCIP/getservicesmenu.asp
[02:00:26 AM]   Backing up URL Services Page.
[02:00:26 AM]   Given URL Server: x.x.x.x
[02:00:26 AM]   Given Backup Server: x.x.x.x
[02:00:26 AM]   Inside function modVirualDir
[02:00:26 AM]   [modVirualDir]: -2147463168 Automation error
[02:00:26 AM]   Failed to copy URL Services page. Couldn't find IIS Root Direct-
                ory for CCMCIP in x.x.x.x

DBLHelper shows no problems

SQL Server error log shows the following error at 2:00:24:29:
"BackupDiskFile::OpenMedia: Backup device 'C:\Ccm0302.dmp' failed to
open. Operating system error = 32(The process cannot access the file
because it is being used by another process.)."

Any ideas?

From MLinsemier at apcapital.com  Fri Aug 24 11:54:43 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Fri, 24 Aug 2007 11:54:43 -0400
Subject: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE after
	installing OS 4.5a SR2
In-Reply-To: <5d093f9a0708232123n6c6bfaeaye0a032f3b18058b1@mail.gmail.com>
References: <AB0A35F2486F81468971AA68558BE71906D9A4AD@smieln01winm01.micoanet.com>
	<5d093f9a0708232123n6c6bfaeaye0a032f3b18058b1@mail.gmail.com>
Message-ID: <AB0A35F2486F81468971AA68558BE71906D9AB98@smieln01winm01.micoanet.com>

Is there a way to look in the .TAR file to see what version of BARS I
was using?  I am pretty sure I was on 4.0(11), but I am not 100%
positive.  

You can't restore with the latest BARS utility?  That seems kind of
ridiculous.

I will be restoring with the same IP, Hostname, and I have all the
passwords so that should be fine.

In the case of a co-resident IPCC Express, do I restore CCM first and
then install IPCC Express?  Honeslty, I only had three scripts running
for prompts (with some Holiday check stuff) so I could reinstall this
from scratch actually without having to restore.

Matt

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, August 24, 2007 12:23 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE after
installing OS 4.5a SR2

If your backup is good (I would say check the logs, but you can't),
then the restore is easy.

A couple of quick things:

the IP address and hostname MUST be identical or the backup will fail.

Also, you must install the EXACT same CCM version (check your
subscriber), and the same OS version that the backup was done (this
may be optional, but why risk it...), and then you MUST have the exact
same version of BARS.

As long as all of that is the same, you should be fine...


Jonathan

On 8/23/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
>
> All,
>
>
>
> This evening while installing the latest OS service pack from Cisco
(OS 4.5a
> SR2) and restarting the CCM server (publisher) I am now greeted with a
BSOD
> displaying Stop 0x0000007B INACCESABLE_BOOT_DEVICE.  This is an IBM
7815-I1
> server with a single SATA drive.  I didn't do anything different then
I
> normally do when I patch servers and have never had a problem before
this
> one.  I have tried booting using the last known good configuration,
safe
> mode with network, safe mode, and command prompt.  I booted with a
Windows
> 2000 Server CD and ran the recovery console and proceeded to run a
CHKDSK
> /P.  The first time through it got to 50% before stopping and stating
that
> "one or more unrecoverable errors have been discovered".  The second
time I
> ran CHKDSK it only got to 25% before displaying the same message.  Is
it
> conceivable that I lost the drive on a reboot and patching the box was
just
> a coincidence?
>
>
>
> I have a recent BARS backup (8/22/07) so I am pretty confident that I
can
> restore the server (even though I have never actually done this
before).  Is
> it difficult to do so?  I am running CCM 4.1(3) with IPCC 3.5(1)
> co-resident.  Any links or experiences (or ideas to get back at my
data)
> would be appreciated.
>
>
>
> Matt
>
>  ________________________________
>
>
>
>
> 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.

From keith.klevenski at rig.net  Fri Aug 24 12:04:05 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 24 Aug 2007 11:04:05 -0500
Subject: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE
	afterinstalling OS 4.5a SR2
In-Reply-To: <AB0A35F2486F81468971AA68558BE71906D9AB98@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE71906D9A4AD@smieln01winm01.micoanet.com><5d093f9a0708232123n6c6bfaeaye0a032f3b18058b1@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE71906D9AB98@smieln01winm01.micoanet.com>
Message-ID: <E0F48CE151B5D440B3649903D7221DA2381CC4@HOUEXH02.rignet.org>

The log file of the backup will have the version of BARS that was used.  

Just like Unity you must use the same version of the backup/restore software for it to work as each version expects to see certain things in the backup file when doing the restore. 

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

"Extending the power of your knowledge"


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Linsemier, Matthew
Sent: Friday, August 24, 2007 10:55 AM
To: Jonathan Charles
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE afterinstalling OS 4.5a SR2

Is there a way to look in the .TAR file to see what version of BARS I
was using?  I am pretty sure I was on 4.0(11), but I am not 100%
positive.  

You can't restore with the latest BARS utility?  That seems kind of
ridiculous.

I will be restoring with the same IP, Hostname, and I have all the
passwords so that should be fine.

In the case of a co-resident IPCC Express, do I restore CCM first and
then install IPCC Express?  Honeslty, I only had three scripts running
for prompts (with some Holiday check stuff) so I could reinstall this
from scratch actually without having to restore.

Matt

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, August 24, 2007 12:23 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE after
installing OS 4.5a SR2

If your backup is good (I would say check the logs, but you can't),
then the restore is easy.

A couple of quick things:

the IP address and hostname MUST be identical or the backup will fail.

Also, you must install the EXACT same CCM version (check your
subscriber), and the same OS version that the backup was done (this
may be optional, but why risk it...), and then you MUST have the exact
same version of BARS.

As long as all of that is the same, you should be fine...


Jonathan

On 8/23/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
>
> All,
>
>
>
> This evening while installing the latest OS service pack from Cisco
(OS 4.5a
> SR2) and restarting the CCM server (publisher) I am now greeted with a
BSOD
> displaying Stop 0x0000007B INACCESABLE_BOOT_DEVICE.  This is an IBM
7815-I1
> server with a single SATA drive.  I didn't do anything different then
I
> normally do when I patch servers and have never had a problem before
this
> one.  I have tried booting using the last known good configuration,
safe
> mode with network, safe mode, and command prompt.  I booted with a
Windows
> 2000 Server CD and ran the recovery console and proceeded to run a
CHKDSK
> /P.  The first time through it got to 50% before stopping and stating
that
> "one or more unrecoverable errors have been discovered".  The second
time I
> ran CHKDSK it only got to 25% before displaying the same message.  Is
it
> conceivable that I lost the drive on a reboot and patching the box was
just
> a coincidence?
>
>
>
> I have a recent BARS backup (8/22/07) so I am pretty confident that I
can
> restore the server (even though I have never actually done this
before).  Is
> it difficult to do so?  I am running CCM 4.1(3) with IPCC 3.5(1)
> co-resident.  Any links or experiences (or ideas to get back at my
data)
> would be appreciated.
>
>
>
> Matt
>
>  ________________________________
>
>
>
>
> 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.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From svoll.voip at gmail.com  Fri Aug 24 12:17:21 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 24 Aug 2007 09:17:21 -0700
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0068CD4B4@USNAEXCH.na.didata.local>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com>
	<f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com>
	<5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com>
	<E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org>
	<5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com>
	<6350516B6159444E9D646523EB4526F403EBA87B@apollo.rbh.local>
	<C1FE15183DA37645BC0633BC604E44F0068CD4B4@USNAEXCH.na.didata.local>
Message-ID: <f84a38d30708240917y175cd089u8cecce93938b7638@mail.gmail.com>

If your running UM and you want Exchange 2007 you have to upgrade Unity
first anyway.... so then you need two people skilled ;-)

Scott


On 8/24/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
> The Exchange 2007 upgrade has it's own caveats, 64-bit hardware, 64-bit
> OS and migration of mailboxes....either way you need someone skilled!
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rhodes, Geoff
> Sent: Friday, August 24, 2007 11:01 AM
> To: cisco voip list
> Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4
>
> Maybe they don't want us going to Unity 5, instead they want to push us
> into using Exchange 2007 Unified Messaging...  :)
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Thursday, August 23, 2007 11:18 PM
> To: Keith Klevenski
> Cc: cisco voip list
> Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4
>
> That is exactly what I did...
>
> Then when I went to run the Unity 5 install, it failed saying there was
> a hostname mismatch... I called TAC, they had no idea what to do and
> said, 'this should work...' and then refused to support it.
>
>
>
> Jonathan
>
> On 8/23/07, Keith Klevenski <keith.klevenski at rig.net> wrote:
> >
> >
> >
> > This does not sound promising as I'm about to embark on the win2k/4.0x
>
> > to
> > win2k3/4.2 to 5.0 myself.  I was going to upgrade to to 4.2 first
> > (since
> > 4.03sr1 isn't supported on win2k3), dirt backup, install fresh win2k3
> > install, dirt restore, upgrade to 5.0.  This is going to be ugly...
> >
> >  ________________________________
> >  From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan
> > Charles
> > Sent: Thu 8/23/2007 6:34 PM
> > To: Scott Voll
> > Cc: cisco voip list
> > Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4
> >
> >
> >
> >
> > The Windows issue is the big one I am going to be running into... most
>
> > of the people who want to upgrade is because they want Windows 2000
> > out of their network (that is why they went to CCM5.1, MPX, etc...)
> >
> > A few customers have Unity 4.2(1) on Win2k3, which means the upgrade
> > should be easy, but if they are running Win2K, that means a complete
> > blow away and start over...
> >
> > I did one back in July, and it was a nightmare, we ended up abandoning
>
> > the old system and recreating everything from scratch...
> >
> >
> > Jonathan
> >
> > On 8/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > > I'm planning on this upgrade October / Novemberish but have not
> > > started
> > the
> > > research on it yet.  If anyone has any comments I would like to know
> also.
> > >
> > > I don't have the Windows issue as I'm already at windows 2k3 with
> > > Exchange
> > > 2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a
> > > little
> > easier
> > > then Jonathan is talking about.
> > >
> > > Scott
> > >
> > >
> > > On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > >
> > > > Has anyone done it successfully?
> > > >
> > > > I have a LOT of customers on Unity 4.X running Windows 2000 who
> > > > want to upgrade to Unity 5 and Windows 2003. However, my first
> > > > attempt was a complete disaster (ended up having to install from
> scratch).
> > > >
> > > > Here is the problem:
> > > >
> > > > You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity
> > > > is installed (says so in the upgrade documentation).
> > > >
> > > > So, you must do a DRT backup, install Windows 2003 clean, install
> > > > your old version of Unity 4.X and restore it. Then you can upgrade
>
> > > > to 5
> > > >
> > > > This is not even slightly as simple as it sounds and has a
> > > > remarkably high failure rate.
> > > >
> > > > (we got an error that the hostname didn't match and the upgrade to
>
> > > > unity
> > > failed)
> > > >
> > > > TAC refused to support the upgrade (this was early July, it had
> > > > come off of NPH that day).
> > > >
> > > > If anyone has, I am curious as to what tricks you did to make it
> work...
> > > >
> > > >
> > > >
> > > > 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
> _______________________________________________
> 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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From MLinsemier at apcapital.com  Fri Aug 24 12:29:44 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Fri, 24 Aug 2007 12:29:44 -0400
Subject: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE
	afterinstalling OS 4.5a SR2
In-Reply-To: <E0F48CE151B5D440B3649903D7221DA2381CC4@HOUEXH02.rignet.org>
References: <AB0A35F2486F81468971AA68558BE71906D9A4AD@smieln01winm01.micoanet.com><5d093f9a0708232123n6c6bfaeaye0a032f3b18058b1@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE71906D9AB98@smieln01winm01.micoanet.com>
	<E0F48CE151B5D440B3649903D7221DA2381CC4@HOUEXH02.rignet.org>
Message-ID: <AB0A35F2486F81468971AA68558BE71906D9ABCA@smieln01winm01.micoanet.com>

The problem is that I can't get at the logfile because its on the server that has the failed drive.  Will I get an error if I try the wrong versions?

-----Original Message-----
From: Keith Klevenski [mailto:keith.klevenski at rig.net] 
Sent: Friday, August 24, 2007 12:04 PM
To: Linsemier, Matthew; Jonathan Charles
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE afterinstalling OS 4.5a SR2

The log file of the backup will have the version of BARS that was used.  

Just like Unity you must use the same version of the backup/restore software for it to work as each version expects to see certain things in the backup file when doing the restore. 

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

"Extending the power of your knowledge"


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Linsemier, Matthew
Sent: Friday, August 24, 2007 10:55 AM
To: Jonathan Charles
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE afterinstalling OS 4.5a SR2

Is there a way to look in the .TAR file to see what version of BARS I
was using?  I am pretty sure I was on 4.0(11), but I am not 100%
positive.  

You can't restore with the latest BARS utility?  That seems kind of
ridiculous.

I will be restoring with the same IP, Hostname, and I have all the
passwords so that should be fine.

In the case of a co-resident IPCC Express, do I restore CCM first and
then install IPCC Express?  Honeslty, I only had three scripts running
for prompts (with some Holiday check stuff) so I could reinstall this
from scratch actually without having to restore.

Matt

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, August 24, 2007 12:23 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE after
installing OS 4.5a SR2

If your backup is good (I would say check the logs, but you can't),
then the restore is easy.

A couple of quick things:

the IP address and hostname MUST be identical or the backup will fail.

Also, you must install the EXACT same CCM version (check your
subscriber), and the same OS version that the backup was done (this
may be optional, but why risk it...), and then you MUST have the exact
same version of BARS.

As long as all of that is the same, you should be fine...


Jonathan

On 8/23/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
>
> All,
>
>
>
> This evening while installing the latest OS service pack from Cisco
(OS 4.5a
> SR2) and restarting the CCM server (publisher) I am now greeted with a
BSOD
> displaying Stop 0x0000007B INACCESABLE_BOOT_DEVICE.  This is an IBM
7815-I1
> server with a single SATA drive.  I didn't do anything different then
I
> normally do when I patch servers and have never had a problem before
this
> one.  I have tried booting using the last known good configuration,
safe
> mode with network, safe mode, and command prompt.  I booted with a
Windows
> 2000 Server CD and ran the recovery console and proceeded to run a
CHKDSK
> /P.  The first time through it got to 50% before stopping and stating
that
> "one or more unrecoverable errors have been discovered".  The second
time I
> ran CHKDSK it only got to 25% before displaying the same message.  Is
it
> conceivable that I lost the drive on a reboot and patching the box was
just
> a coincidence?
>
>
>
> I have a recent BARS backup (8/22/07) so I am pretty confident that I
can
> restore the server (even though I have never actually done this
before).  Is
> it difficult to do so?  I am running CCM 4.1(3) with IPCC 3.5(1)
> co-resident.  Any links or experiences (or ideas to get back at my
data)
> would be appreciated.
>
>
>
> Matt
>
>  ________________________________
>
>
>
>
> 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.
_______________________________________________
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.

From keith.klevenski at rig.net  Fri Aug 24 12:47:26 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 24 Aug 2007 11:47:26 -0500
Subject: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE
	afterinstalling OS 4.5a SR2
In-Reply-To: <AB0A35F2486F81468971AA68558BE71906D9ABCA@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE71906D9A4AD@smieln01winm01.micoanet.com><5d093f9a0708232123n6c6bfaeaye0a032f3b18058b1@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE71906D9AB98@smieln01winm01.micoanet.com>
	<E0F48CE151B5D440B3649903D7221DA2381CC4@HOUEXH02.rignet.org>
	<AB0A35F2486F81468971AA68558BE71906D9ABCA@smieln01winm01.micoanet.com>
Message-ID: <E0F48CE151B5D440B3649903D7221DA2381CD4@HOUEXH02.rignet.org>

I know you will get a warning if you try to restore a different version of CCM than what you are running on the box you're trying to restore to, but I'm not sure if you will get a warning about conflicting BARS versions.  I would think you would since it is required.  I honestly didn't expect it to warn me as I accidentally tried to restore a different version of CCM than what I was running on the box I was restoring to (installed the wrong service release on the new server), but I'm glad it did!  This was in the lab testing, but still...

It would be nice if Cisco included the log file in the tar file.

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

"Extending the power of your knowledge"


-----Original Message-----
From: Linsemier, Matthew [mailto:MLinsemier at apcapital.com] 
Sent: Friday, August 24, 2007 11:30 AM
To: Keith Klevenski; Jonathan Charles
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE afterinstalling OS 4.5a SR2

The problem is that I can't get at the logfile because its on the server that has the failed drive.  Will I get an error if I try the wrong versions?

-----Original Message-----
From: Keith Klevenski [mailto:keith.klevenski at rig.net] 
Sent: Friday, August 24, 2007 12:04 PM
To: Linsemier, Matthew; Jonathan Charles
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE afterinstalling OS 4.5a SR2

The log file of the backup will have the version of BARS that was used.  

Just like Unity you must use the same version of the backup/restore software for it to work as each version expects to see certain things in the backup file when doing the restore. 

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

"Extending the power of your knowledge"


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Linsemier, Matthew
Sent: Friday, August 24, 2007 10:55 AM
To: Jonathan Charles
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE afterinstalling OS 4.5a SR2

Is there a way to look in the .TAR file to see what version of BARS I
was using?  I am pretty sure I was on 4.0(11), but I am not 100%
positive.  

You can't restore with the latest BARS utility?  That seems kind of
ridiculous.

I will be restoring with the same IP, Hostname, and I have all the
passwords so that should be fine.

In the case of a co-resident IPCC Express, do I restore CCM first and
then install IPCC Express?  Honeslty, I only had three scripts running
for prompts (with some Holiday check stuff) so I could reinstall this
from scratch actually without having to restore.

Matt

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, August 24, 2007 12:23 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stop 0x0000007B INACCESABLE_BOOT_DEVICE after
installing OS 4.5a SR2

If your backup is good (I would say check the logs, but you can't),
then the restore is easy.

A couple of quick things:

the IP address and hostname MUST be identical or the backup will fail.

Also, you must install the EXACT same CCM version (check your
subscriber), and the same OS version that the backup was done (this
may be optional, but why risk it...), and then you MUST have the exact
same version of BARS.

As long as all of that is the same, you should be fine...


Jonathan

On 8/23/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
>
> All,
>
>
>
> This evening while installing the latest OS service pack from Cisco
(OS 4.5a
> SR2) and restarting the CCM server (publisher) I am now greeted with a
BSOD
> displaying Stop 0x0000007B INACCESABLE_BOOT_DEVICE.  This is an IBM
7815-I1
> server with a single SATA drive.  I didn't do anything different then
I
> normally do when I patch servers and have never had a problem before
this
> one.  I have tried booting using the last known good configuration,
safe
> mode with network, safe mode, and command prompt.  I booted with a
Windows
> 2000 Server CD and ran the recovery console and proceeded to run a
CHKDSK
> /P.  The first time through it got to 50% before stopping and stating
that
> "one or more unrecoverable errors have been discovered".  The second
time I
> ran CHKDSK it only got to 25% before displaying the same message.  Is
it
> conceivable that I lost the drive on a reboot and patching the box was
just
> a coincidence?
>
>
>
> I have a recent BARS backup (8/22/07) so I am pretty confident that I
can
> restore the server (even though I have never actually done this
before).  Is
> it difficult to do so?  I am running CCM 4.1(3) with IPCC 3.5(1)
> co-resident.  Any links or experiences (or ideas to get back at my
data)
> would be appreciated.
>
>
>
> Matt
>
>  ________________________________
>
>
>
>
> 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.
_______________________________________________
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.

From sweeper1 at gmail.com  Fri Aug 24 13:24:54 2007
From: sweeper1 at gmail.com (sweeper1)
Date: Fri, 24 Aug 2007 13:24:54 -0400
Subject: [cisco-voip] Phone Proxy
In-Reply-To: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>
References: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>
Message-ID: <cf6e8aa90708241024r6076a497s5520bbc2f2f1bf83@mail.gmail.com>

Does anyone know if it's possible to run more than one phone behind a NAT
device?

I have the proxy working fine if I put the phone (behind a Linksys cable
modem router) on the DMZ.
But I have some locations that require multiple phones...

Steve


On 8/14/07, Scott Voll <svoll.voip at gmail.com> wrote:
>
> Hey--
>
> we just got our Phone Proxy setup and it works pretty slick.
>
> I think it only took about 4 hours to unpackage, configure, and get the
> first Phone up and working.
>
> I do however hate Cisco docs as it states you need a null modem.  so I
> look high and low and finally find one to find out that in all reality you
> really don't need one.  Got to love that.
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From keith.klevenski at rig.net  Fri Aug 24 13:48:53 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 24 Aug 2007 12:48:53 -0500
Subject: [cisco-voip] Phone Proxy
In-Reply-To: <cf6e8aa90708241024r6076a497s5520bbc2f2f1bf83@mail.gmail.com>
References: <f84a38d30708141423s24278f6ej5d9dd3bb130f329d@mail.gmail.com>
	<cf6e8aa90708241024r6076a497s5520bbc2f2f1bf83@mail.gmail.com>
Message-ID: <E0F48CE151B5D440B3649903D7221DA2381CE1@HOUEXH02.rignet.org>

Not in reference to phone proxy, but we do have multiple IP phones
behind IOS NAT device registered to CCM:
http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps1834/products
_feature_guide09186a008008052e.html 

 

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 sweeper1
Sent: Friday, August 24, 2007 12:25 PM
To: Scott Voll
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Proxy

 

Does anyone know if it's possible to run more than one phone behind a
NAT device?

 

I have the proxy working fine if I put the phone (behind a Linksys cable
modem router) on the DMZ.  

But I have some locations that require multiple phones...

 

Steve

 

On 8/14/07, Scott Voll <svoll.voip at gmail.com> wrote: 

Hey--

 

we just got our Phone Proxy setup and it works pretty slick.

 

I think it only took about 4 hours to unpackage, configure, and get the
first Phone up and working.

 

I do however hate Cisco docs as it states you need a null modem.  so I
look high and low and finally find one to find out that in all reality
you really don't need one.  Got to love that.

 

Scott


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

 

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From c3voip at nc.rr.com  Fri Aug 24 14:33:24 2007
From: c3voip at nc.rr.com (c3voip)
Date: Fri, 24 Aug 2007 14:33:24 -0400
Subject: [cisco-voip] non-G.729 VC Unit to 7985 (G.729)
Message-ID: <000601c7e67d$414acc70$c3e06550$@rr.com>

Is there any way to make a low-audio-bandwidth video call between a device
that can only handle G.711, G.722, and G.728 and a Cisco 7985 IP Phone
(which can only handle G.711, G.722, and G.729)?  I have tried using a 2851
as a transcoder but the call completes as a voice call only.

 

Is my only solution to use an MCU?

 

Thanks,

-C 

 

 

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From n68851 at hotmail.com  Fri Aug 24 15:06:25 2007
From: n68851 at hotmail.com (Bryan Nowak)
Date: Fri, 24 Aug 2007 15:06:25 -0400
Subject: [cisco-voip] Hiding DistinctiveRingList.xml
Message-ID: <BAY141-W7A676B4DDBFA59EEAD4AD93D70@phx.gbl>




Hello,
 
Is there a way to manipulate 79xx phones so that different ring lists are visible to different phones?
 
Thanks,
Bryan
 
_________________________________________________________________
Find a local pizza place, movie theater, and more?.then map the best route!
http://maps.live.com/default.aspx?v=2&ss=yp.bars~yp.pizza~yp.movie%20theater&cp=42.358996~-71.056691&style=r&lvl=13&tilt=-90&dir=0&alt=-1000&scene=950607&encType=1&FORM=MGAC01
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From keith.klevenski at rig.net  Fri Aug 24 15:18:20 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 24 Aug 2007 14:18:20 -0500
Subject: [cisco-voip] Max CDR Records
Message-ID: <E0F48CE151B5D440B3649903D7221DA2381CF6@HOUEXH02.rignet.org>

Is there any problem with changing the max CDR records service parameter
to a significantly higher number?  1.5 mil records is about 6 weeks for
us which really isn't enough.  I was wondering why the records only went
back 6 weeks since I have automatic purging set to 365 days, but totally
forgot about the max CDR records parameter.  :P  I'd like to keep a year
if possible which would be about 13 mil records.  The call volume is
only going up so I was thinking about setting it to 15 mil or even 20
and let the automatic purge handle anything over 365 days.  

 

Other than the size of the backup and disk space, is there any reason
not to set this so much higher than the 1.5 mil 'recommended' setting?

 

Thanks.

 

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"

 

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From lelio at uoguelph.ca  Fri Aug 24 15:24:58 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 24 Aug 2007 15:24:58 -0400
Subject: [cisco-voip] Max CDR Records
References: <E0F48CE151B5D440B3649903D7221DA2381CF6@HOUEXH02.rignet.org>
Message-ID: <02fa01c7e684$756e8f80$56196883@cfs.uoguelph.ca>

I heard bad things can happen, but am eager to hear what others say. For us, I think it's only about 2 weeks.

  ----- Original Message ----- 
  From: Keith Klevenski 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, August 24, 2007 3:18 PM
  Subject: [cisco-voip] Max CDR Records


  Is there any problem with changing the max CDR records service parameter to a significantly higher number?  1.5 mil records is about 6 weeks for us which really isn't enough.  I was wondering why the records only went back 6 weeks since I have automatic purging set to 365 days, but totally forgot about the max CDR records parameter.  :P  I'd like to keep a year if possible which would be about 13 mil records.  The call volume is only going up so I was thinking about setting it to 15 mil or even 20 and let the automatic purge handle anything over 365 days.  

   

  Other than the size of the backup and disk space, is there any reason not to set this so much higher than the 1.5 mil 'recommended' setting?

   

  Thanks.

   

  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

   

  "Extending the power of your knowledge"

   



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From LuisYrigoyen at bayviewfinancial.com  Fri Aug 24 15:28:27 2007
From: LuisYrigoyen at bayviewfinancial.com (LuisYrigoyen at bayviewfinancial.com)
Date: Fri, 24 Aug 2007 15:28:27 -0400
Subject: [cisco-voip] IPCC JTAPI Trigger limits
Message-ID: <OF83ADFF98.9A6519B7-ON85257341.006A297E-85257341.006AF9FA@bftg.com>

Hi,
Is there a limit to how many JTAPI Triggers an IPCC Express Enhanced 
server can have? 
Is there a limit on how many triggers in an application?

any info or documentation will be greatly appreciated. 

thank you 


 



Luis Yrigoyen

Network Operations
Bayview Financial, L.P.
4425 Ponce de Leon Blvd.
Coral Gables, FL 33146

T 305.817.5083
F 305.646.4744
E luisyrigoyen at bayviewfinancial.com
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From jason.aarons at us.didata.com  Fri Aug 24 15:32:36 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 24 Aug 2007 15:32:36 -0400
Subject: [cisco-voip] Max CDR Records
In-Reply-To: <E0F48CE151B5D440B3649903D7221DA2381CF6@HOUEXH02.rignet.org>
References: <E0F48CE151B5D440B3649903D7221DA2381CF6@HOUEXH02.rignet.org>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0068CD87E@USNAEXCH.na.didata.local>

I would look at a third-party to archive your CDRs, ever try to upgrade
CallManager versions with 13 mil records in the db ? Could take a day or
more just to run the upgrade..

 

InforTel for Windows has some good features, allows offbox archiving of
cdrs -jason

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Friday, August 24, 2007 3:18 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Max CDR Records

 

Is there any problem with changing the max CDR records service parameter
to a significantly higher number?  1.5 mil records is about 6 weeks for
us which really isn't enough.  I was wondering why the records only went
back 6 weeks since I have automatic purging set to 365 days, but totally
forgot about the max CDR records parameter.  :P  I'd like to keep a year
if possible which would be about 13 mil records.  The call volume is
only going up so I was thinking about setting it to 15 mil or even 20
and let the automatic purge handle anything over 365 days.  

 

Other than the size of the backup and disk space, is there any reason
not to set this so much higher than the 1.5 mil 'recommended' setting?

 

Thanks.

 

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"

 




-----------------------------------------
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 jason.aarons at us.didata.com  Fri Aug 24 15:37:18 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 24 Aug 2007 15:37:18 -0400
Subject: [cisco-voip] Hiding DistinctiveRingList.xml
In-Reply-To: <BAY141-W7A676B4DDBFA59EEAD4AD93D70@phx.gbl>
References: <BAY141-W7A676B4DDBFA59EEAD4AD93D70@phx.gbl>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0068CD89C@USNAEXCH.na.didata.local>

In theory you could use multiple tftp servers running on Windows/Linux,
the phones would have to have their own tftp server to get the
RingList.xml, wav files, etc. Unsupported by Cisco, but might work
-jason

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bryan Nowak
Sent: Friday, August 24, 2007 3:06 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Hiding DistinctiveRingList.xml

 

	Hello,

	 

	Is there a way to manipulate 79xx phones so that different ring
lists are visible to different phones?

	 

	Thanks,

	Bryan

	 

 

________________________________

Find a local pizza place, movie theater, and more....then map the best
route! Find it!
<http://maps.live.com/default.aspx?v=2&ss=yp.bars~yp.pizza~yp.movie%20th
eater&cp=42.358996~-71.056691&style=r&lvl=13&tilt=-90&dir=0&alt=-1000&sc
ene=950607&encType=1&FORM=MGAC01> 




-----------------------------------------
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 svoll.voip at gmail.com  Fri Aug 24 15:47:19 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 24 Aug 2007 12:47:19 -0700
Subject: [cisco-voip] CER two different RP for 911.... why?
Message-ID: <f84a38d30708241247x1ded412v69e14e649ec9b370@mail.gmail.com>

I'm installing CER for the first time and I'm a little confused.

in the admin install doc you setup two different Route Points for 911.  one
in the Phone partition and one in the E911 partition.  both of which are in
the E911 CSS which includes both partitions in the first place.  Can some
one shed some light on this as I'm trying my best to get my head rapped
around this install.

TIA

Scott
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From lelio at uoguelph.ca  Fri Aug 24 15:54:36 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 24 Aug 2007 15:54:36 -0400
Subject: [cisco-voip] Hiding DistinctiveRingList.xml
References: <BAY141-W7A676B4DDBFA59EEAD4AD93D70@phx.gbl>
	<C1FE15183DA37645BC0633BC604E44F0068CD89C@USNAEXCH.na.didata.local>
Message-ID: <036001c7e688$9e557590$56196883@cfs.uoguelph.ca>

It would be really cool to setup a TFTP server that could serve out files based on requesting address.
  ----- Original Message ----- 
  From: Jason Aarons (US) 
  To: Bryan Nowak ; cisco-voip at puck.nether.net 
  Sent: Friday, August 24, 2007 3:37 PM
  Subject: Re: [cisco-voip] Hiding DistinctiveRingList.xml


  In theory you could use multiple tftp servers running on Windows/Linux, the phones would have to have their own tftp server to get the RingList.xml, wav files, etc. Unsupported by Cisco, but might work -jason

   

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bryan Nowak
  Sent: Friday, August 24, 2007 3:06 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Hiding DistinctiveRingList.xml

   

    Hello,

     

    Is there a way to manipulate 79xx phones so that different ring lists are visible to different phones?

     

    Thanks,

    Bryan

     

   


------------------------------------------------------------------------------

  Find a local pizza place, movie theater, and more..then map the best route! Find it!



------------------------------------------------------------------------------


  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 keith.klevenski at rig.net  Fri Aug 24 16:16:02 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 24 Aug 2007 15:16:02 -0500
Subject: [cisco-voip] Max CDR Records
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0068CD87E@USNAEXCH.na.didata.local>
References: <E0F48CE151B5D440B3649903D7221DA2381CF6@HOUEXH02.rignet.org>
	<C1FE15183DA37645BC0633BC604E44F0068CD87E@USNAEXCH.na.didata.local>
Message-ID: <E0F48CE151B5D440B3649903D7221DA2381D07@HOUEXH02.rignet.org>

Actually we are using Infortel.  Can't say I'm a big fan of it, but I
suppose it works.  Actually I don't like it at all.  Actually I don't
like dealing with phone bills at all.  Actually I hate dealing with
phone bills.  

 

Anyway... we upgraded back in March to a version that has a redirected
number report.  One problem is I can't go any further back than March.
I've been asked for a report of calls to the NOC that when to voicemail
starting in December in Excel format (Infortel will only give you a PDF,
HTML, or RTF for this particular report.. LAME).  Silly me, I thought
I've been keeping CDRs for a year when actually I only had 6 weeks worth
so I was just going to run a simple SQL query and give them exactly what
they wanted.  Not.

 

I guess 1.5 mil is the recommended setting for a reason.  20 mil CDRs on
box is silly I guess.  It just forces me to lean more on Infortel which
I do not like.  K

 

 

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: Jason Aarons (US) [mailto:jason.aarons at us.didata.com] 
Sent: Friday, August 24, 2007 2:33 PM
To: Keith Klevenski; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Max CDR Records

 

I would look at a third-party to archive your CDRs, ever try to upgrade
CallManager versions with 13 mil records in the db ? Could take a day or
more just to run the upgrade..

 

InforTel for Windows has some good features, allows offbox archiving of
cdrs -jason

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Friday, August 24, 2007 3:18 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Max CDR Records

 

Is there any problem with changing the max CDR records service parameter
to a significantly higher number?  1.5 mil records is about 6 weeks for
us which really isn't enough.  I was wondering why the records only went
back 6 weeks since I have automatic purging set to 365 days, but totally
forgot about the max CDR records parameter.  :P  I'd like to keep a year
if possible which would be about 13 mil records.  The call volume is
only going up so I was thinking about setting it to 15 mil or even 20
and let the automatic purge handle anything over 365 days.  

 

Other than the size of the backup and disk space, is there any reason
not to set this so much higher than the 1.5 mil 'recommended' setting?

 

Thanks.

 

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"

 

________________________________

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
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From etoro at ctio.noao.edu  Fri Aug 24 15:28:21 2007
From: etoro at ctio.noao.edu (Eduardo Toro R)
Date: Fri, 24 Aug 2007 15:28:21 -0400 (CLT)
Subject: [cisco-voip] Avaya CM and Cisco CallManager Express
Message-ID: <3312.172.16.100.6.1187983701.squirrel@www.ctio.noao.edu>


Hi

 I'm trying to make and H323 trunking connection between and Avaya CM PBX
system and a Cisco Call Manager Express V3.1 . Anyone has some
documentation or examples about this kind of  configuration ?

Eduardo Toro R.
CTIO / CISS
AURA-Chile


From n68851 at hotmail.com  Fri Aug 24 16:22:44 2007
From: n68851 at hotmail.com (Bryan Nowak)
Date: Fri, 24 Aug 2007 16:22:44 -0400
Subject: [cisco-voip] Hiding DistinctiveRingList.xml
Message-ID: <BAY141-W4045A164421EE26D3FDBF893D70@phx.gbl>


Many thanks!


Subject: RE: [cisco-voip] Hiding DistinctiveRingList.xmlDate: Fri, 24 Aug 2007 15:37:18 -0400From: jason.aarons at us.didata.comTo: n68851 at hotmail.com; cisco-voip at puck.nether.net






In theory you could use multiple tftp servers running on Windows/Linux, the phones would have to have their own tftp server to get the RingList.xml, wav files, etc. Unsupported by Cisco, but might work -jason
 


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bryan NowakSent: Friday, August 24, 2007 3:06 PMTo: cisco-voip at puck.nether.netSubject: [cisco-voip] Hiding DistinctiveRingList.xml
 


Hello,
 
Is there a way to manipulate 79xx phones so that different ring lists are visible to different phones?
 
Thanks,
Bryan
 
 



Find a local pizza place, movie theater, and more?.then map the best route! Find it!




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. 
_________________________________________________________________
Find a local pizza place, movie theater, and more?.then map the best route!
http://maps.live.com/default.aspx?v=2&ss=yp.bars~yp.pizza~yp.movie%20theater&cp=42.358996~-71.056691&style=r&lvl=13&tilt=-90&dir=0&alt=-1000&scene=950607&encType=1&FORM=MGAC01
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From cpennington at ncaa.org  Fri Aug 24 16:48:14 2007
From: cpennington at ncaa.org (Pennington, Craig)
Date: Fri, 24 Aug 2007 16:48:14 -0400
Subject: [cisco-voip] Maximum number of 7970's on a 3750-48ps-s
Message-ID: <B0353BDEA8A5A14D8E3CA39AADCA46B606A0149B@NCAA-MAIL.ncaa.org>

I noticed we have reached our maximum power output for the 3750 (540
watts).  I then noticed it is set to the default settings of auto with
15.4 watts max on a port with the switch reporting each 7970 as using
10.3 watts.  I tried to do the power inline auto max XXXXX (where XXXXX
is the number of milli-watts between 4000 and 15400) to lower the max
setting in hopes of increasing the number of ports that would give
power. 540/15.4 gives you a max of 35 powered ports.  And that is what I
have, 35 phones, on each switch and I need to add more.  When I tried to
lower the milli-watts below 15400 the port would say deny power and the
phone would not boot.  has anyone had any luck trying to lower the power
on a 3750 to service more 7970's?
 
Craig Pennington
Network Engineer
National Collegiate Athletic Association
Fax: 317-917-6888
cpennington at ncaa.org <mailto:cpennington at ncaa.org> 
www.ncaa.org <http://www.ncaa.org/> 

This email and any attachments may contain confidential and privileged information.  If you are not the intended recipient, please notify the sender immediately by return email, delete this message and destroy any copies.  Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.
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From joe.j.keller at gmail.com  Fri Aug 24 16:51:18 2007
From: joe.j.keller at gmail.com (Joe Keller)
Date: Fri, 24 Aug 2007 16:51:18 -0400
Subject: [cisco-voip] BARS fails due to SQL Server,
	after OS 4.4aSR9 update
In-Reply-To: <963CE94787005149A52C17F33A028E120A10BDF9@EVS1B.boco.co.boulder.co.us>
References: <189b5daa0708240854l22fe2abbn9e85bdc3c4da5de7@mail.gmail.com>
	<963CE94787005149A52C17F33A028E120A10BDF9@EVS1B.boco.co.boulder.co.us>
Message-ID: <189b5daa0708241351i1d98492er329aa09e6616902a@mail.gmail.com>

2000-4-4a-sr9  	 17-JUL-2007
SR10 came out this week:
2000-4-4a-sr10  	21-AUG-2007
Opening a TAC...

On 8/24/07, Bichler, Jeri <jbichler at co.boulder.co.us> wrote:
>
>
>
> That is BS. TAC, and tell them to test this stuff...  When did SR9 come out
> on the site???   I get all the emails about updates....
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Joe Keller
> Sent: Friday, August 24, 2007 9:54 AM
> To: Cisco-VoIP mailing list
> Subject: [cisco-voip] BARS fails due to SQL Server, after OS 4.4aSR9 update
>
> CCM 4.1(3)
> BARS 4.0.12
> OS was 2000.4.3aSR9 updated to 2000.4.4aSR9. Update log showed no errors.
> BARS fails since update.
>
> Bars status window shows:
>  [02:00:19 AM]  Dumping Ccm0302 database on x.x.x.x
>  *** [02:00:24 AM]  Could not verify db backup.
>  [02:00:24 AM]  Staging database dump from x.x.x.x to staging directory.
>  *** [02:00:24 AM]  Failed to stage the database Ccm0302.
>  *** [02:00:24 AM]  Could not backup database Ccm0302 successfully.
>  [02:00:24 AM]  Staging Locales folder C:\Program
> Files\Cisco\CallManager\DBInstallCSV\Products
>  [02:00:26 AM] Backing up URL Services Page.
>  *** [02:00:26 AM]  Failed to copy URL Services page. Couldn't find IIS Root
> Directory for CCMCIP in x.x.x.x
>  [02:00:26 AM]  Skipping CDR database backup on x.x.x.x.
>  [
> BARS trace log shows:
> [02:00:24 AM]   Could not verify database backup successfully:
> -2147218303    [Mic-
>                 rosoft][ODBC SQL Server Driver][SQL Server]Cannot open
> backup d-
>                 evice 'stiFileDump'. Device error or device off-line. See
> the S-
>                 QL Server error log for more details.
> [Microsoft][ODBC SQL Serv-
>                 er Driver][SQL Server]VERIFY DATABASE is terminating
> abnormally.
>                 .
> [02:00:24 AM]   Could not verify db backup.
> [02:00:24 AM]   Moving dump to staging dir
> [02:00:24 AM]   Failed to stage the database Ccm0302.
> [02:00:24 AM]   Error in modBackup DB: 450    Wrong number of
> arguments or invalid-
>                  property assignment
> [02:00:24 AM]   Could not backup database Ccm0302 successfully.
> [02:00:24 AM]   Staging Locales folder C:\Program
> Files\Cisco\CallManager\DBIns-
>                 tallCSV\Products
> [02:00:24 AM]   Locales Folder Created
> [02:00:24 AM]   In function modStageDir
> [02:00:26 AM]   Successfully backed up locales folder.
> [02:00:26 AM]   The CCM database, DBNAME from registry CCM0302 and the
> Connect-
>                 ion String registry value for DBConnection0 is CCM0302 .
> [02:00:26 AM]   Inside function modGetLocalComputerName
> [02:00:26 AM]   Local Computer name is CCMPUB
> [02:00:26 AM]   Exiting function modGetLocalComputerName
> [02:00:26 AM]   Initializing Sockets
> [02:00:26 AM]   Inside function modGetIpAddress
> [02:00:26 AM]   IPAddress = x.x.x.x
> [02:00:26 AM]   Closing Sockets
> [02:00:26 AM]   Exiting function modGetIpAddress
> [02:00:26 AM]   Succesfully wrote the Publisher details to the catalog.ini
> file.
> [02:00:26 AM]   Inside Function modExecuteQuery
> [02:00:26 AM]
> modExecuteQuery=http://x.x.x.x/CCMCIP/getservicesmenu.asp
> [02:00:26 AM]   Backing up URL Services Page.
> [02:00:26 AM]   Given URL Server: x.x.x.x
> [02:00:26 AM]   Given Backup Server: x.x.x.x
> [02:00:26 AM]   Inside function modVirualDir
> [02:00:26 AM]   [modVirualDir]: -2147463168 Automation error
> [02:00:26 AM]   Failed to copy URL Services page. Couldn't find IIS Root
> Direct-
>                 ory for CCMCIP in x.x.x.x
>
> DBLHelper shows no problems
>
> SQL Server error log shows the following error at 2:00:24:29:
> "BackupDiskFile::OpenMedia: Backup device 'C:\Ccm0302.dmp' failed to open.
> Operating system error = 32(The process cannot access the file because it is
> being used by another process.)."
>
> Any ideas?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From MLinsemier at apcapital.com  Sat Aug 25 02:58:04 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Sat, 25 Aug 2007 02:58:04 -0400
Subject: [cisco-voip] BARS Restore - Failed to drop database CCM0300
Message-ID: <AB0A35F2486F81468971AA68558BE71906D9AD8F@smieln01winm01.micoanet.com>

I am attempting to do a BARS restore on a CallManager 4.1(3) publisher
at this time.  When I run the BARS restore Wizard and enter all of the
information (local file, CCM Database, admin ID & PW) the restore
starts, but I always receive a "failed to drop database CCM0300" in the
logs.  The CDR database is dropped and restored without any issues.
Could this mean I am using the wrong version of BARS that it was backed
up with or is there something else going on?

 

Matt


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From msohail741 at yahoo.co.in  Sat Aug 25 07:37:33 2007
From: msohail741 at yahoo.co.in (Mohammed Sohail)
Date: Sat, 25 Aug 2007 12:37:33 +0100 (BST)
Subject: [cisco-voip] Additional Subscriber
Message-ID: <644502.57471.qm@web7808.mail.in.yahoo.com>

      cisco-voip at puck.nether.net        #message893433304226226360219751084135827 { overflow:auto; visibility:hidden }        HI, I have single cluster of publisher and subscriber currently we have 1,000 IP Phones are all phones are registered to subscriber to off load
  Subscriber we added another subscriber. How do I register my existing 500 phones or new IP Phones to new subscriber?
   
  Regards,
  Sohail



       
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From d4rkf1ber at gmail.com  Sat Aug 25 10:35:39 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Sat, 25 Aug 2007 09:35:39 -0500
Subject: [cisco-voip] Opinion Needed - GUI for CUE VOIP Configuration on ISR
Message-ID: <a1b07ea60708250735i7aabb831xe91f8cd60d7bab7@mail.gmail.com>

I have a friend whose team is working on a GUI for configuring VoIP for CME
on the ISR platform.

He emailed me asking to help identify some workflow related to configuring
VoIP and well its been a while nor was I ever an expert.

If any of you have a moment and want to help out your time would be greatly
appreciated.

I can identify some groupings of some of the tasks, but alot of the stuff
listed we don't use.  I am guessing some items could be just be in like an
optional group if you will.


Here is the main context of the email I recieved:


One of the big taskflow issues of Voip is "what do people want to do
next"? We can figure out great starting points. But the variability in
the potential workflows for configuring Voip are endless.

So, I'm trying to push that we have one workflow that is safe for the
end user and never lands them in a "you can't configure XX yet because
YY has not been configured" situation.

I'm going to list some pieces of Voip configuration below. Can you
reorder these in a sequence or order of operations that you feel would
a) never land the user in the "I can't do this yet" situation and b)
describe how you ordered these items in your personal workflow for
configuring a network back in you Voip days.



-System Defaults (including configuring default device login
information, I'm also going to push for all other system-wide defaults
to appear here instead of their corresponding functional locations, such
as CUE Pin, etc).

-Telephony Service Settings (including Enable FXO hookflash for softkey
templates and enable hunt group logout for softkey templates... More may
be added here, otherwise I may recommend they integrate this as part of
System Defaults).

-Call Handling

-Call Park

-Call Pickup Groups

-Directory Services

-Hunt Groups

-Intercom

-Music On Hold

-Night Service Bell

-Paging Groups

-Voice VLAN

-Phone config

-Softkey template config

-Analog phones/fax config

-Extensions

-Paging Extensions

-Import Bulk Data

-Users

-Voice Mail (VM)

-VM Broadcast Messages

-VM General Delivery Mailboxes

-VM Distribution Lists

-VM Networking Locations

-VM Remote Users

-Trunks

-Trunk Groups

-Incoming Call Dial Plan

-Outgoing Call Dial Plan

-Class of Restriction

-Automated Attendant (AA)

-Automatic Call Distribution

-Incoming Calls Prompts

-Incoming Call Scripts
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From paramds at gmail.com  Sat Aug 25 11:31:43 2007
From: paramds at gmail.com (param singh)
Date: Sat, 25 Aug 2007 10:31:43 -0500
Subject: [cisco-voip] Meeting place express Question.
Message-ID: <867640cd0708250831m7e8bf1d0w748e3310930aa0b0@mail.gmail.com>

Hi All,

I am installing Meeting place express  2.0 for the first time and I had few
questions.
The First problem that I am getting is the feature where the system dials
out and calls the invitees.

I have configured the user profile to have the system call out. The system
can call out to the four digit extension within the company but I am trying
to get the system call out to the someone's cell phone or  call out to the
PSTN and thats not working. Has someone done this. I have already integrated
call manager and MPX. I can calling from outside and i can join the voice
meeting and it works fine.
Also this is meeting place express 2.0 , but we have installed only one
server internally in the network , so basically this is for
internal/external voice conference and internal web conference.



I can call out from the IP phone. Its only the meeting place that does not
call out. Any suggestion please will be highly appreciated.

Thanks
Param
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From aizhar at ccbootcamp.com  Sat Aug 25 11:42:16 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Sat, 25 Aug 2007 08:42:16 -0700
Subject: [cisco-voip] Additional Subscriber
References: <644502.57471.qm@web7808.mail.in.yahoo.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C366356116E@nlimail.nlidomain>

Hi,

It is being done using call manager group and device pools.

You need to create another call manager group and in it put both subscribers, but in the opposite order then it was in the current one.

Then add another device pool which is using that call manager group and assign half of the phones to this device pool.

-------------------------------------------
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 Mohammed Sohail
Sent: Sat 8/25/2007 4:37 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Additional Subscriber [bcc][faked-from][html-rem]
 
      cisco-voip at puck.nether.net        #message893433304226226360219751084135827 { overflow:auto; visibility:hidden }        HI, I have single cluster of publisher and subscriber currently we have 1,000 IP Phones are all phones are registered to subscriber to off load
  Subscriber we added another subscriber. How do I register my existing 500 phones or new IP Phones to new subscriber?
   
  Regards,
  Sohail



       
---------------------------------
 Once upon a time there was 1 GB storage in your inbox. Click here for happy ending.
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From dwolgas1 at rochester.rr.com  Sat Aug 25 16:13:34 2007
From: dwolgas1 at rochester.rr.com (Dave Wolgast)
Date: Sat, 25 Aug 2007 16:13:34 -0400
Subject: [cisco-voip] Meeting place express Question.
In-Reply-To: <867640cd0708250831m7e8bf1d0w748e3310930aa0b0@mail.gmail.com>
References: <867640cd0708250831m7e8bf1d0w748e3310930aa0b0@mail.gmail.com>
Message-ID: <e4491a9c0708251313h6d754675ld71d460e1b9d185@mail.gmail.com>

On 8/25/07, param singh <paramds at gmail.com> wrote:

> I have configured the user profile to have the system call out. The system
> can call out to the four digit extension within the company but I am trying
> to get the system call out to the someone's cell phone or  call out to the
> PSTN and thats not working.


 If you have the MPX set up as an H323 gateway, check the outbound Calling
Search Space on the gateway config in CallManager, and make sure it has
permissions to call PSTN destinations.  That has caught me more than once
(actually, probably got me when I was integrating MPX).

Dave Wolgast
Inergex, Inc.
Livonia, NY
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From paramds at gmail.com  Sat Aug 25 16:29:29 2007
From: paramds at gmail.com (param singh)
Date: Sat, 25 Aug 2007 15:29:29 -0500
Subject: [cisco-voip] Meeting place express Question.
In-Reply-To: <e4491a9c0708251313h6d754675ld71d460e1b9d185@mail.gmail.com>
References: <867640cd0708250831m7e8bf1d0w748e3310930aa0b0@mail.gmail.com>
	<e4491a9c0708251313h6d754675ld71d460e1b9d185@mail.gmail.com>
Message-ID: <867640cd0708251329h210e768cwa707effa813fd614@mail.gmail.com>

Great . thanks Dave It worked.. i was using the wrong CSS.. Thanks again.

On 8/25/07, Dave Wolgast <dwolgas1 at rochester.rr.com> wrote:
>
> On 8/25/07, param singh <paramds at gmail.com> wrote:
>
> > I have configured the user profile to have the system call out. The
> > system can call out to the four digit extension within the company but I am
> > trying to get the system call out to the someone's cell phone or  call out
> > to the PSTN and thats not working.
>
>
>  If you have the MPX set up as an H323 gateway, check the outbound Calling
> Search Space on the gateway config in CallManager, and make sure it has
> permissions to call PSTN destinations.  That has caught me more than once
> (actually, probably got me when I was integrating MPX).
>
> Dave Wolgast
> Inergex, Inc.
> Livonia, NY
>
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From Andrew.Riley at getronics.com  Sun Aug 26 08:20:45 2007
From: Andrew.Riley at getronics.com (Riley, Andrew)
Date: Sun, 26 Aug 2007 22:20:45 +1000
Subject: [cisco-voip] Unity Voicemail Only failover architecture
	confirmation with Exchange.
Message-ID: <A49371C6FF101B46BD3E3A8C89985F0201471C@excausy200.australia.unity>

Hi Guys, 
 
I know this topic has been discussed but I could not find the complete
answer from the available posts.
 
I am working on an opportunity for a 5000 seat VM only solution and
would like to know what is the recommended server numbers for a failover
Unity solution with redundant AD/GC and Exchange.
 
>From my research I believe I need the following.
 
2x Unity servers
2x AD/GC servers
2x Exchange servers
 
1) Is the hardware list sufficient to achieve my requirements.
2) Does the 'shipped' exchange version from Cisco support exchange
clustering to achieve the fail over of exchange?
3) Does the 'shipped' windows 2003 version support the exchange
clustering guidelines as outlined by MS.
4) Do I need a 'front end server' Iam not running OWA etc..
5) Is there an alternate or recommended way to achieve my requirements?
 
Cheers, I appreciate any input.
 
Andrew Riley
 

Andrew Riley
Technical Consultant - Communication Services
CCIE # 18285 Voice 

Getronics Australia Pty Limited
Getronics combines the service capabilities of the original Dutch
company with those of Wang Global, acquired in 1999, and of the Olivetti
systems and services division. We are ranked second worldwide in network
and desktop outsourcing and fourth in network consulting and integration
(Source: IDC 2002-2003).

2 Minna Close
Belrose NSW 2085
Australia
Tel:       +61 2 9847 7309
Fax:      +61 2 9847 7378
Mobile:  +61 402 894 793
Email:    andrew.riley at getronics.com

Communication Services | Security Services | Application Services |
Workspace Management Services | Technology Transformation Services

www.getronics.com.au <http://www.getronics.com.au/>  

 

Please note that whilst we take all care, neither Getronics nor the
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From jason.aarons at us.didata.com  Sun Aug 26 08:54:29 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sun, 26 Aug 2007 08:54:29 -0400
Subject: [cisco-voip] Platform OS ftp update
Message-ID: <C1FE15183DA37645BC0633BC604E44F0068CDC22@USNAEXCH.na.didata.local>

I'm doing my first platform ftp upgrade (Mobility Manager 1.2.1 to
1.2.5a) and getting errors, is there a text log file I can view the ftp
details?

 




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From tednugent69 at yahoo.com  Sun Aug 26 08:34:23 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sun, 26 Aug 2007 05:34:23 -0700 (PDT)
Subject: [cisco-voip] CER two different RP for 911.... why?
In-Reply-To: <f84a38d30708241247x1ded412v69e14e649ec9b370@mail.gmail.com>
Message-ID: <47166.98342.qm@web36104.mail.mud.yahoo.com>

One is the RP for Primary CER server, the second one
"912" or whatever is for the standby server in the
event that the primary is down, on the primary 911 RP
you're setting CFwdB, CFwdNA and CFwdF to 912 and
allowing it visibility to the E911CSS. This will send
the call to the standby server in the event the
primary is not reachable. 
Ted



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

> I'm installing CER for the first time and I'm a
> little confused.
> 
> in the admin install doc you setup two different
> Route Points for 911.  one
> in the Phone partition and one in the E911
> partition.  both of which are in
> the E911 CSS which includes both partitions in the
> first place.  Can some
> one shed some light on this as I'm trying my best to
> get my head rapped
> around this install.
> 
> TIA
> 
> Scott
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
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From tednugent69 at yahoo.com  Sun Aug 26 10:46:32 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sun, 26 Aug 2007 07:46:32 -0700 (PDT)
Subject: [cisco-voip] Platform OS ftp update
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0068CDC22@USNAEXCH.na.didata.local>
Message-ID: <529741.98291.qm@web36115.mail.mud.yahoo.com>

I just ran into similar issues the otherday and I also
could not find any logs on CMM but the logs on the ftp
server were enough to track it down eventually.
My issue was that CMM wants a .gz file extension and I
was changing it to a .sgn like I would with CM5.x.
Not sure if thats your issue but figured I'd share
anyway, next time I'm just burning a DVD.


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

> I'm doing my first platform ftp upgrade (Mobility
> Manager 1.2.1 to
> 1.2.5a) and getting errors, is there a text log file
> I can view the ftp
> details?
> 
>  
> 
> 
> 
> 
> -----------------------------------------
> 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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Check out fun summer activities for kids.
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From milay at cisco.com  Sun Aug 26 10:50:13 2007
From: milay at cisco.com (Mike Lay (milay))
Date: Sun, 26 Aug 2007 07:50:13 -0700
Subject: [cisco-voip] Platform OS ftp update
In-Reply-To: <529741.98291.qm@web36115.mail.mud.yahoo.com>
References: <C1FE15183DA37645BC0633BC604E44F0068CDC22@USNAEXCH.na.didata.local>
	<529741.98291.qm@web36115.mail.mud.yahoo.com>
Message-ID: <413FABF5DCE22849AF86F442E472B78B0288BBC7@xmb-sjc-219.amer.cisco.com>

CSCsh20023  

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
Sent: Sunday, August 26, 2007 9:47 AM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Platform OS ftp update

I just ran into similar issues the otherday and I also could not find
any logs on CMM but the logs on the ftp server were enough to track it
down eventually.
My issue was that CMM wants a .gz file extension and I was changing it
to a .sgn like I would with CM5.x.
Not sure if thats your issue but figured I'd share anyway, next time I'm
just burning a DVD.


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

> I'm doing my first platform ftp upgrade (Mobility Manager 1.2.1 to
> 1.2.5a) and getting errors, is there a text log file I can view the 
> ftp details?
> 
>  
> 
> 
> 
> 
> -----------------------------------------
> 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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From tednugent69 at yahoo.com  Sun Aug 26 11:01:45 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sun, 26 Aug 2007 08:01:45 -0700 (PDT)
Subject: [cisco-voip] Platform OS ftp update
In-Reply-To: <413FABF5DCE22849AF86F442E472B78B0288BBC7@xmb-sjc-219.amer.cisco.com>
Message-ID: <287884.97191.qm@web36110.mail.mud.yahoo.com>

Thanks Mike
I'm very much aware of this bug however my issue was
that CMM apparently wants a .gz file extension where
CCM wants a .sgn file extension so I was treating a
CMM upgrade in the same fashion as a CCM upgrade and
that would not work. Renaming the file to
"cisco-ipt-k9-patch1.2.5.2000-1.tar.gz" worked for
me...


--- "Mike Lay (milay)" <milay at cisco.com> wrote:

> CSCsh20023  
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ted Nugent
> Sent: Sunday, August 26, 2007 9:47 AM
> To: Jason Aarons (US); cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Platform OS ftp update
> 
> I just ran into similar issues the otherday and I
> also could not find
> any logs on CMM but the logs on the ftp server were
> enough to track it
> down eventually.
> My issue was that CMM wants a .gz file extension and
> I was changing it
> to a .sgn like I would with CM5.x.
> Not sure if thats your issue but figured I'd share
> anyway, next time I'm
> just burning a DVD.
> 
> 
> --- "Jason Aarons (US)" <jason.aarons at us.didata.com>
> wrote:
> 
> > I'm doing my first platform ftp upgrade (Mobility
> Manager 1.2.1 to
> > 1.2.5a) and getting errors, is there a text log
> file I can view the 
> > ftp details?
> > 
> >  
> > 
> > 
> > 
> > 
> > -----------------------------------------
> > 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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>        
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> ____________
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> Check out fun summer activities for kids.
>
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From wsisk at cisco.com  Sun Aug 26 13:22:21 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Sun, 26 Aug 2007 13:22:21 -0400
Subject: [cisco-voip] Hiding DistinctiveRingList.xml
In-Reply-To: <BAY141-W7A676B4DDBFA59EEAD4AD93D70@phx.gbl>
References: <BAY141-W7A676B4DDBFA59EEAD4AD93D70@phx.gbl>
Message-ID: <39D090E9-6364-451A-B4D2-DF650B90B721@cisco.com>

point different phones to different TFTP servers in your CM cluster.   
you will 'odd' behavior if you configure tftp redundancy (option 150  
as an array), but cosmetically it may server your purpose.

/Wes

On Aug 24, 2007, at 3:06 PM, Bryan Nowak wrote:

Hello,


Is there a way to manipulate 79xx phones so that different ring lists  
are visible to different phones?


Thanks,

Bryan



Find a local pizza place, movie theater, and more?.then map the best  
route! Find it!
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From wsisk at cisco.com  Sun Aug 26 13:24:52 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Sun, 26 Aug 2007 13:24:52 -0400
Subject: [cisco-voip] BARS Restore - Failed to drop database CCM0300
In-Reply-To: <AB0A35F2486F81468971AA68558BE71906D9AD8F@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE71906D9AD8F@smieln01winm01.micoanet.com>
Message-ID: <019F49FF-001C-47EE-95A3-0348A08934A6@cisco.com>

turn off all 3rd party monitoring apps like infortel, prognosis,  
etc.  they make queries to the database that will keep it open and  
locked.

worst case scenario use 'sp_who2' and 'sp_lock' to see what apps have  
open sessions to the database.

/wes

On Aug 25, 2007, at 2:58 AM, Linsemier, Matthew wrote:

I am attempting to do a BARS restore on a CallManager 4.1(3)  
publisher at this time.  When I run the BARS restore Wizard and enter  
all of the information (local file, CCM Database, admin ID & PW) the  
restore starts, but I always receive a ?failed to drop database  
CCM0300? in the logs.  The CDR database is dropped and restored  
without any issues.  Could this mean I am using the wrong version of  
BARS that it was backed up with or is there something else going on?



Matt


CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may be  
protected by one or more legal privileges. It is intended solely for  
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From jason.aarons at us.didata.com  Sun Aug 26 13:41:18 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sun, 26 Aug 2007 13:41:18 -0400
Subject: [cisco-voip] Platform OS ftp update
In-Reply-To: <413FABF5DCE22849AF86F442E472B78B0288BBC7@xmb-sjc-219.amer.cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F0068CDC22@USNAEXCH.na.didata.local>
	<529741.98291.qm@web36115.mail.mud.yahoo.com>
	<413FABF5DCE22849AF86F442E472B78B0288BBC7@xmb-sjc-219.amer.cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0068CDC4E@USNAEXCH.na.didata.local>

Any way to view a log file on what with ftp is wrong? I do miss Network
Monitor/snoop/being able to troubleshoot problems with calling tac
<grin> -jason

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Sunday, August 26, 2007 10:50 AM
To: Ted Nugent; Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Platform OS ftp update

CSCsh20023  

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
Sent: Sunday, August 26, 2007 9:47 AM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Platform OS ftp update

I just ran into similar issues the otherday and I also could not find
any logs on CMM but the logs on the ftp server were enough to track it
down eventually.
My issue was that CMM wants a .gz file extension and I was changing it
to a .sgn like I would with CM5.x.
Not sure if thats your issue but figured I'd share anyway, next time I'm
just burning a DVD.


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

> I'm doing my first platform ftp upgrade (Mobility Manager 1.2.1 to
> 1.2.5a) and getting errors, is there a text log file I can view the 
> ftp details?
> 
>  
> 
> 
> 
> 
> -----------------------------------------
> 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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Disclaimer:

This e-mail communication and any attachments may contain
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From tednugent69 at yahoo.com  Sun Aug 26 14:29:02 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sun, 26 Aug 2007 11:29:02 -0700 (PDT)
Subject: [cisco-voip] Platform OS ftp update
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0068CDC4E@USNAEXCH.na.didata.local>
Message-ID: <433714.86786.qm@web36107.mail.mud.yahoo.com>

I'd recommend looking on the console of the ftp/sftp
server, I use FileZilla
(http://filezilla.sourceforge.net/) even the basic
settings are probably verbose enough to figure it out.


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

> Any way to view a log file on what with ftp is
> wrong? I do miss Network
> Monitor/snoop/being able to troubleshoot problems
> with calling tac
> <grin> -jason
> 
> -----Original Message-----
> From: Mike Lay (milay) [mailto:milay at cisco.com] 
> Sent: Sunday, August 26, 2007 10:50 AM
> To: Ted Nugent; Jason Aarons (US);
> cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Platform OS ftp update
> 
> CSCsh20023  
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ted Nugent
> Sent: Sunday, August 26, 2007 9:47 AM
> To: Jason Aarons (US); cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Platform OS ftp update
> 
> I just ran into similar issues the otherday and I
> also could not find
> any logs on CMM but the logs on the ftp server were
> enough to track it
> down eventually.
> My issue was that CMM wants a .gz file extension and
> I was changing it
> to a .sgn like I would with CM5.x.
> Not sure if thats your issue but figured I'd share
> anyway, next time I'm
> just burning a DVD.
> 
> 
> --- "Jason Aarons (US)" <jason.aarons at us.didata.com>
> wrote:
> 
> > I'm doing my first platform ftp upgrade (Mobility
> Manager 1.2.1 to
> > 1.2.5a) and getting errors, is there a text log
> file I can view the 
> > ftp details?
> > 
> >  
> > 
> > 
> > 
> > 
> > -----------------------------------------
> > 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
> 
> 
> 
>        
>
________________________________________________________________________
> ____________
> Got a little couch potato? 
> Check out fun summer activities for kids.
>
http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+ki
> ds&cs=bz
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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.
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> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
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From MLinsemier at apcapital.com  Sun Aug 26 19:40:10 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Sun, 26 Aug 2007 19:40:10 -0400
Subject: [cisco-voip] BARS Restore - Failed to drop database CCM0300
In-Reply-To: <019F49FF-001C-47EE-95A3-0348A08934A6@cisco.com>
References: <AB0A35F2486F81468971AA68558BE71906D9AD8F@smieln01winm01.micoanet.com>
	<019F49FF-001C-47EE-95A3-0348A08934A6@cisco.com>
Message-ID: <AB0A35F2486F81468971AA68558BE71906D9AE2F@smieln01winm01.micoanet.com>

I wanted to let everyone know that the problem was resolved when I
installed BARS 4.0(9).  This may have been the version that I was using
to make backups all the time, I just couldn't check on the failed hard
disk.  BARS restored the data quickly and flawlessly for both CCM and
CRS.  This is the first time (and hopefully the last) where I had to
restore a publisher.  It's nice to know that the utilities that Cisco
provides do work.  Thanks again for everyone's suggestions and help.

 

Matt

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Sunday, August 26, 2007 1:25 PM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] BARS Restore - Failed to drop database CCM0300

 

turn off all 3rd party monitoring apps like infortel, prognosis, etc.
they make queries to the database that will keep it open and locked.

 

worst case scenario use 'sp_who2' and 'sp_lock' to see what apps have
open sessions to the database.

 

/wes

 

On Aug 25, 2007, at 2:58 AM, Linsemier, Matthew wrote:

 

I am attempting to do a BARS restore on a CallManager 4.1(3) publisher
at this time.  When I run the BARS restore Wizard and enter all of the
information (local file, CCM Database, admin ID & PW) the restore
starts, but I always receive a "failed to drop database CCM0300" in the
logs.  The CDR database is dropped and restored without any issues.
Could this mean I am using the wrong version of BARS that it was backed
up with or is there something else going on?

 

Matt

________________________________

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

 


CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may
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are not the intended recipient, any use, disclosure, copying
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From bhenry33 at adelphia.net  Sun Aug 26 20:37:45 2007
From: bhenry33 at adelphia.net (Brian E. Henry)
Date: Sun, 26 Aug 2007 20:37:45 -0400
Subject: [cisco-voip] Cisco IP Phone Login Software?
Message-ID: <19816981.1188175065060.JavaMail.root@web40>

I saw in the forum about a couple of months back but cannot find the exact 
thread on a software that when upoon loggining into your desktop it would log 
the phone in Extension Mobility at the same time. 
 
Does anyone know of this vendor? 
 
Brian 


From wsisk at cisco.com  Sun Aug 26 21:39:13 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Sun, 26 Aug 2007 21:39:13 -0400
Subject: [cisco-voip] Platform OS ftp update
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0068CDC4E@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0068CDC22@USNAEXCH.na.didata.local>
	<529741.98291.qm@web36115.mail.mud.yahoo.com>
	<413FABF5DCE22849AF86F442E472B78B0288BBC7@xmb-sjc-219.amer.cisco.com>
	<C1FE15183DA37645BC0633BC604E44F0068CDC4E@USNAEXCH.na.didata.local>
Message-ID: <0923B71C-6D43-423F-B38C-E15990515355@cisco.com>

not user if this made it into the mobility mgr platform before it was  
merged back into CM natively (6.0) but:

utils network

CLI offers a capture option.

/Wes

On Aug 26, 2007, at 1:41 PM, Jason Aarons ((US)) wrote:

Any way to view a log file on what with ftp is wrong? I do miss Network
Monitor/snoop/being able to troubleshoot problems with calling tac
<grin> -jason

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com]
Sent: Sunday, August 26, 2007 10:50 AM
To: Ted Nugent; Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Platform OS ftp update

CSCsh20023

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
Sent: Sunday, August 26, 2007 9:47 AM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Platform OS ftp update

I just ran into similar issues the otherday and I also could not find
any logs on CMM but the logs on the ftp server were enough to track it
down eventually.
My issue was that CMM wants a .gz file extension and I was changing it
to a .sgn like I would with CM5.x.
Not sure if thats your issue but figured I'd share anyway, next time I'm
just burning a DVD.


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

> I'm doing my first platform ftp upgrade (Mobility Manager 1.2.1 to
> 1.2.5a) and getting errors, is there a text log file I can view the
> ftp details?
>
>
>
>
>
>
> -----------------------------------------
> 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




________________________________________________________________________
____________
Got a little couch potato?
Check out fun summer activities for kids.
http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+ki
ds&cs=bz
_______________________________________________
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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
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From erickbee at gmail.com  Sun Aug 26 22:20:30 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Sun, 26 Aug 2007 21:20:30 -0500
Subject: [cisco-voip] Max CDR Records
In-Reply-To: <E0F48CE151B5D440B3649903D7221DA2381D07@HOUEXH02.rignet.org>
References: <E0F48CE151B5D440B3649903D7221DA2381CF6@HOUEXH02.rignet.org>
	<C1FE15183DA37645BC0633BC604E44F0068CD87E@USNAEXCH.na.didata.local>
	<E0F48CE151B5D440B3649903D7221DA2381D07@HOUEXH02.rignet.org>
Message-ID: <f4445faf0708261920x5ba19408k3b3f985735c11fb5@mail.gmail.com>

You may want to consider putting the CDR database on it's own server
in the cluster that just does CDR jobs if it's going to be big.  A few
other things I can think of with it being a bigger size are the
nightly CDR/CAR jobs may take longer to run, backups will be larger if
you backup CDR, CAR, etc.

On 8/24/07, Keith Klevenski <keith.klevenski at rig.net> wrote:
>
>
>
>
> Actually we are using Infortel.  Can't say I'm a big fan of it, but I
> suppose it works.  Actually I don't like it at all.  Actually I don't like
> dealing with phone bills at all.  Actually I hate dealing with phone bills.
>
>
>
> Anyway? we upgraded back in March to a version that has a redirected number
> report.  One problem is I can't go any further back than March.  I've been
> asked for a report of calls to the NOC that when to voicemail starting in
> December in Excel format (Infortel will only give you a PDF, HTML, or RTF
> for this particular report.. LAME).  Silly me, I thought I've been keeping
> CDRs for a year when actually I only had 6 weeks worth so I was just going
> to run a simple SQL query and give them exactly what they wanted.  Not.
>
>
>
> I guess 1.5 mil is the recommended setting for a reason.  20 mil CDRs on box
> is silly I guess.  It just forces me to lean more on Infortel which I do not
> like.  K
>
>
>
>
>
>
> 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
>
>
>
> "Extending the power of your knowledge"
>
>
>
>
>
> From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
>  Sent: Friday, August 24, 2007 2:33 PM
>  To: Keith Klevenski; cisco-voip at puck.nether.net
>  Subject: RE: [cisco-voip] Max CDR Records
>
>
>
>
> I would look at a third-party to archive your CDRs, ever try to upgrade
> CallManager versions with 13 mil records in the db ? Could take a day or
> more just to run the upgrade..
>
>
>
> InforTel for Windows has some good features, allows offbox archiving of cdrs
> -jason
>
>
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Keith Klevenski
>  Sent: Friday, August 24, 2007 3:18 PM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] Max CDR Records
>
>
>
> Is there any problem with changing the max CDR records service parameter to
> a significantly higher number?  1.5 mil records is about 6 weeks for us
> which really isn't enough.  I was wondering why the records only went back 6
> weeks since I have automatic purging set to 365 days, but totally forgot
> about the max CDR records parameter.  :P  I'd like to keep a year if
> possible which would be about 13 mil records.  The call volume is only going
> up so I was thinking about setting it to 15 mil or even 20 and let the
> automatic purge handle anything over 365 days.
>
>
>
> Other than the size of the backup and disk space, is there any reason not to
> set this so much higher than the 1.5 mil 'recommended' setting?
>
>
>
> Thanks.
>
>
>
> 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
>
>
>
> "Extending the power of your knowledge"
>
>
>  ________________________________
>
>
> 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 thomas.hartley at gmail.com  Sun Aug 26 23:05:06 2007
From: thomas.hartley at gmail.com (Thomas Hartley)
Date: Mon, 27 Aug 2007 13:05:06 +1000
Subject: [cisco-voip] Cisco IP Phone Login Software?
In-Reply-To: <19816981.1188175065060.JavaMail.root@web40>
References: <19816981.1188175065060.JavaMail.root@web40>
Message-ID: <db324e8c0708262005u50e6e325hb1170823e488ec3a@mail.gmail.com>

This might be what you're looking for:


http://www.rsconnect.net/

Automatic Login Module




On 8/27/07, Brian E. Henry <bhenry33 at adelphia.net> wrote:
>
> I saw in the forum about a couple of months back but cannot find the exact
> thread on a software that when upoon loggining into your desktop it would
> log
> the phone in Extension Mobility at the same time.
>
> Does anyone know of this vendor?
>
> Brian
>
> _______________________________________________
> 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 Aug 27 00:17:11 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 26 Aug 2007 23:17:11 -0500
Subject: [cisco-voip] Is this Unity's death knell?
Message-ID: <5d093f9a0708262117p99d07d0wc8dfe04695fd7b2@mail.gmail.com>

http://www.microsoft.com/exchange/evaluation/features/unifiedmessaging.mspx

Speech-Enabled Auto Attendant

The Attendant answers calls using an automated operator with
customizable menus (for example, "press 1 for sales") and global
address list directory lookups (for example, "who would you like to
contact?"). Callers can interact with the Automated Attendant through
touchtone menus or their voice using speech recognition.

I think the integrated speed recognition is pretty cool, and as soon
as you play with Outlook Voice Access, Unity looks kind of sad in
comparison...






Jonathan

From jeff at ocjtech.us  Mon Aug 27 01:46:32 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Mon, 27 Aug 2007 00:46:32 -0500
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <5d093f9a0708262117p99d07d0wc8dfe04695fd7b2@mail.gmail.com>
References: <5d093f9a0708262117p99d07d0wc8dfe04695fd7b2@mail.gmail.com>
Message-ID: <1188193592.13352.18.camel@lt21223.campus.dmacc.edu>

On Sun, 2007-08-26 at 23:17 -0500, Jonathan Charles wrote:
> http://www.microsoft.com/exchange/evaluation/features/unifiedmessaging.mspx
> 
> Speech-Enabled Auto Attendant
> 
> The Attendant answers calls using an automated operator with
> customizable menus (for example, "press 1 for sales") and global
> address list directory lookups (for example, "who would you like to
> contact?"). Callers can interact with the Automated Attendant through
> touchtone menus or their voice using speech recognition.
> 
> I think the integrated speed recognition is pretty cool, and as soon
> as you play with Outlook Voice Access, Unity looks kind of sad in
> comparison...

The "cool" features may be nice to give to your "C" level users, but the
killer for me would be letting the in-house Exchange admins handle the
voicemail/auto-attendants on their (existing) highly-available Exchange
clusters, rather than depending on my single Unity server.  That way I
can also stop being an Exchange administrator myself (which I have
neither the training or temperament for).

One thing I noticed from [1]:
        Exchange 2007 Unified Messaging has been tested for use with
        Cisco Call Manager version 5.x. If you have Cisco Call Manager
        5.x on your network, you will be able to connect Exchange 2007
        Unified Messaging servers directly to Call Manager without the
        requirement of an IP/VoIP gateway. Many of the features that are
        included with Exchange 2007 Unified Messaging are fully
        functional with Cisco Call Manager 5.x. However, the Message
        Waiting Indicator (MWI) feature does not work and faxing does
        not work because the T.38 faxing protocol is not supported by
        Cisco Call Manager.

T.38 faxing I think I could do without, but I think I'd be in danger of
losing my job if MWI stopped working.  However, according to the
document that you can download from [2] there's a third-party
solution[3] that enables MWI.

[1] http://technet.microsoft.com/en-us/library/2516dac1-dfdc-47eb-8e6f-18b1537a57b2.aspx
[2] http://www.microsoft.com/downloads/details.aspx?FamilyId=68b43d3c-7c84-4c2f-bfd7-98754970d70e&displaylang=en
[3] http://www.mwi2007.com/

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From WOCAMPO at broward.org  Mon Aug 27 08:57:41 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Mon, 27 Aug 2007 08:57:41 -0400
Subject: [cisco-voip] IPCC JTAPI Trigger limits
In-Reply-To: <OF83ADFF98.9A6519B7-ON85257341.006A297E-85257341.006AF9FA@bftg.com>
References: <OF83ADFF98.9A6519B7-ON85257341.006A297E-85257341.006AF9FA@bftg.com>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA4B2@UVEXVS01.bc.broward.cty>

Yes, there is a limit depending of the agents you need.  For example a
7845 server dual processor can handle 150/300* logged in agents
*deployments>150 require Avg. Call handle time> 3 minutes.  While the
7835 single processor can handle 75. 

Here is a Cisco table  that you can play around and it will tell you the
type of server you need.  

 

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> 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
LuisYrigoyen at bayviewfinancial.com
Sent: Friday, August 24, 2007 3:28 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC JTAPI Trigger limits

 


Hi, 
Is there a limit to how many JTAPI Triggers an IPCC Express Enhanced
server can have? 
Is there a limit on how many triggers in an application? 

any info or documentation will be greatly appreciated. 

thank you 


  



Luis Yrigoyen

Network Operations
Bayview Financial, L.P.
4425 Ponce de Leon Blvd.
Coral Gables, FL 33146

T 305.817.5083
F 305.646.4744
E luisyrigoyen at bayviewfinancial.com

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From rsingleton at morsco.com  Mon Aug 27 10:05:32 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Mon, 27 Aug 2007 09:05:32 -0500
Subject: [cisco-voip] Max CDR Records
In-Reply-To: <E0F48CE151B5D440B3649903D7221DA2381D07@HOUEXH02.rignet.org>
References: <E0F48CE151B5D440B3649903D7221DA2381CF6@HOUEXH02.rignet.org>
	<C1FE15183DA37645BC0633BC604E44F0068CD87E@USNAEXCH.na.didata.local>
	<E0F48CE151B5D440B3649903D7221DA2381D07@HOUEXH02.rignet.org>
Message-ID: <1188223532.4879.16.camel@robert.morsco.com>

On Fri, 2007-08-24 at 15:16 -0500, Keith Klevenski wrote:

>  Silly me, I thought I?ve been keeping CDRs for a year when actually I
> only had 6 weeks worth so I was just going to run a simple SQL query
> and give them exactly what they wanted.  Not.
> 
>  
> 
> I guess 1.5 mil is the recommended setting for a reason.  20 mil CDRs
> on box is silly I guess.  It just forces me to lean more on Infortel
> which I do not like.  K

I faced a similar problem and only recently started doing something I
should have been doing all along, archiving my CDRs on another server.

I have not as yet written a script to put them on another SQL, but what
I'm archiving is the results of a "select * from CallDetailRecord" and
"select * from CallDetailRecordDiagnostic" daily. The cron job runs at
04:00 each day and most days are about 3.5MB compressed. I started the
automatic process on Jun 1 and manually archived what was existing at
that time, so I have daily captures back to April 18.

I also found it handy to know what range of CDRs are available and wrote
a quick script that gets min(dateTimeOrigination) and
max(dateTimeOrigination) and returns them in human readable format.

Robert



From jonvoip at gmail.com  Mon Aug 27 11:09:45 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 27 Aug 2007 10:09:45 -0500
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <1188193592.13352.18.camel@lt21223.campus.dmacc.edu>
References: <5d093f9a0708262117p99d07d0wc8dfe04695fd7b2@mail.gmail.com>
	<1188193592.13352.18.camel@lt21223.campus.dmacc.edu>
Message-ID: <5d093f9a0708270809s2108bc7as96084ecb4056ed93@mail.gmail.com>

I am sure someone will come out with a freeware app that does MWI...

However, it probably won't be able to tag just voicemails and will
light MWI on all inbox items.

The reality is that, in a unified messaging environment, most people
don't use their phones to retrieve vmail anyway.

I think a business argument could almost be made to forgo MWI and
issue everyone speakers for their PCs.



J

On 8/27/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote:
> On Sun, 2007-08-26 at 23:17 -0500, Jonathan Charles wrote:
> > http://www.microsoft.com/exchange/evaluation/features/unifiedmessaging.mspx
> >
> > Speech-Enabled Auto Attendant
> >
> > The Attendant answers calls using an automated operator with
> > customizable menus (for example, "press 1 for sales") and global
> > address list directory lookups (for example, "who would you like to
> > contact?"). Callers can interact with the Automated Attendant through
> > touchtone menus or their voice using speech recognition.
> >
> > I think the integrated speed recognition is pretty cool, and as soon
> > as you play with Outlook Voice Access, Unity looks kind of sad in
> > comparison...
>
> The "cool" features may be nice to give to your "C" level users, but the
> killer for me would be letting the in-house Exchange admins handle the
> voicemail/auto-attendants on their (existing) highly-available Exchange
> clusters, rather than depending on my single Unity server.  That way I
> can also stop being an Exchange administrator myself (which I have
> neither the training or temperament for).
>
> One thing I noticed from [1]:
>         Exchange 2007 Unified Messaging has been tested for use with
>         Cisco Call Manager version 5.x. If you have Cisco Call Manager
>         5.x on your network, you will be able to connect Exchange 2007
>         Unified Messaging servers directly to Call Manager without the
>         requirement of an IP/VoIP gateway. Many of the features that are
>         included with Exchange 2007 Unified Messaging are fully
>         functional with Cisco Call Manager 5.x. However, the Message
>         Waiting Indicator (MWI) feature does not work and faxing does
>         not work because the T.38 faxing protocol is not supported by
>         Cisco Call Manager.
>
> T.38 faxing I think I could do without, but I think I'd be in danger of
> losing my job if MWI stopped working.  However, according to the
> document that you can download from [2] there's a third-party
> solution[3] that enables MWI.
>
> [1] http://technet.microsoft.com/en-us/library/2516dac1-dfdc-47eb-8e6f-18b1537a57b2.aspx
> [2] http://www.microsoft.com/downloads/details.aspx?FamilyId=68b43d3c-7c84-4c2f-bfd7-98754970d70e&displaylang=en
> [3] http://www.mwi2007.com/
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From ealeatherman at gmail.com  Mon Aug 27 11:33:13 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 27 Aug 2007 11:33:13 -0400
Subject: [cisco-voip] OT: Call Center web-based IM
Message-ID: <94a1afde0708270833j66674d58kfedf91b92a8b0577@mail.gmail.com>

Anyone use or see a customer that has an online chat system setup for
customer support, in conjunction with IPCC Express? We have a small contact
center that is interested in making live chat sessions available in addition
to their normal 800 number for support and new student information requests.

There are some logistics that they'll need to work out with staffing and who
is taking voice calls vs IM, etc.. i'm just trying to find out what is out
there product-wise for a small scenario (10 agents would be maximum). Maybe
a hosted solution of some sort?

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From keith.klevenski at rig.net  Mon Aug 27 12:29:30 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Mon, 27 Aug 2007 11:29:30 -0500
Subject: [cisco-voip] Max CDR Records
In-Reply-To: <1188223532.4879.16.camel@robert.morsco.com>
References: <E0F48CE151B5D440B3649903D7221DA2381CF6@HOUEXH02.rignet.org><C1FE15183DA37645BC0633BC604E44F0068CD87E@USNAEXCH.na.didata.local><E0F48CE151B5D440B3649903D7221DA2381D07@HOUEXH02.rignet.org>
	<1188223532.4879.16.camel@robert.morsco.com>
Message-ID: <E0F48CE151B5D440B3649903D7221DA2381DE1@HOUEXH02.rignet.org>

Yea, that sounds like a good idea.  I may consider doing something like that as it is awful nice to have the actual CDRs available if you need them rather than a 3rd party's version of them.

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

"Extending the power of your knowledge"


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert Singleton
Sent: Monday, August 27, 2007 9:06 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Max CDR Records

On Fri, 2007-08-24 at 15:16 -0500, Keith Klevenski wrote:

>  Silly me, I thought I?ve been keeping CDRs for a year when actually I
> only had 6 weeks worth so I was just going to run a simple SQL query
> and give them exactly what they wanted.  Not.
> 
>  
> 
> I guess 1.5 mil is the recommended setting for a reason.  20 mil CDRs
> on box is silly I guess.  It just forces me to lean more on Infortel
> which I do not like.  K

I faced a similar problem and only recently started doing something I
should have been doing all along, archiving my CDRs on another server.

I have not as yet written a script to put them on another SQL, but what
I'm archiving is the results of a "select * from CallDetailRecord" and
"select * from CallDetailRecordDiagnostic" daily. The cron job runs at
04:00 each day and most days are about 3.5MB compressed. I started the
automatic process on Jun 1 and manually archived what was existing at
that time, so I have daily captures back to April 18.

I also found it handy to know what range of CDRs are available and wrote
a quick script that gets min(dateTimeOrigination) and
max(dateTimeOrigination) and returns them in human readable format.

Robert


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

From Louis.Koekemoer at za.didata.com  Mon Aug 27 09:25:48 2007
From: Louis.Koekemoer at za.didata.com (Louis Koekemoer (ZA))
Date: Mon, 27 Aug 2007 15:25:48 +0200
Subject: [cisco-voip] MOH server not registering
Message-ID: <90A8622407907445BEE753AC922C82D402A13D42@ZABRYSVCL17A.af.didata.local>

I have a customer that has CallManager 3.3. They are planning new
equipment and an upgrade, but only late this year or possible early next
year. Their current problem is that their software Music-on-hold server
does not register. I have even deleted and recreate it with the exact
same name, but nothing. I have tried to stop and start the "Cisco IP
Voice Media Streaming App" under services which should automatically
create this. No success. Customer is rebooting server tonight to see if
we get any further. 

 

Anyone with any ideas?

 

Kind Regards

 

Louis Koekemoer 
Cisco IPT Systems Engineer

Dimension Data South Africa 

* +27 11 575 6560

*  +27 76 608 7221 

* louis.koekemoer at za.didata.com <mailto:louis.koekemoer at za.didata.com> 

 


This email and all contents are subject to the following disclaimer:

"http://www.dimensiondata.com/emaildisclaimer.htm"
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From smgustafson at gmail.com  Mon Aug 27 13:30:07 2007
From: smgustafson at gmail.com (Steve G)
Date: Mon, 27 Aug 2007 10:30:07 -0700
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <5d093f9a0708270809s2108bc7as96084ecb4056ed93@mail.gmail.com>
References: <5d093f9a0708262117p99d07d0wc8dfe04695fd7b2@mail.gmail.com>
	<1188193592.13352.18.camel@lt21223.campus.dmacc.edu>
	<5d093f9a0708270809s2108bc7as96084ecb4056ed93@mail.gmail.com>
Message-ID: <8a1f34b50708271030r83fc010jf3fa820c188a461c@mail.gmail.com>

With Exchange 2007, does it allow you to listen to your VM through the
phone?  Like with Unity ViewMail you can click play and it will ring your
extension which will then play your message to you after you pick-up.

On 8/27/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> I am sure someone will come out with a freeware app that does MWI...
>
> However, it probably won't be able to tag just voicemails and will
> light MWI on all inbox items.
>
> The reality is that, in a unified messaging environment, most people
> don't use their phones to retrieve vmail anyway.
>
> I think a business argument could almost be made to forgo MWI and
> issue everyone speakers for their PCs.
>
>
>
> J
>
> On 8/27/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote:
> > On Sun, 2007-08-26 at 23:17 -0500, Jonathan Charles wrote:
> > >
> http://www.microsoft.com/exchange/evaluation/features/unifiedmessaging.mspx
> > >
> > > Speech-Enabled Auto Attendant
> > >
> > > The Attendant answers calls using an automated operator with
> > > customizable menus (for example, "press 1 for sales") and global
> > > address list directory lookups (for example, "who would you like to
> > > contact?"). Callers can interact with the Automated Attendant through
> > > touchtone menus or their voice using speech recognition.
> > >
> > > I think the integrated speed recognition is pretty cool, and as soon
> > > as you play with Outlook Voice Access, Unity looks kind of sad in
> > > comparison...
> >
> > The "cool" features may be nice to give to your "C" level users, but the
> > killer for me would be letting the in-house Exchange admins handle the
> > voicemail/auto-attendants on their (existing) highly-available Exchange
> > clusters, rather than depending on my single Unity server.  That way I
> > can also stop being an Exchange administrator myself (which I have
> > neither the training or temperament for).
> >
> > One thing I noticed from [1]:
> >         Exchange 2007 Unified Messaging has been tested for use with
> >         Cisco Call Manager version 5.x. If you have Cisco Call Manager
> >         5.x on your network, you will be able to connect Exchange 2007
> >         Unified Messaging servers directly to Call Manager without the
> >         requirement of an IP/VoIP gateway. Many of the features that are
> >         included with Exchange 2007 Unified Messaging are fully
> >         functional with Cisco Call Manager 5.x. However, the Message
> >         Waiting Indicator (MWI) feature does not work and faxing does
> >         not work because the T.38 faxing protocol is not supported by
> >         Cisco Call Manager.
> >
> > T.38 faxing I think I could do without, but I think I'd be in danger of
> > losing my job if MWI stopped working.  However, according to the
> > document that you can download from [2] there's a third-party
> > solution[3] that enables MWI.
> >
> > [1]
> http://technet.microsoft.com/en-us/library/2516dac1-dfdc-47eb-8e6f-18b1537a57b2.aspx
> > [2]
> http://www.microsoft.com/downloads/details.aspx?FamilyId=68b43d3c-7c84-4c2f-bfd7-98754970d70e&displaylang=en
> > [3] http://www.mwi2007.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 jason.aarons at us.didata.com  Mon Aug 27 14:08:05 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 27 Aug 2007 14:08:05 -0400
Subject: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy
Message-ID: <C1FE15183DA37645BC0633BC604E44F00692A219@USNAEXCH.na.didata.local>

I see a Hunt Pilot 3695 pointed to Hunt List, which points to Line Group
with 2 members (8307,8338) in broadcast.  Hunt List is reset, shows
good, reset a couple more times for good luck.  All CSS have correct
partitions, etc. No shared lines anywhere. No duplicate numbers on Hunt
Pilot or unassigned dns, etc.

 

I get fast busy dialing Hunt Pilot 3695. I tried changing 3695 to 5555
as we have no numbers beginning with 5....reset Hunt List, still fast
busy...

 

I've set this up hundreds of times with no problems, maybe a bug with
4.2.3sr2b?

 




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

This e-mail communication and any attachments may contain
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From svoll.voip at gmail.com  Mon Aug 27 14:22:11 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 27 Aug 2007 11:22:11 -0700
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <8a1f34b50708271030r83fc010jf3fa820c188a461c@mail.gmail.com>
References: <5d093f9a0708262117p99d07d0wc8dfe04695fd7b2@mail.gmail.com>
	<1188193592.13352.18.camel@lt21223.campus.dmacc.edu>
	<5d093f9a0708270809s2108bc7as96084ecb4056ed93@mail.gmail.com>
	<8a1f34b50708271030r83fc010jf3fa820c188a461c@mail.gmail.com>
Message-ID: <f84a38d30708271122q756be48ew988c40bf1a2d2f45@mail.gmail.com>

yes.  it does ring your phone if desired.  I do have to admit, for once M$
has done a pretty good job at a first run application.  it will also play
your email, and calandar with out any extra licenses....... Cisco, are you
paying attention?

Scott


On 8/27/07, Steve G <smgustafson at gmail.com> wrote:
>
> With Exchange 2007, does it allow you to listen to your VM through the
> phone?  Like with Unity ViewMail you can click play and it will ring your
> extension which will then play your message to you after you pick-up.
>
> On 8/27/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > I am sure someone will come out with a freeware app that does MWI...
> >
> > However, it probably won't be able to tag just voicemails and will
> > light MWI on all inbox items.
> >
> > The reality is that, in a unified messaging environment, most people
> > don't use their phones to retrieve vmail anyway.
> >
> > I think a business argument could almost be made to forgo MWI and
> > issue everyone speakers for their PCs.
> >
> >
> >
> > J
> >
> > On 8/27/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote:
> > > On Sun, 2007-08-26 at 23:17 -0500, Jonathan Charles wrote:
> > > >
> > http://www.microsoft.com/exchange/evaluation/features/unifiedmessaging.mspx
> > > >
> > > > Speech-Enabled Auto Attendant
> > > >
> > > > The Attendant answers calls using an automated operator with
> > > > customizable menus (for example, "press 1 for sales") and global
> > > > address list directory lookups (for example, "who would you like to
> > > > contact?"). Callers can interact with the Automated Attendant
> > through
> > > > touchtone menus or their voice using speech recognition.
> > > >
> > > > I think the integrated speed recognition is pretty cool, and as soon
> >
> > > > as you play with Outlook Voice Access, Unity looks kind of sad in
> > > > comparison...
> > >
> > > The "cool" features may be nice to give to your "C" level users, but
> > the
> > > killer for me would be letting the in-house Exchange admins handle the
> >
> > > voicemail/auto-attendants on their (existing) highly-available
> > Exchange
> > > clusters, rather than depending on my single Unity server.  That way I
> > > can also stop being an Exchange administrator myself (which I have
> > > neither the training or temperament for).
> > >
> > > One thing I noticed from [1]:
> > >         Exchange 2007 Unified Messaging has been tested for use with
> > >         Cisco Call Manager version 5.x. If you have Cisco Call Manager
> >
> > >         5.x on your network, you will be able to connect Exchange 2007
> > >         Unified Messaging servers directly to Call Manager without the
> > >         requirement of an IP/VoIP gateway. Many of the features that
> > are
> > >         included with Exchange 2007 Unified Messaging are fully
> > >         functional with Cisco Call Manager 5.x. However, the Message
> > >         Waiting Indicator (MWI) feature does not work and faxing does
> > >         not work because the T.38 faxing protocol is not supported by
> > >         Cisco Call Manager.
> > >
> > > T.38 faxing I think I could do without, but I think I'd be in danger
> > of
> > > losing my job if MWI stopped working.  However, according to the
> > > document that you can download from [2] there's a third-party
> > > solution[3] that enables MWI.
> > >
> > > [1]
> > http://technet.microsoft.com/en-us/library/2516dac1-dfdc-47eb-8e6f-18b1537a57b2.aspx
> > > [2]
> > http://www.microsoft.com/downloads/details.aspx?FamilyId=68b43d3c-7c84-4c2f-bfd7-98754970d70e&displaylang=en
> > > [3] http://www.mwi2007.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
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From reillyipt at gmail.com  Mon Aug 27 14:41:56 2007
From: reillyipt at gmail.com (Mike Reilly)
Date: Mon, 27 Aug 2007 14:41:56 -0400
Subject: [cisco-voip] CallManager Express b acd tcl ivr script
Message-ID: <C2F89334.4AF%reillyipt@gmail.com>


I am trying to do something different with the CME B ACD tcl script. Call
comes into main line 555-1212 rings several phones, no answer or busy
forward to a number 1880.  Set up a dial peer
Dial-peer voice 1881 voip
Service aa
Destination-pattern 1880
Session target ipv4:10.100.100.1
Incoming called-number 1880
Dtmf-relay h245-alphanumeric
Codec g711ulaw
No vad

The above dial peer will trigger the b acd application.  Does anyone know if
this will work?  Remember the 1880 is going to be a forward destination
under the dn.



From jgrace at adcapnet.com  Mon Aug 27 14:43:36 2007
From: jgrace at adcapnet.com (James Grace)
Date: Mon, 27 Aug 2007 14:43:36 -0400
Subject: [cisco-voip] Im getting a Phone Unprovisioned  Error on my 7940
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C756@mail.adcapnetsvc.local>

?
 
can someone tell me why im getting the below error
Im getting a Phone Unprovisioned  Error on my 7940
 
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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From wsisk at cisco.com  Mon Aug 27 16:02:18 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 27 Aug 2007 16:02:18 -0400
Subject: [cisco-voip] MOH server not registering
In-Reply-To: <90A8622407907445BEE753AC922C82D402A13D42@ZABRYSVCL17A.af.didata.local>
References: <90A8622407907445BEE753AC922C82D402A13D42@ZABRYSVCL17A.af.didata.local>
Message-ID: <05A02FA4-DCDA-49DB-AE02-22D57ABF3406@cisco.com>

In the service parameters for IPVMSApp make sure the run flag for MoH  
is set true.  After that, do you see a registration error in the  
event viewer logs or CM traces?  what device pool is configured for  
the MoH server?  Are the CM servers in that device pool/cm group online?

/Wes

On Aug 27, 2007, at 9:25 AM, Louis Koekemoer ((ZA)) wrote:

I have a customer that has CallManager 3.3. They are planning new  
equipment and an upgrade, but only late this year or possible early  
next year. Their current problem is that their software Music-on-hold  
server does not register. I have even deleted and recreate it with  
the exact same name, but nothing. I have tried to stop and start the  
?Cisco IP Voice Media Streaming App? under services which should  
automatically create this. No success. Customer is rebooting server  
tonight to see if we get any further.



Anyone with any ideas?



Kind Regards



Louis Koekemoer
Cisco IPT Systems Engineer

Dimension Data South Africa

( +27 11 575 6560

)  +27 76 608 7221

* louis.koekemoer at za.didata.com



This email and all contents are subject to the following disclaimer:

"http://www.dimensiondata.com/emaildisclaimer.htm"

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From wsisk at cisco.com  Mon Aug 27 16:04:20 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 27 Aug 2007 16:04:20 -0400
Subject: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00692A219@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00692A219@USNAEXCH.na.didata.local>
Message-ID: <5A883DF8-4479-4985-8A5D-277292B5D7EE@cisco.com>

are users configured to login/logout of hunt group?  are users logged  
in?
hunt list actually registers, does CM perfmon counter increment/show  
huntlist registered?

Any snippet of CM SDI traces?  Those should be pretty decent for  
showing the problem, especially if attempt a call from the same CM  
node where the hunt pilot registers.

/Wes

On Aug 27, 2007, at 2:08 PM, Jason Aarons ((US)) wrote:

I see a Hunt Pilot 3695 pointed to Hunt List, which points to Line  
Group with 2 members (8307,8338) in broadcast.  Hunt List is reset,  
shows good, reset a couple more times for good luck.  All CSS have  
correct partitions, etc. No shared lines anywhere. No duplicate  
numbers on Hunt Pilot or unassigned dns, etc.



I get fast busy dialing Hunt Pilot 3695. I tried changing 3695 to  
5555 as we have no numbers beginning with 5....reset Hunt List, still  
fast busy?



I?ve set this up hundreds of times with no problems, maybe a bug with  
4.2.3sr2b?





Disclaimer: This e-mail communication and any attachments may contain  
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or its contents is strictly prohibited and may be unlawful. If you  
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https://puck.nether.net/mailman/listinfo/cisco-voip


From wsisk at cisco.com  Mon Aug 27 16:05:50 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 27 Aug 2007 16:05:50 -0400
Subject: [cisco-voip] Im getting a Phone Unprovisioned Error on my 7940
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C756@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C756@mail.adcapnetsvc.local>
Message-ID: <07E82E3F-929F-4FB2-A168-5B3730D64EE0@cisco.com>

sounds like your phone is running a SIP load and is not configured in  
the CM server you are pointing to?  is the phone really configured in  
CM?  what load is it running?  you've been around this list and CM  
for a while now, what normal registration troubleshooting have you  
performed?

/Wes

On Aug 27, 2007, at 2:43 PM, James Grace wrote:

?

can someone tell me why im getting the below error
Im getting a Phone Unprovisioned  Error on my 7940

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 27 Aug  
2007 14:42:46 -0400
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From dchamberlain at unislumin.com  Mon Aug 27 16:19:07 2007
From: dchamberlain at unislumin.com (Devin Chamberlain)
Date: Mon, 27 Aug 2007 16:19:07 -0400
Subject: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy
In-Reply-To: <mailman.12921.1188244981.1529.cisco-voip@puck.nether.net>
References: <mailman.12921.1188244981.1529.cisco-voip@puck.nether.net>
Message-ID: <18082B35CF0EBD4C8D265874508A615C06C67380@hermes.unislumin.com>

I have found that in the past with that version of call manager that if
you have another line on the phones that you are trying to get in the
hunt group you will get a fast busy.  If that is the case make sure that
there is on only one line on those phones.


-----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: Monday, August 27, 2007 2:03 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 54, Issue 121

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	cisco-voip at puck.nether.net

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Today's Topics:

   1. 4.2.3sr2b Hunt Pilot rings fast busy (Jason Aarons (US))
   2. Re: Is this Unity's death knell? (Scott Voll)
   3. CallManager Express b acd tcl ivr script (Mike Reilly)
   4. Im getting a Phone Unprovisioned  Error on my 7940 (James Grace)
   5. Re: MOH server not registering (Wes Sisk)


----------------------------------------------------------------------

Message: 1
Date: Mon, 27 Aug 2007 14:08:05 -0400
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
Subject: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy
To: <cisco-voip at puck.nether.net>
Message-ID:
	
<C1FE15183DA37645BC0633BC604E44F00692A219 at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="us-ascii"

I see a Hunt Pilot 3695 pointed to Hunt List, which points to Line Group
with 2 members (8307,8338) in broadcast.  Hunt List is reset, shows
good, reset a couple more times for good luck.  All CSS have correct
partitions, etc. No shared lines anywhere. No duplicate numbers on Hunt
Pilot or unassigned dns, etc.

 

I get fast busy dialing Hunt Pilot 3695. I tried changing 3695 to 5555
as we have no numbers beginning with 5....reset Hunt List, still fast
busy...

 

I've set this up hundreds of times with no problems, maybe a bug with
4.2.3sr2b?

 




-----------------------------------------
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------------------------------

Message: 2
Date: Mon, 27 Aug 2007 11:22:11 -0700
From: "Scott Voll" <svoll.voip at gmail.com>
Subject: Re: [cisco-voip] Is this Unity's death knell?
To: "Steve G" <smgustafson at gmail.com>
Cc: cisco voip list <cisco-voip at puck.nether.net>
Message-ID:
	<f84a38d30708271122q756be48ew988c40bf1a2d2f45 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

yes.  it does ring your phone if desired.  I do have to admit, for once
M$
has done a pretty good job at a first run application.  it will also
play
your email, and calandar with out any extra licenses....... Cisco, are
you
paying attention?

Scott


On 8/27/07, Steve G <smgustafson at gmail.com> wrote:
>
> With Exchange 2007, does it allow you to listen to your VM through the
> phone?  Like with Unity ViewMail you can click play and it will ring
your
> extension which will then play your message to you after you pick-up.
>
> On 8/27/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > I am sure someone will come out with a freeware app that does MWI...
> >
> > However, it probably won't be able to tag just voicemails and will
> > light MWI on all inbox items.
> >
> > The reality is that, in a unified messaging environment, most people
> > don't use their phones to retrieve vmail anyway.
> >
> > I think a business argument could almost be made to forgo MWI and
> > issue everyone speakers for their PCs.
> >
> >
> >
> > J
> >
> > On 8/27/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote:
> > > On Sun, 2007-08-26 at 23:17 -0500, Jonathan Charles wrote:
> > > >
> >
http://www.microsoft.com/exchange/evaluation/features/unifiedmessaging.m
spx
> > > >
> > > > Speech-Enabled Auto Attendant
> > > >
> > > > The Attendant answers calls using an automated operator with
> > > > customizable menus (for example, "press 1 for sales") and global
> > > > address list directory lookups (for example, "who would you like
to
> > > > contact?"). Callers can interact with the Automated Attendant
> > through
> > > > touchtone menus or their voice using speech recognition.
> > > >
> > > > I think the integrated speed recognition is pretty cool, and as
soon
> >
> > > > as you play with Outlook Voice Access, Unity looks kind of sad
in
> > > > comparison...
> > >
> > > The "cool" features may be nice to give to your "C" level users,
but
> > the
> > > killer for me would be letting the in-house Exchange admins handle
the
> >
> > > voicemail/auto-attendants on their (existing) highly-available
> > Exchange
> > > clusters, rather than depending on my single Unity server.  That
way I
> > > can also stop being an Exchange administrator myself (which I have
> > > neither the training or temperament for).
> > >
> > > One thing I noticed from [1]:
> > >         Exchange 2007 Unified Messaging has been tested for use
with
> > >         Cisco Call Manager version 5.x. If you have Cisco Call
Manager
> >
> > >         5.x on your network, you will be able to connect Exchange
2007
> > >         Unified Messaging servers directly to Call Manager without
the
> > >         requirement of an IP/VoIP gateway. Many of the features
that
> > are
> > >         included with Exchange 2007 Unified Messaging are fully
> > >         functional with Cisco Call Manager 5.x. However, the
Message
> > >         Waiting Indicator (MWI) feature does not work and faxing
does
> > >         not work because the T.38 faxing protocol is not supported
by
> > >         Cisco Call Manager.
> > >
> > > T.38 faxing I think I could do without, but I think I'd be in
danger
> > of
> > > losing my job if MWI stopped working.  However, according to the
> > > document that you can download from [2] there's a third-party
> > > solution[3] that enables MWI.
> > >
> > > [1]
> >
http://technet.microsoft.com/en-us/library/2516dac1-dfdc-47eb-8e6f-18b15
37a57b2.aspx
> > > [2]
> >
http://www.microsoft.com/downloads/details.aspx?FamilyId=68b43d3c-7c84-4
c2f-bfd7-98754970d70e&displaylang=en
> > > [3] http://www.mwi2007.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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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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------------------------------

Message: 3
Date: Mon, 27 Aug 2007 14:41:56 -0400
From: Mike Reilly <reillyipt at gmail.com>
Subject: [cisco-voip] CallManager Express b acd tcl ivr script
To: <cisco-voip at puck.nether.net>
Message-ID: <C2F89334.4AF%reillyipt at gmail.com>
Content-Type: text/plain;	charset="US-ASCII"


I am trying to do something different with the CME B ACD tcl script.
Call
comes into main line 555-1212 rings several phones, no answer or busy
forward to a number 1880.  Set up a dial peer
Dial-peer voice 1881 voip
Service aa
Destination-pattern 1880
Session target ipv4:10.100.100.1
Incoming called-number 1880
Dtmf-relay h245-alphanumeric
Codec g711ulaw
No vad

The above dial peer will trigger the b acd application.  Does anyone
know if
this will work?  Remember the 1880 is going to be a forward destination
under the dn.




------------------------------

Message: 4
Date: Mon, 27 Aug 2007 14:43:36 -0400
From: "James Grace" <jgrace at adcapnet.com>
Subject: [cisco-voip] Im getting a Phone Unprovisioned  Error on my
	7940
To: <cisco-voip at puck.nether.net>
Message-ID:
	<ECEA9223F645834FB6A65B7CE377B9CD11C756 at mail.adcapnetsvc.local>
Content-Type: text/plain; charset="iso-8859-1"

?
 
can someone tell me why im getting the below error
Im getting a Phone Unprovisioned  Error on my 7940
 
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 27 Aug 2007
14:42:46 -0400
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------------------------------

Message: 5
Date: Mon, 27 Aug 2007 16:02:18 -0400
From: Wes Sisk <wsisk at cisco.com>
Subject: Re: [cisco-voip] MOH server not registering
To: Louis Koekemoer ((ZA)) <Louis.Koekemoer at za.didata.com>
Cc: cisco-voip at puck.nether.net
Message-ID: <05A02FA4-DCDA-49DB-AE02-22D57ABF3406 at cisco.com>
Content-Type: text/plain; charset="windows-1252"

In the service parameters for IPVMSApp make sure the run flag for MoH  
is set true.  After that, do you see a registration error in the  
event viewer logs or CM traces?  what device pool is configured for  
the MoH server?  Are the CM servers in that device pool/cm group online?

/Wes

On Aug 27, 2007, at 9:25 AM, Louis Koekemoer ((ZA)) wrote:

I have a customer that has CallManager 3.3. They are planning new  
equipment and an upgrade, but only late this year or possible early  
next year. Their current problem is that their software Music-on-hold  
server does not register. I have even deleted and recreate it with  
the exact same name, but nothing. I have tried to stop and start the  
?Cisco IP Voice Media Streaming App? under services which should  
automatically create this. No success. Customer is rebooting server  
tonight to see if we get any further.



Anyone with any ideas?



Kind Regards



Louis Koekemoer
Cisco IPT Systems Engineer

Dimension Data South Africa

( +27 11 575 6560

)  +27 76 608 7221

* louis.koekemoer at za.didata.com



This email and all contents are subject to the following disclaimer:

"http://www.dimensiondata.com/emaildisclaimer.htm"

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End of cisco-voip Digest, Vol 54, Issue 121
*******************************************
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 mmilanovic at ctiusa.com  Mon Aug 27 19:11:14 2007
From: mmilanovic at ctiusa.com (Mladen Milanovic)
Date: Mon, 27 Aug 2007 19:11:14 -0400
Subject: [cisco-voip] [WARNING : MESSAGE ENCRYPTED] Re: IPCC JTAPI
	Trigger limits
In-Reply-To: <F4DD4283A3EF194B84D774C60FB7F786024DA4B2@UVEXVS01.bc.broward.cty>
References: <OF83ADFF98.9A6519B7-ON85257341.006A297E-85257341.006AF9FA@bftg.com>
	<F4DD4283A3EF194B84D774C60FB7F786024DA4B2@UVEXVS01.bc.broward.cty>
Message-ID: <4761C60AD1E74A4D8606CAF48582265ACC03CA@EXCHANGE.ctiusa.com>

Per CCE and CCM SRND:
 
You can configure a maximum of 2500 JTAPI monitored devices (agent
phones, CTI Route Points,

or CTI ports) with a maximum of 15,000 BHCA per MCS-7845 server model
with battery-backed

write cache (BBWC). This means a maximum of 10,000 JTAPI monitored
devices with a maximum

of 60K BHCA per cluster. The BHCA maximum assumes a consistent call
arrival of 4.16 calls per

second (CPS) during the busy hour, with only small variations or peaks
(no more than 20% peak or

5 CPS) around the average calls per second.

 

Your CCM system (how cluster is organized) will determine max number of
JTAPI monitored devices. 

 

Mladen


________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ocampo, Walter
Sent: Monday, August 27, 2007 8:58 AM
To: LuisYrigoyen at bayviewfinancial.com; cisco-voip at puck.nether.net
Subject: [WARNING : MESSAGE ENCRYPTED] Re: [cisco-voip] IPCC JTAPI
Trigger limits



Yes, there is a limit depending of the agents you need.  For example a
7845 server dual processor can handle 150/300* logged in agents
*deployments>150 require Avg. Call handle time> 3 minutes.  While the
7835 single processor can handle 75. 

Here is a Cisco table  that you can play around and it will tell you the
type of server you need.  

 

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> 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
LuisYrigoyen at bayviewfinancial.com
Sent: Friday, August 24, 2007 3:28 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC JTAPI Trigger limits

 


Hi, 
Is there a limit to how many JTAPI Triggers an IPCC Express Enhanced
server can have? 
Is there a limit on how many triggers in an application? 

any info or documentation will be greatly appreciated. 

thank you 


  



Luis Yrigoyen

Network Operations
Bayview Financial, L.P.
4425 Ponce de Leon Blvd.
Coral Gables, FL 33146

T 305.817.5083
F 305.646.4744
E luisyrigoyen at bayviewfinancial.com

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From jonvoip at gmail.com  Tue Aug 28 01:28:59 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 28 Aug 2007 00:28:59 -0500
Subject: [cisco-voip] VPN Connects to destination network, but can't ping,
	telnet, or connect to servers...
Message-ID: <5d093f9a0708272228g382f774cr68a265260037fca4@mail.gmail.com>

So, I have a customer I support, I can VPN into their network (Cisco
VPN client, Cisco 3030 VPN Concentrator), but once I connect, I can't
do anything, can't telnet, can't ping, can't connect to shares,
nothing...

However, if I bypass my firewall (ASA 5520), and connect directly to
the internet, I can VPN in and everything works...

I have noticed this a bunch of times, with a lot of customers.

Why does this happen?




Jonathan

From nikola at worldnet.att.net  Tue Aug 28 01:37:03 2007
From: nikola at worldnet.att.net (Nikola Stojsin)
Date: Tue, 28 Aug 2007 01:37:03 -0400
Subject: [cisco-voip] VPN Connects to destination network, but can't ping,
	telnet, or connect to servers...
In-Reply-To: <5d093f9a0708272228g382f774cr68a265260037fca4@mail.gmail.com>
References: <5d093f9a0708272228g382f774cr68a265260037fca4@mail.gmail.com>
Message-ID: <003701c7e935$7787b770$6e9ca8c0@T42p>

One possible scenario is same internal subnets in both networks.

HTH,
Nikola

 
 
------------------------------
Nikola Stojsin
PhD CCIE #12888
Nikola at att.net
(917) 558-1423
(917) 591-9382 (fax)
------------------------------
 
 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Tuesday, August 28, 2007 1:29 AM
To: cisco voip list
Subject: [cisco-voip] VPN Connects to destination network, but can't
ping,telnet, or connect to servers...

So, I have a customer I support, I can VPN into their network (Cisco
VPN client, Cisco 3030 VPN Concentrator), but once I connect, I can't
do anything, can't telnet, can't ping, can't connect to shares,
nothing...

However, if I bypass my firewall (ASA 5520), and connect directly to
the internet, I can VPN in and everything works...

I have noticed this a bunch of times, with a lot of customers.

Why does this happen?




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

From ErasmuE4 at telkom.co.za  Tue Aug 28 02:02:20 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Tue, 28 Aug 2007 08:02:20 +0200
Subject: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00692A219@USNAEXCH.na.didata.local>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F019853A0@CNTRRA20-XCS01.telkom.co.za>


I also had problems with the attendant Q and hunt operation in
4-2-3-sr1.  Now on SR2b. So far the AC queuing and hunt groups /
pilots works better but there is a lot of funnies still in general. I
heard that SR3 is coming in October and hope it will make the solution
more stable overall. 

 

Sorry - can not give you any specific advise. Have you tried using a
different hunting algorithm

 

Erik Erasmus

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Monday, August 27, 2007 8:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy

 

I see a Hunt Pilot 3695 pointed to Hunt List, which points to Line
Group with 2 members (8307,8338) in broadcast.  Hunt List is reset,
shows good, reset a couple more times for good luck.  All CSS have
correct partitions, etc. No shared lines anywhere. No duplicate
numbers on Hunt Pilot or unassigned dns, etc.

 

I get fast busy dialing Hunt Pilot 3695. I tried changing 3695 to 5555
as we have no numbers beginning with 5....reset Hunt List, still fast
busy...

 

I've set this up hundreds of times with no problems, maybe a bug with
4.2.3sr2b?

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
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From Matthew.Baumgartner at getronics.com  Tue Aug 28 02:05:20 2007
From: Matthew.Baumgartner at getronics.com (Baumgartner, Matthew)
Date: Tue, 28 Aug 2007 16:05:20 +1000
Subject: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy
References: <472E195E692EE74BAE4E7B2880A78C1F019853A0@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <A49371C6FF101B46BD3E3A8C89985F0224BF22@excausy200.australia.unity>

Hi Guys,
 
You need to do an update / reset on the lines (8307,8338) that are on the phones within the hunt group.
This should fix the issue.
 
Hope this helps.
 
Matt Baumgartner
Senior Technical Consultant 
Communication Services 
CCIE #14535 Voice

Getronics Australia Pty Limited 
Getronics combines the service capabilities of the original Dutch company with those of Wang Global, acquired in 1999, and of the Olivetti systems and services division. We are ranked second worldwide in network and desktop outsourcing and fourth in network consulting and integration (Source: IDC 2002-2003).

2 Minna Close 
Belrose NSW 2085 
Australia 
T: +61 2 9847 7662 
F: +61 2 9847 7157 
M: + 61 411 126 709 
Email: matthew.baumgartner at getronics.com <mailto:matthew.baumgartner at getronics.com> 

Communication Services | ICT Security | Network Integration Services | Network & Desktop Outsourcing | Application Integration & Management
www.getronics.com.au <http://www.getronics.com.au/> 

Please note that whilst we take all care, neither Getronics nor the sender accepts any responsibility for viruses and it is your responsibility to scan for viruses. The contents are intended only for use by the addressee and may contain confidential and/or privileged material and any use by other than the intended recipient is prohibited. If you received this in error, please inform the sender and/or addressee immediately and delete the material.
 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Erik Erasmus (E)
Sent: Tue 28/08/2007 4:02 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy



I also had problems with the attendant Q and hunt operation in 4-2-3-sr1.  Now on SR2b. So far the AC queuing and hunt groups / pilots works better but there is a lot of funnies still in general. I heard that SR3 is coming in October and hope it will make the solution more stable overall. 

 

Sorry ? can not give you any specific advise. Have you tried using a different hunting algorithm

 

Erik Erasmus

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (US)
Sent: Monday, August 27, 2007 8:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 4.2.3sr2b Hunt Pilot rings fast busy

 

I see a Hunt Pilot 3695 pointed to Hunt List, which points to Line Group with 2 members (8307,8338) in broadcast.  Hunt List is reset, shows good, reset a couple more times for good luck.  All CSS have correct partitions, etc. No shared lines anywhere. No duplicate numbers on Hunt Pilot or unassigned dns, etc.

 

I get fast busy dialing Hunt Pilot 3695. I tried changing 3695 to 5555 as we have no numbers beginning with 5....reset Hunt List, still fast busy?

 

I?ve set this up hundreds of times with no problems, maybe a bug with 4.2.3sr2b?

 

________________________________

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 tele at plexialab.it  Tue Aug 28 02:43:25 2007
From: tele at plexialab.it (tele)
Date: Tue, 28 Aug 2007 08:43:25 +0200
Subject: [cisco-voip] CallManager Express b acd tcl ivr script
In-Reply-To: <C2F89334.4AF%reillyipt@gmail.com>
References: <C2F89334.4AF%reillyipt@gmail.com>
Message-ID: <1188283405.6257.60.camel@mda>

Hi,

Yes this works, remember to set the correct pilot number for the AA.

param aa-pilot 1880


:tele

On Mon, 2007-08-27 at 14:41 -0400, Mike Reilly wrote:
> I am trying to do something different with the CME B ACD tcl script. Call
> comes into main line 555-1212 rings several phones, no answer or busy
> forward to a number 1880.  Set up a dial peer
> Dial-peer voice 1881 voip
> Service aa
> Destination-pattern 1880
> Session target ipv4:10.100.100.1
> Incoming called-number 1880
> Dtmf-relay h245-alphanumeric
> Codec g711ulaw
> No vad
> 
> The above dial peer will trigger the b acd application.  Does anyone know if
> this will work?  Remember the 1880 is going to be a forward destination
> under the dn.
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From pmarques74 at gmail.com  Tue Aug 28 07:15:56 2007
From: pmarques74 at gmail.com (pedro marques)
Date: Tue, 28 Aug 2007 12:15:56 +0100
Subject: [cisco-voip] MCS-7825-I3
Message-ID: <d2216e4e0708280415s1f9b64a9ma5f3f4d58450811e@mail.gmail.com>

Hi list,

I currently have two of these MCS-7825-I3 servers. According to this URL:

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_sheet0900aecd805efb09.html<javascript:newWin('http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_sheet0900aecd805efb09.html')>

it is stated that these are servers for 5.1 or 6.0 Cisco Unified
Communications Manager Application.

My problem is that my sales team sold a MCS 7825-I3, 1000 User
CM4.2-K9-7825-I3 software package which seems to be compatible with the
aforementioned servers. Considering this SW, I tried to do the installation
with the installation DVDs included in the package. Nothing worked. With
Disk 1 (78XX-I1/I2) the hardware detection clearly states that the hardware
is not compatible. With Disk 2 (7825-I3, 7845-I1/I2) it begins the
installation and then gets to a point where I get a notification that it is
"gathering information" and ejects the DVD. Nothing else happens after that
point.

Am I looking at this problem with the wrong approach?

Any help would be much appreciated.

cheers,
pedro

-- 
Pedro Miguel Marques
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From ErasmuE4 at telkom.co.za  Tue Aug 28 08:12:40 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Tue, 28 Aug 2007 14:12:40 +0200
Subject: [cisco-voip] MCS-7825-I3
In-Reply-To: <d2216e4e0708280415s1f9b64a9ma5f3f4d58450811e@mail.gmail.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01985422@CNTRRA20-XCS01.telkom.co.za>


The URL below has a matrix showing the MCS server - CCM software
support

 

 

http://www.cisco.com/en/US/customer/products/hw/voiceapp/ps378/prod_br
ochure0900aecd8062a4f9.html

 

looks helpful

 

Erik Erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of pedro marques
Sent: Tuesday, August 28, 2007 1:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MCS-7825-I3

 

Hi list,

I currently have two of these MCS-7825-I3 servers. According to this
URL: 

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_sh
eet0900aecd805efb09.html
<javascript:newWin('http://www.cisco.com/en/US/products/hw/voiceapp/ps
378/products_data_sheet0900aecd805efb09.html')> 

it is stated that these are servers for 5.1 or 6.0 Cisco Unified
Communications Manager Application. 

My problem is that my sales team sold a MCS 7825-I3, 1000 User
CM4.2-K9-7825-I3 software package which seems to be compatible with
the aforementioned servers. Considering this SW, I tried to do the
installation with the installation DVDs included in the package.
Nothing worked. With Disk 1 (78XX-I1/I2) the hardware detection
clearly states that the hardware is not compatible. With Disk 2
(7825-I3, 7845-I1/I2) it begins the installation and then gets to a
point where I get a notification that it is "gathering information"
and ejects the DVD. Nothing else happens after that point. 

Am I looking at this problem with the wrong approach? 

Any help would be much appreciated. 

cheers, 
pedro

-- 
Pedro Miguel Marques 



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From jonvoip at gmail.com  Tue Aug 28 09:08:59 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 28 Aug 2007 08:08:59 -0500
Subject: [cisco-voip] VPN Connects to destination network, but can't ping,
	telnet, or connect to servers...
In-Reply-To: <003701c7e935$7787b770$6e9ca8c0@T42p>
References: <5d093f9a0708272228g382f774cr68a265260037fca4@mail.gmail.com>
	<003701c7e935$7787b770$6e9ca8c0@T42p>
Message-ID: <5d093f9a0708280608k357f9198ueaea2c709581da1@mail.gmail.com>

I suspected this as well, however, I have tried doing a route add on
the PC to the specific host I was trying to reach and had no luck...

There has to be a way around this...



Jonathan

On 8/28/07, Nikola Stojsin <nikola at worldnet.att.net> wrote:
> One possible scenario is same internal subnets in both networks.
>
> HTH,
> Nikola
>
>
>
> ------------------------------
> Nikola Stojsin
> PhD CCIE #12888
> Nikola at att.net
> (917) 558-1423
> (917) 591-9382 (fax)
> ------------------------------
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
> Sent: Tuesday, August 28, 2007 1:29 AM
> To: cisco voip list
> Subject: [cisco-voip] VPN Connects to destination network, but can't
> ping,telnet, or connect to servers...
>
> So, I have a customer I support, I can VPN into their network (Cisco
> VPN client, Cisco 3030 VPN Concentrator), but once I connect, I can't
> do anything, can't telnet, can't ping, can't connect to shares,
> nothing...
>
> However, if I bypass my firewall (ASA 5520), and connect directly to
> the internet, I can VPN in and everything works...
>
> I have noticed this a bunch of times, with a lot of customers.
>
> Why does this happen?
>
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From MLinsemier at apcapital.com  Tue Aug 28 09:03:25 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 28 Aug 2007 09:03:25 -0400
Subject: [cisco-voip] VPN Connects to destination network, but can't ping,
	telnet, or connect to servers...
In-Reply-To: <5d093f9a0708272228g382f774cr68a265260037fca4@mail.gmail.com>
References: <5d093f9a0708272228g382f774cr68a265260037fca4@mail.gmail.com>
Message-ID: <AB0A35F2486F81468971AA68558BE71906E0D623@smieln01winm01.micoanet.com>

Johnathan,

A few things you should check.  

* Make sure that the target network does not overlap your own.  If it
does you will have to implement NAT-T to resolve the issue.
* Check to see if you are connecting using IPSec or IPSec tunneled over
UDP (port 4500).  If you are trying to use pure IPSec behind a firewall,
you may need to configure a fixup or map a static address to your
machine for a one to one translation.
* Use the following ACL as a reference of what to let back in trough the
ASA.  Most likely you will only need to allow GRE, ESP, ISAKMP, and
non500-ISAKMP (UDP 4500).  

   access-list 101 permit gre any host x.x.x.x
   access-list 101 permit ahp any host x.x.x.x
   access-list 101 permit esp any host x.x.x.x
   access-list 101 permit udp any host x.x.x.x eq isakmp
   access-list 101 permit udp any host x.x.x.x eq non500-isakmp
   access-list 101 permit udp any host x.x.x.x eq 10000

In the past when I have had similar issues, it seems that opening
GRE/ESP fixed my problems.  You could enable all of them in your ASA and
then connect over VPN, and then eliminate them one at a time until you
know which resolves your problem.  You can also review the ACL hit count
to see which ones are not being used and remove them as well.  Hope this
helps.

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: Tuesday, August 28, 2007 1:29 AM
To: cisco voip list
Subject: [cisco-voip] VPN Connects to destination network, but can't
ping,telnet, or connect to servers...

So, I have a customer I support, I can VPN into their network (Cisco
VPN client, Cisco 3030 VPN Concentrator), but once I connect, I can't
do anything, can't telnet, can't ping, can't connect to shares,
nothing...

However, if I bypass my firewall (ASA 5520), and connect directly to
the internet, I can VPN in and everything works...

I have noticed this a bunch of times, with a lot of customers.

Why does this happen?




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

From asobihoudai at yahoo.com  Tue Aug 28 09:18:12 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 28 Aug 2007 06:18:12 -0700 (PDT)
Subject: [cisco-voip] CCME using hunt groups across the WAN is not possible
In-Reply-To: <5d093f9a0708280608k357f9198ueaea2c709581da1@mail.gmail.com>
Message-ID: <572815.85392.qm@web57206.mail.re3.yahoo.com>

Hi, I asked Cisco if using a CCME hunt group across
the WAN is possible and apparently it isn't.

Just a friendly reminder in case your project leader
comes up with some hairbrained idea like this without
checking any facts....

Paul


       
____________________________________________________________________________________
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 smgustafson at gmail.com  Tue Aug 28 11:20:50 2007
From: smgustafson at gmail.com (Steve G)
Date: Tue, 28 Aug 2007 08:20:50 -0700
Subject: [cisco-voip] Deploying Centralized CUE for Multisite WAN
Message-ID: <8a1f34b50708280820v4736d322n1fe0132ed557e1e3@mail.gmail.com>

Does anyone have experience with CUE servicing 3 to 4 WAN sites?  I am
curious how this has worked and if there are any pitfalls I need to worry
about.  3 of my sites use a common gateway for PSTN (PRI) access, but one
has it's own PSTN connection with a PRI.  I have read on CCO that it is only
supported as 1 SRST site per CUE Module.  They also need 4 different
autoattendants, which CUE allows for 3 + the default.  Can I modify the
default AA for my 4th AA?  Yea, this is another one of those hairbrained
Sales Engineer ideas.
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From tettokalle at hotmail.com  Tue Aug 28 11:27:43 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Tue, 28 Aug 2007 17:27:43 +0200
Subject: [cisco-voip] no called party info on isdn q931
Message-ID: <fb1ete$vhc$1@sea.gmane.org>

hi,

I have the following problem, there is no called Party Info in the ISDN 
debug, therefore the call could not be connectd correctly. The call will 
be connected but to the NULL number. I think this i why you need at 
least .T for incoming called number at the POTS dial peer. Is this a 
Cable and Wireless, which is the provider, problem? Or do I have done 
anything wrong.

*Aug 28 15:02:54.187: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref 
= 0x0005
         Sending Complete
         Bearer Capability i = 0x9090A3
                 Standard = CCITT
                 Transfer Capability = 3.1kHz Audio
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have 
in-band info
         Calling Party Number i = 0x0083, '07970132009'
                 Plan:Unknown, Type:Unknown
*Aug 28 15:02:54.199: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 
callref = 0x8005
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Progress Ind i = 0x8188 - In-band info or appropriate now 
available
*Aug 28 15:02:54.199: ISDN Se0/0/0:15 Q931: TX -> CONNECT pd = 8 
callref = 0x8005
*Aug 28 15:02:54.315: ISDN Se0/0/0:15 Q931: RX <- CONNECT_ACK pd = 8 
callref = 0x0005
*Aug 28 15:02:54.315: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now 
connected to 07970132009 N/A
*Aug 28 15:03:00.315: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now 
connected to 07970132009 N/A
*Aug 28 15:03:03.595: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 
callref = 0x0005
         Cause i = 0x8290 - Normal call clearing
         Display i = '*AA*CLEARED#'
*Aug 28 15:03:03.599: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 
disconnected from 07970132009 , call lasted 9 seconds
*Aug 28 15:03:03.599: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 
callref = 0x8005
*Aug 28 15:03:03.615: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 
callref = 0x0005
         Cause i = 0x8290 - Normal call clearing

cheers


From wsisk at cisco.com  Tue Aug 28 11:37:15 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 28 Aug 2007 11:37:15 -0400
Subject: [cisco-voip] no called party info on isdn q931
In-Reply-To: <fb1ete$vhc$1@sea.gmane.org>
References: <fb1ete$vhc$1@sea.gmane.org>
Message-ID: <4AF433E9-9B7F-42CD-845C-207F02752F1B@cisco.com>

contact your provider and ask for the digits.   otherwise, configure  
'connection plar' on the voice port to route the call to a default  
destination.

/wes

On Aug 28, 2007, at 11:27 AM, Kalle Tetto wrote:

hi,

I have the following problem, there is no called Party Info in the ISDN
debug, therefore the call could not be connectd correctly. The call will
be connected but to the NULL number. I think this i why you need at
least .T for incoming called number at the POTS dial peer. Is this a
Cable and Wireless, which is the provider, problem? Or do I have done
anything wrong.

*Aug 28 15:02:54.187: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref
= 0x0005
          Sending Complete
          Bearer Capability i = 0x9090A3
                  Standard = CCITT
                  Transfer Capability = 3.1kHz Audio
                  Transfer Mode = Circuit
                  Transfer Rate = 64 kbit/s
          Channel ID i = 0xA98381
                  Exclusive, Channel 1
          Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have
in-band info
          Calling Party Number i = 0x0083, '07970132009'
                  Plan:Unknown, Type:Unknown
*Aug 28 15:02:54.199: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
callref = 0x8005
          Channel ID i = 0xA98381
                  Exclusive, Channel 1
          Progress Ind i = 0x8188 - In-band info or appropriate now
available
*Aug 28 15:02:54.199: ISDN Se0/0/0:15 Q931: TX -> CONNECT pd = 8
callref = 0x8005
*Aug 28 15:02:54.315: ISDN Se0/0/0:15 Q931: RX <- CONNECT_ACK pd = 8
callref = 0x0005
*Aug 28 15:02:54.315: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now
connected to 07970132009 N/A
*Aug 28 15:03:00.315: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now
connected to 07970132009 N/A
*Aug 28 15:03:03.595: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8
callref = 0x0005
          Cause i = 0x8290 - Normal call clearing
          Display i = '*AA*CLEARED#'
*Aug 28 15:03:03.599: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0
disconnected from 07970132009 , call lasted 9 seconds
*Aug 28 15:03:03.599: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8
callref = 0x8005
*Aug 28 15:03:03.615: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x0005
          Cause i = 0x8290 - Normal call clearing

cheers

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


From bob at smalltime.com  Tue Aug 28 11:38:27 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Tue, 28 Aug 2007 10:38:27 -0500
Subject: [cisco-voip] Maximum number of 7970's on a 3750-48ps-s
In-Reply-To: <B0353BDEA8A5A14D8E3CA39AADCA46B606A0149B@NCAA-MAIL.ncaa.org>
References: <B0353BDEA8A5A14D8E3CA39AADCA46B606A0149B@NCAA-MAIL.ncaa.org>
Message-ID: <46D44173.5070101@smalltime.com>

Pennington, Craig wrote:
> I noticed we have reached our maximum power output for the 3750 (540 
> watts).  I then noticed it is set to the default settings of auto with 
> 15.4 watts max on a port with the switch reporting each 7970 as using 
> 10.3 watts.  I tried to do the power inline auto max XXXXX (where XXXXX 
> is the number of milli-watts between 4000 and 15400) to lower the max 
> setting in hopes of increasing the number of ports that would give 
> power. 540/15.4 gives you a max of 35 powered ports.  And that is what I 
> have, 35 phones, on each switch and I need to add more.  When I tried to 
> lower the milli-watts below 15400 the port would say deny power and the 
> phone would not boot.  has anyone had any luck trying to lower the power 
> on a 3750 to service more 7970's?

Yes, but notice that the phones aren't using 15.4 watts in the first place:

# sh cdp neigh det
Device ID: SEP0015C65C6D56
Entry address(es):
   IP address: 10.255.5.62
Platform: Cisco IP Phone 7970,  Capabilities: Host Phone
Interface: FastEthernet1/0/15,  Port ID (outgoing port): Port 1
Holdtime : 124 sec

Version :
SCCP70.8-2-1S

advertisement version: 2
Duplex: full
Power drawn: 6.300 Watts
Power request id: 27990, Power management id: 1
Power request levels are:10250 6300 0 0 0
Management address(es):


So when we're configuring a switch, we do:
# interface range fa1/0/1-48
# power inline consumption 6300

To lock all the ports to 6300 mw.

And a 3750 has 370 available watts:
# sh power inline

Module   Available     Used     Remaining
           (Watts)     (Watts)    (Watts)
------   ---------   --------   ---------
1           370.0      227.8       142.2
2           370.0      134.0       236.0

so 370 divided by 10.250 is what gives you the 36 phone limit.

From wsisk at cisco.com  Tue Aug 28 11:40:10 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 28 Aug 2007 11:40:10 -0400
Subject: [cisco-voip] MCS-7825-I3
In-Reply-To: <d2216e4e0708280415s1f9b64a9ma5f3f4d58450811e@mail.gmail.com>
References: <d2216e4e0708280415s1f9b64a9ma5f3f4d58450811e@mail.gmail.com>
Message-ID: <DE25671E-B621-4098-986F-72E82AFB81B6@cisco.com>

next gen hardware was only added in the 5.1(2b) by CSCsj57609.

for windows OS you need 2000.4.5,2000.4.5a, or 2000.4.5b if you want  
CM4.1/4.2
or 2003.1.2 or 2003.1.2a if you want CM 4.3

The software kit CM4.2-K9-7825-I3 bill of materials does contain the  
correct disks.  Contact whomever your purchased the software from to  
get a replacement software kit as yours appears incomplete/incorrect.

/Wes

The part number CM4.2-K9-7825-I3 appears correct

On Aug 28, 2007, at 7:15 AM, pedro marques wrote:

Hi list,

I currently have two of these MCS-7825-I3 servers. According to this  
URL:

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/ 
products_data_sheet0900aecd805efb09.html

it is stated that these are servers for 5.1 or 6.0 Cisco Unified  
Communications Manager Application.

My problem is that my sales team sold a MCS 7825-I3, 1000 User CM4.2- 
K9-7825-I3 software package which seems to be compatible with the  
aforementioned servers. Considering this SW, I tried to do the  
installation with the installation DVDs included in the package.  
Nothing worked. With Disk 1 (78XX-I1/I2) the hardware detection  
clearly states that the hardware is not compatible. With Disk 2 (7825- 
I3, 7845-I1/I2) it begins the installation and then gets to a point  
where I get a notification that it is "gathering information" and  
ejects the DVD. Nothing else happens after that point.

Am I looking at this problem with the wrong approach?

Any help would be much appreciated.

cheers,
pedro

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

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From kyle.hansen at hci.utah.edu  Tue Aug 28 11:50:19 2007
From: kyle.hansen at hci.utah.edu (Kyle Hansen (HCH))
Date: Tue, 28 Aug 2007 09:50:19 -0600
Subject: [cisco-voip] Masking Caller ID
Message-ID: <916324E1C885C54989449104CB7225B60140FF68@EMAIL1.hci.utah.edu>

Does anyone know how to easily mask a caller ID name/number in Call
Manager.  I am trying to hide a few number internally when they call
other staff members.  Any ideas??

 

Thank you,

Kyle Hansen

Computing and Technology Group (C.A.T.G.)

Customer Support Lead/Project Manager

Huntsman Cancer Hospital

1950 Circle of Hope - Salt Lake City, UT  84112.  801.585.0330

If you need computer support please call the HELPDESK @ 585-0330 or

Submit a request for support at:
www.hci.utah.edu/internal/computer_support.html

 

 

 

 

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From balbler at nts.at  Tue Aug 28 11:57:40 2007
From: balbler at nts.at (Bernhard Albler)
Date: Tue, 28 Aug 2007 17:57:40 +0200
Subject: [cisco-voip] no called party info on isdn q931
In-Reply-To: <4AF433E9-9B7F-42CD-845C-207F02752F1B@cisco.com>
References: <fb1ete$vhc$1@sea.gmane.org>
	<4AF433E9-9B7F-42CD-845C-207F02752F1B@cisco.com>
Message-ID: <395e03520000cc79@nts.at>

Hi Kalle,

where are you located? This looks very much like our most excellent Austrian ISDN Setup. Which has cost me about 10 years of my total lifespan. 


Basically we solve this by matching ^$ in a translation-profile on the voice-port and doing some stunts with overlap receiving and dialpeers to get this working.

Ping me offlist for details.

Regards
bernhard
------------------------------------------
DI (FH) Bernhard Albler, NTS AG 
Tel.: +43 (1) 867 67 76 -0, email: balbler at nts.at 
 

> -----Urspr?ngliche Nachricht-----
> Von: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] Im Auftrag von Wes Sisk
> Gesendet: Dienstag, 28. August 2007 17:37
> An: Kalle Tetto
> Cc: cisco-voip at puck.nether.net
> Betreff: Re: [cisco-voip] no called party info on isdn q931
> 
> contact your provider and ask for the digits.   otherwise, configure
> 'connection plar' on the voice port to route the call to a default
> destination.
> 
> /wes
> 
> On Aug 28, 2007, at 11:27 AM, Kalle Tetto wrote:
> 
> hi,
> 
> I have the following problem, there is no called Party Info in the ISDN
> debug, therefore the call could not be connectd correctly. The call will
> be connected but to the NULL number. I think this i why you need at
> least .T for incoming called number at the POTS dial peer. Is this a
> Cable and Wireless, which is the provider, problem? Or do I have done
> anything wrong.
> 
> *Aug 28 15:02:54.187: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref
> = 0x0005
>           Sending Complete
>           Bearer Capability i = 0x9090A3
>                   Standard = CCITT
>                   Transfer Capability = 3.1kHz Audio
>                   Transfer Mode = Circuit
>                   Transfer Rate = 64 kbit/s
>           Channel ID i = 0xA98381
>                   Exclusive, Channel 1
>           Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have
> in-band info
>           Calling Party Number i = 0x0083, '07970132009'
>                   Plan:Unknown, Type:Unknown
> *Aug 28 15:02:54.199: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> callref = 0x8005
>           Channel ID i = 0xA98381
>                   Exclusive, Channel 1
>           Progress Ind i = 0x8188 - In-band info or appropriate now
> available
> *Aug 28 15:02:54.199: ISDN Se0/0/0:15 Q931: TX -> CONNECT pd = 8
> callref = 0x8005
> *Aug 28 15:02:54.315: ISDN Se0/0/0:15 Q931: RX <- CONNECT_ACK pd = 8
> callref = 0x0005
> *Aug 28 15:02:54.315: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now
> connected to 07970132009 N/A
> *Aug 28 15:03:00.315: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now
> connected to 07970132009 N/A
> *Aug 28 15:03:03.595: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8
> callref = 0x0005
>           Cause i = 0x8290 - Normal call clearing
>           Display i = '*AA*CLEARED#'
> *Aug 28 15:03:03.599: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0
> disconnected from 07970132009 , call lasted 9 seconds
> *Aug 28 15:03:03.599: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8
> callref = 0x8005
> *Aug 28 15:03:03.615: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8
> callref = 0x0005
>           Cause i = 0x8290 - Normal call clearing
> 
> cheers
> 
> _______________________________________________
> 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 tettokalle at hotmail.com  Tue Aug 28 12:06:13 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Tue, 28 Aug 2007 18:06:13 +0200
Subject: [cisco-voip] no called party info on isdn q931
Message-ID: <fb1h5k$54j$1@sea.gmane.org>

hi,

I have the following problem, there is no called Party Info in the ISDN 
debug, therefore the call could not be connectd correctly. The call will 
be connected but to the NULL number. I think this i why you need at 
least .T for incoming called number at the POTS dial peer. Is this a 
Cable and Wireless, which is the provider, problem? Or do I have done 
anything wrong.

*Aug 28 15:02:54.187: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref 
= 0x0005
         Sending Complete
         Bearer Capability i = 0x9090A3
                 Standard = CCITT
                 Transfer Capability = 3.1kHz Audio
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have 
in-band info
         Calling Party Number i = 0x0083, '07970132009'
                 Plan:Unknown, Type:Unknown
*Aug 28 15:02:54.199: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 
callref = 0x8005
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Progress Ind i = 0x8188 - In-band info or appropriate now available
*Aug 28 15:02:54.199: ISDN Se0/0/0:15 Q931: TX -> CONNECT pd = 8 callref 
= 0x8005
*Aug 28 15:02:54.315: ISDN Se0/0/0:15 Q931: RX <- CONNECT_ACK pd = 8 
callref = 0x0005
*Aug 28 15:02:54.315: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now 
connected to 07970132009 N/A
*Aug 28 15:03:00.315: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now 
connected to 07970132009 N/A
*Aug 28 15:03:03.595: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 
callref = 0x0005
         Cause i = 0x8290 - Normal call clearing
         Display i = '*AA*CLEARED#'
*Aug 28 15:03:03.599: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 
disconnected from 07970132009 , call lasted 9 seconds
*Aug 28 15:03:03.599: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 callref 
= 0x8005
*Aug 28 15:03:03.615: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 
callref = 0x0005
         Cause i = 0x8290 - Normal call clearing

cheers


From lelio at uoguelph.ca  Tue Aug 28 12:52:58 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 28 Aug 2007 12:52:58 -0400
Subject: [cisco-voip] Masking Caller ID
References: <916324E1C885C54989449104CB7225B60140FF68@EMAIL1.hci.utah.edu>
Message-ID: <001901c7e993$e3571d70$56196883@cfs.uoguelph.ca>

This has been discussed on the list a few times....searching the archives will help with details.

Essentially, you create a translation with the calling party name/number restricted and then put that translation into a partition that callers have access to.

You can do this so that all calls from a specific set of lines will always be masked, i.e. the translation pattern is the only pattern reachable and then the translation pattern's calling search space can reach all the rest of the lines. Or, you can use a prefix, say *44 XXXXX (for 5 digit dialplans)


  ----- Original Message ----- 
  From: Kyle Hansen (HCH) 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, August 28, 2007 11:50 AM
  Subject: [cisco-voip] Masking Caller ID


  Does anyone know how to easily mask a caller ID name/number in Call Manager.  I am trying to hide a few number internally when they call other staff members.  Any ideas??

   

  Thank you,

  Kyle Hansen

  Computing and Technology Group (C.A.T.G.)

  Customer Support Lead/Project Manager

  Huntsman Cancer Hospital

  1950 Circle of Hope - Salt Lake City, UT  84112.  801.585.0330

  If you need computer support please call the HELPDESK @ 585-0330 or

  Submit a request for support at:  www.hci.utah.edu/internal/computer_support.html

   

   

   

   



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From ez4me2c3d at gmail.com  Tue Aug 28 13:04:35 2007
From: ez4me2c3d at gmail.com (Anthony H)
Date: Tue, 28 Aug 2007 12:04:35 -0500
Subject: [cisco-voip] Maximum number of 7970's on a 3750-48ps-s
In-Reply-To: <B0353BDEA8A5A14D8E3CA39AADCA46B606A0149B@NCAA-MAIL.ncaa.org>
References: <AcfmkBdCyFJwZVlhRaGKRf/c5VrHBw==>
	<B0353BDEA8A5A14D8E3CA39AADCA46B606A0149B@NCAA-MAIL.ncaa.org>
Message-ID: <49921b8d0708281004x151430d4y689f336699663a1@mail.gmail.com>

Perhaps this information could be of some use.  Though Robert did give some
great detailed information on the subject.

http://www.cisco.com/en/US/products/hw/switches/ps5023/products_qanda_item09186a00801b0971.shtml

*Q.* How does Power Management work in the Cisco Catalyst 3750 PoE switches?
*A.* The Cisco Catalyst 3750-24PS switch supports a maximum of 15.4W per
port of -48V DC power on all 24 ports over standard Category 3 and 5
unshielded twisted-pair (UTP) cable up to 100 meters, however all 48 10/100
ports in Cisco Catalyst 3750-48PS switch will be able to deliver
7.7Wsimultaneouslyand any given port can be configurable to support up
to the maximum of
15.4 watts as long as the switch's - 48V power capacity is not exceeded. The
Cisco Catalyst 3750 PoE switches monitor and track the requests for power,
and only grants power when it is available. If the connected PD draws more
power, which results in exceeding the system power budget that is available
on the switch, then power will be denied to that port and the PD will not
power up. If power is denied, the switch re-checks the power budget
periodically (every 15 seconds), and continues to attempt to grant the
request for power. If power is granted, the switch updates the power budget.
Additional power management features are described later in this section.

Anthony

On 8/24/07, Pennington, Craig <cpennington at ncaa.org> wrote:
>
>  I noticed we have reached our maximum power output for the 3750 (540
> watts).  I then noticed it is set to the default settings of auto with
> 15.4 watts max on a port with the switch reporting each 7970 as using 10.3watts.  I tried to do the power inline auto max XXXXX (where XXXXX is the
> number of milli-watts between 4000 and 15400) to lower the max setting in
> hopes of increasing the number of ports that would give power. 540/15.4
> gives you a max of 35 powered ports.  And that is what I have, 35 phones, on
> each switch and I need to add more.  When I tried to lower the milli-watts
> below 15400 the port would say deny power and the phone would not boot.  has
> anyone had any luck trying to lower the power on a 3750 to service more
> 7970's?
>
>  Craig Pennington
> Network Engineer
> *N*ational *C*ollegiate *A*thletic *A*ssociation
> Fax: 317-917-6888
> cpennington at ncaa.org
> www.ncaa.org
> This email and any attachments may contain confidential and privileged
> information. If you are not the intended recipient, please notify the sender
> immediately by return email, delete this message and destroy any copies. Any
> dissemination or use of this information by a person other than the intended
> recipient is unauthorized and may be illegal.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Anthony
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From rschuknecht at gmx.de  Tue Aug 28 13:28:18 2007
From: rschuknecht at gmx.de (rschuknecht at gmx.de)
Date: Tue, 28 Aug 2007 19:28:18 +0200
Subject: [cisco-voip] CUCM 5.X/6.X and LDAP
Message-ID: <20070828172818.116950@gmx.net>

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
-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt?
Der kanns mit allen: http://www.gmx.net/de/go/multimessenger

From bob at smalltime.com  Tue Aug 28 13:36:23 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Tue, 28 Aug 2007 12:36:23 -0500
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <1188193592.13352.18.camel@lt21223.campus.dmacc.edu>
References: <5d093f9a0708262117p99d07d0wc8dfe04695fd7b2@mail.gmail.com>
	<1188193592.13352.18.camel@lt21223.campus.dmacc.edu>
Message-ID: <46D45D17.3020006@smalltime.com>

> On Sun, 2007-08-26 at 23:17 -0500, Jonathan Charles wrote:
> http://www.microsoft.com/exchange/evaluation/features/unifiedmessaging.mspx

So, does Cisco have a "competitive analysis" document that will provide 
a compelling reason to my management to deploy Unity 5 rather than 
migrating everyone to E2K7?  (Since we're going to be deploying E2K7 
anyway?)


From wsisk at cisco.com  Tue Aug 28 14:35:56 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 28 Aug 2007 14:35:56 -0400
Subject: [cisco-voip] CUCM 5.X/6.X and LDAP
In-Reply-To: <20070828172818.116950@gmx.net>
References: <20070828172818.116950@gmx.net>
Message-ID: <46D46B0C.2060308@cisco.com>

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
>   

From jd80301 at yahoo.com  Tue Aug 28 13:43:19 2007
From: jd80301 at yahoo.com (Jake Doe)
Date: Tue, 28 Aug 2007 10:43:19 -0700 (PDT)
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <8a1f34b50708271030r83fc010jf3fa820c188a461c@mail.gmail.com>
Message-ID: <477364.83090.qm@web50805.mail.re2.yahoo.com>

Yes, Exchange 2007 UM allows you to play your voice
mail through the phone as well.  However, the Play on
Phone feature is only available with Outlook 2007 or
Outlook Web Access 2007 (e.g. no plug-in required). 
Otherwise, the voice mail is just another WMA/WAV file
attachment in older versions of Outlook.

Also, if Exchange 2007 UM dialing rules have been
configured, you could actually Play on Phone to any
number (e.g. home, mobile, etc.).

--- Steve G <smgustafson at gmail.com> wrote:

> With Exchange 2007, does it allow you to listen to
> your VM through the
> phone?  Like with Unity ViewMail you can click play
> and it will ring your
> extension which will then play your message to you
> after you pick-up.
> 
> On 8/27/07, Jonathan Charles <jonvoip at gmail.com>
> wrote:
> >
> > I am sure someone will come out with a freeware
> app that does MWI...
> >
> > However, it probably won't be able to tag just
> voicemails and will
> > light MWI on all inbox items.
> >
> > The reality is that, in a unified messaging
> environment, most people
> > don't use their phones to retrieve vmail anyway.
> >
> > I think a business argument could almost be made
> to forgo MWI and
> > issue everyone speakers for their PCs.
> >
> >
> >
> > J
> >
> > On 8/27/07, Jeffrey C. Ollie <jeff at ocjtech.us>
> wrote:
> > > On Sun, 2007-08-26 at 23:17 -0500, Jonathan
> Charles wrote:
> > > >
> >
>
http://www.microsoft.com/exchange/evaluation/features/unifiedmessaging.mspx
> > > >
> > > > Speech-Enabled Auto Attendant
> > > >
> > > > The Attendant answers calls using an automated
> operator with
> > > > customizable menus (for example, "press 1 for
> sales") and global
> > > > address list directory lookups (for example,
> "who would you like to
> > > > contact?"). Callers can interact with the
> Automated Attendant through
> > > > touchtone menus or their voice using speech
> recognition.
> > > >
> > > > I think the integrated speed recognition is
> pretty cool, and as soon
> > > > as you play with Outlook Voice Access, Unity
> looks kind of sad in
> > > > comparison...
> > >
> > > The "cool" features may be nice to give to your
> "C" level users, but the
> > > killer for me would be letting the in-house
> Exchange admins handle the
> > > voicemail/auto-attendants on their (existing)
> highly-available Exchange
> > > clusters, rather than depending on my single
> Unity server.  That way I
> > > can also stop being an Exchange administrator
> myself (which I have
> > > neither the training or temperament for).
> > >
> > > One thing I noticed from [1]:
> > >         Exchange 2007 Unified Messaging has been
> tested for use with
> > >         Cisco Call Manager version 5.x. If you
> have Cisco Call Manager
> > >         5.x on your network, you will be able to
> connect Exchange 2007
> > >         Unified Messaging servers directly to
> Call Manager without the
> > >         requirement of an IP/VoIP gateway. Many
> of the features that are
> > >         included with Exchange 2007 Unified
> Messaging are fully
> > >         functional with Cisco Call Manager 5.x.
> However, the Message
> > >         Waiting Indicator (MWI) feature does not
> work and faxing does
> > >         not work because the T.38 faxing
> protocol is not supported by
> > >         Cisco Call Manager.
> > >
> > > T.38 faxing I think I could do without, but I
> think I'd be in danger of
> > > losing my job if MWI stopped working.  However,
> according to the
> > > document that you can download from [2] there's
> a third-party
> > > solution[3] that enables MWI.
> > >
> > > [1]
> >
>
http://technet.microsoft.com/en-us/library/2516dac1-dfdc-47eb-8e6f-18b1537a57b2.aspx
> > > [2]
> >
>
http://www.microsoft.com/downloads/details.aspx?FamilyId=68b43d3c-7c84-4c2f-bfd7-98754970d70e&displaylang=en
> > > [3] http://www.mwi2007.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
> >
> > _______________________________________________
> 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 Aug 28 14:59:56 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 28 Aug 2007 14:59:56 -0400
Subject: [cisco-voip] Call FWD during SRST
Message-ID: <OF06CD3469.343CC6E5-ON85257345.00669B21-85257345.00685D39@bftg.com>

I am looking to accomplish the following with SRST. 

In SRST I want 2 DID's that route to 2 different hunt groups (2 business 
units same location).  If the call is not answered then I want to CFWD 
each DID to it's own  external number.  One per business unit.

I thought this could be done with the alias command but upon reading 
further it appears that an alias is only valid with locally REGISTERED 
numbers on the gateway.  Config was going to be something like this:

alias 1 4700 to 4701 preference 1
alias 2 4700 to 4702 preference 2
alias 3 4700 to 4703 preference 3
alias 4 4700 to 4704 preference 4
alias 5 4700 to 9XXXXXXXXXX preference 5  --- Want to send this to PSTN
 
alias 15 4749 to 4745 preference 1
alias 16 4749 to 4744 preference 2
alias 17 4749 to 4740 preference 3
alias 18 4749 to 9XXXXXXXXXX preference 4  --- Want to send this to the 
PSTN

call-forward busy 9XXXXXXXXXX
call-forward noan 9XXXXXXXXXX timeout 16

I believe the CFWD busy or CFWD noan can be to the PSTN but in this case I 
need specific calls to go to different PSTN destinations.  Any help would 
be appreciated.

Carlos
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From d4rkf1ber at gmail.com  Tue Aug 28 15:02:53 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Tue, 28 Aug 2007 14:02:53 -0500
Subject: [cisco-voip] Cisco Emergency Responder Install ?
Message-ID: <a1b07ea60708281202u382bff12pa45a51e47b61cdbf@mail.gmail.com>

I am in the process of installing CER for the first time, right now I
am focusing on just one CER server "publisher" obviously and want to
get it up and working in a very limited fashion just for testing
purposes and to prove that it can and will do all we need.

During the install it is asking for the publisher CallManagers info
for being able to login to it.  As I understand it, it will eventually
use this to poll CM for a list of phones and such.

I want to make sure that as I intall this for the first time and work
through the learning process and such that it "CER" does not in any
way tie into Callmanagers database or anything.  Its 'integration'
with callmanager is just at a read-only type of level right?

I know at some point I will need to setup some integration between the
two, but that would require me to actually do something and not having
CER automatically tie into CallManager and manipulate it right?

Sorry for sounding and being paranoid,  I just don't want CER to some
how make some change to CM that I am not aware of and all of a sudden
create a problem for my production CallManagers.

From svoll.voip at gmail.com  Tue Aug 28 15:40:18 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Tue, 28 Aug 2007 12:40:18 -0700
Subject: [cisco-voip] Cisco Emergency Responder Install ?
In-Reply-To: <a1b07ea60708281202u382bff12pa45a51e47b61cdbf@mail.gmail.com>
References: <a1b07ea60708281202u382bff12pa45a51e47b61cdbf@mail.gmail.com>
Message-ID: <f84a38d30708281240i135d9bc5vf66fde9430c4b2e6@mail.gmail.com>

I'm right there with you on the first install but a week or so down the
road.

CER only polls the CM and switches as far as I know.  all the tieing in is
done via Route Points, cti ports, and translations.

Scott


On 8/28/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
>
> I am in the process of installing CER for the first time, right now I
> am focusing on just one CER server "publisher" obviously and want to
> get it up and working in a very limited fashion just for testing
> purposes and to prove that it can and will do all we need.
>
> During the install it is asking for the publisher CallManagers info
> for being able to login to it.  As I understand it, it will eventually
> use this to poll CM for a list of phones and such.
>
> I want to make sure that as I intall this for the first time and work
> through the learning process and such that it "CER" does not in any
> way tie into Callmanagers database or anything.  Its 'integration'
> with callmanager is just at a read-only type of level right?
>
> I know at some point I will need to setup some integration between the
> two, but that would require me to actually do something and not having
> CER automatically tie into CallManager and manipulate it right?
>
> Sorry for sounding and being paranoid,  I just don't want CER to some
> how make some change to CM that I am not aware of and all of a sudden
> create a problem for my production CallManagers.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From bob at smalltime.com  Tue Aug 28 15:52:16 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Tue, 28 Aug 2007 14:52:16 -0500
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <477364.83090.qm@web50805.mail.re2.yahoo.com>
References: <477364.83090.qm@web50805.mail.re2.yahoo.com>
Message-ID: <46D47CF0.6070209@smalltime.com>

Jake Doe wrote:
> Yes, Exchange 2007 UM allows you to play your voice
> mail through the phone as well.  However, the Play on
> Phone feature is only available with Outlook 2007 or
> Outlook Web Access 2007 (e.g. no plug-in required). 
> Otherwise, the voice mail is just another WMA/WAV file
> attachment in older versions of Outlook.

 From what I recall at Interop, they don't have a pitch-adjusted tempo 
feature during playback, so you can't slow down the audio to catch a 
phone number which may have been spoken too quickly to make out.

ObPetPeeve:  People who delight in giving you a phone number in a 
voicemail as quickly as possible as some sort of personal contest.

From su1droot at gmail.com  Tue Aug 28 17:16:59 2007
From: su1droot at gmail.com (Ben Mason)
Date: Tue, 28 Aug 2007 17:16:59 -0400
Subject: [cisco-voip] Software phone Mac OS X Cal Manager Express
Message-ID: <f83dffc80708281416u1d107222v90890cf9799fec22@mail.gmail.com>

Hello all,
I am wondering if anyone has any recommendation on softphone software
for Mac OS X that will work with CallManager Express.  google hasn't
been too helpful.

I would also take good recommendations for softphone that work with
normal CallManager also.

-- 
--
Ben
email: locutus at the-collective.net
web: http://locutus.the-collective.net/
irc: su1droot; aim: su1droot; y!: su1droot

From lelio at uoguelph.ca  Tue Aug 28 17:28:17 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 28 Aug 2007 17:28:17 -0400
Subject: [cisco-voip] Display Logging Parameter on 7911
Message-ID: <002a01c7e9ba$59934100$56196883@cfs.uoguelph.ca>

Can anyone help describe what this is and whether it is a security concern? The second sentence says it does not control the generation of logs, but the last sentence says you can use it to force logs. I wanted to turn it off for a bunch of phones but BAT conveniently does not have this setting in the list. Grrrr.


      Logging Display : This option selects what type of console logging is allowed. This option does not control the generation of logs - just whether the logs display. Disabled indicates that logging does not display to the console, nor to the connected downstream port. PC Controlled indicates that the workstation attached to the PC port will control whether logging is enabled. Enabled indicates that logs are always sent both to the console and to downstream port. Use Enabled to force logs on so they can be captured with a packet sniffer. 
      This is a required field.
      Default: PC Controlled.
     
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From tednugent69 at yahoo.com  Tue Aug 28 17:47:28 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Tue, 28 Aug 2007 14:47:28 -0700 (PDT)
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
Message-ID: <482648.21929.qm@web36115.mail.mud.yahoo.com>

Well folks it?s been a long time coming but I think we
have a new winner in the WORST CISCO DOCUMENTATION
award. I actually wasn?t sure it was possible but the
entire Mobility Manager 1.2 documentation set sucks so
bad that its beaten out Personal Assistant 1.3 which
has held the title for almost 3 yrs now!! A special
congratulation and kiss my ass goes out to the lazy
losers that wrote crap!

Sorry for the rant, Cisco Docs usually aren't too bad
but the 2nd graders that wrote this one have out done
themselves!



       
____________________________________________________________________________________
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From jonvoip at gmail.com  Tue Aug 28 18:33:05 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 28 Aug 2007 17:33:05 -0500
Subject: [cisco-voip] Phone Hostname
Message-ID: <5d093f9a0708281533h77d08972m775fb777ecb22e55@mail.gmail.com>

I have a customer who has a requirement that all devices on the IP
Network conform to the following hostname:

USCHIPHN001

or whatever...

Anyway, is there any way to change the hostname of a Cisco IP Phone to
conform to their network-wide naming convention?

Obviously I can match up DHCP and DNS together, but the device itself
will still have SEP000318397489 as its hostname...



Jonathan

From jd80301 at yahoo.com  Tue Aug 28 20:51:41 2007
From: jd80301 at yahoo.com (Jake Doe)
Date: Tue, 28 Aug 2007 17:51:41 -0700 (PDT)
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <46D47CF0.6070209@smalltime.com>
Message-ID: <170044.24423.qm@web50811.mail.re2.yahoo.com>

Perhaps not through Play on Phone (since it's just an
embedded Windows Media Player).  However, if you use
the TUI or VUI, then you have the ability to slow down
or speed up the message.  See the Outlook Voice Access
(OVA) Quick Reference Guide for those options:

http://technet.microsoft.com/en-us/library/bb397228.aspx

--- Robert Kulagowski <bob at smalltime.com> wrote:

> Jake Doe wrote:
> > Yes, Exchange 2007 UM allows you to play your
> voice
> > mail through the phone as well.  However, the Play
> on
> > Phone feature is only available with Outlook 2007
> or
> > Outlook Web Access 2007 (e.g. no plug-in
> required). 
> > Otherwise, the voice mail is just another WMA/WAV
> file
> > attachment in older versions of Outlook.
> 
>  From what I recall at Interop, they don't have a
> pitch-adjusted tempo 
> feature during playback, so you can't slow down the
> audio to catch a 
> phone number which may have been spoken too quickly
> to make out.
> 
> ObPetPeeve:  People who delight in giving you a
> phone number in a 
> voicemail as quickly as possible as some sort of
> personal contest.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________
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that gives answers, not web links. 
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From jd80301 at yahoo.com  Tue Aug 28 20:57:26 2007
From: jd80301 at yahoo.com (Jake Doe)
Date: Tue, 28 Aug 2007 17:57:26 -0700 (PDT)
Subject: [cisco-voip] CUPS (1.x or 6.x) Evaluation
Message-ID: <110872.84254.qm@web50801.mail.re2.yahoo.com>

Does anyone know if there is a trial version of the
Cisco Unified Presence Server (1.x or 6.x) that one
can download for evaluation?


       
____________________________________________________________________________________Ready for the edge of your seat? 
Check out tonight's top picks on Yahoo! TV. 
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From jgrace at adcapnet.com  Tue Aug 28 22:16:10 2007
From: jgrace at adcapnet.com (James Grace)
Date: Tue, 28 Aug 2007 22:16:10 -0400
Subject: [cisco-voip] our cm 6.0 admin page takes a long time to resolve page
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C75E@mail.adcapnetsvc.local>

?
our cm 6.0 admin page takes a long time to resolve page. For some reason we have to wait about 10 sec to change the next line on the administrator page.  Its like the server is having a issue resolving the page after each change.  example i can select a CSS and then i have to wait 10 sec to move to the next parameter.
 
James Grace 
 
Systems Engineer
jgrace at 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 28 Aug 2007 22:15:15 -0400
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From Nick.Kassel at Charles-Stanley.co.uk  Wed Aug 29 03:56:21 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed, 29 Aug 2007 08:56:21 +0100
Subject: [cisco-voip] our cm 6.0 admin page takes a long time to resolve
	page
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC014520E2@CSEXCH01.charles-stanley.co.uk>

I had some very weird issues with CM 6.0 when the switchport was not set
to 100mb full duplex and the server settings were. It was producing very
inconsistent results.

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: 29 August 2007 03:16
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] our cm 6.0 admin page takes a long time to resolve
page

 

our cm 6.0 admin page takes a long time to resolve page. For some reason
we have to wait about 10 sec to change the next line on the
administrator page.  Its like the server is having a issue resolving the
page after each change.  example i can select a CSS and then i have to
wait 10 sec to move to the next parameter.

 

James Grace 

 

Systems Engineer

jgrace at 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 28 Aug 2007
22:15:15 -0400
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From ErasmuE4 at telkom.co.za  Wed Aug 29 05:12:41 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 29 Aug 2007 11:12:41 +0200
Subject: [cisco-voip] Deploying Centralized CUE for Multisite WAN
In-Reply-To: <8a1f34b50708280820v4736d322n1fe0132ed557e1e3@mail.gmail.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F0198549F@CNTRRA20-XCS01.telkom.co.za>


In a UC 500 update cisco mentioned something about running remote
phones on UC500. maybe you can try and get a presentation on UC500. As
far as I know it is based on CME so the functionality will be the same

 

 

 

Erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Steve G
Sent: Tuesday, August 28, 2007 5:21 PM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] Deploying Centralized CUE for Multisite WAN

 

Does anyone have experience with CUE servicing 3 to 4 WAN sites?  I am
curious how this has worked and if there are any pitfalls I need to
worry about.  3 of my sites use a common gateway for PSTN (PRI)
access, but one has it's own PSTN connection with a PRI.  I have read
on CCO that it is only supported as 1 SRST site per CUE Module.  They
also need 4 different autoattendants, which CUE allows for 3 + the
default.  Can I modify the default AA for my 4th AA?  Yea, this is
another one of those hairbrained Sales Engineer ideas. 



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From ErasmuE4 at telkom.co.za  Wed Aug 29 05:15:08 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 29 Aug 2007 11:15:08 +0200
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
In-Reply-To: <482648.21929.qm@web36115.mail.mud.yahoo.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F019854A0@CNTRRA20-XCS01.telkom.co.za>


Did you consider the unity docs scattered all over the show - any
person making first contact with Unity will probably also feel quite
lost


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
Sent: Tuesday, August 28, 2007 11:47 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!

Well folks it's been a long time coming but I think we
have a new winner in the WORST CISCO DOCUMENTATION
award. I actually wasn't sure it was possible but the
entire Mobility Manager 1.2 documentation set sucks so
bad that its beaten out Personal Assistant 1.3 which
has held the title for almost 3 yrs now!! A special
congratulation and kiss my ass goes out to the lazy
losers that wrote crap!

Sorry for the rant, Cisco Docs usually aren't too bad
but the 2nd graders that wrote this one have out done
themselves!



       
______________________________________________________________________
______________
Choose the right car based on your needs.  Check out Yahoo! Autos new
Car Finder tool.
http://autos.yahoo.com/carfinder/
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From d4rkf1ber at gmail.com  Wed Aug 29 09:46:07 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Wed, 29 Aug 2007 08:46:07 -0500
Subject: [cisco-voip] federal NRO number assignment rules ?
Message-ID: <a1b07ea60708290646n7eadd32ege276323b2921569a@mail.gmail.com>

Recently I asked the group about an internal issue I am facing
regarding how we are leveraging DID's and central PRI's.

Basically, we have asked for and recieved a block of DID's that all
fall into one range.  Those are all tied to a couple of PRI's that
come into a central location.

We have not only assigned DID's to employees who work out of the
central location, but through the magic of VoIP "ohhh ahhh" we have
users with Cisco phones and DID's from the corporate office who work
remotely.

Nothing new here, and nothing I have not seen done every place I have
worked out in the last oh 10 years or so.

However for the first time I am at a company who has some lawyer /
regulatory person on staff who is questioning whether this is legal or
not.

The current statement from this person experiencing concern is,
"compliance with federal NRO number assignment rules".

Sounds awfully legal and well thats what I asked for.  I previously
told this person that for the most part everyone I have polled
regarding these concerns have basically agreed in saying its very
common to do this and if it is illegal then show me where and what
makes it illegal?  So thats what they replied with, and well I am
googling and trying to make sense of what I read but I can't say I
fully understand it.

Anyone know more about this and understand the implications and how it
affects a business doing what I am doing internally for employee
phones?

From tednugent69 at yahoo.com  Wed Aug 29 09:47:33 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 29 Aug 2007 06:47:33 -0700 (PDT)
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F019854A0@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <504001.12611.qm@web36106.mail.mud.yahoo.com>

Ahh yes however once you finally do track down the
Unity Doc you need its typically pretty good.

For instance they wouldn't call a DN (Distinguished
Name) a Directory Number like they do in the CMM
docs..




--- "Erik Erasmus (E)" <ErasmuE4 at telkom.co.za> wrote:

> 
> Did you consider the unity docs scattered all over
> the show - any
> person making first contact with Unity will probably
> also feel quite
> lost
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ted Nugent
> Sent: Tuesday, August 28, 2007 11:47 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] New Winner-Worst Cisco
> Documentation Award!!!
> 
> Well folks it's been a long time coming but I think
> we
> have a new winner in the WORST CISCO DOCUMENTATION
> award. I actually wasn't sure it was possible but
> the
> entire Mobility Manager 1.2 documentation set sucks
> so
> bad that its beaten out Personal Assistant 1.3 which
> has held the title for almost 3 yrs now!! A special
> congratulation and kiss my ass goes out to the lazy
> losers that wrote crap!
> 
> Sorry for the rant, Cisco Docs usually aren't too
> bad
> but the 2nd graders that wrote this one have out
> done
> themselves!
> 
> 
> 
>        
>
______________________________________________________________________
> ______________
> Choose the right car based on your needs.  Check out
> Yahoo! Autos new
> Car Finder tool.
> http://autos.yahoo.com/carfinder/
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> <html>
> <body>
>
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>
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> 



      ____________________________________________________________________________________
Luggage? GPS? Comic books? 
Check out fitting gifts for grads at Yahoo! Search
http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz

From smgustafson at gmail.com  Wed Aug 29 09:51:20 2007
From: smgustafson at gmail.com (Steve G)
Date: Wed, 29 Aug 2007 06:51:20 -0700
Subject: [cisco-voip] Deploying Centralized CUE for Multisite WAN
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F0198549F@CNTRRA20-XCS01.telkom.co.za>
References: <8a1f34b50708280820v4736d322n1fe0132ed557e1e3@mail.gmail.com>
	<472E195E692EE74BAE4E7B2880A78C1F0198549F@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <8a1f34b50708290651t33442a5w77fcec71dee425d9@mail.gmail.com>

Sorry, I failed to mention that I was installing this CUE module integrated
with CCM, not CME.

On 8/29/07, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote:
>
>  In a UC 500 update cisco mentioned something about running remote phones
> on UC500. maybe you can try and get a presentation on UC500. As far as I
> know it is based on CME so the functionality will be the same
>
>
>
>
>
>
>
> Erik erasmus
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Steve G
> *Sent:* Tuesday, August 28, 2007 5:21 PM
> *To:* cisco-voip at puck-nether.net
> *Subject:* [cisco-voip] Deploying Centralized CUE for Multisite WAN
>
>
>
> Does anyone have experience with CUE servicing 3 to 4 WAN sites?  I am
> curious how this has worked and if there are any pitfalls I need to worry
> about.  3 of my sites use a common gateway for PSTN (PRI) access, but one
> has it's own PSTN connection with a PRI.  I have read on CCO that it is only
> supported as 1 SRST site per CUE Module.  They also need 4 different
> autoattendants, which CUE allows for 3 + the default.  Can I modify the
> default AA for my 4th AA?  Yea, this is another one of those hairbrained
> Sales Engineer ideas.
>
>
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> This e-mail is subject to an electronic communication legal notice,
> available at :
> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>
>
>
>
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From d4rkf1ber at gmail.com  Wed Aug 29 10:42:15 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Wed, 29 Aug 2007 09:42:15 -0500
Subject: [cisco-voip] federal NRO number assignment rules ?
In-Reply-To: <9552cdf0708290709m50b0f46fg90230ed7f8bc9f20@mail.gmail.com>
References: <a1b07ea60708290646n7eadd32ege276323b2921569a@mail.gmail.com>
	<9552cdf0708290709m50b0f46fg90230ed7f8bc9f20@mail.gmail.com>
Message-ID: <a1b07ea60708290742g6a111062sf5c02cdafbd25f13@mail.gmail.com>

I agree 911 needs to be addressed, and believe me it is.  911 is not
our issue in this case, steps have been taken to ensure that the
correct address is applied for 911 calls and routed to the correct
PSAP.

The issue and question we are facing is simply in determining whether
its ok to assign a user a DID who is located in a area serviced by a
different NPA-XXX



On 8/29/07, charles george <cisco1.charles at gmail.com> wrote:
> users with Cisco phones and DID's from the corporate office who work
> remotely.
>
> Probably has to do with the fact that if someone dials 911 and the DID's
> state that the address for these DID's are at 1 location, the remote users
> will not get help when they call 911 from their Voip phone.  Emergency
> Sevices will go to the address  for the DID not the location of the Call.
>
> They (Emergency Service) will not know.
>
> Might need enhanced 911 for this to comply.
>
> I may be wrong, but right away Federal and remote and DID's this comes to my
> mind with these combination of words.
>
>
> On 8/29/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
> >
> > Recently I asked the group about an internal issue I am facing
> > regarding how we are leveraging DID's and central PRI's.
> >
> > Basically, we have asked for and recieved a block of DID's that all
> > fall into one range.  Those are all tied to a couple of PRI's that
> > come into a central location.
> >
> > We have not only assigned DID's to employees who work out of the
> > central location, but through the magic of VoIP "ohhh ahhh" we have
> > users with Cisco phones and DID's from the corporate office who work
> > remotely.
> >
> > Nothing new here, and nothing I have not seen done every place I have
> > worked out in the last oh 10 years or so.
> >
> > However for the first time I am at a company who has some lawyer /
> > regulatory person on staff who is questioning whether this is legal or
> > not.
> >
> > The current statement from this person experiencing concern is,
> > "compliance with federal NRO number assignment rules".
> >
> > Sounds awfully legal and well thats what I asked for.  I previously
> > told this person that for the most part everyone I have polled
> > regarding these concerns have basically agreed in saying its very
> > common to do this and if it is illegal then show me where and what
> > makes it illegal?  So thats what they replied with, and well I am
> > googling and trying to make sense of what I read but I can't say I
> > fully understand it.
> >
> > Anyone know more about this and understand the implications and how it
> > affects a business doing what I am doing internally for employee
> > phones?
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From amlibk at gmail.com  Wed Aug 29 10:45:19 2007
From: amlibk at gmail.com (billy keller)
Date: Wed, 29 Aug 2007 09:45:19 -0500
Subject: [cisco-voip] Voicemail Indicator Light not Shutting off
Message-ID: <becebc230708290745l1d3c5ae2o3ea45eb0e53b100@mail.gmail.com>

Hi,

There was an email outage yesterday at my office due to the exchange
datastore unexpectedly filling up, and this obviously impacted our ability
to retrieve voicemail messages.  Exchange was restored, however the message
indicator lights on all of our phones will not turn off.  Is there a global
fix for this?

Thanks a bunch!
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From bob at smalltime.com  Wed Aug 29 10:47:22 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 29 Aug 2007 09:47:22 -0500
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
In-Reply-To: <504001.12611.qm@web36106.mail.mud.yahoo.com>
References: <504001.12611.qm@web36106.mail.mud.yahoo.com>
Message-ID: <46D586FA.8070604@smalltime.com>

Ted Nugent wrote:
> Ahh yes however once you finally do track down the
> Unity Doc you need its typically pretty good.
> 
> For instance they wouldn't call a DN (Distinguished
> Name) a Directory Number like they do in the CMM
> docs..

You are in a twisty maze of passageways, all alike...

From kenjohnson at letu.edu  Wed Aug 29 10:51:19 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Wed, 29 Aug 2007 09:51:19 -0500
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <f84a38d30708240917y175cd089u8cecce93938b7638@mail.gmail.com>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com><f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com><5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com><E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org><5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com><6350516B6159444E9D646523EB4526F403EBA87B@apollo.rbh.local><C1FE15183DA37645BC0633BC604E44F0068CD4B4@USNAEXCH.na.didata.local>
	<f84a38d30708240917y175cd089u8cecce93938b7638@mail.gmail.com>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EFB7C63E@rachael-reborn.letnet.net>

I sadly am stuck on 2k for our 5.x move unless support for the Dialogic
cards has materialized on 2003 and I'm just out of the loop - I'm
assuming that things haven't changed since the last time I looked at the
docs unless 5.x includes some proprietary 2k3 drivers but since my
understanding was those were largely OS-level drivers from Intel I'm
assuming if they were available they'd be available to all versions of
Unity......

 

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 Scott Voll
Sent: Friday, August 24, 2007 11:17 AM
To: Jason Aarons (US)
Cc: Rhodes, Geoff; cisco voip list
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4

 

If your running UM and you want Exchange 2007 you have to upgrade Unity
first anyway.... so then you need two people skilled ;-)

 

Scott

 

On 8/24/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote: 

The Exchange 2007 upgrade has it's own caveats, 64-bit hardware, 64-bit
OS and migration of mailboxes....either way you need someone skilled! 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Rhodes, Geoff
Sent: Friday, August 24, 2007 11:01 AM
To: cisco voip list
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4

Maybe they don't want us going to Unity 5, instead they want to push us 
into using Exchange 2007 Unified Messaging...  :)


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Jonathan
Charles
Sent: Thursday, August 23, 2007 11:18 PM
To: Keith Klevenski
Cc: cisco voip list
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4 

That is exactly what I did...

Then when I went to run the Unity 5 install, it failed saying there was
a hostname mismatch... I called TAC, they had no idea what to do and
said, 'this should work...' and then refused to support it. 



Jonathan

On 8/23/07, Keith Klevenski <keith.klevenski at rig.net> wrote:
>
>
>
> This does not sound promising as I'm about to embark on the win2k/4.0x


> to
> win2k3/4.2 to 5.0 myself.  I was going to upgrade to to 4.2 first
> (since
> 4.03sr1 isn't supported on win2k3), dirt backup, install fresh win2k3
> install, dirt restore, upgrade to 5.0.  This is going to be ugly...
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan
> Charles 
> Sent: Thu 8/23/2007 6:34 PM
> To: Scott Voll
> Cc: cisco voip list
> Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4
>
>
>
>
> The Windows issue is the big one I am going to be running into... most


> of the people who want to upgrade is because they want Windows 2000
> out of their network (that is why they went to CCM5.1, MPX, etc...)
>
> A few customers have Unity 4.2(1) on Win2k3, which means the upgrade 
> should be easy, but if they are running Win2K, that means a complete
> blow away and start over...
>
> I did one back in July, and it was a nightmare, we ended up abandoning

> the old system and recreating everything from scratch... 
>
>
> Jonathan
>
> On 8/23/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > I'm planning on this upgrade October / Novemberish but have not 
> > started
> the
> > research on it yet.  If anyone has any comments I would like to know
also.
> >
> > I don't have the Windows issue as I'm already at windows 2k3 with 
> > Exchange
> > 2k3 and Unity 4.0.5 UM.  I'm hoping upgrading to 5.0 will be a
> > little
> easier
> > then Jonathan is talking about.
> >
> > Scott
> > 
> >
> > On 8/23/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > > Has anyone done it successfully?
> > >
> > > I have a LOT of customers on Unity 4.X running Windows 2000 who
> > > want to upgrade to Unity 5 and Windows 2003. However, my first
> > > attempt was a complete disaster (ended up having to install from
scratch).
> > > 
> > > Here is the problem:
> > >
> > > You CANNOT upgrade a Windows 2000 box to Windows 2003 once Unity
> > > is installed (says so in the upgrade documentation).
> > > 
> > > So, you must do a DRT backup, install Windows 2003 clean, install
> > > your old version of Unity 4.X and restore it. Then you can upgrade

> > > to 5
> > >
> > > This is not even slightly as simple as it sounds and has a 
> > > remarkably high failure rate.
> > >
> > > (we got an error that the hostname didn't match and the upgrade to

> > > unity
> > failed)
> > > 
> > > TAC refused to support the upgrade (this was early July, it had
> > > come off of NPH that day).
> > >
> > > If anyone has, I am curious as to what tricks you did to make it 
work...
> > >
> > >
> > >
> > > 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
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________________________________


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From bob at smalltime.com  Wed Aug 29 11:08:16 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 29 Aug 2007 10:08:16 -0500
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <04B6D8CDD22FD34993C47E33E21F82EFB7C63E@rachael-reborn.letnet.net>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com><f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com><5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com><E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org><5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com><6350516B6159444E9D646523EB4526F403EBA87B@apollo.rbh.local><C1FE15183DA37645BC0633BC604E44F0068CD4B4@USNAEXCH.na.didata.local>	<f84a38d30708240917y175cd089u8cecce93938b7638@mail.gmail.com>
	<04B6D8CDD22FD34993C47E33E21F82EFB7C63E@rachael-reborn.letnet.net>
Message-ID: <46D58BE0.8080107@smalltime.com>

Johnson, Ken wrote:
> I sadly am stuck on 2k for our 5.x move unless support for the Dialogic 
> cards has materialized on 2003 and I?m just out of the loop ? I?m 
> assuming that things haven?t changed since the last time I looked at the 
> docs unless 5.x includes some proprietary 2k3 drivers but since my 
> understanding was those were largely OS-level drivers from Intel I?m 
> assuming if they were available they?d be available to all versions of 
> Unity??

I'm assuming that you're talking about the dialogic terminating PRIs; 
can you use an external gateway (2821 / 2851) to terminate the PRIs and 
convert to IP / RTP at that point?

From rsingleton at morsco.com  Wed Aug 29 11:18:15 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Wed, 29 Aug 2007 10:18:15 -0500
Subject: [cisco-voip] Voicemail Indicator Light not Shutting off
In-Reply-To: <becebc230708290745l1d3c5ae2o3ea45eb0e53b100@mail.gmail.com>
References: <becebc230708290745l1d3c5ae2o3ea45eb0e53b100@mail.gmail.com>
Message-ID: <1188400695.24829.9.camel@robert.morsco.com>

On Wed, 2007-08-29 at 09:45 -0500, billy keller wrote:
> Hi,
>  
> There was an email outage yesterday at my office due to the exchange
> datastore unexpectedly filling up, and this obviously impacted our
> ability to retrieve voicemail messages.  Exchange was restored,
> however the message indicator lights on all of our phones will not
> turn off.  Is there a global fix for this?

Not sure about global fix, but there is a tool to resynchronize MWI,
assuming Exchange has their current state. BTW, this resynch process
probably ran overnight, so if the lights are not correct now, this may
not help.

Run the Integration Manager and most systems will have an integration
called "Cisco CallManager Integration". Look at the properties tab for
this integration and there is a setting for MWI synchronization time and
a button to "Resynchronize Now". It will take a little while if you have
a lot of phones.

Robert





From wsisk at cisco.com  Wed Aug 29 11:18:09 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 29 Aug 2007 11:18:09 -0400
Subject: [cisco-voip] Display Logging Parameter on 7911
In-Reply-To: <002a01c7e9ba$59934100$56196883@cfs.uoguelph.ca>
References: <002a01c7e9ba$59934100$56196883@cfs.uoguelph.ca>
Message-ID: <4835945F-3AAC-4462-A6B8-DB00DB5A6A0D@cisco.com>

I believe this is used in conjunction with IP Phone Doctor.  Rather  
than collecting logs by console port to each phone (scalable?) this  
would have phone logs sent out in an L2 multicast to collect by a  
central server.

You probably want it off.

/wes

On Aug 28, 2007, at 5:28 PM, Lelio Fulgenzi wrote:

Can anyone help describe what this is and whether it is a security  
concern? The second sentence says it does not control the generation  
of logs, but the last sentence says you can use it to force logs. I  
wanted to turn it off for a bunch of phones but BAT conveniently does  
not have this setting in the list. Grrrr.


Logging Display :	This option selects what type of console logging is  
allowed. This option does not control the generation of logs - just  
whether the logs display. Disabled indicates that logging does not  
display to the console, nor to the connected downstream port. PC  
Controlled indicates that the       workstation attached to the PC  
port will control whether logging is enabled. Enabled indicates that  
logs are always sent both to the console and to downstream port. Use  
Enabled to force logs on so they can be captured with a packet sniffer.
This is a required field.
Default: PC Controlled.
_______________________________________________
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 Aug 29 11:18:30 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 29 Aug 2007 11:18:30 -0400
Subject: [cisco-voip] Phone Hostname
In-Reply-To: <5d093f9a0708281533h77d08972m775fb777ecb22e55@mail.gmail.com>
References: <5d093f9a0708281533h77d08972m775fb777ecb22e55@mail.gmail.com>
Message-ID: <363E5C63-6FAC-4DF2-9974-FD9B87D88DD0@cisco.com>

Nope.

On Aug 28, 2007, at 6:33 PM, Jonathan Charles wrote:

I have a customer who has a requirement that all devices on the IP
Network conform to the following hostname:

USCHIPHN001

or whatever...

Anyway, is there any way to change the hostname of a Cisco IP Phone to
conform to their network-wide naming convention?

Obviously I can match up DHCP and DNS together, but the device itself
will still have SEP000318397489 as its hostname...



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


From bcarter at sentinel.com  Wed Aug 29 12:20:22 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed, 29 Aug 2007 11:20:22 -0500
Subject: [cisco-voip] OT: Digital Dictation System
Message-ID: <6A0744639B86DD459433108DBF101D26018BDDB4@MAIL01.sentinel.com>

Customer is asking for recommendations on Digital Dictation systems.
Any suggestions?  Anything that works with Cisco VoIP?
 
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From svoll.voip at gmail.com  Wed Aug 29 13:04:34 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 29 Aug 2007 10:04:34 -0700
Subject: [cisco-voip] CER -- Switch port Details
Message-ID: <f84a38d30708291004u193c419k1fc157cdd1ff1c63@mail.gmail.com>

I have configured the SNMP community strings / and ran the switch / phone
update.  I see the ports on ths switches but I do not see the phone
extensions / IP / Type.  is this a problem?  or do I have to assign the ERL
fist?  I don't have the ERL's yet as I'm still waiting on the service
provider.

TIA

Scott
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From svoll.voip at gmail.com  Wed Aug 29 13:21:32 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 29 Aug 2007 10:21:32 -0700
Subject: [cisco-voip] CER -- Switch port Details
In-Reply-To: <f84a38d30708291004u193c419k1fc157cdd1ff1c63@mail.gmail.com>
References: <f84a38d30708291004u193c419k1fc157cdd1ff1c63@mail.gmail.com>
Message-ID: <f84a38d30708291021t5671e421xd21b8bc09ec1db44@mail.gmail.com>

Never mind...... impatience.

Scott


On 8/29/07, Scott Voll <svoll.voip at gmail.com> wrote:
>
> I have configured the SNMP community strings / and ran the switch / phone
> update.  I see the ports on ths switches but I do not see the phone
> extensions / IP / Type.  is this a problem?  or do I have to assign the ERL
> fist?  I don't have the ERL's yet as I'm still waiting on the service
> provider.
>
> TIA
>
> Scott
>
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From jason.aarons at us.didata.com  Wed Aug 29 13:27:27 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 29 Aug 2007 13:27:27 -0400
Subject: [cisco-voip] OT: Digital Dictation System
In-Reply-To: <6A0744639B86DD459433108DBF101D26018BDDB4@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26018BDDB4@MAIL01.sentinel.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069A920B@USNAEXCH.na.didata.local>

Most are USB Mic Applications back to file server, similar to how you
upload/download music files from iTunes, only it's Dictaphone instead of
iTunes, and you get to record the music with your USB microphone. Thus
no integration with the phone system.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Wednesday, August 29, 2007 12:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] OT: Digital Dictation System

 

Customer is asking for recommendations on Digital Dictation systems.
Any suggestions?  Anything that works with Cisco VoIP?

 




-----------------------------------------
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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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From jonvoip at gmail.com  Wed Aug 29 13:21:41 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 29 Aug 2007 12:21:41 -0500
Subject: [cisco-voip] Phone Hostname
In-Reply-To: <363E5C63-6FAC-4DF2-9974-FD9B87D88DD0@cisco.com>
References: <5d093f9a0708281533h77d08972m775fb777ecb22e55@mail.gmail.com>
	<363E5C63-6FAC-4DF2-9974-FD9B87D88DD0@cisco.com>
Message-ID: <5d093f9a0708291021l285fb0f8u86badecb8cfa2bfd@mail.gmail.com>

Cool.


Jonathan

On 8/29/07, Wes Sisk <wsisk at cisco.com> wrote:
> Nope.
>
> On Aug 28, 2007, at 6:33 PM, Jonathan Charles wrote:
>
> I have a customer who has a requirement that all devices on the IP
> Network conform to the following hostname:
>
> USCHIPHN001
>
> or whatever...
>
> Anyway, is there any way to change the hostname of a Cisco IP Phone to
> conform to their network-wide naming convention?
>
> Obviously I can match up DHCP and DNS together, but the device itself
> will still have SEP000318397489 as its hostname...
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From ealeatherman at gmail.com  Wed Aug 29 13:53:49 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 29 Aug 2007 13:53:49 -0400
Subject: [cisco-voip] OT: Digital Dictation System
In-Reply-To: <6A0744639B86DD459433108DBF101D26018BDDB4@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26018BDDB4@MAIL01.sentinel.com>
Message-ID: <94a1afde0708291053n5afe47fak65cf3bb0f5a34825@mail.gmail.com>

I just had the exact same question today asked to us.. user's weren't happy
with the price tag an outside vendor had provided. I offered to set them up
a voicemail box specifically for the VIP to record his dictations to, but
that didn't fly. Was the only thing I could think of on the voip system.

On 8/29/07, Carter, Bill <bcarter at sentinel.com> wrote:
>
>  Customer is asking for recommendations on Digital Dictation systems.  Any
> suggestions?  Anything that works with Cisco VoIP?
>
>

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From Peterson_Erik at aac.com  Wed Aug 29 13:21:30 2007
From: Peterson_Erik at aac.com (Peterson_Erik at aac.com)
Date: Wed, 29 Aug 2007 13:21:30 -0400
Subject: [cisco-voip] federal NRO number assignment rules ?
Message-ID: <OFA0022A33.FAE3D7B7-ON85257346.005F38E8-85257346.005F5A7B@aac.com>

You can find some details on the Numbering Resource Optimization order 
here:

http://www.fcc.gov/omd/pra/3060-0895.html

From WOCAMPO at broward.org  Wed Aug 29 14:19:28 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Wed, 29 Aug 2007 14:19:28 -0400
Subject: [cisco-voip] Cisco Unity
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA4BE@UVEXVS01.bc.broward.cty>

One of the remote location had a 7800HP server running Cisco Unity. The
server has the following alarm,

Anyone had this before?  

 

Error: Event ID 1186

IDE ATA disk status change. The ATA disk drive with model MAXTOR 6e040l0
& serial number E1LZFB1E has new status 3 of 4

ATA disk status values

*  1-other

*  2-ok

*  3=smart error

*  4=failed

 

Thanks,

 

 

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 lmarasci at cisco.com  Wed Aug 29 14:25:50 2007
From: lmarasci at cisco.com (Louis Marascio (lmarasci))
Date: Wed, 29 Aug 2007 11:25:50 -0700
Subject: [cisco-voip] OT: Digital Dictation System
In-Reply-To: <94a1afde0708291053n5afe47fak65cf3bb0f5a34825@mail.gmail.com>
Message-ID: <6EE7932CC71C8D49A20C0E639BCD2165028C012D@xmb-sjc-21c.amer.cisco.com>

Without knowing the specific requirements of such a system, if all you
need to do is handle inbound phone calls and record a message based on
the user then this would be a very straight forward application to build
with the Cisco Unified Application Environment. Answering an incoming
call, gathering a PIN for authorization (if required), and recording the
audio should take all of about 10 minutes to wire together.
 
More information (and free software for download) can be found here:
http://www.cisco.com/web/developer/cuae/
 
Best regards,
 
Louis

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Wednesday, August 29, 2007 12:54 PM
To: Carter, Bill
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] OT: Digital Dictation System


I just had the exact same question today asked to us.. user's weren't
happy with the price tag an outside vendor had provided. I offered to
set them up a voicemail box specifically for the VIP to record his
dictations to, but that didn't fly. Was the only thing I could think of
on the voip system. 


On 8/29/07, Carter, Bill <bcarter at sentinel.com> wrote: 

	Customer is asking for recommendations on Digital Dictation
systems.  Any suggestions?  Anything that works with Cisco VoIP?
	 


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 
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From GRhodes at rbh.com  Wed Aug 29 15:19:00 2007
From: GRhodes at rbh.com (Rhodes, Geoff)
Date: Wed, 29 Aug 2007 15:19:00 -0400
Subject: [cisco-voip] Syn Server failed?
In-Reply-To: <6EE7932CC71C8D49A20C0E639BCD2165028C012D@xmb-sjc-21c.amer.cisco.com>
References: <94a1afde0708291053n5afe47fak65cf3bb0f5a34825@mail.gmail.com>
	<6EE7932CC71C8D49A20C0E639BCD2165028C012D@xmb-sjc-21c.amer.cisco.com>
Message-ID: <6350516B6159444E9D646523EB4526F403EBA8EE@apollo.rbh.local>

I keep seeing this error on my Subscriber (not on the Publisher):
 
Event Type: Information
Event Source: Cisco Desktop Enterprise Server
Event Category: None
Event ID: 2
Date:  8/29/2007
Time:  3:01:47 PM
User:  N/A
Computer: ADONIS
Description:
SS3024 Syn Server failed to get device information, return error <Could
not get resource description (COMM_FAILURE)>, try config file
 
In fact, I don't even see a service on my Publisher called "Cisco
Desktop Sync Server".  There are other "Cisco Desktop" services running
on the Subscriber too that aren't on the Publisher, such as "Cisco
Desktop Call Chat Server", "Cisco Desktop TAI Server" and Cisco Desktop
RASCAL Server".
 
Any ideas on why this is so and why it's having issues?  Both servers
are running OS 2000.4.5a and are Call Manager 4.2.3sr1.
 
Many thanks in advance.
 
Geoff Rhodes
Director, Information Technology
Robinson, Bradshaw & Hinson, P.A.
704 / 377-8188
grhodes at rbh.com
 
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From jason.aarons at us.didata.com  Wed Aug 29 15:32:20 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 29 Aug 2007 15:32:20 -0400
Subject: [cisco-voip] Syn Server failed?
In-Reply-To: <6350516B6159444E9D646523EB4526F403EBA8EE@apollo.rbh.local>
References: <94a1afde0708291053n5afe47fak65cf3bb0f5a34825@mail.gmail.com><6EE7932CC71C8D49A20C0E639BCD2165028C012D@xmb-sjc-21c.amer.cisco.com>
	<6350516B6159444E9D646523EB4526F403EBA8EE@apollo.rbh.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069A94E2@USNAEXCH.na.didata.local>

UCCX Co-resident? Using TAPS? What version UCCX? These services are
related to Contact Center Express.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rhodes, Geoff
Sent: Wednesday, August 29, 2007 3:19 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Syn Server failed?

 

I keep seeing this error on my Subscriber (not on the Publisher):

 

Event Type: Information
Event Source: Cisco Desktop Enterprise Server
Event Category: None
Event ID: 2
Date:  8/29/2007
Time:  3:01:47 PM
User:  N/A
Computer: ADONIS
Description:
SS3024 Syn Server failed to get device information, return error <Could
not get resource description (COMM_FAILURE)>, try config file

 

In fact, I don't even see a service on my Publisher called "Cisco
Desktop Sync Server".  There are other "Cisco Desktop" services running
on the Subscriber too that aren't on the Publisher, such as "Cisco
Desktop Call Chat Server", "Cisco Desktop TAI Server" and Cisco Desktop
RASCAL Server".

 

Any ideas on why this is so and why it's having issues?  Both servers
are running OS 2000.4.5a and are Call Manager 4.2.3sr1.

 

Many thanks in advance.

 

Geoff Rhodes

Director, Information Technology

Robinson, Bradshaw & Hinson, P.A.

704 / 377-8188

grhodes at rbh.com

 




-----------------------------------------
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 GRhodes at rbh.com  Wed Aug 29 15:41:34 2007
From: GRhodes at rbh.com (Rhodes, Geoff)
Date: Wed, 29 Aug 2007 15:41:34 -0400
Subject: [cisco-voip] Syn Server failed?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0069A94E2@USNAEXCH.na.didata.local>
References: <94a1afde0708291053n5afe47fak65cf3bb0f5a34825@mail.gmail.com><6EE7932CC71C8D49A20C0E639BCD2165028C012D@xmb-sjc-21c.amer.cisco.com>
	<6350516B6159444E9D646523EB4526F403EBA8EE@apollo.rbh.local>
	<C1FE15183DA37645BC0633BC604E44F0069A94E2@USNAEXCH.na.didata.local>
Message-ID: <6350516B6159444E9D646523EB4526F403EBA8F1@apollo.rbh.local>

Ah, that must be it, IPCC Express.  I'm going to just disable that one
service and see what happens.

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com] 
Sent: Wednesday, August 29, 2007 3:32 PM
To: Rhodes, Geoff; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Syn Server failed?



UCCX Co-resident? Using TAPS? What version UCCX? These services are
related to Contact Center Express.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rhodes, Geoff
Sent: Wednesday, August 29, 2007 3:19 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Syn Server failed?

 

I keep seeing this error on my Subscriber (not on the Publisher):

 

Event Type: Information
Event Source: Cisco Desktop Enterprise Server
Event Category: None
Event ID: 2
Date:  8/29/2007
Time:  3:01:47 PM
User:  N/A
Computer: ADONIS
Description:
SS3024 Syn Server failed to get device information, return error <Could
not get resource description (COMM_FAILURE)>, try config file

 

In fact, I don't even see a service on my Publisher called "Cisco
Desktop Sync Server".  There are other "Cisco Desktop" services running
on the Subscriber too that aren't on the Publisher, such as "Cisco
Desktop Call Chat Server", "Cisco Desktop TAI Server" and Cisco Desktop
RASCAL Server".

 

Any ideas on why this is so and why it's having issues?  Both servers
are running OS 2000.4.5a and are Call Manager 4.2.3sr1.

 

Many thanks in advance.

 

Geoff Rhodes

Director, Information Technology

Robinson, Bradshaw & Hinson, P.A.

704 / 377-8188

grhodes at rbh.com

 

________________________________

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 c3voip at nc.rr.com  Wed Aug 29 15:48:07 2007
From: c3voip at nc.rr.com (c3voip)
Date: Wed, 29 Aug 2007 15:48:07 -0400
Subject: [cisco-voip] RSS Feeds on IP Phones
In-Reply-To: <f48ccm$to6$2@sea.gmane.org>
References: <f46jql$lh1$1@sea.gmane.org> <f48ccm$to6$2@sea.gmane.org>
Message-ID: <000001c7ea75$856ced10$9046c730$@rr.com>

Any perl experts out there?  I keep getting an XML/Parser.pm error when
trying to pull up one of the RSS Feeds that I have setup on this script.
The script itself will load and I see the menu, but when I select one of the
Feeds I get an error.

Here is the error:
no element found at line 1, column 0, byte -1 at
/usr/lib/perl5/vendor_perl/5.8.8/i386-linux-thread-multi/XML/Parser.pm line
187

Thanks,
-C

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kalle Tetto
Sent: Thursday, June 07, 2007 3:34 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RSS Feeds on IP Phones


I've very successfully installed this:

http://dontpokebadgers.com/RSS2Cisco/how-to.html

just install it on a seperate linux or windows machine and let the ip 
phone service point to this machine.

Stift schrieb:
> I heard a lot about that it is possible to read RSS feeds on a Cisco IP 
> Phone. But I can't find a working IP Phone Service. Has anybody that 
> configured?

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


From kenjohnson at letu.edu  Wed Aug 29 16:30:43 2007
From: kenjohnson at letu.edu (Johnson, Ken)
Date: Wed, 29 Aug 2007 15:30:43 -0500
Subject: [cisco-voip] Unity 5 upgrading from Unity 4
In-Reply-To: <46D58BE0.8080107@smalltime.com>
References: <5d093f9a0708231510y1f40d9ccu9ef8d9053f27cd14@mail.gmail.com><f84a38d30708231555q3709813asbe9efce7a351b5c7@mail.gmail.com><5d093f9a0708231634l5e8fda35x96504db8edd23872@mail.gmail.com><E0F48CE151B5D440B3649903D7221DA20752A4@HOUEXH02.rignet.org><5d093f9a0708232018o57d506f0ie2438467c66cab7f@mail.gmail.com><6350516B6159444E9D646523EB4526F403EBA87B@apollo.rbh.local><C1FE15183DA37645BC0633BC604E44F0068CD4B4@USNAEXCH.na.didata.local>	<f84a38d30708240917y175cd089u8cecce93938b7638@mail.gmail.com><04B6D8CDD22FD34993C47E33E21F82EFB7C63E@rachael-reborn.letnet.net>
	<46D58BE0.8080107@smalltime.com>
Message-ID: <04B6D8CDD22FD34993C47E33E21F82EFB7C7EE@rachael-reborn.letnet.net>

No - I wish it was a PRI but they stopped support for purchasing the PRI
cards (as far as I could tell) before we upgraded our hardware last year
- it's still on pots lines into a Dialogic D/120JCT-LS.

(This is a split system with unity serving the administrative
callmanager cluster and our legacy NEAX serving our residence halls.

I don't know that I'd ever considered converting those lines in an
external gateway - admittedly we haven't revisited the preferred method
for connecting legacy systems to Unity since we originally deployed in
2001 :-/  I might need to read up more on non-Dialogic options for
pulling that off it sounds like and move to something a little more
standard :-)

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 Robert
Kulagowski
Sent: Wednesday, August 29, 2007 10:08 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity 5 upgrading from Unity 4

Johnson, Ken wrote:
> I sadly am stuck on 2k for our 5.x move unless support for the
Dialogic 
> cards has materialized on 2003 and I'm just out of the loop - I'm 
> assuming that things haven't changed since the last time I looked at
the 
> docs unless 5.x includes some proprietary 2k3 drivers but since my 
> understanding was those were largely OS-level drivers from Intel I'm 
> assuming if they were available they'd be available to all versions of

> Unity......

I'm assuming that you're talking about the dialogic terminating PRIs; 
can you use an external gateway (2821 / 2851) to terminate the PRIs and 
convert to IP / RTP at that point?
_______________________________________________
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 Aug 29 16:56:37 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 29 Aug 2007 16:56:37 -0400
Subject: [cisco-voip] CER -- Switch port Details
In-Reply-To: <f84a38d30708291004u193c419k1fc157cdd1ff1c63@mail.gmail.com>
References: <f84a38d30708291004u193c419k1fc157cdd1ff1c63@mail.gmail.com>
Message-ID: <46D5DD85.4040102@cisco.com>

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From wsisk at cisco.com  Wed Aug 29 16:58:17 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 29 Aug 2007 16:58:17 -0400
Subject: [cisco-voip] OT: Digital Dictation System
In-Reply-To: <6EE7932CC71C8D49A20C0E639BCD2165028C012D@xmb-sjc-21c.amer.cisco.com>
References: <6EE7932CC71C8D49A20C0E639BCD2165028C012D@xmb-sjc-21c.amer.cisco.com>
Message-ID: <46D5DDE9.8050002@cisco.com>

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From manuel.cristobal at us.didata.com  Wed Aug 29 18:01:01 2007
From: manuel.cristobal at us.didata.com (Manuel Cristobal (US))
Date: Wed, 29 Aug 2007 18:01:01 -0400
Subject: [cisco-voip] isdn switch-type for E1 in London-UK
In-Reply-To: <46D5DDE9.8050002@cisco.com>
References: <6EE7932CC71C8D49A20C0E639BCD2165028C012D@xmb-sjc-21c.amer.cisco.com>
	<46D5DDE9.8050002@cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069A9778@USNAEXCH.na.didata.local>

 

Anyone can tell me what type  of isdn switch-type to configure in a 2800
voice gateway for a E1 in London from the ones below?

 

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)

 

Thanks. 

Regards,

 

Manuel Cristobal

Solutions Architect




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From keli at carocomp.ro  Wed Aug 29 19:12:12 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Thu, 30 Aug 2007 02:12:12 +0300
Subject: [cisco-voip] Cisco Unity
In-Reply-To: <F4DD4283A3EF194B84D774C60FB7F786024DA4BE@UVEXVS01.bc.broward.cty>
References: <F4DD4283A3EF194B84D774C60FB7F786024DA4BE@UVEXVS01.bc.broward.cty>
Message-ID: <20070830021212.326nuks9z4s88wgo@mail.carocomp.ro>

If I were you, I'd upload a S.M.A.R.T. monitoring software (you should  
find some freebies on the net I guess) to he server, and check out  
what's going on.

However, if the disk has a smart error, you should be getting ready to  
change it sometime soon.

regards,
   Zoltan

Quoting "Ocampo, Walter" <WOCAMPO at broward.org>:

> One of the remote location had a 7800HP server running Cisco Unity. The
> server has the following alarm,
>
> Anyone had this before?
>
>
>
> Error: Event ID 1186
>
> IDE ATA disk status change. The ATA disk drive with model MAXTOR 6e040l0
> & serial number E1LZFB1E has new status 3 of 4
>
> ATA disk status values
>
> *  1-other
>
> *  2-ok
>
> *  3=smart error
>
> *  4=failed
>
>
>
> Thanks,
>
>
>
>
>
> 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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This message was sent using IMP, the Internet Messaging Program.


From jgrace at adcapnet.com  Wed Aug 29 19:16:39 2007
From: jgrace at adcapnet.com (James Grace)
Date: Wed, 29 Aug 2007 19:16:39 -0400
Subject: [cisco-voip] i can not configure ip comunicator on cm 6.0
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C763@mail.adcapnetsvc.local>

?
I can not configure ip communicator on cm 6.0 .  For some reason i get a registration rejected error.  I have the ip com phone enter in cm.  I also have other phone in the Database.  but when I start up my softphone, it thinks its not registered.
 
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
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or taking of any action in reliance upon this information by persons or entities other than the intended
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From kspellman02 at yahoo.com  Wed Aug 29 19:14:05 2007
From: kspellman02 at yahoo.com (K S)
Date: Wed, 29 Aug 2007 16:14:05 -0700 (PDT)
Subject: [cisco-voip] Is this Unity's death knell?
In-Reply-To: <170044.24423.qm@web50811.mail.re2.yahoo.com>
Message-ID: <118256.3913.qm@web53809.mail.re2.yahoo.com>

Ahhh the "Free" Unified Messaging from MS.  Free is a
bundled price....something else in that bundle could
also be called free. Check out the Cisco Workspace
License.....lots of "free" there as well.

Free from MS is a Software license that requires MS
users to buy software support on for for the rest of
it's natural life....or it becomes illegal. 

Free is calling MS tech support when the Intel Gateway
they recommend won't talk to your NEC/Nortel?Avaya
PBX...thats a trip in itself. 

T.38 IS supported with CUCM in H.323 mode so that
shouldn't be an issue.

The new versions of Unity (4.2, 5.0) offer a ton of
enhancements that MS doesn't have. MWI for starters
(there is 3rd party for MS that charges per user plus
the application). Also Unity users don't have to
change their TUI...they can use what they are already
using with the custom TUI settings in Unity so there
is no re-learning by the user. 

Cisco's encryption of messages means security that MS
doesn't have. Imagine having someone forward an
internal message that had critical business info on it
to someone off the network. Cisco Unity prevents that.

Cisco has failover for Unity that is at least as
reliable as the MS clustering....probably more so.

Speech Rec on Unity is in its first phase in the 5.0
product and will be getting better over the next 3
months from what I hear.

Again, there are many enhancements to Unity that make
"free" not so attractive.

You've got to look at the overall offer and how it
impacts the business. 




--- Jake Doe <jd80301 at yahoo.com> wrote:

> Perhaps not through Play on Phone (since it's just
> an
> embedded Windows Media Player).  However, if you use
> the TUI or VUI, then you have the ability to slow
> down
> or speed up the message.  See the Outlook Voice
> Access
> (OVA) Quick Reference Guide for those options:
> 
>
http://technet.microsoft.com/en-us/library/bb397228.aspx
> 
> --- Robert Kulagowski <bob at smalltime.com> wrote:
> 
> > Jake Doe wrote:
> > > Yes, Exchange 2007 UM allows you to play your
> > voice
> > > mail through the phone as well.  However, the
> Play
> > on
> > > Phone feature is only available with Outlook
> 2007
> > or
> > > Outlook Web Access 2007 (e.g. no plug-in
> > required). 
> > > Otherwise, the voice mail is just another
> WMA/WAV
> > file
> > > attachment in older versions of Outlook.
> > 
> >  From what I recall at Interop, they don't have a
> > pitch-adjusted tempo 
> > feature during playback, so you can't slow down
> the
> > audio to catch a 
> > phone number which may have been spoken too
> quickly
> > to make out.
> > 
> > ObPetPeeve:  People who delight in giving you a
> > phone number in a 
> > voicemail as quickly as possible as some sort of
> > personal contest.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> 
>        
>
____________________________________________________________________________________
> 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
> 



       
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Need a vacation? Get great deals
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From jason.aarons at us.didata.com  Wed Aug 29 21:54:17 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 29 Aug 2007 21:54:17 -0400
Subject: [cisco-voip] Display at top of 7961 alternating
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069A986C@USNAEXCH.na.didata.local>

I noticed on a new 4.2.3 install the 7961 phone at the top of display
alternated between date/time and the external mask. 

 

 Is this a new feature with 4.2.3? I don't ever recall seeing this
before?

 




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From dalejandro_lm at yahoo.es  Wed Aug 29 23:34:08 2007
From: dalejandro_lm at yahoo.es (david)
Date: Thu, 30 Aug 2007 03:34:08 +0000 (GMT)
Subject: [cisco-voip] CME 4.1-unknown number
Message-ID: <682383.66064.qm@web27106.mail.ukl.yahoo.com>

Hi folks,
I'd been configuring a CME, it has 4FXS/DID and 4FXO ports, FXO ports  has a standard configuration with a DIAL PEER and a conection PLAR, but when someone call through PSTN, the IP PHONES (7940 and 7912) display UNKNOWN NUMBER.
If some body could give me a clue?
Thanks and best regards.
David


       
____________________________________________________________________________________
S? un Mejor Amante del Cine                         
?Quieres saber c?mo? ?Deja que otras personas te ayuden!
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From ml at iridescent.org  Thu Aug 30 03:31:20 2007
From: ml at iridescent.org (Lawrence E. Bakst)
Date: Thu, 30 Aug 2007 00:31:20 -0700
Subject: [cisco-voip] CME 4.1-unknown number
In-Reply-To: <682383.66064.qm@web27106.mail.ukl.yahoo.com>
References: <682383.66064.qm@web27106.mail.ukl.yahoo.com>
Message-ID: <p062408acc2fc226609c7@[192.168.0.7]>

My config is different than yours, but do you have "caller-id enable" on the "voice-port"?

leb


At 3:34 AM +0000 8/30/07, david wrote:
>Hi folks,
>I'd been configuring a CME, it has 4FXS/DID and 4FXO ports, FXO ports  has a standard configuration with a DIAL PEER and a conection PLAR, but when someone call through PSTN, the IP PHONES (7940 and 7912) display UNKNOWN NUMBER.
>If some body could give me a clue?
>Thanks and best regards.
>David
> 
>
>
>
>S? un Mejor Amante del Cine
>?Quieres saber c?mo? <http://us.rd.yahoo.com/mail/es/tagline/beabetter/*http://advision.webevents.yahoo.com/reto/entretenimiento.html>?Deja que otras personas te ayuden! .
>
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip



From Nick.Kassel at Charles-Stanley.co.uk  Thu Aug 30 03:49:55 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 30 Aug 2007 08:49:55 +0100
Subject: [cisco-voip] isdn switch-type for E1 in London-UK
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC014520E8@CSEXCH01.charles-stanley.co.uk>

Primary-net5

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manuel
Cristobal (US)
Sent: 29 August 2007 23:01
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] isdn switch-type for E1 in London-UK

 

 

Anyone can tell me what type  of isdn switch-type to configure in a 2800
voice gateway for a E1 in London from the ones below?

 

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)

 

Thanks. 

Regards,

 

Manuel Cristobal

Solutions Architect

________________________________

Disclaimer: This e-mail communication and any attachments may contain
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From cbomba at s4nets.com  Thu Aug 30 09:10:08 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Thu, 30 Aug 2007 09:10:08 -0400 (EDT)
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
Message-ID: <54732.192.168.1.70.1188479408.webmail@192.168.1.70>

I had to install Mobility Manager last year when it was still pretty new and when I called TAC the engineer asked me to send him the document because they sent me to a TAC engineer that hadn't even seen CMM yet.  After working with the guy for a full day they finally escalated to someone that knew what they were talking about and I got it working.  The documentation was Poo back then and haven't dealt with it since, but I guess people didn't complain enough to have them go back and rework the docs.  I was glad to get that behind me.  

Can't wait to see what we have to do to get that functionality working in CUCM 6.0.  It is built into the software now, no need for separate server.

Chris


Date: Wed, 29 Aug 2007 09:47:22 -0500
From: Robert Kulagowski <bob at smalltime.com>
Subject: Re: [cisco-voip] New Winner-Worst Cisco Documentation
	Award!!!
To: cisco-voip at puck.nether.net
Message-ID: <46D586FA.8070604 at smalltime.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Ted Nugent wrote:
> Ahh yes however once you finally do track down the
> Unity Doc you need its typically pretty good.
> 
> For instance they wouldn't call a DN (Distinguished
> Name) a Directory Number like they do in the CMM
> docs..

You are in a twisty maze of passageways, all alike...




From MillerS at DicksteinShapiro.COM  Thu Aug 30 09:14:44 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Thu, 30 Aug 2007 09:14:44 -0400
Subject: [cisco-voip] Question About Stopping CSVPIMconnector Services!
References: <253FAE7A06B11B49A3A29770B2B8BAFC014520E8@CSEXCH01.charles-stanley.co.uk>
Message-ID: <421CAA1A5300B84A87D200D709985782041F27@DCEX2.DSMO.COM>

I am receiving errors on my Unity boxes concerning VPIM and I am told
that I can eliminate those errors by simply disabling/turning off the
CSVPIM Connector service.  We have a a primary and failover Unity box
and we do not use anything other than Unity for voicemail.  We have
Unified Messaging through Exchange, but no other voicemail systems that
are connected.
 
I just want to be sure that stopping the CSVPIM connector service will
not have any unintended consequences and that it is not necessary for
the regular Exchange Unified Messaging integration.  If anyone has any
thoughts or insight, I would greatly appreciate it.  I am overly
cautious, I know.
 

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: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Thursday, August 30, 2007 3:50 AM
To: Manuel Cristobal (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] isdn switch-type for E1 in London-UK



Primary-net5

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manuel
Cristobal (US)
Sent: 29 August 2007 23:01
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] isdn switch-type for E1 in London-UK

 

 

Anyone can tell me what type  of isdn switch-type to configure in a 2800
voice gateway for a E1 in London from the ones below?

 

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)

 

Thanks. 

Regards,

 

Manuel Cristobal

Solutions Architect




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From baryonyx5 at gmail.com  Thu Aug 30 10:47:40 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Thu, 30 Aug 2007 10:47:40 -0400
Subject: [cisco-voip] Deploying Centralized CUE for Multisite WAN
In-Reply-To: <8a1f34b50708280820v4736d322n1fe0132ed557e1e3@mail.gmail.com>
References: <8a1f34b50708280820v4736d322n1fe0132ed557e1e3@mail.gmail.com>
Message-ID: <a749d0620708300747l3b7a5995h37766e3983f6fe05@mail.gmail.com>

Unless something has changed recently, CUE does not support G.729 so if
you're using that codec over the WAN you'll need to have transcoder
resources available to the CUE.

On 8/28/07, Steve G <smgustafson at gmail.com> wrote:
>
> Does anyone have experience with CUE servicing 3 to 4 WAN sites?  I am
> curious how this has worked and if there are any pitfalls I need to worry
> about.  3 of my sites use a common gateway for PSTN (PRI) access, but one
> has it's own PSTN connection with a PRI.  I have read on CCO that it is only
> supported as 1 SRST site per CUE Module.  They also need 4 different
> autoattendants, which CUE allows for 3 + the default.  Can I modify the
> default AA for my 4th AA?  Yea, this is another one of those hairbrained
> Sales Engineer ideas.
> _______________________________________________
> 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 c3voip at nc.rr.com  Thu Aug 30 10:48:09 2007
From: c3voip at nc.rr.com (c3voip)
Date: Thu, 30 Aug 2007 10:48:09 -0400
Subject: [cisco-voip] RSS Feeds on IP Phones
In-Reply-To: <000001c7ea75$856ced10$9046c730$@rr.com>
References: <f46jql$lh1$1@sea.gmane.org> <f48ccm$to6$2@sea.gmane.org>
	<000001c7ea75$856ced10$9046c730$@rr.com>
Message-ID: <000b01c7eb14$c8703aa0$5950afe0$@rr.com>

Found the problem:

SELinux httpd_can_network_connect Boolean was set by default to off, just
had to turn it on.

Thanks,
-C

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of c3voip
Sent: Wednesday, August 29, 2007 3:48 PM
To: 'Kalle Tetto'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RSS Feeds on IP Phones

Any perl experts out there?  I keep getting an XML/Parser.pm error when
trying to pull up one of the RSS Feeds that I have setup on this script.
The script itself will load and I see the menu, but when I select one of the
Feeds I get an error.

Here is the error:
no element found at line 1, column 0, byte -1 at
/usr/lib/perl5/vendor_perl/5.8.8/i386-linux-thread-multi/XML/Parser.pm line
187

Thanks,
-C

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kalle Tetto
Sent: Thursday, June 07, 2007 3:34 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RSS Feeds on IP Phones


I've very successfully installed this:

http://dontpokebadgers.com/RSS2Cisco/how-to.html

just install it on a seperate linux or windows machine and let the ip 
phone service point to this machine.

Stift schrieb:
> I heard a lot about that it is possible to read RSS feeds on a Cisco IP 
> Phone. But I can't find a working IP Phone Service. Has anybody that 
> configured?

_______________________________________________
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 baryonyx5 at gmail.com  Thu Aug 30 10:50:05 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Thu, 30 Aug 2007 10:50:05 -0400
Subject: [cisco-voip] Call FWD during SRST
In-Reply-To: <OF06CD3469.343CC6E5-ON85257345.00669B21-85257345.00685D39@bftg.com>
References: <OF06CD3469.343CC6E5-ON85257345.00669B21-85257345.00685D39@bftg.com>
Message-ID: <a749d0620708300750p58e23883r5b82b97457581f30@mail.gmail.com>

You should be able to use the cfw option on the alias command like this;

* alias 4 4700 to 4704 preference 4 cfw 9xxxxxxxxxx timeout 16*

On 8/28/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote:
>
>
> I am looking to accomplish the following with SRST.
>
> In SRST I want 2 DID's that route to 2 different hunt groups (2 business
> units same location).  If the call is not answered then I want to CFWD each
> DID to it's own  external number.  One per business unit.
>
> I thought this could be done with the alias command but upon reading
> further it appears that an alias is only valid with locally REGISTERED
> numbers on the gateway.  Config was going to be something like this:
>
> *alias 1 4700 to 4701 preference 1
> alias 2 4700 to 4702 preference 2
> alias 3 4700 to 4703 preference 3
> alias 4 4700 to 4704 preference 4
> alias 5 4700 to 9XXXXXXXXXX preference 5  **--- **Want to send this to
> PSTN**
>
> alias 15 4749 to 4745 preference 1
> alias 16 4749 to 4744 preference 2
> alias 17 4749 to 4740 preference 3
> alias 18 4749 to 9XXXXXXXXXX preference 4  --- **Want to send this to the
> PSTN**
>
> call-forward busy 9XXXXXXXXXX
> call-forward noan 9XXXXXXXXXX timeout 16*
>
>
> I believe the CFWD busy or CFWD noan can be to the PSTN but in this case I
> need specific calls to go to different PSTN destinations.  Any help would be
> appreciated.
>
> Carlos
>
>
> _______________________________________________
> 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 manoj.kalpage at gmail.com  Thu Aug 30 11:24:04 2007
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Fri, 31 Aug 2007 00:24:04 +0900
Subject: [cisco-voip] ISDN SIMULATOR ISDN BRI-U and BRI-S/T
Message-ID: <bbe09c1d0708300824o211cd9ddl880c4a4ee883a923@mail.gmail.com>

Hi All,
I am looking for ISDN simulator for CCIE voice lab. I found there are
two type of ISDN simulators at http://www.vconsole.com/ as bellow.

ISDN BRI/ST
http://www.vconsole.com/2-Port-T1/E1/PRI-+-8-port-BRI-S/T-ISDN-Simulator-p-15.html

ISDN BRI-U
http://www.vconsole.com/2-Port-T1/E1/PRI-+-8-port-BRI-U-ISDN-Simulator-p-14.html

First thing, I cannot understand is what is the different between
BRI-U and BRI-S/T and next thing is which one is good for CCIE voice
LAB.

Are these good or anyone know any other product which is better than this?

Best regards,
PERE

From jason.aarons at us.didata.com  Thu Aug 30 11:28:46 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 30 Aug 2007 11:28:46 -0400
Subject: [cisco-voip] 4.2 Shared line chime
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069A9DEC@USNAEXCH.na.didata.local>

Randy Manager - Line1 - 4000 (shared)

 

Sharon Receptionist - Line 1 - 4001

Sharon Receptionist - Line14 - 4000 (shared)

 

Elizabeth Employee - Line1 - 6600

 

If Sharon uses 4001 to dial 6600 then Randy's 7961 main display (same
area as You have Voicemail) shows "Sharon > Elizabeth" and makes a
chime.  Any way to disabled this feature system wide service parameter?

 

CallManager 4.2.3(SR2b) with 7961/7914s running SCCP41.8-2-2SR1S

 




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From CDAmore at jackhenry.com  Thu Aug 30 11:49:46 2007
From: CDAmore at jackhenry.com (Chris D'amore)
Date: Thu, 30 Aug 2007 10:49:46 -0500
Subject: [cisco-voip] 4.2 Shared line chime
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0069A9DEC@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0069A9DEC@USNAEXCH.na.didata.local>
Message-ID: <2B3745BC9D31514381A6FCC5E72D9373022B5D73@SMOMAIL.jhacorp.com>

Are all of these DN's in the same pickup group? This sounds like Call
Pickup notification. It can be disabled on the respective pickup group.
Not sure about Service Parameter.

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Thursday, August 30, 2007 10:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 4.2 Shared line chime

 

Randy Manager - Line1 - 4000 (shared)

 

Sharon Receptionist - Line 1 - 4001

Sharon Receptionist - Line14 - 4000 (shared)

 

Elizabeth Employee - Line1 - 6600

 

If Sharon uses 4001 to dial 6600 then Randy's 7961 main display (same
area as You have Voicemail) shows "Sharon > Elizabeth" and makes a
chime.  Any way to disabled this feature system wide service parameter?

 

CallManager 4.2.3(SR2b) with 7961/7914s running SCCP41.8-2-2SR1S

 

________________________________

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
strictly prohibited and may be unlawful. If you have received this
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From manoj.kalpage at gmail.com  Thu Aug 30 12:01:19 2007
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Fri, 31 Aug 2007 01:01:19 +0900
Subject: [cisco-voip] ISDN SIMULATOR ISDN BRI-U and BRI-S/T
In-Reply-To: <46D6E34B.6040100@arizona.edu>
References: <bbe09c1d0708300824o211cd9ddl880c4a4ee883a923@mail.gmail.com>
	<46D6E34B.6040100@arizona.edu>
Message-ID: <bbe09c1d0708300901w3e98fddck347c48feb4f001d1@mail.gmail.com>

Thanks Walt,

Does DSU convert BRI "U" to BRI "S/T"?

If I get BRI "U" simulator, do I need the "U" to "S/T" converter for
connect VIC2-2BRI-NT/TE card?

Best regards,
Manoj


On 8/31/07, Walt Moody <moody at arizona.edu> wrote:
> Howdy.
>
> The BRI "U" interface is the two-wire interface that would take service
> from a central office to a subscriber's premises.  It normally
> terminates on an NT-1.
>
> The BRI "S/T" interface is the four-wire interface from the NT-1 to the
> telephone equipment.
>
> -walt
>
> Manoj Kalpage wrote:
>
> > Hi All,
> > I am looking for ISDN simulator for CCIE voice lab. I found there are
> > two type of ISDN simulators at http://www.vconsole.com/ as bellow.
> >
> > ISDN BRI/ST
> > http://www.vconsole.com/2-Port-T1/E1/PRI-+-8-port-BRI-S/T-ISDN-Simulator-p-15.html
> >
> > ISDN BRI-U
> > http://www.vconsole.com/2-Port-T1/E1/PRI-+-8-port-BRI-U-ISDN-Simulator-p-14.html
> >
> > First thing, I cannot understand is what is the different between
> > BRI-U and BRI-S/T and next thing is which one is good for CCIE voice
> > LAB.
> >
> > Are these good or anyone know any other product which is better than this?
> >
> > Best regards,
> > PERE
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From Brian.VanBenschoten at inacom.com  Thu Aug 30 11:33:50 2007
From: Brian.VanBenschoten at inacom.com (Brian Van Benschoten)
Date: Thu, 30 Aug 2007 10:33:50 -0500
Subject: [cisco-voip] CM6 - super copy breaks source phone
Message-ID: <153D0FAADC2C90459D202D16D24A94AE0A109E28@exch1.inacom-msn.com>

Anyone run into this issue yet ?  I opened a TAC case # 60674145

 

 

CM 6.01a

System version: 6.0.1.2000-3

 

Using the new "super copy" feature to quickly create a phone. 

The feature creates a working "target" handset but breaks the source
handset used in the copy feature.

 

Calls to the extension of the source handset ring fast busy.  I added
the extension of the source phone to another device as a shared
extension and same result.  It appears to effect the directory number
not the device.

 

Usage of the super copy....

I find phone with extension 5555 in the device list, on the far right
select "super copy" supply a new MAC address. change the DN, and other
unique features and save it.

 

The new phone works, the original phone that was copied rings fast busy.

A reboot of CM seems to fix the source phone.

 

 

 

________________________________________

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 d4rkf1ber at gmail.com  Thu Aug 30 12:37:45 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Thu, 30 Aug 2007 11:37:45 -0500
Subject: [cisco-voip] DIDs / ELINs for use in CER ?
Message-ID: <a1b07ea60708300937h1832f541yaebd3cee5e7b62fd@mail.gmail.com>

I am getting a bit confused about something regarding the use of DIDs in CER.

The documents basically state that you need to get some DID's from the
service provider for each ERL you plan to define, these DID's will be
used as ELINs to both identify the location and for the PSAP to return
the call if need be.

So a 911 call is placed, and once things are setup CER will basically
do or have CM do a translation of the users extension to one of these
DIDs you specify so that the service provider  will know what PSAP
that call needs to get routed to and so the PSAP will have a number to
call back on and such.

All sounds fine and dandy I guess.

However, my experience (which isn't much) has led me to think that
usually when dealing with  DID ranges from a service provider that on
911 calls the service provider does not usually pass the DID of the
phone to the PSAP but rather the billing number which is usually the
first number of the range I thought?

This isn't a problem as far as passing the location, but it would be a
problem for a return call from the PSAP.  CER would be looking for a
call coming into one number which would be mapped to the phone which
called 911, however if the billing number was passed to the PSAP thats
the number that would be called back.

Hopefully I am just blanking on something here and this is not the case.

Can anyone enlighten me on what I am missing here?  Thanks!

From jason.aarons at us.didata.com  Thu Aug 30 12:42:49 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 30 Aug 2007 12:42:49 -0400
Subject: [cisco-voip] DIDs / ELINs for use in CER ?
In-Reply-To: <a1b07ea60708300937h1832f541yaebd3cee5e7b62fd@mail.gmail.com>
References: <a1b07ea60708300937h1832f541yaebd3cee5e7b62fd@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069A9F8B@USNAEXCH.na.didata.local>

You call the telco and ask them to stop sending the main billing number
for calls where you provide calling party number.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dark Fiber
Sent: Thursday, August 30, 2007 12:38 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] DIDs / ELINs for use in CER ?

I am getting a bit confused about something regarding the use of DIDs in
CER.

The documents basically state that you need to get some DID's from the
service provider for each ERL you plan to define, these DID's will be
used as ELINs to both identify the location and for the PSAP to return
the call if need be.

So a 911 call is placed, and once things are setup CER will basically
do or have CM do a translation of the users extension to one of these
DIDs you specify so that the service provider  will know what PSAP
that call needs to get routed to and so the PSAP will have a number to
call back on and such.

All sounds fine and dandy I guess.

However, my experience (which isn't much) has led me to think that
usually when dealing with  DID ranges from a service provider that on
911 calls the service provider does not usually pass the DID of the
phone to the PSAP but rather the billing number which is usually the
first number of the range I thought?

This isn't a problem as far as passing the location, but it would be a
problem for a return call from the PSAP.  CER would be looking for a
call coming into one number which would be mapped to the phone which
called 911, however if the billing number was passed to the PSAP thats
the number that would be called back.

Hopefully I am just blanking on something here and this is not the case.

Can anyone enlighten me on what I am missing here?  Thanks!
_______________________________________________
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 bob at smalltime.com  Thu Aug 30 13:17:04 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Thu, 30 Aug 2007 12:17:04 -0500
Subject: [cisco-voip] CCM log translator
Message-ID: <46D6FB90.3020305@smalltime.com>

I'm trying to investigate a call issue after-the-fact.  The users at the 
site reported that they were expecting a call to come in to reception, 
which they then said "Did a half-ring, then went to VM"

I've got CCM traces from that time period.  Is there a tool that will 
take 4 MB of text files and in plain-English translate what actually 
_happened_?  The log files are a mix of human readable text, hex codes, 
etc, etc.  Add in that there are other calls going on at other sites at 
the same time and it's practically impossible to follow the thread of 
what's going on.

From baryonyx5 at gmail.com  Thu Aug 30 13:42:23 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Thu, 30 Aug 2007 13:42:23 -0400
Subject: [cisco-voip] CCM log translator
In-Reply-To: <46D6FB90.3020305@smalltime.com>
References: <46D6FB90.3020305@smalltime.com>
Message-ID: <a749d0620708301042t1023725w455555283111e15a@mail.gmail.com>

There's a couple of tools you could try.

http://www.employees.org/~tiryaki/tc/

If you have access to CCO downloads you can look for VLT (voice log
translator).

On 8/30/07, Robert Kulagowski <bob at smalltime.com> wrote:
>
> I'm trying to investigate a call issue after-the-fact.  The users at the
> site reported that they were expecting a call to come in to reception,
> which they then said "Did a half-ring, then went to VM"
>
> I've got CCM traces from that time period.  Is there a tool that will
> take 4 MB of text files and in plain-English translate what actually
> _happened_?  The log files are a mix of human readable text, hex codes,
> etc, etc.  Add in that there are other calls going on at other sites at
> the same time and it's practically impossible to follow the thread of
> what's going on.
> _______________________________________________
> 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 tednugent69 at yahoo.com  Thu Aug 30 13:55:16 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 30 Aug 2007 10:55:16 -0700 (PDT)
Subject: [cisco-voip] CCM log translator
In-Reply-To: <46D6FB90.3020305@smalltime.com>
Message-ID: <466665.13865.qm@web36107.mail.mud.yahoo.com>

http://www.employees.org/~pgiralt/TranslatorX/


--- Robert Kulagowski <bob at smalltime.com> wrote:

> I'm trying to investigate a call issue
> after-the-fact.  The users at the 
> site reported that they were expecting a call to
> come in to reception, 
> which they then said "Did a half-ring, then went to
> VM"
> 
> I've got CCM traces from that time period.  Is there
> a tool that will 
> take 4 MB of text files and in plain-English
> translate what actually 
> _happened_?  The log files are a mix of human
> readable text, hex codes, 
> etc, etc.  Add in that there are other calls going
> on at other sites at 
> the same time and it's practically impossible to
> follow the thread of 
> what's going on.
> _______________________________________________
> 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 tednugent69 at yahoo.com  Thu Aug 30 13:10:26 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 30 Aug 2007 10:10:26 -0700 (PDT)
Subject: [cisco-voip] DIDs / ELINs for use in CER ?
In-Reply-To: <a1b07ea60708300937h1832f541yaebd3cee5e7b62fd@mail.gmail.com>
Message-ID: <472383.16574.qm@web36112.mail.mud.yahoo.com>


If the provider is sending the BTN to the PSAP they
need to fix that, that?s NOT how it?s supposed to
work. However you are right 90% of the installs that
I've have done have run into this problem but its
usually resolved with a call to the provider telling
them to honor the calling number sent to them.

And actually it would be a problem with passing
location info especially if you were working in a
multi-building or multisite environment because your
ERLs and ELINS are tied together in the PSAP ALI DB.
If the BTN is being sent as opposed to the actual ELIN
 then emergency personnel would always be sent to the
bill to address even if the ERL is in another building
2 blocks over on the second floor. So you absolutely
need to get that fixed for CER to work correctly.


--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> I am getting a bit confused about something
> regarding the use of DIDs in CER.
> 
> The documents basically state that you need to get
> some DID's from the
> service provider for each ERL you plan to define,
> these DID's will be
> used as ELINs to both identify the location and for
> the PSAP to return
> the call if need be.
> 
> So a 911 call is placed, and once things are setup
> CER will basically
> do or have CM do a translation of the users
> extension to one of these
> DIDs you specify so that the service provider  will
> know what PSAP
> that call needs to get routed to and so the PSAP
> will have a number to
> call back on and such.
> 
> All sounds fine and dandy I guess.
> 
> However, my experience (which isn't much) has led me
> to think that
> usually when dealing with  DID ranges from a service
> provider that on
> 911 calls the service provider does not usually pass
> the DID of the
> phone to the PSAP but rather the billing number
> which is usually the
> first number of the range I thought?
> 
> This isn't a problem as far as passing the location,
> but it would be a
> problem for a return call from the PSAP.  CER would
> be looking for a
> call coming into one number which would be mapped to
> the phone which
> called 911, however if the billing number was passed
> to the PSAP thats
> the number that would be called back.
> 
> Hopefully I am just blanking on something here and
> this is not the case.
> 
> Can anyone enlighten me on what I am missing here? 
> Thanks!
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________
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http://autos.yahoo.com/carfinder/

From wsisk at cisco.com  Thu Aug 30 14:29:13 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 14:29:13 -0400
Subject: [cisco-voip] CCM log translator
In-Reply-To: <46D6FB90.3020305@smalltime.com>
References: <46D6FB90.3020305@smalltime.com>
Message-ID: <46D70C79.3020703@cisco.com>

so long as it is a not a SIP phone you can:

1. find the tcp handle of the phone you expect the call
2. follow that TCP handle through the logs watching events on the phone, 
they are pretty intuitive.  The biggest thing you need is a decoder for 
the call states:
typedef enum {
    TsOffHook = 1,
    TsOnHook = 2,
    TsRingOut = 3,
    TsRingIn = 4,
    TsConnected = 5,
    TsBusy = 6,
    TsCongestion = 7,
    TsHold = 8,
    TsCallWaiting = 9,
    TsCallTransfer = 10,
    TsCallPark = 11,
    TsProceed = 12,
    TsCallRemoteMultiline = 13,
    TsInvalidNumber = 14
} StationDCallState;


When the specific call (identified by callerid in the callinfo message) 
stops ringing, look just above that to see why the call was rerouted.

/Wes

Robert Kulagowski wrote:
> I'm trying to investigate a call issue after-the-fact.  The users at the 
> site reported that they were expecting a call to come in to reception, 
> which they then said "Did a half-ring, then went to VM"
>
> I've got CCM traces from that time period.  Is there a tool that will 
> take 4 MB of text files and in plain-English translate what actually 
> _happened_?  The log files are a mix of human readable text, hex codes, 
> etc, etc.  Add in that there are other calls going on at other sites at 
> the same time and it's practically impossible to follow the thread of 
> what's going on.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From lelio at uoguelph.ca  Thu Aug 30 14:30:28 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 30 Aug 2007 14:30:28 -0400
Subject: [cisco-voip] Display Logging Parameter on 7911
References: <002a01c7e9ba$59934100$56196883@cfs.uoguelph.ca>
	<4835945F-3AAC-4462-A6B8-DB00DB5A6A0D@cisco.com>
Message-ID: <00aa01c7eb33$d714c2f0$56196883@cfs.uoguelph.ca>

So you think it's a bad thing having it enabled?

  ----- Original Message ----- 
  From: Wes Sisk 
  To: Lelio Fulgenzi 
  Cc: Cisco VoIPoE List 
  Sent: Wednesday, August 29, 2007 11:18 AM
  Subject: Re: [cisco-voip] Display Logging Parameter on 7911


  I believe this is used in conjunction with IP Phone Doctor.  Rather than collecting logs by console port to each phone (scalable?) this would have phone logs sent out in an L2 multicast to collect by a central server.


  You probably want it off.


  /wes


  On Aug 28, 2007, at 5:28 PM, Lelio Fulgenzi wrote:


  Can anyone help describe what this is and whether it is a security concern? The second sentence says it does not control the generation of logs, but the last sentence says you can use it to force logs. I wanted to turn it off for a bunch of phones but BAT conveniently does not have this setting in the list. Grrrr.


        Logging Display : This option selects what type of console logging is allowed. This option does not control the generation of logs - just whether the logs display. Disabled indicates that logging does not display to the console, nor to the connected downstream port. PC Controlled indicates that the workstation attached to the PC port will control whether logging is enabled. Enabled indicates that logs are always sent both to the console and to downstream port. Use Enabled to force logs on so they can be captured with a packet sniffer. 
        This is a required field.
        Default: PC Controlled.
       

  _______________________________________________
  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 Aug 30 14:34:07 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 14:34:07 -0400
Subject: [cisco-voip] Display Logging Parameter on 7911
In-Reply-To: <00aa01c7eb33$d714c2f0$56196883@cfs.uoguelph.ca>
References: <002a01c7e9ba$59934100$56196883@cfs.uoguelph.ca>
	<4835945F-3AAC-4462-A6B8-DB00DB5A6A0D@cisco.com>
	<00aa01c7eb33$d714c2f0$56196883@cfs.uoguelph.ca>
Message-ID: <46D70D9F.6080105@cisco.com>

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From wsisk at cisco.com  Thu Aug 30 14:35:13 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 14:35:13 -0400
Subject: [cisco-voip] ISDN SIMULATOR ISDN BRI-U and BRI-S/T
In-Reply-To: <bbe09c1d0708300824o211cd9ddl880c4a4ee883a923@mail.gmail.com>
References: <bbe09c1d0708300824o211cd9ddl880c4a4ee883a923@mail.gmail.com>
Message-ID: <46D70DE1.1070700@cisco.com>

dig carefully and long enough and you will find the pinouts for a BRI 
crossover cable for Cisco BRI cards and the hidden CLI commands to make 
IOS emulate a BRI switch.

/Wes

Manoj Kalpage wrote:
> Hi All,
> I am looking for ISDN simulator for CCIE voice lab. I found there are
> two type of ISDN simulators at http://www.vconsole.com/ as bellow.
>
> ISDN BRI/ST
> http://www.vconsole.com/2-Port-T1/E1/PRI-+-8-port-BRI-S/T-ISDN-Simulator-p-15.html
>
> ISDN BRI-U
> http://www.vconsole.com/2-Port-T1/E1/PRI-+-8-port-BRI-U-ISDN-Simulator-p-14.html
>
> First thing, I cannot understand is what is the different between
> BRI-U and BRI-S/T and next thing is which one is good for CCIE voice
> LAB.
>
> Are these good or anyone know any other product which is better than this?
>
> Best regards,
> PERE
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Thu Aug 30 14:37:40 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 14:37:40 -0400
Subject: [cisco-voip] CM6 - super copy breaks source phone
In-Reply-To: <153D0FAADC2C90459D202D16D24A94AE0A109E28@exch1.inacom-msn.com>
References: <153D0FAADC2C90459D202D16D24A94AE0A109E28@exch1.inacom-msn.com>
Message-ID: <46D70E74.8090309@cisco.com>

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From wsisk at cisco.com  Thu Aug 30 14:38:44 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 14:38:44 -0400
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
In-Reply-To: <54732.192.168.1.70.1188479408.webmail@192.168.1.70>
References: <54732.192.168.1.70.1188479408.webmail@192.168.1.70>
Message-ID: <46D70EB4.3000801@cisco.com>

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From svoll.voip at gmail.com  Thu Aug 30 14:42:13 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 30 Aug 2007 11:42:13 -0700
Subject: [cisco-voip] DIDs / ELINs for use in CER ?
In-Reply-To: <472383.16574.qm@web36112.mail.mud.yahoo.com>
References: <a1b07ea60708300937h1832f541yaebd3cee5e7b62fd@mail.gmail.com>
	<472383.16574.qm@web36112.mail.mud.yahoo.com>
Message-ID: <f84a38d30708301142j30f1733dof8b32ee56a6569e7@mail.gmail.com>

Good to know because I've dealt with the same issues.

I just thought I had to add the PS-ALI / PSAP contract to get it fixed.
Guess I'll work on that.

Thanks

Scott


On 8/30/07, Ted Nugent <tednugent69 at yahoo.com> wrote:
>
>
> If the provider is sending the BTN to the PSAP they
> need to fix that, that's NOT how it's supposed to
> work. However you are right 90% of the installs that
> I've have done have run into this problem but its
> usually resolved with a call to the provider telling
> them to honor the calling number sent to them.
>
> And actually it would be a problem with passing
> location info especially if you were working in a
> multi-building or multisite environment because your
> ERLs and ELINS are tied together in the PSAP ALI DB.
> If the BTN is being sent as opposed to the actual ELIN
> then emergency personnel would always be sent to the
> bill to address even if the ERL is in another building
> 2 blocks over on the second floor. So you absolutely
> need to get that fixed for CER to work correctly.
>
>
> --- Dark Fiber <d4rkf1ber at gmail.com> wrote:
>
> > I am getting a bit confused about something
> > regarding the use of DIDs in CER.
> >
> > The documents basically state that you need to get
> > some DID's from the
> > service provider for each ERL you plan to define,
> > these DID's will be
> > used as ELINs to both identify the location and for
> > the PSAP to return
> > the call if need be.
> >
> > So a 911 call is placed, and once things are setup
> > CER will basically
> > do or have CM do a translation of the users
> > extension to one of these
> > DIDs you specify so that the service provider  will
> > know what PSAP
> > that call needs to get routed to and so the PSAP
> > will have a number to
> > call back on and such.
> >
> > All sounds fine and dandy I guess.
> >
> > However, my experience (which isn't much) has led me
> > to think that
> > usually when dealing with  DID ranges from a service
> > provider that on
> > 911 calls the service provider does not usually pass
> > the DID of the
> > phone to the PSAP but rather the billing number
> > which is usually the
> > first number of the range I thought?
> >
> > This isn't a problem as far as passing the location,
> > but it would be a
> > problem for a return call from the PSAP.  CER would
> > be looking for a
> > call coming into one number which would be mapped to
> > the phone which
> > called 911, however if the billing number was passed
> > to the PSAP thats
> > the number that would be called back.
> >
> > Hopefully I am just blanking on something here and
> > this is not the case.
> >
> > Can anyone enlighten me on what I am missing here?
> > Thanks!
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
>
>
> ____________________________________________________________________________________
> Choose the right car based on your needs.  Check out Yahoo! Autos new Car
> Finder tool.
> http://autos.yahoo.com/carfinder/
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From paramds at gmail.com  Thu Aug 30 14:42:47 2007
From: paramds at gmail.com (param singh)
Date: Thu, 30 Aug 2007 13:42:47 -0500
Subject: [cisco-voip] Unity Connection Web Password
Message-ID: <867640cd0708301142p573ef4dal62f2c590448cd634@mail.gmail.com>

Hi All,

Has anybody unlocked Unity connection's Administration password. This is the
password that you use to login into the Unity Connection Administration Web
page.  I am trying to log into Unity connection administration page and  I
guess I typed wrong passwords two many times and  know it says " account
Locked" . Before this I never thought this account could be locked. I have
already rebooted the server  but no use. Any help will be appreciated.


Param
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From jgrace at adcapnet.com  Thu Aug 30 14:58:33 2007
From: jgrace at adcapnet.com (James Grace)
Date: Thu, 30 Aug 2007 14:58:33 -0400
Subject: [cisco-voip] Im getting a resource unavailable error code with SIP
	trunk
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C769@mail.adcapnetsvc.local>

?
Im getting a resource unavailable error code in my trace logs when i make inbound and outbound calls.  This is with a SIP trunk.  the trunk is comming direct to my callmanager.  What causes this error
 
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."

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From wsisk at cisco.com  Thu Aug 30 15:01:33 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 15:01:33 -0400
Subject: [cisco-voip] CM6 - super copy breaks source phone
In-Reply-To: <153D0FAADC2C90459D202D16D24A94AE0A109E28@exch1.inacom-msn.com>
References: <153D0FAADC2C90459D202D16D24A94AE0A109E28@exch1.inacom-msn.com>
Message-ID: <46D7140D.3090808@cisco.com>

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From wsisk at cisco.com  Thu Aug 30 15:04:23 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 15:04:23 -0400
Subject: [cisco-voip] Im getting a resource unavailable error code with
 SIP	trunk
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C769@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C769@mail.adcapnetsvc.local>
Message-ID: <46D714B7.70901@cisco.com>

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From lelio at uoguelph.ca  Thu Aug 30 15:08:20 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 30 Aug 2007 15:08:20 -0400
Subject: [cisco-voip] Im getting a resource unavailable error code with
	SIP	trunk
References: <ECEA9223F645834FB6A65B7CE377B9CD11C769@mail.adcapnetsvc.local>
	<46D714B7.70901@cisco.com>
Message-ID: <001d01c7eb39$2253d0d0$56196883@cfs.uoguelph.ca>

who said you can't get 5 nine's with Cisco... ;)

  ----- Original Message ----- 
  From: Wes Sisk 
  To: James Grace 
  Cc: cisco-voip at puck.nether.net 
  Sent: Thursday, August 30, 2007 3:04 PM
  Subject: Re: [cisco-voip] Im getting a resource unavailable error code with SIP trunk


  99.999% of time resource unavailable is a problem with media negotiation,  this comes in several forms:
  with SIP, both endpoints may not support rfc2833 which causes an MTP to be invoked, are MTP's avialable?
  if an MTP is required for any reason, then CM builin MTP only does g711.  If you need MTP for g.729 you must have hardware MTP


  James Grace wrote: 
    ? 
    Im getting a resource unavailable error code in my trace logs when i make inbound and outbound calls.  This is with a SIP trunk.  the trunk is comming direct to my callmanager.  What causes this error

    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 30 Aug 2007 14:57:33 -0400
  ----------------------------------------------------------------------------
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

------------------------------------------------------------------------------


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From craig at staffin.org  Thu Aug 30 15:13:29 2007
From: craig at staffin.org (Craig Staffin)
Date: Thu, 30 Aug 2007 14:13:29 -0500
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
In-Reply-To: <46D70EB4.3000801@cisco.com>
References: <54732.192.168.1.70.1188479408.webmail@192.168.1.70>
	<46D70EB4.3000801@cisco.com>
Message-ID: <9b76d88c0708301213p59111d89v123fcbc39484a19a@mail.gmail.com>

now the real question has anyone actually gotten the mobility features to
work correctly??

On 8/30/07, Wes Sisk <wsisk at cisco.com> wrote:
>
>  CM6, mobility manager is included natively.  /Wes
>
> Christopher M. Bomba wrote:
>
> I had to install Mobility Manager last year when it was still pretty new and when I called TAC the engineer asked me to send him the document because they sent me to a TAC engineer that hadn't even seen CMM yet.  After working with the guy for a full day they finally escalated to someone that knew what they were talking about and I got it working.  The documentation was Poo back then and haven't dealt with it since, but I guess people didn't complain enough to have them go back and rework the docs.  I was glad to get that behind me.
>
> Can't wait to see what we have to do to get that functionality working in CUCM 6.0.  It is built into the software now, no need for separate server.
>
> Chris
>
>
> Date: Wed, 29 Aug 2007 09:47:22 -0500
> From: Robert Kulagowski <bob at smalltime.com> <bob at smalltime.com>
> Subject: Re: [cisco-voip] New Winner-Worst Cisco Documentation
> 	Award!!!
> To: cisco-voip at puck.nether.net
> Message-ID: <46D586FA.8070604 at smalltime.com> <46D586FA.8070604 at smalltime.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Ted Nugent wrote:
>
>
>  Ahh yes however once you finally do track down the
> Unity Doc you need its typically pretty good.
>
> For instance they wouldn't call a DN (Distinguished
> Name) a Directory Number like they do in the CMM
> docs..
>
>  You are in a twisty maze of passageways, all alike...
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-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
>



-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003
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From d4rkf1ber at gmail.com  Thu Aug 30 15:33:22 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Thu, 30 Aug 2007 14:33:22 -0500
Subject: [cisco-voip] DIDs / ELINs for use in CER ?
In-Reply-To: <f84a38d30708301142j30f1733dof8b32ee56a6569e7@mail.gmail.com>
References: <a1b07ea60708300937h1832f541yaebd3cee5e7b62fd@mail.gmail.com>
	<472383.16574.qm@web36112.mail.mud.yahoo.com>
	<f84a38d30708301142j30f1733dof8b32ee56a6569e7@mail.gmail.com>
Message-ID: <a1b07ea60708301233q73c66ce0hd7d49115c8272083@mail.gmail.com>

Is it common practice for telco service providers to offer more
detailed location information then just the address if you request it?

I am running into a bunch of walls in getting this requested.

For instance, if I identify I have 10 locations I want to specify and
I want two DID's for each location then I would ask for 20 DID's, but
they just want to associated these 20 DID's to one location, and one
BTN and say they have no way to provide more detailed info for the
location so say like Floor 1 or Floor 2.

Not to mention the fact that some of these DID's are suppose to be for
different physical locations all together.  And then on top of that is
the issue I mentioned earlier where they only send the BTN and don't
honor our calling info.

Is there something specific you have to ask for?  Maybe its a
different service that I have to ask for then just asking for new
DID's on a PRI?

Anyway, some of you would mind sharing some info with me such as what
did you have to ask for, maybe the telco or service providers name and
what the service is called if it has a specific name, I would greatly
appreciate the info.

I didn't think this was going to be the hardest part of setting all
this up to be honest.



On 8/30/07, Scott Voll <svoll.voip at gmail.com> wrote:
> Good to know because I've dealt with the same issues.
>
> I just thought I had to add the PS-ALI / PSAP contract to get it fixed.
> Guess I'll work on that.
>
> Thanks
>
> Scott
>
>
> On 8/30/07, Ted Nugent <tednugent69 at yahoo.com> wrote:
> >
> > If the provider is sending the BTN to the PSAP they
> > need to fix that, that's NOT how it's supposed to
> > work. However you are right 90% of the installs that
> > I've have done have run into this problem but its
> > usually resolved with a call to the provider telling
> > them to honor the calling number sent to them.
> >
> > And actually it would be a problem with passing
> > location info especially if you were working in a
> > multi-building or multisite environment because your
> > ERLs and ELINS are tied together in the PSAP ALI DB.
> > If the BTN is being sent as opposed to the actual ELIN
> > then emergency personnel would always be sent to the
> > bill to address even if the ERL is in another building
> > 2 blocks over on the second floor. So you absolutely
> > need to get that fixed for CER to work correctly.
> >
> >
> > --- Dark Fiber <d4rkf1ber at gmail.com> wrote:
> >
> > > I am getting a bit confused about something
> > > regarding the use of DIDs in CER.
> > >
> > > The documents basically state that you need to get
> > > some DID's from the
> > > service provider for each ERL you plan to define,
> > > these DID's will be
> > > used as ELINs to both identify the location and for
> > > the PSAP to return
> > > the call if need be.
> > >
> > > So a 911 call is placed, and once things are setup
> > > CER will basically
> > > do or have CM do a translation of the users
> > > extension to one of these
> > > DIDs you specify so that the service provider  will
> > > know what PSAP
> > > that call needs to get routed to and so the PSAP
> > > will have a number to
> > > call back on and such.
> > >
> > > All sounds fine and dandy I guess.
> > >
> > > However, my experience (which isn't much) has led me
> > > to think that
> > > usually when dealing with  DID ranges from a service
> > > provider that on
> > > 911 calls the service provider does not usually pass
> > > the DID of the
> > > phone to the PSAP but rather the billing number
> > > which is usually the
> > > first number of the range I thought?
> > >
> > > This isn't a problem as far as passing the location,
> > > but it would be a
> > > problem for a return call from the PSAP.  CER would
> > > be looking for a
> > > call coming into one number which would be mapped to
> > > the phone which
> > > called 911, however if the billing number was passed
> > > to the PSAP thats
> > > the number that would be called back.
> > >
> > > Hopefully I am just blanking on something here and
> > > this is not the case.
> > >
> > > Can anyone enlighten me on what I am missing here?
> > > Thanks!
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> >
> >
> >
> ____________________________________________________________________________________
> > Choose the right car based on your needs.  Check out Yahoo! Autos new Car
> Finder tool.
> > http://autos.yahoo.com/carfinder/
> > _______________________________________________
> > 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 Aug 30 15:34:29 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 30 Aug 2007 15:34:29 -0400
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
In-Reply-To: <9b76d88c0708301213p59111d89v123fcbc39484a19a@mail.gmail.com>
References: <54732.192.168.1.70.1188479408.webmail@192.168.1.70><46D70EB4.3000801@cisco.com>
	<9b76d88c0708301213p59111d89v123fcbc39484a19a@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069E1887@USNAEXCH.na.didata.local>

Yes, works fine. It's more the 1 call in, 4 calls out that uses PRI
resources.  Need to plan Erlang type stats carefully..

 

It would be better if the CMMUser was more simple like Vonage simulcast
website, etc.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Craig Staffin
Sent: Thursday, August 30, 2007 3:13 PM
To: Wes Sisk
Cc: cbomba at s4nets.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!

 

now the real question has anyone actually gotten the mobility features
to work correctly??

On 8/30/07, Wes Sisk <wsisk at cisco.com > wrote:

CM6, mobility manager is included natively.  /Wes

Christopher M. Bomba wrote: 

I had to install Mobility Manager last year when it was still pretty new
and when I called TAC the engineer asked me to send him the document
because they sent me to a TAC engineer that hadn't even seen CMM yet.
After working with the guy for a full day they finally escalated to
someone that knew what they were talking about and I got it working.
The documentation was Poo back then and haven't dealt with it since, but
I guess people didn't complain enough to have them go back and rework
the docs.  I was glad to get that behind me.  




Can't wait to see what we have to do to get that functionality working
in CUCM 6.0.  It is built into the software now, no need for separate
server.



Chris





Date: Wed, 29 Aug 2007 09:47:22 -0500


From: Robert Kulagowski <bob at smalltime.com> <mailto:bob at smalltime.com> 
Subject: Re: [cisco-voip] New Winner-Worst Cisco Documentation
        Award!!!
To: cisco-voip at puck.nether.net
Message-ID: <46D586FA.8070604 at smalltime.com>
<mailto:46D586FA.8070604 at smalltime.com> 
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
 
Ted Nugent wrote:
  

	Ahh yes however once you finally do track down the
	
	Unity Doc you need its typically pretty good.
	
	
	
	For instance they wouldn't call a DN (Distinguished
	
	Name) a Directory Number like they do in the CMM
	
	
	docs..
	
	    

You are in a twisty maze of passageways, all alike...







_______________________________________________

cisco-voip mailing list

<mailto:cisco-voip at puck.nether.net> 
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




-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003 




-----------------------------------------
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 rschuknecht at gmx.de  Thu Aug 30 15:47:59 2007
From: rschuknecht at gmx.de (Robert Schuknecht)
Date: Thu, 30 Aug 2007 21:47:59 +0200
Subject: [cisco-voip] Antw: Re:  Display Logging Parameter on 7911
In-Reply-To: 2a95b484bff604154255d89b1e1d2824
References: 2a95b484bff604154255d89b1e1d2824
Message-ID: <46D73B0F.5E3C.0004.0@gmx.de>

Wes,

in a post of an other list, you mentioned that the IP Phone Doctor Application
should be available via the CCO Downloads. But i didn?t find it.

Regards

Robert

>>> Wes Sisk<wsisk at cisco.com> schrieb am Mittwoch, 29. August 2007 um 17:18 in
Nachricht 2a95b484bff604154255d89b1e1d2824:
> I believe this is used in conjunction with IP Phone Doctor.  Rather  
> than collecting logs by console port to each phone (scalable?) this  
> would have phone logs sent out in an L2 multicast to collect by a  
> central server.
> 
> You probably want it off.
> 
> /wes
> 
> On Aug 28, 2007, at 5:28 PM, Lelio Fulgenzi wrote:
> 
> Can anyone help describe what this is and whether it is a security  
> concern? The second sentence says it does not control the generation  
> of logs, but the last sentence says you can use it to force logs. I  
> wanted to turn it off for a bunch of phones but BAT conveniently does  
> not have this setting in the list. Grrrr.
> 
> 
> Logging Display :	This option selects what type of console logging is  
> allowed. This option does not control the generation of logs - just  
> whether the logs display. Disabled indicates that logging does not  
> display to the console, nor to the connected downstream port. PC  
> Controlled indicates that the       workstation attached to the PC  
> port will control whether logging is enabled. Enabled indicates that  
> logs are always sent both to the console and to downstream port. Use  
> Enabled to force logs on so they can be captured with a packet sniffer.
> This is a required field.
> Default: PC Controlled.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net 
> https://puck.nether.net/mailman/listinfo/cisco-voip

From wsisk at cisco.com  Thu Aug 30 16:11:42 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 16:11:42 -0400
Subject: [cisco-voip] Antw: Re:  Display Logging Parameter on 7911
In-Reply-To: <46D73B0F.5E3C.0004.0@gmx.de>
References: 2a95b484bff604154255d89b1e1d2824 <46D73B0F.5E3C.0004.0@gmx.de>
Message-ID: <46D7247E.6000602@cisco.com>

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From Brian.VanBenschoten at inacom.com  Thu Aug 30 16:21:21 2007
From: Brian.VanBenschoten at inacom.com (Brian Van Benschoten)
Date: Thu, 30 Aug 2007 15:21:21 -0500
Subject: [cisco-voip] CM6 - super copy breaks source phone
In-Reply-To: <46D7140D.3090808@cisco.com>
References: <153D0FAADC2C90459D202D16D24A94AE0A109E28@exch1.inacom-msn.com>
	<46D7140D.3090808@cisco.com>
Message-ID: <153D0FAADC2C90459D202D16D24A94AE0A10A1AC@exch1.inacom-msn.com>

SR: 606741451

SUMMARY: CM6 - Super copy feature breaks source phone

 

 

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Thursday, August 30, 2007 2:02 PM
To: Brian Van Benschoten
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM6 - super copy breaks source phone

 

Hi Brian,

I tried the steps you describe and did not observe the problem.

I suspect your problem may lie more in the DN modification than in use
of supercopy.

CSCsj32735 describes a very similar situation and is only fixed in
6.0(1)es3 i.e.: 006.000(001.1103.001)

6.0(1) was 6.0(0.9901.222) before the renumber so you can see this
defect is not fixed in 6.0(1) base as the integrated in field only
contains 006.000(000.9901.324)

Please let me know the complete TAC case number.

/Wes

Brian Van Benschoten wrote: 

Anyone run into this issue yet ?  I opened a TAC case # 60674145

 

 

CM 6.01a

System version: 6.0.1.2000-3

 

Using the new "super copy" feature to quickly create a phone. 

The feature creates a working "target" handset but breaks the source
handset used in the copy feature.

 

Calls to the extension of the source handset ring fast busy.  I added
the extension of the source phone to another device as a shared
extension and same result.  It appears to effect the directory number
not the device.

 

Usage of the super copy....

I find phone with extension 5555 in the device list, on the far right
select "super copy" supply a new MAC address. change the DN, and other
unique features and save it.

 

The new phone works, the original phone that was copied rings fast busy.

A reboot of CM seems to fix the source phone.

 

 

 

________________________________________

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/> 

 

 


________________________________



 
_______________________________________________
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  Thu Aug 30 15:43:11 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Thu, 30 Aug 2007 15:43:11 -0400 (EDT)
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
Message-ID: <40762.192.168.1.70.1188502991.webmail@192.168.1.70>

Yes, I had it working when I left, I could call the office phone, which in turn rang my cell phone.  I could answer my cell phone walk up to my desk, hang up with the caller and pick the call up on my office phone.  Works, but I just don't see having someone spend money for a MCS server and software for that functionality.  I am glad it is integrated with CUCM 6.0.

Chris


-----Original Message-----
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
Sent: Thu, August 30, 2007 3:34 pm
To: Craig Staffin <craig at staffin.org>, Wes Sisk <wsisk at cisco.com>
Cc: cbomba at s4nets.com, cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!

Yes, works fine. It's more the 1 call in, 4 calls out that uses PRI
resources.  Need to plan Erlang type stats carefully..

 

It would be better if the CMMUser was more simple like Vonage simulcast
website, etc.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Craig Staffin
Sent: Thursday, August 30, 2007 3:13 PM
To: Wes Sisk
Cc: cbomba at s4nets.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!

 

now the real question has anyone actually gotten the mobility features
to work correctly??

On 8/30/07, Wes Sisk <wsisk at cisco.com > wrote:

CM6, mobility manager is included natively.  /Wes

Christopher M. Bomba wrote: 

I had to install Mobility Manager last year when it was still pretty new
and when I called TAC the engineer asked me to send him the document
because they sent me to a TAC engineer that hadn't even seen CMM yet.
After working with the guy for a full day they finally escalated to
someone that knew what they were talking about and I got it working.
The documentation was Poo back then and haven't dealt with it since, but
I guess people didn't complain enough to have them go back and rework
the docs.  I was glad to get that behind me.  




Can't wait to see what we have to do to get that functionality working
in CUCM 6.0.  It is built into the software now, no need for separate
server.



Chris





Date: Wed, 29 Aug 2007 09:47:22 -0500


From: Robert Kulagowski <bob at smalltime.com> <mailto:bob at smalltime.com> 
Subject: Re: [cisco-voip] New Winner-Worst Cisco Documentation
        Award!!!
To: cisco-voip at puck.nether.net
Message-ID: <46D586FA.8070604 at smalltime.com>
<mailto:46D586FA.8070604 at smalltime.com> 
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
 
Ted Nugent wrote:
  

	Ahh yes however once you finally do track down the
	
	Unity Doc you need its typically pretty good.
	
	
	
	For instance they wouldn't call a DN (Distinguished
	
	Name) a Directory Number like they do in the CMM
	
	
	docs..
	
	    

You are in a twisty maze of passageways, all alike...







_______________________________________________

cisco-voip mailing list

<mailto:cisco-voip at puck.nether.net> 
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




-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003 




-----------------------------------------
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 ealeatherman at gmail.com  Thu Aug 30 16:47:06 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 30 Aug 2007 16:47:06 -0400
Subject: [cisco-voip] IPCC Express 5.0.1
Message-ID: <94a1afde0708301347l787a51e5sceca5134bb097969@mail.gmail.com>

Anyone running this or seen it install successfully? Thoughts?

Currently running callmanager 4.1 and IPCC Express 4.0. I'm kinda getting
painted into a corner as far as desired (read: demanded) features like BLF
etc from CCM 5. Need to upgrade sometime this calendar year from callmanager
4, but our IPCC express 4.0 with high availability is holding us back. IPCC
5 is the next version that supports both callmanager 5 and high
availability.. but i'm pretty nervous about migrating to a new version like
that so soon after it comes out. I've been burned before running initial
release software, don't relish the thought of doing it again but I may not
have a choice.

Interested in anyone's experience with it thus far.. wasn't getting a warm
and fuzzy feeling from comments I heard at networkers this year about the
initial release.

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From svoll.voip at gmail.com  Thu Aug 30 17:42:58 2007
From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 30 Aug 2007 14:42:58 -0700
Subject: [cisco-voip] auto dial
Message-ID: <f84a38d30708301442q396a41eeyb675e490c82c281@mail.gmail.com>

How do I set an ATA port (elevator phone) to dial a number on pick up?

Sorry for the simple question. I just don't have time to research.

TIA

Scott
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From Nesen.Kisten at za.didata.com  Thu Aug 30 17:45:10 2007
From: Nesen.Kisten at za.didata.com (Nesen Kisten (ZA))
Date: Thu, 30 Aug 2007 23:45:10 +0200
Subject: [cisco-voip] auto dial
References: <f84a38d30708301442q396a41eeyb675e490c82c281@mail.gmail.com>
Message-ID: <DA507CEAA8B853469A9A74AD432B58F41A553F@ZABRYSVCL17A.af.didata.local>

Hi,
 
Create a blank translation pattern and set your number to dial within the translation patttern.
 
Regards

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Scott Voll
Sent: Thu 2007/08/30 11:42 PM
To: cisco voip list
Subject: [cisco-voip] auto dial


How do I set an ATA port (elevator phone) to dial a number on pick up?
 
Sorry for the simple question. I just don't have time to research.
 
TIA
 
Scott
 

This email and all contents are subject to the following disclaimer:

"http://www.dimensiondata.com/emaildisclaimer.htm"
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From pat-cv at wcyv.com  Thu Aug 30 17:49:45 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Thu, 30 Aug 2007 17:49:45 -0400
Subject: [cisco-voip] Unity Connection Web Password
In-Reply-To: <867640cd0708301142p573ef4dal62f2c590448cd634@mail.gmail.com>
References: <867640cd0708301142p573ef4dal62f2c590448cd634@mail.gmail.com>
Message-ID: <46D73B79.4020100@wcyv.com>

The account should unlock after the interval defined in the 
authentication rule associated with your administrator account (I think 
the default is 30 minutes). What version is this? Assuming 1.2, you can 
force it to reset by resetting the password from the command prompt:

G:\Cisco Unity Connection\TechTools\pwreset.bat Administrator <new password>

-------- Original Message  --------
From: "param singh" <paramds at gmail.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Connection Web Password
Date: 8/30/2007 2:42 PM

> Hi All,
> 
> Has anybody unlocked Unity connection's Administration password. This is 
> the password that you use to login into the Unity Connection 
> Administration Web page.  I am trying to log into Unity connection 
> administration page and  I guess I typed wrong passwords two many times 
> and  know it says " account Locked" . Before this I never thought this 
> account could be locked. I have already rebooted the server  but no use. 
> Any help will be appreciated.
> 
> 
> Param
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From rsingleton at morsco.com  Thu Aug 30 17:51:27 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Thu, 30 Aug 2007 16:51:27 -0500
Subject: [cisco-voip] auto dial
In-Reply-To: <DA507CEAA8B853469A9A74AD432B58F41A553F@ZABRYSVCL17A.af.didata.local>
References: <f84a38d30708301442q396a41eeyb675e490c82c281@mail.gmail.com>
	<DA507CEAA8B853469A9A74AD432B58F41A553F@ZABRYSVCL17A.af.didata.local>
Message-ID: <1188510687.4869.21.camel@robert.morsco.com>

On Thu, 2007-08-30 at 23:45 +0200, Nesen Kisten (ZA) wrote:
> Hi,
>  
> Create a blank translation pattern and set your number to dial within
> the translation patttern.

I would expand that to say create a separate calling search space for
the phone, a separate partition for the translation pattern and make
sure that no other calling search space includes the partition used for
the translation pattern, or you'll suddenly have all phones dialing that
number upon picking up and nobody will be able to call you about
it... :)

I'm not saying that's exactly what happened to me or anything...

Robert



From rsingleton at morsco.com  Thu Aug 30 18:08:47 2007
From: rsingleton at morsco.com (Robert Singleton)
Date: Thu, 30 Aug 2007 17:08:47 -0500
Subject: [cisco-voip] Unity Connection Web Password
In-Reply-To: <867640cd0708301142p573ef4dal62f2c590448cd634@mail.gmail.com>
References: <867640cd0708301142p573ef4dal62f2c590448cd634@mail.gmail.com>
Message-ID: <1188511727.4869.38.camel@robert.morsco.com>

> I have already rebooted the server but no use.

Whenever I hear the story about someone who endlessly tolerates a
malfunction without really fixing it, whether it's a PC or a car or a
blender, I refer to it as "the standard Microsoft fix", essentially:
'ignore it until it won't go any more then reboot', followed closely by
'reboot till it works'.

I don't mean to pick on anyone, but it's incredibly frustrating that in
this culture, it is not only tolerated but even expected to periodically
take a mallet to the brains of our critical systems in the hope that it
will be better when they regain consciousness. You would yell at your
LEC if they pulled something like that....  oops...

Robert


From asobihoudai at yahoo.com  Thu Aug 30 17:09:11 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 30 Aug 2007 14:09:11 -0700 (PDT)
Subject: [cisco-voip] CCME GUI Error Solution "Configuration change: Please
	report the following error: No Matched Command."
Message-ID: <553965.37766.qm@web57201.mail.re3.yahoo.com>

This is a TACACS problem.

If you have this line in your CCME box

aaa authorization config-commands

You will encounter this command. This will also
prevent any non-TACACS authenticated login from making
any changes to the router meaning no end user speed
dials, no customer web admin, nothing.

Remove this command and you should be okay...waiting
on response from TAC for a workaround.

Cheers,
Paul


       
____________________________________________________________________________________
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 tednugent69 at yahoo.com  Thu Aug 30 19:56:36 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 30 Aug 2007 16:56:36 -0700 (PDT)
Subject: [cisco-voip] New Winner-Worst Cisco Documentation Award!!!
In-Reply-To: <40762.192.168.1.70.1188502991.webmail@192.168.1.70>
Message-ID: <912658.85741.qm@web36103.mail.mud.yahoo.com>

Does anyone have CMM working on CM5.1 with the CM LDAP
pointing to AD? I've got to set this up in a couple
weeks and I'm having a hard time finding anything
about where to point CMM. I can only guess with the
current docs that the directory config is ignored in
CM5.x and I don't really need to worry about it
providing AXL is working between CMM and CCM? Sound
about right?



--- "Christopher M. Bomba" <cbomba at s4nets.com> wrote:

> Yes, I had it working when I left, I could call the
> office phone, which in turn rang my cell phone.  I
> could answer my cell phone walk up to my desk, hang
> up with the caller and pick the call up on my office
> phone.  Works, but I just don't see having someone
> spend money for a MCS server and software for that
> functionality.  I am glad it is integrated with CUCM
> 6.0.
> 
> Chris
> 
> 
> -----Original Message-----
> From: "Jason Aarons \(US\)"
> <jason.aarons at us.didata.com>
> Sent: Thu, August 30, 2007 3:34 pm
> To: Craig Staffin <craig at staffin.org>, Wes Sisk
> <wsisk at cisco.com>
> Cc: cbomba at s4nets.com, cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] New Winner-Worst Cisco
> Documentation Award!!!
> 
> Yes, works fine. It's more the 1 call in, 4 calls
> out that uses PRI
> resources.  Need to plan Erlang type stats
> carefully..
> 
>  
> 
> It would be better if the CMMUser was more simple
> like Vonage simulcast
> website, etc.
> 
>  
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Craig Staffin
> Sent: Thursday, August 30, 2007 3:13 PM
> To: Wes Sisk
> Cc: cbomba at s4nets.com; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] New Winner-Worst Cisco
> Documentation Award!!!
> 
>  
> 
> now the real question has anyone actually gotten the
> mobility features
> to work correctly??
> 
> On 8/30/07, Wes Sisk <wsisk at cisco.com > wrote:
> 
> CM6, mobility manager is included natively.  /Wes
> 
> Christopher M. Bomba wrote: 
> 
> I had to install Mobility Manager last year when it
> was still pretty new
> and when I called TAC the engineer asked me to send
> him the document
> because they sent me to a TAC engineer that hadn't
> even seen CMM yet.
> After working with the guy for a full day they
> finally escalated to
> someone that knew what they were talking about and I
> got it working.
> The documentation was Poo back then and haven't
> dealt with it since, but
> I guess people didn't complain enough to have them
> go back and rework
> the docs.  I was glad to get that behind me.  
> 
> 
> 
> 
> Can't wait to see what we have to do to get that
> functionality working
> in CUCM 6.0.  It is built into the software now, no
> need for separate
> server.
> 
> 
> 
> Chris
> 
> 
> 
> 
> 
> Date: Wed, 29 Aug 2007 09:47:22 -0500
> 
> 
> From: Robert Kulagowski <bob at smalltime.com>
> <mailto:bob at smalltime.com> 
> Subject: Re: [cisco-voip] New Winner-Worst Cisco
> Documentation
>         Award!!!
> To: cisco-voip at puck.nether.net
> Message-ID: <46D586FA.8070604 at smalltime.com>
> <mailto:46D586FA.8070604 at smalltime.com> 
> Content-Type: text/plain; charset=ISO-8859-1;
> format=flowed
>  
> Ted Nugent wrote:
>   
> 
> 	Ahh yes however once you finally do track down the
> 	
> 	Unity Doc you need its typically pretty good.
> 	
> 	
> 	
> 	For instance they wouldn't call a DN (Distinguished
> 	
> 	Name) a Directory Number like they do in the CMM
> 	
> 	
> 	docs..
> 	
> 	    
> 
> You are in a twisty maze of passageways, all
> alike...
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> 
> cisco-voip mailing list
> 
> <mailto:cisco-voip at puck.nether.net> 
> 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
> 
> 
> 
> 
> -- 
> Craig Staffin
> Craig at staffin.org
> (H) 262-437-7313
> (C) 262-613-6003 
> 
> 
> 
> 
> -----------------------------------------
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> 
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> contain
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> use by the
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> are not the
> intended addressee, you are hereby notified that you
> have received
> this communication in error and that any use or
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> and may be
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> in error, please
> notify us immediately by replying to this message
> and deleting it
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> 
> _______________________________________________
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> 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 tednugent69 at yahoo.com  Thu Aug 30 20:31:45 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 30 Aug 2007 17:31:45 -0700 (PDT)
Subject: [cisco-voip] DIDs / ELINs for use in CER ?
In-Reply-To: <a1b07ea60708301233q73c66ce0hd7d49115c8272083@mail.gmail.com>
Message-ID: <672711.87313.qm@web36104.mail.mud.yahoo.com>

Yes there is, it?s a special E911 service you'll more
then likely need to subscribe to, for example Pinpoint
service is what BellSouth offers. You need to work
with your provider to update thier PS-ALI (e911)
database with you're ERL/ELIN Mappings which in turn
is popped to the 911 operator when the call is sent to
the PSAP. If you're not talking to this group you?re
wasting your time because the entire idea is foreign
to them to anyone but them. Not too long ago you could
actually export your CER DB into a NENA compliant ALI
format and just send it off to your provider however
it appears that more recently things have become more
automated (at least for them) and you're required to
manually entire your information. At any rate the key
here is the "E" in E911, this is what allows you to
get granular within you organization, Address,
building, floor, section etc.

HTH
Ted








--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Is it common practice for telco service providers to
> offer more
> detailed location information then just the address
> if you request it?
> 
> I am running into a bunch of walls in getting this
> requested.
> 
> For instance, if I identify I have 10 locations I
> want to specify and
> I want two DID's for each location then I would ask
> for 20 DID's, but
> they just want to associated these 20 DID's to one
> location, and one
> BTN and say they have no way to provide more
> detailed info for the
> location so say like Floor 1 or Floor 2.
> 
> Not to mention the fact that some of these DID's are
> suppose to be for
> different physical locations all together.  And then
> on top of that is
> the issue I mentioned earlier where they only send
> the BTN and don't
> honor our calling info.
> 
> Is there something specific you have to ask for? 
> Maybe its a
> different service that I have to ask for then just
> asking for new
> DID's on a PRI?
> 
> Anyway, some of you would mind sharing some info
> with me such as what
> did you have to ask for, maybe the telco or service
> providers name and
> what the service is called if it has a specific
> name, I would greatly
> appreciate the info.
> 
> I didn't think this was going to be the hardest part
> of setting all
> this up to be honest.
> 
> 
> 
> On 8/30/07, Scott Voll <svoll.voip at gmail.com> wrote:
> > Good to know because I've dealt with the same
> issues.
> >
> > I just thought I had to add the PS-ALI / PSAP
> contract to get it fixed.
> > Guess I'll work on that.
> >
> > Thanks
> >
> > Scott
> >
> >
> > On 8/30/07, Ted Nugent <tednugent69 at yahoo.com>
> wrote:
> > >
> > > If the provider is sending the BTN to the PSAP
> they
> > > need to fix that, that's NOT how it's supposed
> to
> > > work. However you are right 90% of the installs
> that
> > > I've have done have run into this problem but
> its
> > > usually resolved with a call to the provider
> telling
> > > them to honor the calling number sent to them.
> > >
> > > And actually it would be a problem with passing
> > > location info especially if you were working in
> a
> > > multi-building or multisite environment because
> your
> > > ERLs and ELINS are tied together in the PSAP ALI
> DB.
> > > If the BTN is being sent as opposed to the
> actual ELIN
> > > then emergency personnel would always be sent to
> the
> > > bill to address even if the ERL is in another
> building
> > > 2 blocks over on the second floor. So you
> absolutely
> > > need to get that fixed for CER to work
> correctly.
> > >
> > >
> > > --- Dark Fiber <d4rkf1ber at gmail.com> wrote:
> > >
> > > > I am getting a bit confused about something
> > > > regarding the use of DIDs in CER.
> > > >
> > > > The documents basically state that you need to
> get
> > > > some DID's from the
> > > > service provider for each ERL you plan to
> define,
> > > > these DID's will be
> > > > used as ELINs to both identify the location
> and for
> > > > the PSAP to return
> > > > the call if need be.
> > > >
> > > > So a 911 call is placed, and once things are
> setup
> > > > CER will basically
> > > > do or have CM do a translation of the users
> > > > extension to one of these
> > > > DIDs you specify so that the service provider 
> will
> > > > know what PSAP
> > > > that call needs to get routed to and so the
> PSAP
> > > > will have a number to
> > > > call back on and such.
> > > >
> > > > All sounds fine and dandy I guess.
> > > >
> > > > However, my experience (which isn't much) has
> led me
> > > > to think that
> > > > usually when dealing with  DID ranges from a
> service
> > > > provider that on
> > > > 911 calls the service provider does not
> usually pass
> > > > the DID of the
> > > > phone to the PSAP but rather the billing
> number
> > > > which is usually the
> > > > first number of the range I thought?
> > > >
> > > > This isn't a problem as far as passing the
> location,
> > > > but it would be a
> > > > problem for a return call from the PSAP.  CER
> would
> > > > be looking for a
> > > > call coming into one number which would be
> mapped to
> > > > the phone which
> > > > called 911, however if the billing number was
> passed
> > > > to the PSAP thats
> > > > the number that would be called back.
> > > >
> > > > Hopefully I am just blanking on something here
> and
> > > > this is not the case.
> > > >
> > > > Can anyone enlighten me on what I am missing
> here?
> > > > Thanks!
> > > >
> _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > >
> > >
> > >
> > >
> >
>
____________________________________________________________________________________
> > > Choose the right car based on your needs.  Check
> out Yahoo! Autos new Car
> > Finder tool.
> > > http://autos.yahoo.com/carfinder/
> > > _______________________________________________
> > > 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 tednugent69 at yahoo.com  Thu Aug 30 21:05:33 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 30 Aug 2007 18:05:33 -0700 (PDT)
Subject: [cisco-voip] Unity Connection Web Password
In-Reply-To: <1188511727.4869.38.camel@robert.morsco.com>
Message-ID: <932193.64061.qm@web36106.mail.mud.yahoo.com>

BAHAAA! Good stuff... sad thing is that most become
numb to it...


--- Robert Singleton <rsingleton at morsco.com> wrote:

> > I have already rebooted the server but no use.
> 
> Whenever I hear the story about someone who
> endlessly tolerates a
> malfunction without really fixing it, whether it's a
> PC or a car or a
> blender, I refer to it as "the standard Microsoft
> fix", essentially:
> 'ignore it until it won't go any more then reboot',
> followed closely by
> 'reboot till it works'.
> 
> I don't mean to pick on anyone, but it's incredibly
> frustrating that in
> this culture, it is not only tolerated but even
> expected to periodically
> take a mallet to the brains of our critical systems
> in the hope that it
> will be better when they regain consciousness. You
> would yell at your
> LEC if they pulled something like that....  oops...
> 
> Robert
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



      ____________________________________________________________________________________
Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/ 

From wsisk at cisco.com  Thu Aug 30 22:15:40 2007
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 30 Aug 2007 22:15:40 -0400
Subject: [cisco-voip] auto dial
In-Reply-To: <f84a38d30708301442q396a41eeyb675e490c82c281@mail.gmail.com>
References: <f84a38d30708301442q396a41eeyb675e490c82c281@mail.gmail.com>
Message-ID: <46D779CC.8040109@cisco.com>

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From MillerS at DicksteinShapiro.COM  Thu Aug 30 22:52:40 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Thu, 30 Aug 2007 22:52:40 -0400
Subject: [cisco-voip] De-Frag on Call Managers and Unity?
Message-ID: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM>

Does anyone have any words of wisdom about whether/how defrags can be
run on these servers?
 

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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From bcarter at sentinel.com  Thu Aug 30 23:08:29 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 30 Aug 2007 22:08:29 -0500
Subject: [cisco-voip] De-Frag on Call Managers and Unity?
In-Reply-To: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM>
References: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM>
Message-ID: <6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com>

If you are onsite and have time....I sometimes reboot into Windows Safe
mode and run defrag.  It can take 30 minutes.  Can't tell if it makes a
difference.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Thursday, August 30, 2007 9:53 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] De-Frag on Call Managers and Unity?


Does anyone have any words of wisdom about whether/how defrags can be
run on these servers?
 

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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responsible for delivering this confidential communication 
to the intended recipient, you have received this 
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From bcarter at sentinel.com  Thu Aug 30 23:11:41 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 30 Aug 2007 22:11:41 -0500
Subject: [cisco-voip] CM Caller Name PRI support
Message-ID: <6A0744639B86DD459433108DBF101D26018BDF0E@MAIL01.sentinel.com>

I have tried talking to a couple of Telco's regarding Caller Name
support on PRI trunk.  Both vendors have been confused on the terms
"Display ie" and "Facility ie".  Any suggestions on this for talking to
Verizon or SBC/ATT ?
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From erickbee at gmail.com  Fri Aug 31 01:59:02 2007
From: erickbee at gmail.com (Erick Bergquist)
Date: Fri, 31 Aug 2007 00:59:02 -0500
Subject: [cisco-voip] De-Frag on Call Managers and Unity?
In-Reply-To: <6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com>
References: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM>
	<6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com>
Message-ID: <f4445faf0708302259n791d75dl5509a08313ebd28c@mail.gmail.com>

I have seen a few places where the HD was highly fragmented it caused
problems, and defragging fixed things up.

I just ran into one 2-3 weeks ago where Unity would not start up on a
reboot. The drive was 50% fragmented. Had to run defrag a few times
(took a whole day almost) and since then when they reboot they have
not had problems.

On 8/30/07, Carter, Bill <bcarter at sentinel.com> wrote:
>
>
> If you are onsite and have time....I sometimes reboot into Windows Safe mode
> and run defrag.  It can take 30 minutes.  Can't tell if it makes a
> difference.
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Miller, Steve
> Sent: Thursday, August 30, 2007 9:53 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] De-Frag on Call Managers and Unity?
>
>
>
> Does anyone have any words of wisdom about whether/how defrags can be run on
> these servers?
>
>
> 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
>

From Mudassar.khalid at waridtel.com  Fri Aug 31 02:19:10 2007
From: Mudassar.khalid at waridtel.com (Mudassar.Khalid/NE/Lahore)
Date: Fri, 31 Aug 2007 11:19:10 +0500
Subject: [cisco-voip] CRS 5.0 and Call manager 4.1(3) integration
Message-ID: <6D5F57C11421A54DBC708162DF74F2F867CFA2@LHR-MAIL.WT.WI.Pri>

Dear all,
       I am using Call Manager 4.1(3) Sr2 installed in my cluster. With IPCC express 3.5(2) , I test my scripts successfully run in lab with call manager 4.1(3). 
 Now i want to deploy a new MCS 7835 H2 server with latest version of CRS 5.0 installed into it and integreate with my existing call manager cluster.
Does any body known to this situation or anyone can guide me about the problems facing while integreating CRS 5.0 with Call Manager 4.1(3) Sr2?
Please tell me if there would be no problem to integerate CRS version 5.0 with my existing call manager cluster.
 
Kind Regards,
Mudassar Khalid
Network support Engineer
O&M Lahore NEPL
Cell : (0092) 0322-4481004
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From rschuknecht at gmx.de  Fri Aug 31 05:19:57 2007
From: rschuknecht at gmx.de (Robert Schuknecht)
Date: Fri, 31 Aug 2007 11:19:57 +0200
Subject: [cisco-voip] CUPS 6.0 and IP Phone Messenger
Message-ID: <46D7F95D.5E3C.0004.0@gmx.de>

Hi List,

i have an CUCM 6.0 and an CUPS 6.0, with the IP Phone Messnger Service, installed in my LAB. When i send an IM Message, with IPPM, from one Phone to another, nothing happens. (The Message will not Pop-Up automatically) When i want to see the message i have to go to "Messages" within the IPPM.

In my old LAB, with an CCM5.1 and CUPS1.03, it worked.

Does anybody know how to troubleshoot such an issue, or what the error could be?

Any ideas or hints are welcome.

Regards

Robert

From Nick.Kassel at Charles-Stanley.co.uk  Fri Aug 31 05:41:24 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri, 31 Aug 2007 10:41:24 +0100
Subject: [cisco-voip] CM 6.0 Upgrade to 2000.4
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC03B4825E@CSEXCH01.charles-stanley.co.uk>

My 6.0 cluster is currently on 6.0.1.1000.37, I have downloaded the 2
split files for 6.0.1.2000.4 and combined them into one ISO file,
however when I try and upgrade, it fails at the point after it has
downloaded the file. 

 

Has anyone come across this?

 

 

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From jason.aarons at us.didata.com  Fri Aug 31 07:16:12 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 31 Aug 2007 07:16:12 -0400
Subject: [cisco-voip] CM Caller Name PRI support
In-Reply-To: <6A0744639B86DD459433108DBF101D26018BDF0E@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26018BDF0E@MAIL01.sentinel.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069E1CB2@USNAEXCH.na.didata.local>

My understanding is it's a switch change, versus a per port.

 

I haven't found a telco that would change to accommodate Cisco, in 2001
I had my Cisco AM submit a PERS requesting callmanager 7900 phones
receive calling party name under h323 in any method delivered by telco
to gateway, check with your Cisco AM/SE on having them escalate the
resolution.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Thursday, August 30, 2007 11:12 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CM Caller Name PRI support

 

I have tried talking to a couple of Telco's regarding Caller Name
support on PRI trunk.  Both vendors have been confused on the terms
"Display ie" and "Facility ie".  Any suggestions on this for talking to
Verizon or SBC/ATT ?




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

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From MillerS at DicksteinShapiro.COM  Fri Aug 31 07:17:16 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Fri, 31 Aug 2007 07:17:16 -0400
Subject: [cisco-voip] De-Frag on Call Managers and Unity?
References: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM>
	<6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com>
	<f4445faf0708302259n791d75dl5509a08313ebd28c@mail.gmail.com>
Message-ID: <421CAA1A5300B84A87D200D709985782041F4D@DCEX2.DSMO.COM>

Excellent info.  Thanks guys! 


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: Erick Bergquist [mailto:erickbee at gmail.com] 
Sent: Friday, August 31, 2007 1:59 AM
To: Carter, Bill
Cc: Miller, Steve; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] De-Frag on Call Managers and Unity?

I have seen a few places where the HD was highly fragmented it caused
problems, and defragging fixed things up.

I just ran into one 2-3 weeks ago where Unity would not start up on a
reboot. The drive was 50% fragmented. Had to run defrag a few times
(took a whole day almost) and since then when they reboot they have
not had problems.

On 8/30/07, Carter, Bill <bcarter at sentinel.com> wrote:
>
>
> If you are onsite and have time....I sometimes reboot into Windows
Safe mode
> and run defrag.  It can take 30 minutes.  Can't tell if it makes a
> difference.
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Miller, Steve
> Sent: Thursday, August 30, 2007 9:53 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] De-Frag on Call Managers and Unity?
>
>
>
> Does anyone have any words of wisdom about whether/how defrags can be
run on
> these servers?
>
>
> 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
>


--------------------------------------------------------
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

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


From MillerS at DicksteinShapiro.COM  Fri Aug 31 07:30:36 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Fri, 31 Aug 2007 07:30:36 -0400
Subject: [cisco-voip] De-Frag on Call Managers and Unity?
References: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM>
	<6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com>
	<f4445faf0708302259n791d75dl5509a08313ebd28c@mail.gmail.com>
Message-ID: <421CAA1A5300B84A87D200D709985782041F4E@DCEX2.DSMO.COM>

I have a primary and a failover in one location.  Should I assume that
it's OK to run one server while I defrag the other or should they both
be shut down? 


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: Erick Bergquist [mailto:erickbee at gmail.com] 
Sent: Friday, August 31, 2007 1:59 AM
To: Carter, Bill
Cc: Miller, Steve; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] De-Frag on Call Managers and Unity?

I have seen a few places where the HD was highly fragmented it caused
problems, and defragging fixed things up.

I just ran into one 2-3 weeks ago where Unity would not start up on a
reboot. The drive was 50% fragmented. Had to run defrag a few times
(took a whole day almost) and since then when they reboot they have
not had problems.

On 8/30/07, Carter, Bill <bcarter at sentinel.com> wrote:
>
>
> If you are onsite and have time....I sometimes reboot into Windows
Safe mode
> and run defrag.  It can take 30 minutes.  Can't tell if it makes a
> difference.
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Miller, Steve
> Sent: Thursday, August 30, 2007 9:53 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] De-Frag on Call Managers and Unity?
>
>
>
> Does anyone have any words of wisdom about whether/how defrags can be
run on
> these servers?
>
>
> 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
>


--------------------------------------------------------
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 ealeatherman at gmail.com  Fri Aug 31 08:00:02 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 31 Aug 2007 08:00:02 -0400
Subject: [cisco-voip] CRS 5.0 and Call manager 4.1(3) integration
In-Reply-To: <6D5F57C11421A54DBC708162DF74F2F867CFA2@LHR-MAIL.WT.WI.Pri>
References: <6D5F57C11421A54DBC708162DF74F2F867CFA2@LHR-MAIL.WT.WI.Pri>
Message-ID: <94a1afde0708310500s26475d94g77a428bbb289a1ef@mail.gmail.com>

Hello Mudassar,

CRS 5.0 is not compatible with callmanager version 4, take a look at the
compatibility guide at this link for details:
http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1683/ccmigration_09186a008086dfe9.pdf

You would need to upgrade to callmanager version 5 or 6.

Hope that helps,

Ed

On 8/31/07, Mudassar.Khalid/NE/Lahore <Mudassar.khalid at waridtel.com> wrote:
>
>  Dear all,
>        I am using Call Manager 4.1(3) Sr2 installed in my cluster. With
> IPCC express 3.5(2) , I test my scripts successfully run in lab with call
> manager 4.1(3).
>  Now i want to deploy a new MCS 7835 H2 server with latest version of CRS
> 5.0 installed into it and integreate with my existing call manager
> cluster.
> Does any body known to this situation or anyone can guide me about the
> problems facing while integreating CRS 5.0 with Call Manager 4.1(3) Sr2?
> Please tell me if there would be no problem to integerate CRS version 5.0with my existing call manager cluster.
>
>  Kind Regards,
> Mudassar Khalid
> Network support Engineer
> O&M Lahore NEPL
> Cell : (0092) 0322-4481004
>
> _______________________________________________
> 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 CarlosOrtiz at bayviewfinancial.com  Fri Aug 31 08:59:50 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 31 Aug 2007 08:59:50 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
Message-ID: <OFFE84AEAC.023B852F-ON85257348.004404D2-85257348.00476586@bftg.com>

I am not able to send International calls through our PRI's at a location 
outside of Philly.  Local and LD work fine.  debugs at the router show the 
number being sent to the telco are fine. 

Debug ISDN q931

Calling Party Number i = 0x2181, '215664XXXX' 
                Plan:ISDN, Type:National 
        Called Party Number i = 0x91, '011448458787000'

I found a CCO TAC case where this exact problem is described and addressed 
by changing the calling plan type from national to unknown using the 
following command under the D channel interface.

isdn map address .* plan isdn type unknown

This did in fact change the plan type but I was unable to complete local, 
LD and Intl after adding this.

Any suggestions?

Carlos







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From jsteinberg at gmail.com  Fri Aug 31 09:49:40 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 31 Aug 2007 09:49:40 -0400
Subject: [cisco-voip] CM Caller Name PRI support
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0069E1CB2@USNAEXCH.na.didata.local>
References: <6A0744639B86DD459433108DBF101D26018BDF0E@MAIL01.sentinel.com>
	<C1FE15183DA37645BC0633BC604E44F0069E1CB2@USNAEXCH.na.didata.local>
Message-ID: <f1b65f880708310649v2a7cd185g55f9e9679e61bddd@mail.gmail.com>

Whenever I've asked for CNAM on new PRI orders I havent had any issue
with it and the inbound caller ID name works fine.

However, I've only used CNAM on NI2 PRI circuits but I've done it with
Bellsouth, ICG, Qwest, MCI, ATT and a couple others.

However, I am never recommending or ordering CNAM from CLECs anymore.
I've been burned before because their database is totally messed up
and has more incorrect information than correct information.

Justin

On 8/31/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> My understanding is it's a switch change, versus a per port.
>
>
>
> I haven't found a telco that would change to accommodate Cisco, in 2001 I
> had my Cisco AM submit a PERS requesting callmanager 7900 phones receive
> calling party name under h323 in any method delivered by telco to gateway,
> check with your Cisco AM/SE on having them escalate the resolution.
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Carter, Bill
>  Sent: Thursday, August 30, 2007 11:12 PM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] CM Caller Name PRI support
>
>
>
>
> I have tried talking to a couple of Telco's regarding Caller Name support on
> PRI trunk.  Both vendors have been confused on the terms "Display ie" and
> "Facility ie".  Any suggestions on this for talking to Verizon or SBC/ATT ?
>
>
> ________________________________
>
>
>
>
>  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 jsteinberg at gmail.com  Fri Aug 31 09:51:10 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 31 Aug 2007 09:51:10 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <OFFE84AEAC.023B852F-ON85257348.004404D2-85257348.00476586@bftg.com>
References: <OFFE84AEAC.023B852F-ON85257348.004404D2-85257348.00476586@bftg.com>
Message-ID: <f1b65f880708310651w749d46ctdf1f0f75aa13f9a7@mail.gmail.com>

isdn map address 011* plan isdn type unknown

On 8/31/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote:
>
>
> I am not able to send International calls through our PRI's at a location
> outside of Philly.  Local and LD work fine.  debugs at the router show the
> number being sent to the telco are fine.
>
> Debug ISDN q931
>
> Calling Party Number i = 0x2181, '215664XXXX'
>                 Plan:ISDN, Type:National
>         Called Party Number i = 0x91, '011448458787000'
>
> I found a CCO TAC case where this exact problem is described and addressed
> by changing the calling plan type from national to unknown using the
> following command under the D channel interface.
>
> *isdn map address .* plan isdn type unknown*
>
> This did in fact change the plan type but I was unable to complete local,
> LD and Intl after adding this.
>
> Any suggestions?
>
> Carlos
>
>
>
>
>
>
> _______________________________________________
> 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 Aug 31 09:51:36 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 31 Aug 2007 09:51:36 -0400
Subject: [cisco-voip] Inter-Tel Phones using CallManager/Unity as voicemail?
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069E1E99@USNAEXCH.na.didata.local>

Does anyone have Inter-Tel phones hanging off callmanager via ISDN PRI
and using Unity Voicemail Only for voicemail? Can you describe your
setup and how it's working?

 

I'm not sure if the Inter-Tel phones MWI light will work right. Maybe I
can use stutter dial tone with Unity.

 

I don't see any PBX Interoperability guides with Intertel on CCO!

 




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From jsteinberg at gmail.com  Fri Aug 31 09:53:26 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 31 Aug 2007 09:53:26 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <f1b65f880708310651w749d46ctdf1f0f75aa13f9a7@mail.gmail.com>
References: <OFFE84AEAC.023B852F-ON85257348.004404D2-85257348.00476586@bftg.com>
	<f1b65f880708310651w749d46ctdf1f0f75aa13f9a7@mail.gmail.com>
Message-ID: <f1b65f880708310653s5f2c22b6i996b301ceaf125bc@mail.gmail.com>

actually i didnt read your entire email.  you need to make both plan and
isdn unknown for international since you are sending the leading '011'.

 isdn map address 011.* plan unknown type unknown



On 8/31/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> isdn map address 011* plan isdn type unknown
>
> On 8/31/07, CarlosOrtiz at bayviewfinancial.com <CarlosOrtiz at bayviewfinancial.com>
> wrote:
>
> >
> > I am not able to send International calls through our PRI's at a
> > location outside of Philly.  Local and LD work fine.  debugs at the router
> > show the number being sent to the telco are fine.
> >
> > Debug ISDN q931
> >
> > Calling Party Number i = 0x2181, '215664XXXX'
> >                 Plan:ISDN, Type:National
> >         Called Party Number i = 0x91, '011448458787000'
> >
> > I found a CCO TAC case where this exact problem is described and
> > addressed by changing the calling plan type from national to unknown using
> > the following command under the D channel interface.
> >
> > *isdn map address .* plan isdn type unknown*
> >
> > This did in fact change the plan type but I was unable to complete
> > local, LD and Intl after adding this.
> >
> > Any suggestions?
> >
> > Carlos
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
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From bhenry33 at adelphia.net  Fri Aug 31 09:56:35 2007
From: bhenry33 at adelphia.net (Brian E. Henry)
Date: Fri, 31 Aug 2007 9:56:35 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
Message-ID: <31233762.1188568595797.JavaMail.root@web22>

What is the cause code that you receive back from the telco?

I have used translation rules in the past on H323 gateways to do what you were doing and I do believe that 44 is United Kingdom and found that the provider was prefixing 011 also and thus the call failing.  I just sent them, for United Kingdom calls, 44XXXXX and it worked just fine.

Brian
---- CarlosOrtiz at bayviewfinancial.com wrote: 
> I am not able to send International calls through our PRI's at a location 
> outside of Philly.  Local and LD work fine.  debugs at the router show the 
> number being sent to the telco are fine. 
> 
> Debug ISDN q931
> 
> Calling Party Number i = 0x2181, '215664XXXX' 
>                 Plan:ISDN, Type:National 
>         Called Party Number i = 0x91, '011448458787000'
> 
> I found a CCO TAC case where this exact problem is described and addressed 
> by changing the calling plan type from national to unknown using the 
> following command under the D channel interface.
> 
> isdn map address .* plan isdn type unknown
> 
> This did in fact change the plan type but I was unable to complete local, 
> LD and Intl after adding this.
> 
> Any suggestions?
> 
> Carlos
> 
> 
> 
> 
> 
> 
> 


From jason.aarons at us.didata.com  Fri Aug 31 10:11:17 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 31 Aug 2007 10:11:17 -0400
Subject: [cisco-voip] CM Caller Name PRI support
In-Reply-To: <f1b65f880708310649v2a7cd185g55f9e9679e61bddd@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26018BDF0E@MAIL01.sentinel.com>
	<C1FE15183DA37645BC0633BC604E44F0069E1CB2@USNAEXCH.na.didata.local>
	<f1b65f880708310649v2a7cd185g55f9e9679e61bddd@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069E1F0C@USNAEXCH.na.didata.local>

MGCP I assume?

-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Friday, August 31, 2007 9:50 AM
To: Jason Aarons (US)
Cc: Carter, Bill; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM Caller Name PRI support

Whenever I've asked for CNAM on new PRI orders I havent had any issue
with it and the inbound caller ID name works fine.

However, I've only used CNAM on NI2 PRI circuits but I've done it with
Bellsouth, ICG, Qwest, MCI, ATT and a couple others.

However, I am never recommending or ordering CNAM from CLECs anymore.
I've been burned before because their database is totally messed up
and has more incorrect information than correct information.

Justin

On 8/31/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> My understanding is it's a switch change, versus a per port.
>
>
>
> I haven't found a telco that would change to accommodate Cisco, in
2001 I
> had my Cisco AM submit a PERS requesting callmanager 7900 phones
receive
> calling party name under h323 in any method delivered by telco to
gateway,
> check with your Cisco AM/SE on having them escalate the resolution.
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Carter, Bill
>  Sent: Thursday, August 30, 2007 11:12 PM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] CM Caller Name PRI support
>
>
>
>
> I have tried talking to a couple of Telco's regarding Caller Name
support on
> PRI trunk.  Both vendors have been confused on the terms "Display ie"
and
> "Facility ie".  Any suggestions on this for talking to Verizon or
SBC/ATT ?
>
>
> ________________________________
>
>
>
>
>  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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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
from your computer. Thank you.

From jefflin.choi at gmail.com  Fri Aug 31 10:39:05 2007
From: jefflin.choi at gmail.com (Jefflin Choi)
Date: Fri, 31 Aug 2007 22:39:05 +0800
Subject: [cisco-voip] CCME GUI Error Solution "Configuration change:
	Please report the following error: No Matched Command."
In-Reply-To: <553965.37766.qm@web57201.mail.re3.yahoo.com>
References: <553965.37766.qm@web57201.mail.re3.yahoo.com>
Message-ID: <5c72afa80708310739y39dc6f44l82d47cd91cb0ffa0@mail.gmail.com>

Hi Paul,

Removed and Reapplied aaa authorization configuration works fine for me.

I was just showing the customer that it was that command causing the problem
and after reapplied, it is still working.
I was expecting to have the problem after reapplying the "aaa authorization"
command but no more problems on GUI.

Cheers,
Jeff

On 8/31/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> This is a TACACS problem.
>
> If you have this line in your CCME box
>
> aaa authorization config-commands
>
> You will encounter this command. This will also
> prevent any non-TACACS authenticated login from making
> any changes to the router meaning no end user speed
> dials, no customer web admin, nothing.
>
> Remove this command and you should be okay...waiting
> on response from TAC for a workaround.
>
> Cheers,
> Paul
>
>
>
>
> ____________________________________________________________________________________
> 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
> _______________________________________________
> 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 Aug 31 10:59:10 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 31 Aug 2007 10:59:10 -0400
Subject: [cisco-voip] funny things with cat4000/cat4500
Message-ID: <001d01c7ebdf$7c83b830$56196883@cfs.uoguelph.ca>

We've had some funy things with our cat4000/cat4500s whereby phones have to cycle a few times before they register. We've never seen this, and if you connect a phone that was already connected, it usually doesn't cycle more than once. We also have a 7912 that refuses to come up on a 4000/4500 but works on a 37xx. 

Lelio
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From CarlosOrtiz at bayviewfinancial.com  Fri Aug 31 11:06:07 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 31 Aug 2007 11:06:07 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <31233762.1188568595797.JavaMail.root@web22>
Message-ID: <OFAEF8CFA5.1959462D-ON85257348.0052A8DF-85257348.0052F533@bftg.com>

Here is a test call :

ug 31 11:04:48.367 EDT: ISDN Se0/1/0:23 Q931: pak_private_number: Invalid 
type/plan 0x0 0x0 may be overriden; sw-type 13
Aug 31 11:04:48.367 EDT: ISDN Se0/1/0:23 Q931: pak_private_number: Invalid 
type/plan 0x0 0x0 may be overriden; sw-type 13
Aug 31 11:04:48.371 EDT: ISDN Se0/1/0:23 Q931: Applying typeplan for 
sw-type 0xD is 0x1 0x1, Called num 011448458787000
Aug 31 11:04:48.371 EDT: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref 
= 0x0243 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transfer Capability = Speech 
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98393 
                Exclusive, Channel 19 
        Progress Ind i = 0x8183 - Origination address is non-ISDN 
        Calling Party Number i = 0x2181, '2156641300' 
                Plan:ISDN, Type:National 
        Called Party Number i = 0x91, '011448458787000' 
                Plan:ISDN, Type:International
Aug 31 11:04:48.467 EDT: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8 
callref = 0x8243 
        Channel ID i = 0xA98393 
                Exclusive, Channel 19
Aug 31 11:04:48.583 EDT: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8 
callref = 0x8243 
        Cause i = 0xC484 - Vacant code or prefix digit not dialed 
        Progress Ind i = 0x8288 - In-band info or appropriate now 
available 
Aug 31 11:04:50.855 EDT: ISDN Se0/2/0:23 Q931: Ux_DLRelInd: DL_REL_IND 
received from L2







"Brian E. Henry" <bhenry33 at adelphia.net> 
08/31/2007 09:56 AM
Please respond to
bhenry33 at adelphia.net


To
CarlosOrtiz at bayviewfinancial.com
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] Intl Calls with Verizon in Philadelphia area






What is the cause code that you receive back from the telco?

I have used translation rules in the past on H323 gateways to do what you 
were doing and I do believe that 44 is United Kingdom and found that the 
provider was prefixing 011 also and thus the call failing.  I just sent 
them, for United Kingdom calls, 44XXXXX and it worked just fine.

Brian
---- CarlosOrtiz at bayviewfinancial.com wrote: 
> I am not able to send International calls through our PRI's at a 
location 
> outside of Philly.  Local and LD work fine.  debugs at the router show 
the 
> number being sent to the telco are fine. 
> 
> Debug ISDN q931
> 
> Calling Party Number i = 0x2181, '215664XXXX' 
>                 Plan:ISDN, Type:National 
>         Called Party Number i = 0x91, '011448458787000'
> 
> I found a CCO TAC case where this exact problem is described and 
addressed 
> by changing the calling plan type from national to unknown using the 
> following command under the D channel interface.
> 
> isdn map address .* plan isdn type unknown
> 
> This did in fact change the plan type but I was unable to complete 
local, 
> LD and Intl after adding this.
> 
> Any suggestions?
> 
> Carlos
> 
> 
> 
> 
> 
> 
> 


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From CarlosOrtiz at bayviewfinancial.com  Fri Aug 31 11:22:04 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 31 Aug 2007 11:22:04 -0400
Subject: [cisco-voip] Shared Line phone Limit
Message-ID: <OF798C418A.B81A0902-ON85257348.0053E86C-85257348.00546AEF@bftg.com>

Is their a limit on how many phones can share a line?  I have seen it set 
as high as 30 but had a request for about 60.

Carlos
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From jason.aarons at us.didata.com  Fri Aug 31 11:27:48 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 31 Aug 2007 11:27:48 -0400
Subject: [cisco-voip] Shared Lines in 4.2.3?
In-Reply-To: <OF798C418A.B81A0902-ON85257348.0053E86C-85257348.00546AEF@bftg.com>
References: <OF798C418A.B81A0902-ON85257348.0053E86C-85257348.00546AEF@bftg.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0069E20D6@USNAEXCH.na.didata.local>

Carlos, what version of CallManager are you running? I'm not sure about
the limit, but I am seeing something new in 4.2.3 with shared lines,
have you seen this? I've set up shared lines before in 4.1.3, this seems
new. If any of the 60 make a call your phone will chime and the display
will show whom they are calling! This could drive someone nuts!

 

Randy Manager - Line1 - 4000 (shared)

 

Sharon Receptionist - Line 1 - 4001

Sharon Receptionist - Line14 - 4000 (shared)

 

Elizabeth Employee - Line1 - 6600

 

If Sharon uses 4001 to dial 6600 then Randy's 7961 main display (same
area as You have Voicemail) shows "Sharon > Elizabeth" and makes a
chime.  Any way to disabled this feature system wide service parameter?

 

 

 

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Friday, August 31, 2007 11:22 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Shared Line phone Limit

 


Is their a limit on how many phones can share a line?  I have seen it
set as high as 30 but had a request for about 60. 

Carlos




-----------------------------------------
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This e-mail communication and any attachments may contain
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From CarlosOrtiz at bayviewfinancial.com  Fri Aug 31 11:33:56 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 31 Aug 2007 11:33:56 -0400
Subject: [cisco-voip] Shared Lines in 4.2.3?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0069E20D6@USNAEXCH.na.didata.local>
Message-ID: <OFD7D735D7.F6DC200B-ON85257348.00555EB2-85257348.00558143@bftg.com>

CM 4.1.3.    Did you mean Sharon dial from 4000 and not 4001?   In my case 
this will only be an inbound line for many people so the chime would not 
be an issue.   There has to be a way to disable that.



"Jason Aarons \(US\)" <jason.aarons at us.didata.com> 
08/31/2007 11:27 AM

To
<CarlosOrtiz at bayviewfinancial.com>, <cisco-voip at puck.nether.net>
cc

Subject
Shared Lines in 4.2.3?






Carlos, what version of CallManager are you running? I?m not sure about 
the limit, but I am seeing something new in 4.2.3 with shared lines, have 
you seen this? I?ve set up shared lines before in 4.1.3, this seems new. 
If any of the 60 make a call your phone will chime and the display will 
show whom they are calling! This could drive someone nuts!
 
Randy Manager ? Line1 ? 4000 (shared)
 
Sharon Receptionist ? Line 1 - 4001
Sharon Receptionist ? Line14 ? 4000 (shared)
 
Elizabeth Employee ? Line1 ? 6600
 
If Sharon uses 4001 to dial 6600 then Randy?s 7961 main display (same area 
as You have Voicemail) shows ?Sharon > Elizabeth? and makes a chime.  Any 
way to disabled this feature system wide service parameter?
 
 
 
 
 
 
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Friday, August 31, 2007 11:22 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Shared Line phone Limit
 

Is their a limit on how many phones can share a line?  I have seen it set 
as high as 30 but had a request for about 60. 

Carlos

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 jsteinberg at gmail.com  Fri Aug 31 11:36:01 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 31 Aug 2007 11:36:01 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <OFAEF8CFA5.1959462D-ON85257348.0052A8DF-85257348.0052F533@bftg.com>
References: <31233762.1188568595797.JavaMail.root@web22>
	<OFAEF8CFA5.1959462D-ON85257348.0052A8DF-85257348.0052F533@bftg.com>
Message-ID: <f1b65f880708310836qa47d49fi41b0f465add9086e@mail.gmail.com>

It is more than likely rejecting the call because you have international
access code '011' prefixed AND you have the isdn/international designation.
Some providers just want one or the other.

Did you try the following under the serial interface for your PRI?

 isdn map address 011.* plan unknown type unknown

If you are using MGCP you might change the dot in your route pattern to
9011.! and do digit discard predot.



On 8/31/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote:
>
>
> Here is a test call :
>
> ug 31 11:04:48.367 EDT: ISDN Se0/1/0:23 Q931: pak_private_number: Invalid
> type/plan 0x0 0x0 may be overriden; sw-type 13
> Aug 31 11:04:48.367 EDT: ISDN Se0/1/0:23 Q931: pak_private_number: Invalid
> type/plan 0x0 0x0 may be overriden; sw-type 13
> Aug 31 11:04:48.371 EDT: ISDN Se0/1/0:23 Q931: Applying typeplan for
> sw-type 0xD is 0x1 0x1, Called num 011448458787000
> Aug 31 11:04:48.371 EDT: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref
> = 0x0243
>         Bearer Capability i = 0x8090A2
>                 Standard = CCITT
>                 Transfer Capability = Speech
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA98393
>                 Exclusive, Channel 19
>         Progress Ind i = 0x8183 - Origination address is non-ISDN
>         Calling Party Number i = 0x2181, '2156641300'
>                 Plan:ISDN, Type:National
>         Called Party Number i = 0x91, '011448458787000'
>                 Plan:ISDN, Type:International
> Aug 31 11:04:48.467 EDT: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8
>  callref = 0x8243
>         Channel ID i = 0xA98393
>                 Exclusive, Channel 19
> Aug 31 11:04:48.583 EDT: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8
>  callref = 0x8243
>         Cause i = 0xC484 - Vacant code or prefix digit not dialed
>         Progress Ind i = 0x8288 - In-band info or appropriate now
> available
> Aug 31 11:04:50.855 EDT: ISDN Se0/2/0:23 Q931: Ux_DLRelInd: DL_REL_IND
> received from L2
>
>
>
>
>
>
>  *"Brian E. Henry" <bhenry33 at adelphia.net>*
>
> 08/31/2007 09:56 AM  Please respond to
> bhenry33 at adelphia.net
>
>   To
> CarlosOrtiz at bayviewfinancial.com  cc
> cisco-voip at puck.nether.net  Subject
> Re: [cisco-voip] Intl Calls with Verizon in Philadelphia area
>
>
>
>
>
>
> What is the cause code that you receive back from the telco?
>
> I have used translation rules in the past on H323 gateways to do what you
> were doing and I do believe that 44 is United Kingdom and found that the
> provider was prefixing 011 also and thus the call failing.  I just sent
> them, for United Kingdom calls, 44XXXXX and it worked just fine.
>
> Brian
> ---- CarlosOrtiz at bayviewfinancial.com wrote:
> > I am not able to send International calls through our PRI's at a
> location
> > outside of Philly.  Local and LD work fine.  debugs at the router show
> the
> > number being sent to the telco are fine.
> >
> > Debug ISDN q931
> >
> > Calling Party Number i = 0x2181, '215664XXXX'
> >                 Plan:ISDN, Type:National
> >         Called Party Number i = 0x91, '011448458787000'
> >
> > I found a CCO TAC case where this exact problem is described and
> addressed
> > by changing the calling plan type from national to unknown using the
> > following command under the D channel interface.
> >
> > isdn map address .* plan isdn type unknown
> >
> > This did in fact change the plan type but I was unable to complete
> local,
> > LD and Intl after adding this.
> >
> > Any suggestions?
> >
> > Carlos
> >
> >
> >
> >
> >
> >
> >
>
>
>
> _______________________________________________
> 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  Fri Aug 31 11:39:57 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 31 Aug 2007 11:39:57 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <f1b65f880708310836qa47d49fi41b0f465add9086e@mail.gmail.com>
Message-ID: <OF6FB769FA.B48F47EE-ON85257348.00560196-85257348.00560E19@bftg.com>

I'll try it out tonight as I can't make changes now......

Thanks for the suggestions......



"Justin Steinberg" <jsteinberg at gmail.com> 
08/31/2007 11:36 AM

To
"CarlosOrtiz at bayviewfinancial.com" <CarlosOrtiz at bayviewfinancial.com>
cc
bhenry33 at adelphia.net, cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] Intl Calls with Verizon in Philadelphia area






It is more than likely rejecting the call because you have international 
access code '011' prefixed AND you have the isdn/international 
designation.  Some providers just want one or the other.

Did you try the following under the serial interface for your PRI? 

 isdn map address 011.* plan unknown type unknown

If you are using MGCP you might change the dot in your route pattern to 
9011.! and do digit discard predot.



On 8/31/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote: 

Here is a test call : 

ug 31 11:04:48.367 EDT: ISDN Se0/1/0:23 Q931: pak_private_number: Invalid 
type/plan 0x0 0x0 may be overriden; sw-type 13 
Aug 31 11:04:48.367 EDT: ISDN Se0/1/0:23 Q931: pak_private_number: Invalid 
type/plan 0x0 0x0 may be overriden; sw-type 13 
Aug 31 11:04:48.371 EDT: ISDN Se0/1/0:23 Q931: Applying typeplan for 
sw-type 0xD is 0x1 0x1, Called num 011448458787000 
Aug 31 11:04:48.371 EDT: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref 
= 0x0243 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transfer Capability = Speech   
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98393 
                Exclusive, Channel 19 
        Progress Ind i = 0x8183 - Origination address is non-ISDN   
        Calling Party Number i = 0x2181, '2156641300' 
                Plan:ISDN, Type:National 
        Called Party Number i = 0x91, '011448458787000' 
                Plan:ISDN, Type:International 
Aug 31 11:04:48.467 EDT: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8 
callref = 0x8243 
        Channel ID i = 0xA98393 
                Exclusive, Channel 19 
Aug 31 11:04:48.583 EDT: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8 
callref = 0x8243 
        Cause i = 0xC484 - Vacant code or prefix digit not dialed 
        Progress Ind i = 0x8288 - In-band info or appropriate now 
available 
Aug 31 11:04:50.855 EDT: ISDN Se0/2/0:23 Q931: Ux_DLRelInd: DL_REL_IND 
received from L2 






"Brian E. Henry" <bhenry33 at adelphia.net> 
08/31/2007 09:56 AM 

Please respond to
bhenry33 at adelphia.net



To
CarlosOrtiz at bayviewfinancial.com 
cc
cisco-voip at puck.nether.net 
Subject
Re: [cisco-voip] Intl Calls with Verizon in Philadelphia area








What is the cause code that you receive back from the telco?

I have used translation rules in the past on H323 gateways to do what you 
were doing and I do believe that 44 is United Kingdom and found that the 
provider was prefixing 011 also and thus the call failing.  I just sent 
them, for United Kingdom calls, 44XXXXX and it worked just fine.

Brian
---- CarlosOrtiz at bayviewfinancial.com wrote: 
> I am not able to send International calls through our PRI's at a 
location 
> outside of Philly.  Local and LD work fine.  debugs at the router show 
the 
> number being sent to the telco are fine. 
> 
> Debug ISDN q931
> 
> Calling Party Number i = 0x2181, '215664XXXX' 
>                 Plan:ISDN, Type:National 
>         Called Party Number i = 0x91, '011448458787000'
> 
> I found a CCO TAC case where this exact problem is described and 
addressed 
> by changing the calling plan type from national to unknown using the 
> following command under the D channel interface.
> 
> isdn map address .* plan isdn type unknown
> 
> This did in fact change the plan type but I was unable to complete 
local, 
> LD and Intl after adding this.
> 
> Any suggestions?
> 
> Carlos
> 
> 
> 
> 
> 
> 
> 



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

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From tednugent69 at yahoo.com  Fri Aug 31 13:15:53 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 31 Aug 2007 10:15:53 -0700 (PDT)
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <OFFE84AEAC.023B852F-ON85257348.004404D2-85257348.00476586@bftg.com>
Message-ID: <599918.53357.qm@web36115.mail.mud.yahoo.com>

I had similar issues a few weeks back with an H323
gateway and tried leaving plan type as is (defaulting
to international like yours) and just striping the 011
and for whatever reason I was still getting the error.
Eventually I ended up using translation profiles to
change the plan and that worked.

voice translation-rule 1
 rule 1 /^9011/ /011/ type any national plan any isdn
!
voice translation-profile pstn-out
 translate called 1
!
dial-peer voice 30 pots
 translation-profile outgoing pstn-out
 destination-pattern 9011T
 port 0/0/0:23
 forward-digits all




--- CarlosOrtiz at bayviewfinancial.com wrote:

> I am not able to send International calls through
> our PRI's at a location 
> outside of Philly.  Local and LD work fine.  debugs
> at the router show the 
> number being sent to the telco are fine. 
> 
> Debug ISDN q931
> 
> Calling Party Number i = 0x2181, '215664XXXX' 
>                 Plan:ISDN, Type:National 
>         Called Party Number i = 0x91,
> '011448458787000'
> 
> I found a CCO TAC case where this exact problem is
> described and addressed 
> by changing the calling plan type from national to
> unknown using the 
> following command under the D channel interface.
> 
> isdn map address .* plan isdn type unknown
> 
> This did in fact change the plan type but I was
> unable to complete local, 
> LD and Intl after adding this.
> 
> Any suggestions?
> 
> Carlos
> 
> 
> 
> 
> 
> 
> 
> > _______________________________________________
> 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 

From CarlosOrtiz at bayviewfinancial.com  Fri Aug 31 14:26:48 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 31 Aug 2007 14:26:48 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <599918.53357.qm@web36115.mail.mud.yahoo.com>
Message-ID: <OF892FC6A6.55F121C8-ON85257348.00652EF7-85257348.0065580E@bftg.com>

So when the calls exit what are they marked as?   What LEC was it? 

Carlos



Ted Nugent <tednugent69 at yahoo.com> 
08/31/2007 01:15 PM

To
CarlosOrtiz at bayviewfinancial.com, cisco-voip at puck.nether.net
cc

Subject
Re: [cisco-voip] Intl Calls with Verizon in Philadelphia area






I had similar issues a few weeks back with an H323
gateway and tried leaving plan type as is (defaulting
to international like yours) and just striping the 011
and for whatever reason I was still getting the error.
Eventually I ended up using translation profiles to
change the plan and that worked.

voice translation-rule 1
 rule 1 /^9011/ /011/ type any national plan any isdn
!
voice translation-profile pstn-out
 translate called 1
!
dial-peer voice 30 pots
 translation-profile outgoing pstn-out
 destination-pattern 9011T
 port 0/0/0:23
 forward-digits all




--- CarlosOrtiz at bayviewfinancial.com wrote:

> I am not able to send International calls through
> our PRI's at a location 
> outside of Philly.  Local and LD work fine.  debugs
> at the router show the 
> number being sent to the telco are fine. 
> 
> Debug ISDN q931
> 
> Calling Party Number i = 0x2181, '215664XXXX' 
>                 Plan:ISDN, Type:National 
>         Called Party Number i = 0x91,
> '011448458787000'
> 
> I found a CCO TAC case where this exact problem is
> described and addressed 
> by changing the calling plan type from national to
> unknown using the 
> following command under the D channel interface.
> 
> isdn map address .* plan isdn type unknown
> 
> This did in fact change the plan type but I was
> unable to complete local, 
> LD and Intl after adding this.
> 
> Any suggestions?
> 
> Carlos
> 
> 
> 
> 
> 
> 
> 
> > _______________________________________________
> 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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From lelio at uoguelph.ca  Fri Aug 31 14:31:54 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 31 Aug 2007 14:31:54 -0400
Subject: [cisco-voip] MGCP dial-peer order changing
Message-ID: <00ab01c7ebfd$34a94430$56196883@cfs.uoguelph.ca>

OK, this is now the second time that I find my MGCP dial-peers at the bottom of my dial-peer list after I modified things to put them at the top.

What is going on?

Have others seen this?

Lelio
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From tednugent69 at yahoo.com  Fri Aug 31 15:05:06 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 31 Aug 2007 12:05:06 -0700 (PDT)
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <OF892FC6A6.55F121C8-ON85257348.00652EF7-85257348.0065580E@bftg.com>
Message-ID: <187435.28291.qm@web36110.mail.mud.yahoo.com>

I don't have the debugs anymore and they weren't much
help anyway since I was only really seeing a LEC
disconnect but you could see that the plan and type
would switch to "Plan:ISDN, Type:International" for
international calls. The LEC was Time Warner here in
Raleigh NC and the TW switch tech was saying basically
the same thing. Either drop the 011 and send it as
international or send it as national and keep the 011.
Like I said I could not get it to work dropping the
011 and sending it as international so I had to change
it to national and send the 011 which worked.
ALso another weird thing was it was only happening on
certain international calls, some would go through
fine others would get dropped with no really good ISDN
cause code other then far end rejected or something to
that affect.



--- CarlosOrtiz at bayviewfinancial.com wrote:

> So when the calls exit what are they marked as?  
> What LEC was it? 
> 
> Carlos
> 
> 
> 
> Ted Nugent <tednugent69 at yahoo.com> 
> 08/31/2007 01:15 PM
> 
> To
> CarlosOrtiz at bayviewfinancial.com,
> cisco-voip at puck.nether.net
> cc
> 
> Subject
> Re: [cisco-voip] Intl Calls with Verizon in
> Philadelphia area
> 
> 
> 
> 
> 
> 
> I had similar issues a few weeks back with an H323
> gateway and tried leaving plan type as is
> (defaulting
> to international like yours) and just striping the
> 011
> and for whatever reason I was still getting the
> error.
> Eventually I ended up using translation profiles to
> change the plan and that worked.
> 
> voice translation-rule 1
>  rule 1 /^9011/ /011/ type any national plan any
> isdn
> !
> voice translation-profile pstn-out
>  translate called 1
> !
> dial-peer voice 30 pots
>  translation-profile outgoing pstn-out
>  destination-pattern 9011T
>  port 0/0/0:23
>  forward-digits all
> 
> 
> 
> 
> --- CarlosOrtiz at bayviewfinancial.com wrote:
> 
> > I am not able to send International calls through
> > our PRI's at a location 
> > outside of Philly.  Local and LD work fine. 
> debugs
> > at the router show the 
> > number being sent to the telco are fine. 
> > 
> > Debug ISDN q931
> > 
> > Calling Party Number i = 0x2181, '215664XXXX' 
> >                 Plan:ISDN, Type:National 
> >         Called Party Number i = 0x91,
> > '011448458787000'
> > 
> > I found a CCO TAC case where this exact problem is
> > described and addressed 
> > by changing the calling plan type from national to
> > unknown using the 
> > following command under the D channel interface.
> > 
> > isdn map address .* plan isdn type unknown
> > 
> > This did in fact change the plan type but I was
> > unable to complete local, 
> > LD and Intl after adding this.
> > 
> > Any suggestions?
> > 
> > Carlos
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > > _______________________________________________
> > 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 
> 
> 



       
____________________________________________________________________________________
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 jeff at ocjtech.us  Fri Aug 31 15:05:14 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Fri, 31 Aug 2007 14:05:14 -0500
Subject: [cisco-voip] LowActivePartitionAvailableDiskSpace?
Message-ID: <1188587114.3432.3.camel@lt21223.campus.dmacc.edu>

I tried to export too many CDRs on my CM5.1 publisher and it filled up a
partition and the CDR export failed.  RTMT has been complaining ever
since.  I haven't figured out what I need to do to clean up the mess and
free up the space (hopefully I'm not losing CDRs now).  What's the
proper procedure for cleaning this up?

Jeff

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From SCASPER at mtb.com  Fri Aug 31 15:04:39 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Fri, 31 Aug 2007 15:04:39 -0400
Subject: [cisco-voip] No Hold Key
In-Reply-To: <f4445faf0708302259n791d75dl5509a08313ebd28c@mail.gmail.com>
References: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM>
	<6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com>
	<f4445faf0708302259n791d75dl5509a08313ebd28c@mail.gmail.com>
Message-ID: <46D82E09.9874.0039.0@mtb.com>

 I have a few sites where users are complaining that after answering an inbound call no Hold softkey is available. This does not happen all the time only a couple of times a day. These sites use multiple shared lines. We are on 4.1.3.
  I was wondering if the "maximum number of calls" setting could have an effect on this but my understanding is that this setting is per device so even if you have a call on a line that shows up on 15 phones it should only subtract one call from the available call pool. Then again I may be on the completely wrong track. What would cause the Hold soft key to not apear sporadically?
 
Steve

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

************************************
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From tednugent69 at yahoo.com  Fri Aug 31 15:07:34 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 31 Aug 2007 12:07:34 -0700 (PDT)
Subject: [cisco-voip] MGCP dial-peer order changing
In-Reply-To: <00ab01c7ebfd$34a94430$56196883@cfs.uoguelph.ca>
Message-ID: <65647.80201.qm@web36106.mail.mud.yahoo.com>

This was a bug a LONG time ago.. the fix at the time
was to remove "ccm-manager config" to prevent MGCP
from recreating the dialpeers and writing them at the
bottom... I have not had this problem in years though?


--- Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

> OK, this is now the second time that I find my MGCP
> dial-peers at the bottom of my dial-peer list after
> I modified things to put them at the top.
> 
> What is going on?
> 
> Have others seen this?
> 
> Lelio
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



       
____________________________________________________________________________________
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Comedy with an Edge to see what's on, when. 
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From lelio at uoguelph.ca  Fri Aug 31 15:11:54 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 31 Aug 2007 15:11:54 -0400
Subject: [cisco-voip] MGCP dial-peer order changing
References: <65647.80201.qm@web36106.mail.mud.yahoo.com>
Message-ID: <00ed01c7ec02$cb137c10$56196883@cfs.uoguelph.ca>

dear lord. some of us want that ccm-manager config statement in there!

i've got pretty recent code, 12.4T something, so why it's not fixed, who 
knows.


----- Original Message ----- 
From: "Ted Nugent" <tednugent69 at yahoo.com>
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>; "Cisco VoIPoE List" 
<cisco-voip at puck.nether.net>
Sent: Friday, August 31, 2007 3:07 PM
Subject: Re: [cisco-voip] MGCP dial-peer order changing


> This was a bug a LONG time ago.. the fix at the time
> was to remove "ccm-manager config" to prevent MGCP
> from recreating the dialpeers and writing them at the
> bottom... I have not had this problem in years though?
>
>
> --- Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>> OK, this is now the second time that I find my MGCP
>> dial-peers at the bottom of my dial-peer list after
>> I modified things to put them at the top.
>>
>> What is going on?
>>
>> Have others seen this?
>>
>> Lelio
>> > _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ____________________________________________________________________________________
> Sick sense of humor? Visit Yahoo! TV's
> Comedy with an Edge to see what's on, when.
> http://tv.yahoo.com/collections/222
> 


From tednugent69 at yahoo.com  Fri Aug 31 15:15:07 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 31 Aug 2007 12:15:07 -0700 (PDT)
Subject: [cisco-voip] CM6.1 released??
Message-ID: <265724.96392.qm@web36115.mail.mud.yahoo.com>

I just heard from one of our Cisco guys that 6.1 was
released? I told him that he was smoking crack because
they just released 6.01 like 2 weeks ago and there?s
no mention of 6.1 anywhere on CCO
 Is it aprilfools
day or something or what?s the deal?


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

From CarlosOrtiz at bayviewfinancial.com  Fri Aug 31 16:09:00 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 31 Aug 2007 16:09:00 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <187435.28291.qm@web36110.mail.mud.yahoo.com>
Message-ID: <OFE249FB10.1A9D0D34-ON85257348.006E7E8D-85257348.006EB05D@bftg.com>

I just got off the phone with Verizon and the tech was giving me some 
stories that could explain your problem.  He says that once the calls get 
across the pond they sometimes are sent to different international 
carriers.  Some switches will pass it fine while others not.  That could 
explain your hit or miss scenario.  Anyway this is pretty aggravating.

Carlos



Ted Nugent <tednugent69 at yahoo.com> 
Sent by: cisco-voip-bounces at puck.nether.net
08/31/2007 03:05 PM

To
CarlosOrtiz at bayviewfinancial.com
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] Intl Calls with Verizon in Philadelphia area






I don't have the debugs anymore and they weren't much
help anyway since I was only really seeing a LEC
disconnect but you could see that the plan and type
would switch to "Plan:ISDN, Type:International" for
international calls. The LEC was Time Warner here in
Raleigh NC and the TW switch tech was saying basically
the same thing. Either drop the 011 and send it as
international or send it as national and keep the 011.
Like I said I could not get it to work dropping the
011 and sending it as international so I had to change
it to national and send the 011 which worked.
ALso another weird thing was it was only happening on
certain international calls, some would go through
fine others would get dropped with no really good ISDN
cause code other then far end rejected or something to
that affect.



--- CarlosOrtiz at bayviewfinancial.com wrote:

> So when the calls exit what are they marked as? 
> What LEC was it? 
> 
> Carlos
> 
> 
> 
> Ted Nugent <tednugent69 at yahoo.com> 
> 08/31/2007 01:15 PM
> 
> To
> CarlosOrtiz at bayviewfinancial.com,
> cisco-voip at puck.nether.net
> cc
> 
> Subject
> Re: [cisco-voip] Intl Calls with Verizon in
> Philadelphia area
> 
> 
> 
> 
> 
> 
> I had similar issues a few weeks back with an H323
> gateway and tried leaving plan type as is
> (defaulting
> to international like yours) and just striping the
> 011
> and for whatever reason I was still getting the
> error.
> Eventually I ended up using translation profiles to
> change the plan and that worked.
> 
> voice translation-rule 1
>  rule 1 /^9011/ /011/ type any national plan any
> isdn
> !
> voice translation-profile pstn-out
>  translate called 1
> !
> dial-peer voice 30 pots
>  translation-profile outgoing pstn-out
>  destination-pattern 9011T
>  port 0/0/0:23
>  forward-digits all
> 
> 
> 
> 
> --- CarlosOrtiz at bayviewfinancial.com wrote:
> 
> > I am not able to send International calls through
> > our PRI's at a location 
> > outside of Philly.  Local and LD work fine. 
> debugs
> > at the router show the 
> > number being sent to the telco are fine. 
> > 
> > Debug ISDN q931
> > 
> > Calling Party Number i = 0x2181, '215664XXXX' 
> >                 Plan:ISDN, Type:National 
> >         Called Party Number i = 0x91,
> > '011448458787000'
> > 
> > I found a CCO TAC case where this exact problem is
> > described and addressed 
> > by changing the calling plan type from national to
> > unknown using the 
> > following command under the D channel interface.
> > 
> > isdn map address .* plan isdn type unknown
> > 
> > This did in fact change the plan type but I was
> > unable to complete local, 
> > LD and Intl after adding this.
> > 
> > Any suggestions?
> > 
> > Carlos
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > > _______________________________________________
> > 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 
> 
> 



 
____________________________________________________________________________________
Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, 
news, photos & more. 
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From jsteinberg at gmail.com  Fri Aug 31 16:17:49 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 31 Aug 2007 16:17:49 -0400
Subject: [cisco-voip] Intl Calls with Verizon in Philadelphia area
In-Reply-To: <OFE249FB10.1A9D0D34-ON85257348.006E7E8D-85257348.006EB05D@bftg.com>
References: <187435.28291.qm@web36110.mail.mud.yahoo.com>
	<OFE249FB10.1A9D0D34-ON85257348.006E7E8D-85257348.006EB05D@bftg.com>
Message-ID: <f1b65f880708311317w6851df35waee34a12a9ac0aab@mail.gmail.com>

my situation was just like Tim's.  Hit or miss.  When you add the isdn map
command under your serial interface that will fix your problem.



On 8/31/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote:
>
>
> I just got off the phone with Verizon and the tech was giving me some
> stories that could explain your problem.  He says that once the calls get
> across the pond they sometimes are sent to different international carriers.
>  Some switches will pass it fine while others not.  That could explain your
> hit or miss scenario.  Anyway this is pretty aggravating.
>
> Carlos
>
>
>  *Ted Nugent <tednugent69 at yahoo.com>*
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 08/31/2007 03:05 PM
>   To
> CarlosOrtiz at bayviewfinancial.com  cc
> cisco-voip at puck.nether.net  Subject
> Re: [cisco-voip] Intl Calls with Verizon in Philadelphia area
>
>
>
>
>
>
> I don't have the debugs anymore and they weren't much
> help anyway since I was only really seeing a LEC
> disconnect but you could see that the plan and type
> would switch to "Plan:ISDN, Type:International" for
> international calls. The LEC was Time Warner here in
> Raleigh NC and the TW switch tech was saying basically
> the same thing. Either drop the 011 and send it as
> international or send it as national and keep the 011.
> Like I said I could not get it to work dropping the
> 011 and sending it as international so I had to change
> it to national and send the 011 which worked.
> ALso another weird thing was it was only happening on
> certain international calls, some would go through
> fine others would get dropped with no really good ISDN
> cause code other then far end rejected or something to
> that affect.
>
>
>
> --- CarlosOrtiz at bayviewfinancial.com wrote:
>
> > So when the calls exit what are they marked as?
> > What LEC was it?
> >
> > Carlos
> >
> >
> >
> > Ted Nugent <tednugent69 at yahoo.com>
> > 08/31/2007 01:15 PM
> >
> > To
> > CarlosOrtiz at bayviewfinancial.com,
> > cisco-voip at puck.nether.net
> > cc
> >
> > Subject
> > Re: [cisco-voip] Intl Calls with Verizon in
> > Philadelphia area
> >
> >
> >
> >
> >
> >
> > I had similar issues a few weeks back with an H323
> > gateway and tried leaving plan type as is
> > (defaulting
> > to international like yours) and just striping the
> > 011
> > and for whatever reason I was still getting the
> > error.
> > Eventually I ended up using translation profiles to
> > change the plan and that worked.
> >
> > voice translation-rule 1
> >  rule 1 /^9011/ /011/ type any national plan any
> > isdn
> > !
> > voice translation-profile pstn-out
> >  translate called 1
> > !
> > dial-peer voice 30 pots
> >  translation-profile outgoing pstn-out
> >  destination-pattern 9011T
> >  port 0/0/0:23
> >  forward-digits all
> >
> >
> >
> >
> > --- CarlosOrtiz at bayviewfinancial.com wrote:
> >
> > > I am not able to send International calls through
> > > our PRI's at a location
> > > outside of Philly.  Local and LD work fine.
> > debugs
> > > at the router show the
> > > number being sent to the telco are fine.
> > >
> > > Debug ISDN q931
> > >
> > > Calling Party Number i = 0x2181, '215664XXXX'
> > >                 Plan:ISDN, Type:National
> > >         Called Party Number i = 0x91,
> > > '011448458787000'
> > >
> > > I found a CCO TAC case where this exact problem is
> > > described and addressed
> > > by changing the calling plan type from national to
> > > unknown using the
> > > following command under the D channel interface.
> > >
> > > isdn map address .* plan isdn type unknown
> > >
> > > This did in fact change the plan type but I was
> > > unable to complete local,
> > > LD and Intl after adding this.
> > >
> > > Any suggestions?
> > >
> > > Carlos
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > > _______________________________________________
> > > 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
> >
> >
>
>
>
>
>
> ____________________________________________________________________________________
> Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail,
> news, photos & more.
> http://mobile.yahoo.com/go?refer=1GNXIC
> _______________________________________________
> 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 jason.aarons at us.didata.com  Fri Aug 31 17:15:30 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 31 Aug 2007 17:15:30 -0400
Subject: [cisco-voip] No Hold Key
In-Reply-To: <46D82E09.9874.0039.0@mtb.com>
References: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM><6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com><f4445faf0708302259n791d75dl5509a08313ebd28c@mail.gmail.com>
	<46D82E09.9874.0039.0@mtb.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006A20D0E@USNAEXCH.na.didata.local>

I heard this the other day under 4.3.2, I thought it was end-user
training issue, they were doing something wrong, same deal the site has
everyone sharing everyone's lines using 7914 sidecars.

I like your idea, max number of calls per line must be too low...I had
left it at 4 lines I think..

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of STEVEN CASPER
Sent: Friday, August 31, 2007 3:05 PM
To: cisco-voip at puck.nether.net
Cc: LINDA ROSENBLUTH; RAWLE GORDON; KATHERINE FOX
Subject: [cisco-voip] No Hold Key

 

 I have a few sites where users are complaining that after answering an
inbound call no Hold softkey is available. This does not happen all the
time only a couple of times a day. These sites use multiple shared
lines. We are on 4.1.3.

  I was wondering if the "maximum number of calls" setting could have an
effect on this but my understanding is that this setting is per device
so even if you have a call on a line that shows up on 15 phones it
should only subtract one call from the available call pool. Then again I
may be on the completely wrong track. What would cause the Hold soft key
to not apear sporadically?

 

Steve

 

 

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

************************************
This email may contain privileged and/or confidential information that
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From tednugent69 at yahoo.com  Fri Aug 31 16:58:03 2007
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 31 Aug 2007 13:58:03 -0700 (PDT)
Subject: [cisco-voip] CM6.1 released??
Message-ID: <872543.65723.qm@web36109.mail.mud.yahoo.com>

ok maybe its true after all, but I'm pretty sure that
guy still smokes crack. :-)? Anyone have confirmation
or a feature listing?

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_bulletin0900aecd80693a40.html




--- Ted Nugent <tednugent69 at yahoo.com> wrote:

> I just heard from one of our Cisco guys that 6.1 was
> released? I told him that he was smoking crack
> because
> they just released 6.01 like 2 weeks ago and there?s
> no mention of 6.1 anywhere on CCO
 Is it aprilfools
> day or something or what?s the deal?
> 
> 
>        
>
____________________________________________________________________________________
> Need a vacation? Get great deals
> to amazing places on Yahoo! Travel.
> http://travel.yahoo.com/
> 



      ____________________________________________________________________________________
Luggage? GPS? Comic books? 
Check out fitting gifts for grads at Yahoo! Search
http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz

From SCASPER at mtb.com  Fri Aug 31 18:25:44 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Fri, 31 Aug 2007 18:25:44 -0400
Subject: [cisco-voip] No Hold Key
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F006A20D0E@USNAEXCH.na.didata.local>
References: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM><6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com><f4445faf0708302259n791d75dl5509a08313ebd28c@mail.gmail.com>
	<46D82E09.9874.0039.0@mtb.com>
	<C1FE15183DA37645BC0633BC604E44F006A20D0E@USNAEXCH.na.didata.local>
Message-ID: <46D85D29.9874.0039.0@mtb.com>

 That is what I thought also. We do not turn on speaker phone at our
branch locations which leads to much confusion. Should be easy enough to
re-create. I'll set it up in our lab and post the results

>>> "Jason Aarons (US)" <jason.aarons at us.didata.com> 8/31/2007 5:15 PM
>>>

I heard this the other day under 4.3.2, I thought it was end-user
training issue, they were doing something wrong, same deal the site has
everyone sharing everyone?s lines using 7914 sidecars.
I like your idea, max number of calls per line must be too low?I had
left it at 4 lines I think..
 

From:cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of STEVEN
CASPER
Sent: Friday, August 31, 2007 3:05 PM
To: cisco-voip at puck.nether.net 
Cc: LINDA ROSENBLUTH; RAWLE GORDON; KATHERINE FOX
Subject: [cisco-voip] No Hold Key

 

 I have a few sites where users are complaining that after answering an
inbound call no Hold softkey is available. This does not happen all the
time only a couple of times a day. These sites use multiple shared
lines. We are on 4.1.3.

  I was wondering if the "maximum number of calls" setting could have
an effect on this but my understanding is that this setting is per
device so even if you have a call on a line that shows up on 15 phones
it should only subtract one call from the available call pool. Then
again I may be on the completely wrong track. What would cause the Hold
soft key to not apear sporadically?

 

Steve

 

 

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026 
************************************
This email may contain privileged and/or confidential information that
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From asobihoudai at yahoo.com  Fri Aug 31 19:54:28 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 31 Aug 2007 16:54:28 -0700 (PDT)
Subject: [cisco-voip] De-Frag on Call Managers and Unity?
In-Reply-To: <6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com>
Message-ID: <205510.37747.qm@web57208.mail.re3.yahoo.com>

I read somewhere a while back that defragmentation
tools should never be run on an Exchange server. I
guess this means that defrag should never be run on a
windows-based Unity server. Can anyone comment on
this?

Paul

--- "Carter, Bill" <bcarter at sentinel.com> wrote:

> If you are onsite and have time....I sometimes
> reboot into Windows Safe
> mode and run defrag.  It can take 30 minutes.  Can't
> tell if it makes a
> difference.
> 
> ________________________________
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Miller, Steve
> Sent: Thursday, August 30, 2007 9:53 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] De-Frag on Call Managers and
> Unity?
> 
> 
> Does anyone have any words of wisdom about
> whether/how defrags can be
> run on these servers?
>  
> 
> 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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From SCASPER at mtb.com  Fri Aug 31 20:29:18 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Fri, 31 Aug 2007 20:29:18 -0400
Subject: [cisco-voip] No Hold Key
In-Reply-To: <46D85D29.9874.0039.0@mtb.com>
References: <421CAA1A5300B84A87D200D709985782041F46@DCEX2.DSMO.COM><6A0744639B86DD459433108DBF101D26018BDF0D@MAIL01.sentinel.com><f4445faf0708302259n791d75dl5509a08313ebd28c@mail.gmail.com>
	<46D82E09.9874.0039.0@mtb.com>
	<C1FE15183DA37645BC0633BC604E44F006A20D0E@USNAEXCH.na.didata.local>
	<46D85D29.9874.0039.0@mtb.com>
Message-ID: <46D87A1F.9874.0039.0@mtb.com>

Got a response on the CCO forum. Looks like this may be a chronic
problem.
 
http://forums.cisco.com/eforum/servlet/NetProf?page=netprof&type=Subscriptions&loc=.1ddfaafe/0&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony


>>> "STEVEN CASPER" <SCASPER at mtb.com> 8/31/2007 6:25 PM >>>
 That is what I thought also. We do not turn on speaker phone at our
branch locations which leads to much confusion. Should be easy enough to
re-create. I'll set it up in our lab and post the results

>>> "Jason Aarons (US)" <jason.aarons at us.didata.com> 8/31/2007 5:15 PM
>>>

I heard this the other day under 4.3.2, I thought it was end-user
training issue, they were doing something wrong, same deal the site has
everyone sharing everyone?s lines using 7914 sidecars.
I like your idea, max number of calls per line must be too low?I had
left it at 4 lines I think..
 

From:cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of STEVEN
CASPER
Sent: Friday, August 31, 2007 3:05 PM
To: cisco-voip at puck.nether.net 
Cc: LINDA ROSENBLUTH; RAWLE GORDON; KATHERINE FOX
Subject: [cisco-voip] No Hold Key

 

 I have a few sites where users are complaining that after answering an
inbound call no Hold softkey is available. This does not happen all the
time only a couple of times a day. These sites use multiple shared
lines. We are on 4.1.3.

  I was wondering if the "maximum number of calls" setting could have
an effect on this but my understanding is that this setting is per
device so even if you have a call on a line that shows up on 15 phones
it should only subtract one call from the available call pool. Then
again I may be on the completely wrong track. What would cause the Hold
soft key to not apear sporadically?

 

Steve

 

 

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.
************************************



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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n
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From jason.aarons at us.didata.com  Fri Aug 31 20:43:16 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 31 Aug 2007 20:43:16 -0400
Subject: [cisco-voip] Call Pickup Notification  4.2 Shared line chime
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0069A9DEC@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0069A9DEC@USNAEXCH.na.didata.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F006A20DFE@USNAEXCH.na.didata.local>

Turns out I think this is Call Pickup Notification new in 4.2, we have
Pickup Groups setup, doing some more testing -jason

 

New and Change Features in CallManager - Call Pickup Notification

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_note09
186a00805f1470.html#wp2448574

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Thursday, August 30, 2007 11:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 4.2 Shared line chime

 

Randy Manager - Line1 - 4000 (shared)

 

Sharon Receptionist - Line 1 - 4001

Sharon Receptionist - Line14 - 4000 (shared)

 

Elizabeth Employee - Line1 - 6600

 

If Sharon uses 4001 to dial 6600 then Randy's 7961 main display (same
area as You have Voicemail) shows "Sharon > Elizabeth" and makes a
chime.  Any way to disabled this feature system wide service parameter?

 

CallManager 4.2.3(SR2b) with 7961/7914s running SCCP41.8-2-2SR1S

 

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