From hcb at unco.de  Mon Sep  1 23:47:17 2003
From: hcb at unco.de (Hielke Christian Braun)
Date: Tue Sep  2 01:48:08 2003
Subject: [cisco-voip] Restricting CLID.
In-Reply-To: <20030827104008.GE7354@bofh.pl>
References: <20030827100921.GD7354@bofh.pl>
	<20030827102853.GA36443@frogman.motivity.ca>
	<20030827104008.GE7354@bofh.pl>
Message-ID: <20030902054717.GA5990@pacserv>

Jakob,

i am using a rule like this:

!
translation-rule 1
 Rule 1 any null
!

to remove all calling numbers on outgoing calls. Also depending on
the IOS version you run, there might be a clid option in the pots
dial-peer:

xxx-xxxx(config-dial-peer)#clid ?
  network-number  To provide Network Provided Number
  restrict        To restrict presentation of caller ID
  strip           To remove calling party number entirely


It is in 12.3. In earlier versions i think it was invisible.

Regards,
 Christian.



On Wed, Aug 27, 2003 at 12:40:08PM +0200, Jakub Klausa wrote:
> On Wed, Aug 27, 2003 at 06:28:53AM -0400, Vance Shipley wrote:
> => Jakub,
> => 
> => Do you want to "set the presentation indicator to Restricted" or do
> => you want to remove the Calling Party Number IE?  The former will
> => continue to send the Calling Party Number IE, including the number
> => of the caller, but will specify that it should not be displayed on
> => the called party's telephone.  This allows services such as Call
> => Return to continue to work.
> => 
> => In either case I don't have an answer handy but it something to
> => consider.
> => 
> => 	-Vance
> 
> I'd rather have the secound one, ie not sending anything in the CPA number IE,
> but setting the restricted indicator would be also very nice.
> 
> The only way i somehow thought of doing the former is to do some tricky
> translation-rule, but despite my problems with the translation-rules logic
> and grammar earlier, i don't know how to make it to leave the rewritten
> field empty.
> 
> Any sugguestions?
> 
> -- 
> k.



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

From Scott.Voll at wesd.org  Tue Sep  2 16:03:17 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Sep  2 18:03:25 2003
Subject: [cisco-voip] 7905 and DHCP?
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D437@venus.wesd.org>

Anyone else having problems with 7905's getting DHCP from a router running DHCP???  The 7940sand 60 don't seem to have any problems.  Will be opening a tac case.

Scott Voll
Network Analyst, CCNA 
Willamette ESD
Phone: (503) 385-4571
scott.voll@wesd.org
?


From wsisk at cisco.com  Wed Sep  3 16:11:09 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Sep  3 15:11:24 2003
Subject: [cisco-voip] 7905 and DHCP?
In-Reply-To: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D437@venus.wesd.org>
Message-ID: <ELEMLGDGDBLINAJEIOCLIEBHMMAA.wsisk@cisco.com>

possibly:
CSCea26503 7905 phone failed to get IP Address from DHCP reply packet
Fixed in the latest version (available whenever we have next
software release). 79XX will now take DHCP offer packets from
native VLAN if native VLAN is the same as aux VLAN.

Pls span the switchport the phone is using and get a sniffertrace of phone
reboot to confirm dhcpoffer is coming back.

On your router you might run 'deb dhcp detial' if not too busy to see
if DHCP discover is making it to the router.

/Wes

> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> Sent: Tuesday, September 02, 2003 6:03 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] 7905 and DHCP?
>
>
> Anyone else having problems with 7905's getting DHCP from a
> router running DHCP???  The 7940sand 60 don't seem to have any
> problems.  Will be opening a tac case.
>
> Scott Voll
> Network Analyst, CCNA
> Willamette ESD
> Phone: (503) 385-4571
> scott.voll@wesd.org
> ?
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From Scott.Voll at wesd.org  Wed Sep  3 15:50:07 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Sep  3 17:50:12 2003
Subject: [cisco-voip] 7905 and DHCP?
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDA7@venus.wesd.org>

Wes--

Looks like the bug I'm up against. TAC told me to disable cdp.  When I did that, it put the phone on the data network with the DHCP coming from the windows server rather then the IP range from the 3640 on the voice vlan.  

When is this latest release going to be out???? I could use it now!!!

I did some sniffing and the phone(7905) is sending DHCP request and discoveries but the 3640 isn't getting anything from what I can tell. I see no dhcp offers. 

Tac then had me put the 7905s on a port just set for the voice vlan.  Still didn't work.  Any other ideas????  I was supposed to have this IP Phone Solution in place a week ago.

Thanks

Scott

PS.  The cat 3550's think the 7905s are HOST as far as cdp knows.  

-----Original Message-----
From: Wes Sisk [mailto:wsisk@cisco.com] 
Sent: Wednesday, September 03, 2003 12:11 PM
To: Voll, Scott; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?

possibly:
CSCea26503 7905 phone failed to get IP Address from DHCP reply packet
Fixed in the latest version (available whenever we have next
software release). 79XX will now take DHCP offer packets from
native VLAN if native VLAN is the same as aux VLAN.

Pls span the switchport the phone is using and get a sniffertrace of phone
reboot to confirm dhcpoffer is coming back.

On your router you might run 'deb dhcp detial' if not too busy to see
if DHCP discover is making it to the router.

/Wes

> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> Sent: Tuesday, September 02, 2003 6:03 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] 7905 and DHCP?
>
>
> Anyone else having problems with 7905's getting DHCP from a
> router running DHCP???  The 7940sand 60 don't seem to have any
> problems.  Will be opening a tac case.
>
> Scott Voll
> Network Analyst, CCNA
> Willamette ESD
> Phone: (503) 385-4571
> scott.voll@wesd.org
> ?
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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

From Scott.Voll at wesd.org  Wed Sep  3 16:23:40 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Sep  3 18:23:49 2003
Subject: [cisco-voip] 7905 and DHCP?
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D43E@venus.wesd.org>

I just tried that load and Its still isn't working?  Any other ideas?  I'm wandering about an IOS UPGrade for the cat 3550 to understand the cdp so it sets the right Vlan?!?  

---scott

-----Original Message-----
From: Wes Sisk [mailto:wsisk@cisco.com] 
Sent: Wednesday, September 03, 2003 3:00 PM
To: Voll, Scott
Subject: RE: [cisco-voip] 7905 and DHCP?

verified version is 1.1(030404)

there is a package bundle for the low end phones on CCO:
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser

that includes CP7905010101SCCP030818A which I think has the
fix.  Have you tried this load?

/Wes


> -----Original Message-----
> From: Voll, Scott [mailto:Scott.Voll@wesd.org]
> Sent: Wednesday, September 03, 2003 5:50 PM
> To: Wes Sisk; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] 7905 and DHCP?
>
>
> Wes--
>
> Looks like the bug I'm up against. TAC told me to disable cdp.
> When I did that, it put the phone on the data network with the
> DHCP coming from the windows server rather then the IP range from
> the 3640 on the voice vlan.
>
> When is this latest release going to be out???? I could use it now!!!
>
> I did some sniffing and the phone(7905) is sending DHCP request
> and discoveries but the 3640 isn't getting anything from what I
> can tell. I see no dhcp offers.
>
> Tac then had me put the 7905s on a port just set for the voice
> vlan.  Still didn't work.  Any other ideas????  I was supposed to
> have this IP Phone Solution in place a week ago.
>
> Thanks
>
> Scott
>
> PS.  The cat 3550's think the 7905s are HOST as far as cdp knows.
>
> -----Original Message-----
> From: Wes Sisk [mailto:wsisk@cisco.com]
> Sent: Wednesday, September 03, 2003 12:11 PM
> To: Voll, Scott; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] 7905 and DHCP?
>
> possibly:
> CSCea26503 7905 phone failed to get IP Address from DHCP reply packet
> Fixed in the latest version (available whenever we have next
> software release). 79XX will now take DHCP offer packets from
> native VLAN if native VLAN is the same as aux VLAN.
>
> Pls span the switchport the phone is using and get a sniffertrace of phone
> reboot to confirm dhcpoffer is coming back.
>
> On your router you might run 'deb dhcp detial' if not too busy to see
> if DHCP discover is making it to the router.
>
> /Wes
>
> > -----Original Message-----
> > From: cisco-voip-bounces@puck.nether.net
> > [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> > Sent: Tuesday, September 02, 2003 6:03 PM
> > To: cisco-voip@puck.nether.net
> > Subject: [cisco-voip] 7905 and DHCP?
> >
> >
> > Anyone else having problems with 7905's getting DHCP from a
> > router running DHCP???  The 7940sand 60 don't seem to have any
> > problems.  Will be opening a tac case.
> >
> > Scott Voll
> > Network Analyst, CCNA
> > Willamette ESD
> > Phone: (503) 385-4571
> > scott.voll@wesd.org
> > ?
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Thu Sep  4 08:52:08 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Sep  4 10:52:19 2003
Subject: [cisco-voip] 7905 and DHCP?
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D440@venus.wesd.org>

What IOS are you using?

Thanks

Scott

-----Original Message-----
From: Anderson, Ian [mailto:i.anderson@lancaster.ac.uk] 
Sent: Thursday, September 04, 2003 12:54 AM
To: Voll, Scott
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?


I'm running 7905s with build CP79050101SCCP030404A these are connected through 3550-PWR-SMIs to a 3550-12G-EMI which is serving as the router and DHCP.
I can provide IOS versions of the two 3550s if required.

No problems with DHCP here.

Ian

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll@wesd.org]
Sent: 03 September 2003 23:24
To: Wes Sisk
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?


I just tried that load and Its still isn't working?  Any other ideas?  I'm wandering about an IOS UPGrade for the cat 3550 to understand the cdp so it sets the right Vlan?!?  

---scott

-----Original Message-----
From: Wes Sisk [mailto:wsisk@cisco.com] 
Sent: Wednesday, September 03, 2003 3:00 PM
To: Voll, Scott
Subject: RE: [cisco-voip] 7905 and DHCP?

verified version is 1.1(030404)

there is a package bundle for the low end phones on CCO:
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser

that includes CP7905010101SCCP030818A which I think has the
fix.  Have you tried this load?

/Wes


> -----Original Message-----
> From: Voll, Scott [mailto:Scott.Voll@wesd.org]
> Sent: Wednesday, September 03, 2003 5:50 PM
> To: Wes Sisk; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] 7905 and DHCP?
>
>
> Wes--
>
> Looks like the bug I'm up against. TAC told me to disable cdp.
> When I did that, it put the phone on the data network with the
> DHCP coming from the windows server rather then the IP range from
> the 3640 on the voice vlan.
>
> When is this latest release going to be out???? I could use it now!!!
>
> I did some sniffing and the phone(7905) is sending DHCP request
> and discoveries but the 3640 isn't getting anything from what I
> can tell. I see no dhcp offers.
>
> Tac then had me put the 7905s on a port just set for the voice
> vlan.  Still didn't work.  Any other ideas????  I was supposed to
> have this IP Phone Solution in place a week ago.
>
> Thanks
>
> Scott
>
> PS.  The cat 3550's think the 7905s are HOST as far as cdp knows.
>
> -----Original Message-----
> From: Wes Sisk [mailto:wsisk@cisco.com]
> Sent: Wednesday, September 03, 2003 12:11 PM
> To: Voll, Scott; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] 7905 and DHCP?
>
> possibly:
> CSCea26503 7905 phone failed to get IP Address from DHCP reply packet
> Fixed in the latest version (available whenever we have next
> software release). 79XX will now take DHCP offer packets from
> native VLAN if native VLAN is the same as aux VLAN.
>
> Pls span the switchport the phone is using and get a sniffertrace of phone
> reboot to confirm dhcpoffer is coming back.
>
> On your router you might run 'deb dhcp detial' if not too busy to see
> if DHCP discover is making it to the router.
>
> /Wes
>
> > -----Original Message-----
> > From: cisco-voip-bounces@puck.nether.net
> > [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> > Sent: Tuesday, September 02, 2003 6:03 PM
> > To: cisco-voip@puck.nether.net
> > Subject: [cisco-voip] 7905 and DHCP?
> >
> >
> > Anyone else having problems with 7905's getting DHCP from a
> > router running DHCP???  The 7940sand 60 don't seem to have any
> > problems.  Will be opening a tac case.
> >
> > Scott Voll
> > Network Analyst, CCNA
> > Willamette ESD
> > Phone: (503) 385-4571
> > scott.voll@wesd.org
> > ?
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


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

From rsahafi at ipsmarx.com  Thu Sep  4 15:52:27 2003
From: rsahafi at ipsmarx.com (Ramin Sahafi)
Date: Thu Sep  4 14:52:58 2003
Subject: [cisco-voip] 7905 and DHCP?
In-Reply-To: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D440@venus.wesd.org>
Message-ID: <001f01c37315$b031d870$1006a8c0@tech>

Dear Scott,

I've tested this matter on 3 different 5350 with 3 different IOS.
12.2.13T3 , 12.2.15T5 , 12.3.1

Here is more information about my case.
Hope you can find a solution .

I have seen a problem that I still can not find any solution for it . It
is under 5350 ONLY. 
When we have two carrier and carrier number one could not terminate the
call by any reason and Carrier number two terminate the call, after the
call has been finished, I have two START/STOP signal from the router
side with the same Prefix of Carrier number 1. I am sure that call is
terminating on carrier number two, but the STOP signal shows that call
terminated at carrier number ONE. It seems that information for
START/STOP will repeated for carrier number two. Any one have any idea. 
you only need to active some debugs:

debug voip aaa 

debug radius accounting

debug aaa accounting

debug voice aaa 

debug voip ccapi inout

debug vtsp all

debug aaa authorization

debug vtsp all

and you can see the result in debug.

just for your info, I tested this matter on 3 different 5350 with 3
different IOS.
12.2.13T3 , 12.2.15T5 , 12.3.1

Cisco Internetwork Operating System Software
IOS (tm) 5350 Software (C5350-JS-M), Version 12.3(1), RELEASE SOFTWARE
(fc3)
Copyright (c) 1986-2003 by cisco Systems, Inc.
Compiled Thu 15-May-03 07:56 by dchih
Image text-base: 0x6000895C, data-base: 0x61900000

ROM: System Bootstrap, Version 12.2(1r)1, RELEASE SOFTWARE (fc1)
BOOTLDR: 5350 Software (C5350-BOOT-M), Version 12.2(2)XB2, EARLY
DEPLOYMENT RELE
ASE SOFTWARE (fc1)

IITC_GW uptime is 5 days, 18 hours, 4 minutes
System returned to ROM by reload at 18:53:53 UTC Fri Feb 4 2000
System image file is "flash:c5350-js-mz.123-1.bin"

cisco AS5350 (R7K) processor (revision T) with 131072K/65536K bytes of
memory.
Processor board ID JAE0711084S
R7000 CPU at 250Mhz, Implementation 39, Rev 1.0, 256KB L2, 2048KB L3
Cache
Last reset from IOS reload
Bridging software.
X.25 software, Version 3.0.0.
SuperLAT software (copyright 1990 by Meridian Technology Corp).
TN3270 Emulation software.
Primary Rate ISDN software, Version 1.1.
Manufacture Cookie Info:
EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x32,
Board Hardware Version 3.34, Item Number 800-5171-02,
Board Revision C0, Serial Number JAE0711084S,
PLD/ISP Version 2.2, Manufacture Date 11-Mar-2003.
Processor 0x14, MAC Address 0x0BFDA28C6
Backplane HW Revision 1.0, Flash Type 5V
2 FastEthernet/IEEE 802.3 interface(s)
34 Serial network interface(s)
108 terminal line(s)
4 Channelized T1/PRI port(s)
512K bytes of non-volatile configuration memory.
32768K bytes of processor board System flash (Read/Write)
8192K bytes of processor board Boot flash (Read/Write)

Configuration register is 0x2102



in below example, call can not terminate at dial-peer 111, and will go
to dial-peer 222 and will terminate. but the STOP signal shows the
destination number as 11101144...... instead of 22201144.... 

even the h323-remote-address=198.198.104.1 is correct , I mean it shows
the second carrier ip address, but the destination number (
Called-Station-Id ) is not correct. 

I have tested this thing, on 3 different 5350 and different IOS . Result
is the same. But the same config undser 5300 working well !!! 
any one can help me? 

here is the relative config, 




aaa authentication login h323 group radius 
aaa authentication login telnet local 
aaa authentication login voip group radius 
aaa authorization exec h323 group radius 
aaa authorization exec telnet if-authenticated 
aaa accounting send stop-record authentication failure 
aaa accounting exec default start-stop group radius 
aaa accounting connection default start-stop group radius 
aaa accounting connection h323 start-stop group radius 
aaa accounting connection voip start-stop group radius 


gw-accounting aaa 
acct-template callhistory-detail 


radius-server host 192.168.1.20 auth-port 1845 acct-port 1846 
radius-server retransmit 1 
radius-server key 5555 
radius-server authorization permit missing Service-Type 
radius-server vsa send accounting 
radius-server vsa send authentication 


translation-rule 111 
Rule 0 ^01144 11144 

dial-peer voice 111 voip 
preference 1 
destination-pattern 01144T 
translate-outgoing called 111 
session target ipv4:200.200.102.2 
dtmf-relay h245-signal h245-alphanumeric 



translation-rule 222 
Rule 0 ^01144 22244 

dial-peer voice 222 voip 
preference 2 
destination-pattern 01144T 
translate-outgoing called 222 
session target ipv4:198.198.104.1 
dtmf-relay h245-signal h245-alphanumeric












As you can see in below, all stages working fine and call can go
through,  but the problem is not shown here. In Debug Radius accounting,


you can see that wrong destination number will return by Router .


I have dialed: 011914428269323

As you can see, the first START was not successful and was stopped
correctly,

"peer-address=7230914428269323"


But in Second START, you can see the same destination number that should
not be same as the first one. it should change based on the second
dial-peer , interesting part is that, router report :

"gw-final-xlated-cdn=ton:2,npi:1,#:34258914428269323"
"peer-address=34258914428269323"

that means the final  number  is correct, but again in STOP signal we
will receive the wrong info from Router.



**************** First Start *******************

*Feb  8 20:52:44.163: RADIUS:  Acct-Authentic      [45]  6   RADIUS
[1]
*Feb  8 20:52:44.163: RADIUS:  User-Name           [1]   12
"4166657510"
*Feb  8 20:52:44.163: RADIUS:  Acct-Status-Type    [40]  6   Start
[1]
*Feb  8 20:52:44.163: RADIUS:  Calling-Station-Id  [31]  12
"4166657510"
*Feb  8 20:52:44.163: RADIUS:  Called-Station-Id   [30]  18
"7230914428269323"
*Feb  8 20:52:44.163: RADIUS:  Service-Type        [6]   6   Login
[1]

"peer-address=7230914428269323"


**************** First Stop *******************

*Feb  8 20:52:59.171: RADIUS:  User-Name           [1]   12
"4166657510"
*Feb  8 20:52:59.171: RADIUS:  Acct-Status-Type    [40]  6   Stop
[2]
*Feb  8 20:52:59.171: RADIUS:  Calling-Station-Id  [31]  12
"4166657510"
*Feb  8 20:52:59.171: RADIUS:  Called-Station-Id   [30]  18
"7230914428269323"
*Feb  8 20:52:59.171: RADIUS:  Service-Type        [6]   6   Login
[1]



**************** Second Start *******************

*Feb  8 20:52:59.175: RADIUS:  User-Name           [1]   12
"4166657510"
*Feb  8 20:52:59.175: RADIUS:  Acct-Status-Type    [40]  6   Start
[1]
*Feb  8 20:52:59.175: RADIUS:  Calling-Station-Id  [31]  12
"4166657510"
*Feb  8 20:52:59.175: RADIUS:  Called-Station-Id   [30]  18
"7230914428269323"


"gw-final-xlated-cdn=ton:2,npi:1,#:34258914428269323"
"peer-address=34258914428269323"



**************** Seconf Stop *******************

*Feb  8 20:53:49.747: RADIUS:  Acct-Status-Type    [40]  6   Stop
[2]
*Feb  8 20:53:49.747: RADIUS:  Calling-Station-Id  [31]  12
"4166657510"
*Feb  8 20:53:49.747: RADIUS:  Called-Station-Id   [30]  18
"7230914428269323"
*Feb  8 20:53:49.747: RADIUS:  Service-Type        [6]   6   Login
[1]







Here is the dial-peer & result of "" debug translation detail ""

translation-rule 7230011
 Rule 0 ^0111.% 72301
 Rule 1 ^0112.% 72302
 Rule 2 ^0113.% 72303
 Rule 3 ^0114.% 72304
 Rule 4 ^0115.% 72305
 Rule 5 ^0116.% 72306
 Rule 6 ^0117.% 72307
 Rule 7 ^0118.% 72308
 Rule 8 ^0119.% 72309
!

translation-rule 34258
 Rule 0 ^0111.% 342581
 Rule 1 ^0112.% 342582
 Rule 2 ^0113.% 342583
 Rule 3 ^0114.% 342584
 Rule 4 ^0115.% 342585
 Rule 5 ^0116.% 342586
 Rule 6 ^0117.% 342587
 Rule 7 ^0118.% 342588
 Rule 8 ^0119.% 342589
!

dial-peer voice 7230 voip
 preference 1
 destination-pattern 01191T
 translate-outgoing called 7230011
 session target ipv4:198.198.104.1 
 dtmf-relay h245-signal h245-alphanumeric
!

dial-peer voice 34258 voip
 preference 2
 destination-pattern 01191T
 translate-outgoing called 34258
 session target ipv4:200.200.102.2 
 dtmf-relay h245-signal h245-alphanumeric
!



*Feb 10 13:10:37.609: xrule_checking
*Feb 10 13:10:37.609: xrule_checking calling 4166657510, called
011914428269323
*Feb 10 13:10:37.609: xrule_checking peer_tag 723001191, direction 2,
protocol 0
*Feb 10 13:10:37.609: xrule_translation
*Feb 10 13:10:37.609: xrule_translation callednumber 011914428269323,
strlen 15
*Feb 10 13:10:37.609: xrule_translation callednumber 011914428269323
xruleCalledTag=7230011
*Feb 10 13:10:37.609: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.609: xrule_translation Xrule index 0, Numpertype 0x9
*Feb 10 13:10:37.609: dpMatchString, target_number 011914428269323,
match_number ^0111.%
*Feb 10 13:10:37.609: dpMatchString match_tmp 0111, match_len 4
*Feb 10 13:10:37.609: dpMatchString beginning_replace 1, match_tmp
0111,target 011914428269323
*Feb 10 13:10:37.609: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.609: xrule_translation Xrule index 1, Numpertype 0x9
*Feb 10 13:10:37.609: dpMatchString, target_number 011914428269323,
match_number ^0112.%
*Feb 10 13:10:37.609: dpMatchString match_tmp 0112, match_len 4
*Feb 10 13:10:37.609: dpMatchString beginning_replace 1, match_tmp
0112,target 011914428269323
*Feb 10 13:10:37.609: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.609: xrule_translation Xrule index 2, Numpertype 0x9
*Feb 10 13:10:37.609: dpMatchString, target_number 011914428269323,
match_number ^0113.%
*Feb 10 13:10:37.609: dpMatchString match_tmp 0113, match_len 4
*Feb 10 13:10:37.609: dpMatchString beginning_replace 1, match_tmp
0113,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 3, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0114.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0114, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0114,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 4, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0115.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0115, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0115,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 5, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0116.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0116, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0116,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 6, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0117.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0117, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0117,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 7, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0118.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0118, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0118,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 8, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0119.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0119, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0119,target 011914428269323
*Feb 10 13:10:37.613: dpMatchString 0. target 011914428269323,match_tmp
0119,match_len 4
*Feb 10 13:10:37.613: replace_string
*Feb 10 13:10:37.613: replace_string match ^0119.%, replace 72309
*Feb 10 13:10:37.613: translation_format replace_rule ^0119.%,
strip_proceeding 0
*Feb 10 13:10:37.613: replace_string match_tmp ^0119.%, strip_proceeding
0
*Feb 10 13:10:37.613: replace_string match_tmp 0119
*Feb 10 13:10:37.613: replace_string direction 2, callparty  2
*Feb 10 13:10:37.613: replace_string direction 2, callparty 2, target
011914428269323
*Feb 10 13:10:37.613: replace_string match_tmp 0119,replace 72309 
*Feb 10 13:10:37.613:
replace_string0.replace10,target11914428269323,current11914428269323,mat
ch_tmp119
*Feb 10 13:10:37.613: replace_string0.1 compare_len 1,match_len 4
*Feb 10 13:10:37.613:
replace_string0.replace101,target1914428269323,current1914428269323,matc
h_tmp19
*Feb 10 13:10:37.613: replace_string0.1 compare_len 2,match_len 4
*Feb 10 13:10:37.613:
replace_string0.replace1011,target914428269323,current914428269323,match
_tmp9
*Feb 10 13:10:37.613: replace_string0.1 compare_len 3,match_len 4
*Feb 10 13:10:37.613:
replace_string0.replace10119,target14428269323,current14428269323,match_
tmp
*Feb 10 13:10:37.613: replace_string0.1 compare_len 4,match_len 4
*Feb 10 13:10:37.613: replace_string 3. replace1 0119, compare_len 4
*Feb 10 13:10:37.613: replace_string 4. replace1 7119,compare_len
3,replace 2309
*Feb 10 13:10:37.613: replace_string 4. replace1 7219,compare_len
2,replace 309
*Feb 10 13:10:37.613: replace_string 4. replace1 7239,compare_len
1,replace 09
*Feb 10 13:10:37.613: replace_string 4. replace1 7230,compare_len
0,replace 9
*Feb 10 13:10:37.613: replace_string 4. replace1 72309,compare_len
-1,replace 
*Feb 10 13:10:37.613: replace_string 5. replace1 72309, compare_len
-1,match_len 4
*Feb 10 13:10:37.613: replace_string 6. replace1 72309,compare_len
-1,current 14428269323
*Feb 10 13:10:37.613: replace_string 7. replace1 723091
*Feb 10 13:10:37.617: replace_string 7. replace1 7230914
*Feb 10 13:10:37.617: replace_string 7. replace1 72309144
*Feb 10 13:10:37.617: replace_string 7. replace1 723091442
*Feb 10 13:10:37.617: replace_string 7. replace1 7230914428
*Feb 10 13:10:37.617: replace_string 7. replace1 72309144282
*Feb 10 13:10:37.617: replace_string 7. replace1 723091442826
*Feb 10 13:10:37.617: replace_string 7. replace1 7230914428269
*Feb 10 13:10:37.617: replace_string 7. replace1 72309144282693
*Feb 10 13:10:37.617: replace_string 7. replace1 723091442826932
*Feb 10 13:10:37.617: replace_string 7. replace1 7230914428269323
*Feb 10 13:10:37.617: replace_string buffer 7230914428269323
*Feb 10 13:10:37.617: xrule_translation index 8,xrule_number
7230914428269323, callparty 2
*Feb 10 13:10:37.617: xrule_translation Return rc = 0
*Feb 10 13:10:37.617: xrule_checking Return rc = 0
*Feb 10 13:10:52.617: xrule_checking
*Feb 10 13:10:52.617: xrule_checking calling 4166657510, called
011914428269323
*Feb 10 13:10:52.617: xrule_checking peer_tag 342581191, direction 2,
protocol 0
*Feb 10 13:10:52.617: xrule_translation
*Feb 10 13:10:52.617: xrule_translation callednumber 011914428269323,
strlen 15
*Feb 10 13:10:52.617: xrule_translation callednumber 011914428269323
xruleCalledTag=34258
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 0, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0111.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0111, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0111,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 1, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0112.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0112, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0112,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 2, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0113.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0113, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0113,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 3, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0114.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0114, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0114,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 4, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0115.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0115, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0115,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 5, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0116.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0116, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0116,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 6, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0117.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0117, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0117,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 7, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0118.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0118, match_len 4
*Feb 10 13:10:52.621: dpMatchString beginning_replace 1, match_tmp
0118,target 011914428269323
*Feb 10 13:10:52.621: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.621: xrule_translation Xrule index 8, Numpertype 0x9
*Feb 10 13:10:52.621: dpMatchString, target_number 011914428269323,
match_number ^0119.%
*Feb 10 13:10:52.621: dpMatchString match_tmp 0119, match_len 4
*Feb 10 13:10:52.621: dpMatchString beginning_replace 1, match_tmp
0119,target 011914428269323
*Feb 10 13:10:52.621: dpMatchString 0. target 011914428269323,match_tmp
0119,match_len 4
*Feb 10 13:10:52.621: replace_string
*Feb 10 13:10:52.621: replace_string match ^0119.%, replace 342589
*Feb 10 13:10:52.621: translation_format replace_rule ^0119.%,
strip_proceeding 0
*Feb 10 13:10:52.621: replace_string match_tmp ^0119.%, strip_proceeding
0
*Feb 10 13:10:52.621: replace_string match_tmp 0119
*Feb 10 13:10:52.621: replace_string direction 2, callparty  2
*Feb 10 13:10:52.621: replace_string direction 2, callparty 2, target
011914428269323
*Feb 10 13:10:52.621: replace_string match_tmp 0119,replace 342589 
*Feb 10 13:10:52.621:
replace_string0.replace10,target11914428269323,current11914428269323,mat
ch_tmp119
*Feb 10 13:10:52.621: replace_string0.1 compare_len 1,match_len 4
*Feb 10 13:10:52.621:
replace_string0.replace101,target1914428269323,current1914428269323,matc
h_tmp19
*Feb 10 13:10:52.621: replace_string0.1 compare_len 2,match_len 4
*Feb 10 13:10:52.621:
replace_string0.replace1011,target914428269323,current914428269323,match
_tmp9
*Feb 10 13:10:52.621: replace_string0.1 compare_len 3,match_len 4
*Feb 10 13:10:52.621:
replace_string0.replace10119,target14428269323,current14428269323,match_
tmp
*Feb 10 13:10:52.621: replace_string0.1 compare_len 4,match_len 4
*Feb 10 13:10:52.621: replace_string 3. replace1 0119, compare_len 4
*Feb 10 13:10:52.621: replace_string 4. replace1 3119,compare_len
3,replace 42589
*Feb 10 13:10:52.621: replace_string 4. replace1 3419,compare_len
2,replace 2589
*Feb 10 13:10:52.621: replace_string 4. replace1 3429,compare_len
1,replace 589
*Feb 10 13:10:52.621: replace_string 4. replace1 3425,compare_len
0,replace 89
*Feb 10 13:10:52.621: replace_string 4. replace1 34258,compare_len
-1,replace 9
*Feb 10 13:10:52.621: replace_string 4. replace1 342589,compare_len
-2,replace 
*Feb 10 13:10:52.621: replace_string 5. replace1 342589, compare_len
-2,match_len 4
*Feb 10 13:10:52.621: replace_string 6. replace1 342589,compare_len
-2,current 14428269323
*Feb 10 13:10:52.621: replace_string 7. replace1 3425891
*Feb 10 13:10:52.621: replace_string 7. replace1 34258914
*Feb 10 13:10:52.621: replace_string 7. replace1 342589144
*Feb 10 13:10:52.621: replace_string 7. replace1 3425891442
*Feb 10 13:10:52.621: replace_string 7. replace1 34258914428
*Feb 10 13:10:52.621: replace_string 7. replace1 342589144282
*Feb 10 13:10:52.621: replace_string 7. replace1 3425891442826
*Feb 10 13:10:52.621: replace_string 7. replace1 34258914428269
*Feb 10 13:10:52.621: replace_string 7. replace1 342589144282693
*Feb 10 13:10:52.621: replace_string 7. replace1 3425891442826932
*Feb 10 13:10:52.621: replace_string 7. replace1 34258914428269323
*Feb 10 13:10:52.621: replace_string buffer 34258914428269323
*Feb 10 13:10:52.621: xrule_translation index 8,xrule_number
34258914428269323, callparty 2
*Feb 10 13:10:52.621: xrule_translation Return rc = 0
*Feb 10 13:10:52.621: xrule_checking Return rc = 0










-----Original Message-----
From: cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, September 04, 2003 10:52 AM
To: Anderson, Ian
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?

What IOS are you using?

Thanks

Scott

-----Original Message-----
From: Anderson, Ian [mailto:i.anderson@lancaster.ac.uk] 
Sent: Thursday, September 04, 2003 12:54 AM
To: Voll, Scott
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?


I'm running 7905s with build CP79050101SCCP030404A these are connected
through 3550-PWR-SMIs to a 3550-12G-EMI which is serving as the router
and DHCP.
I can provide IOS versions of the two 3550s if required.

No problems with DHCP here.

Ian

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll@wesd.org]
Sent: 03 September 2003 23:24
To: Wes Sisk
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?


I just tried that load and Its still isn't working?  Any other ideas?
I'm wandering about an IOS UPGrade for the cat 3550 to understand the
cdp so it sets the right Vlan?!?  

---scott

-----Original Message-----
From: Wes Sisk [mailto:wsisk@cisco.com] 
Sent: Wednesday, September 03, 2003 3:00 PM
To: Voll, Scott
Subject: RE: [cisco-voip] 7905 and DHCP?

verified version is 1.1(030404)

there is a package bundle for the low end phones on CCO:
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser

that includes CP7905010101SCCP030818A which I think has the
fix.  Have you tried this load?

/Wes


> -----Original Message-----
> From: Voll, Scott [mailto:Scott.Voll@wesd.org]
> Sent: Wednesday, September 03, 2003 5:50 PM
> To: Wes Sisk; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] 7905 and DHCP?
>
>
> Wes--
>
> Looks like the bug I'm up against. TAC told me to disable cdp.
> When I did that, it put the phone on the data network with the
> DHCP coming from the windows server rather then the IP range from
> the 3640 on the voice vlan.
>
> When is this latest release going to be out???? I could use it now!!!
>
> I did some sniffing and the phone(7905) is sending DHCP request
> and discoveries but the 3640 isn't getting anything from what I
> can tell. I see no dhcp offers.
>
> Tac then had me put the 7905s on a port just set for the voice
> vlan.  Still didn't work.  Any other ideas????  I was supposed to
> have this IP Phone Solution in place a week ago.
>
> Thanks
>
> Scott
>
> PS.  The cat 3550's think the 7905s are HOST as far as cdp knows.
>
> -----Original Message-----
> From: Wes Sisk [mailto:wsisk@cisco.com]
> Sent: Wednesday, September 03, 2003 12:11 PM
> To: Voll, Scott; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] 7905 and DHCP?
>
> possibly:
> CSCea26503 7905 phone failed to get IP Address from DHCP reply packet
> Fixed in the latest version (available whenever we have next
> software release). 79XX will now take DHCP offer packets from
> native VLAN if native VLAN is the same as aux VLAN.
>
> Pls span the switchport the phone is using and get a sniffertrace of
phone
> reboot to confirm dhcpoffer is coming back.
>
> On your router you might run 'deb dhcp detial' if not too busy to see
> if DHCP discover is making it to the router.
>
> /Wes
>
> > -----Original Message-----
> > From: cisco-voip-bounces@puck.nether.net
> > [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> > Sent: Tuesday, September 02, 2003 6:03 PM
> > To: cisco-voip@puck.nether.net
> > Subject: [cisco-voip] 7905 and DHCP?
> >
> >
> > Anyone else having problems with 7905's getting DHCP from a
> > router running DHCP???  The 7940sand 60 don't seem to have any
> > problems.  Will be opening a tac case.
> >
> > Scott Voll
> > Network Analyst, CCNA
> > Willamette ESD
> > Phone: (503) 385-4571
> > scott.voll@wesd.org
> > ?
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


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

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



From WCarter at sentinel.com  Wed Sep 10 16:09:44 2003
From: WCarter at sentinel.com (Carter, Bill)
Date: Wed Sep 10 16:09:49 2003
Subject: [cisco-voip] 7206VXR Random Echo During Business hours only
Message-ID: <F4059FD8E8F9D141BE8C48C0DCF1FA1579425A@rodan.sentinel.com>

I have a 7206 with a VXC-2TE1+ and a Channelized DS-3.  6 T-1's from the DS-3 connect to another location for Data IP only.  2 PRIs connect to the voice card.  

I am getting random reports of Echo throughout the day.  Some times on local or LD or Cell phone calls.  At night I go through the echo troubleshooting procedures and all of my ERLs look great.  I never see or hear echo at night.  On some calls it is echo for the duration of the call, and some calls the echo comes and goes during the call.

The 6 T-1s are load balanced via EIGRP to the main site.  Again, there is no VoIP going across these T's.  I have noticed that the T-1's are getting hit pretty hard throughout the day. 

Could the echo be caused by the gateway itself when data traffic is hitting it hard.  I am thinking this slows the gateways ability to process VoIP/PSTN traffic.  Gateway is running H.323.  Someone has suggested MGCP could help my echo but it is not available.

Any thoughts??

Bill Carter


From eugen at romtelecom.net  Thu Sep 11 11:44:05 2003
From: eugen at romtelecom.net (Eugen Pantica)
Date: Thu Sep 11 03:42:58 2003
Subject: [cisco-voip] Traffic VOIP
Message-ID: <3F6027C5.9030303@romtelecom.net>

Is someone interested for VOIP traffic terminated in Romania (Europe) or 
do you know any
company that would want to terminate VOIP traffic in Romania?

Eugen

From Delikato at aol.com  Thu Sep 11 15:31:25 2003
From: Delikato at aol.com (Delikato@aol.com)
Date: Thu Sep 11 14:31:48 2003
Subject: [cisco-voip] Inquiry
Message-ID: <a7.35b24379.2c92197d@aol.com>

Gentlemen :


Our group is considering implementing a Voip system to reduce our hotel 
telephone costs in the Bahamas. Prior to committing ourselves to any specific 
business venture to do this for us, we prefer to educate ourselves what exactly is 
required .

Is there a web site we can find this kind of information ?

Delikato 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030911/4c2bcc64/attachment.htm
From Scott.Voll at wesd.org  Thu Sep 11 12:41:26 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Sep 11 14:41:37 2003
Subject: [cisco-voip] Inquiry
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D465@venus.wesd.org>

Do you have a cisco cco login?

 

-----Original Message-----
From: Delikato@aol.com [mailto:Delikato@aol.com] 
Sent: Thursday, September 11, 2003 11:31 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Inquiry

 

Gentlemen :


Our group is considering implementing a Voip system to reduce our hotel
telephone costs in the Bahamas. Prior to committing ourselves to any
specific business venture to do this for us, we prefer to educate
ourselves what exactly is required .

Is there a web site we can find this kind of information ?

Delikato 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030911/38d4588c/attachment.htm
From Scott.Voll at wesd.org  Fri Sep 12 10:46:57 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Sep 12 12:47:05 2003
Subject: [cisco-voip] How do I get a persons name on the phone
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDB2@venus.wesd.org>

OK another newby question.

I have my call manager 3.3.3sr1 up and running, routing and happy.  So
now I'm working on the nice little things.  

I do have corporate directories working and I know how to associate a
user to a phone, but I don't know how to get the persons name on the
phone display.  Any help would be great..... I'm having a hard time
finding any info on Cisco CCO.



Scott Voll
Network Analyst, CCNA 
Willamette ESD


From Scovin at Symmetricom.com  Fri Sep 12 10:55:43 2003
From: Scovin at Symmetricom.com (Steve Covin)
Date: Fri Sep 12 12:56:01 2003
Subject: [cisco-voip] Fax over Cisco VoIP
Message-ID: <27BBC8307A2C994092D90693E15CDA94E639DD@irvexchange.datum.com>

I'm very new to asking questions on the list.
Have there been any issues with the Fax and VoIP with the Call manager 3.xx?
thanks

Steve Covin
Symmetricom Austin
Office: 512-721-4527 
Cell :   512-694-0981


-----Original Message-----
From: cisco-voip-request@puck.nether.net
[mailto:cisco-voip-request@puck.nether.net]
Sent: Friday, September 05, 2003 11:00 AM
To: cisco-voip@puck.nether.net
Subject: cisco-voip Digest, Vol 7, Issue 4


Send cisco-voip mailing list submissions to
	cisco-voip@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@puck.nether.net

You can reach the person managing the list at
	cisco-voip-owner@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: 7905 and DHCP? (Ramin Sahafi)


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

Message: 1
Date: Thu, 4 Sep 2003 14:52:27 -0400
From: "Ramin Sahafi" <rsahafi@ipsmarx.com>
Subject: RE: [cisco-voip] 7905 and DHCP?
To: "'Voll, Scott'" <Scott.Voll@wesd.org>,	"'Anderson, Ian'"
	<i.anderson@lancaster.ac.uk>
Cc: cisco-voip@puck.nether.net
Message-ID: <001f01c37315$b031d870$1006a8c0@tech>
Content-Type: text/plain;	charset="iso-8859-1"

Dear Scott,

I've tested this matter on 3 different 5350 with 3 different IOS.
12.2.13T3 , 12.2.15T5 , 12.3.1

Here is more information about my case.
Hope you can find a solution .

I have seen a problem that I still can not find any solution for it . It
is under 5350 ONLY. 
When we have two carrier and carrier number one could not terminate the
call by any reason and Carrier number two terminate the call, after the
call has been finished, I have two START/STOP signal from the router
side with the same Prefix of Carrier number 1. I am sure that call is
terminating on carrier number two, but the STOP signal shows that call
terminated at carrier number ONE. It seems that information for
START/STOP will repeated for carrier number two. Any one have any idea. 
you only need to active some debugs:

debug voip aaa 

debug radius accounting

debug aaa accounting

debug voice aaa 

debug voip ccapi inout

debug vtsp all

debug aaa authorization

debug vtsp all

and you can see the result in debug.

just for your info, I tested this matter on 3 different 5350 with 3
different IOS.
12.2.13T3 , 12.2.15T5 , 12.3.1

Cisco Internetwork Operating System Software
IOS (tm) 5350 Software (C5350-JS-M), Version 12.3(1), RELEASE SOFTWARE
(fc3)
Copyright (c) 1986-2003 by cisco Systems, Inc.
Compiled Thu 15-May-03 07:56 by dchih
Image text-base: 0x6000895C, data-base: 0x61900000

ROM: System Bootstrap, Version 12.2(1r)1, RELEASE SOFTWARE (fc1)
BOOTLDR: 5350 Software (C5350-BOOT-M), Version 12.2(2)XB2, EARLY
DEPLOYMENT RELE
ASE SOFTWARE (fc1)

IITC_GW uptime is 5 days, 18 hours, 4 minutes
System returned to ROM by reload at 18:53:53 UTC Fri Feb 4 2000
System image file is "flash:c5350-js-mz.123-1.bin"

cisco AS5350 (R7K) processor (revision T) with 131072K/65536K bytes of
memory.
Processor board ID JAE0711084S
R7000 CPU at 250Mhz, Implementation 39, Rev 1.0, 256KB L2, 2048KB L3
Cache
Last reset from IOS reload
Bridging software.
X.25 software, Version 3.0.0.
SuperLAT software (copyright 1990 by Meridian Technology Corp).
TN3270 Emulation software.
Primary Rate ISDN software, Version 1.1.
Manufacture Cookie Info:
EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x32,
Board Hardware Version 3.34, Item Number 800-5171-02,
Board Revision C0, Serial Number JAE0711084S,
PLD/ISP Version 2.2, Manufacture Date 11-Mar-2003.
Processor 0x14, MAC Address 0x0BFDA28C6
Backplane HW Revision 1.0, Flash Type 5V
2 FastEthernet/IEEE 802.3 interface(s)
34 Serial network interface(s)
108 terminal line(s)
4 Channelized T1/PRI port(s)
512K bytes of non-volatile configuration memory.
32768K bytes of processor board System flash (Read/Write)
8192K bytes of processor board Boot flash (Read/Write)

Configuration register is 0x2102



in below example, call can not terminate at dial-peer 111, and will go
to dial-peer 222 and will terminate. but the STOP signal shows the
destination number as 11101144...... instead of 22201144.... 

even the h323-remote-address=198.198.104.1 is correct , I mean it shows
the second carrier ip address, but the destination number (
Called-Station-Id ) is not correct. 

I have tested this thing, on 3 different 5350 and different IOS . Result
is the same. But the same config undser 5300 working well !!! 
any one can help me? 

here is the relative config, 




aaa authentication login h323 group radius 
aaa authentication login telnet local 
aaa authentication login voip group radius 
aaa authorization exec h323 group radius 
aaa authorization exec telnet if-authenticated 
aaa accounting send stop-record authentication failure 
aaa accounting exec default start-stop group radius 
aaa accounting connection default start-stop group radius 
aaa accounting connection h323 start-stop group radius 
aaa accounting connection voip start-stop group radius 


gw-accounting aaa 
acct-template callhistory-detail 


radius-server host 192.168.1.20 auth-port 1845 acct-port 1846 
radius-server retransmit 1 
radius-server key 5555 
radius-server authorization permit missing Service-Type 
radius-server vsa send accounting 
radius-server vsa send authentication 


translation-rule 111 
Rule 0 ^01144 11144 

dial-peer voice 111 voip 
preference 1 
destination-pattern 01144T 
translate-outgoing called 111 
session target ipv4:200.200.102.2 
dtmf-relay h245-signal h245-alphanumeric 



translation-rule 222 
Rule 0 ^01144 22244 

dial-peer voice 222 voip 
preference 2 
destination-pattern 01144T 
translate-outgoing called 222 
session target ipv4:198.198.104.1 
dtmf-relay h245-signal h245-alphanumeric












As you can see in below, all stages working fine and call can go
through,  but the problem is not shown here. In Debug Radius accounting,


you can see that wrong destination number will return by Router .


I have dialed: 011914428269323

As you can see, the first START was not successful and was stopped
correctly,

"peer-address=7230914428269323"


But in Second START, you can see the same destination number that should
not be same as the first one. it should change based on the second
dial-peer , interesting part is that, router report :

"gw-final-xlated-cdn=ton:2,npi:1,#:34258914428269323"
"peer-address=34258914428269323"

that means the final  number  is correct, but again in STOP signal we
will receive the wrong info from Router.



**************** First Start *******************

*Feb  8 20:52:44.163: RADIUS:  Acct-Authentic      [45]  6   RADIUS
[1]
*Feb  8 20:52:44.163: RADIUS:  User-Name           [1]   12
"4166657510"
*Feb  8 20:52:44.163: RADIUS:  Acct-Status-Type    [40]  6   Start
[1]
*Feb  8 20:52:44.163: RADIUS:  Calling-Station-Id  [31]  12
"4166657510"
*Feb  8 20:52:44.163: RADIUS:  Called-Station-Id   [30]  18
"7230914428269323"
*Feb  8 20:52:44.163: RADIUS:  Service-Type        [6]   6   Login
[1]

"peer-address=7230914428269323"


**************** First Stop *******************

*Feb  8 20:52:59.171: RADIUS:  User-Name           [1]   12
"4166657510"
*Feb  8 20:52:59.171: RADIUS:  Acct-Status-Type    [40]  6   Stop
[2]
*Feb  8 20:52:59.171: RADIUS:  Calling-Station-Id  [31]  12
"4166657510"
*Feb  8 20:52:59.171: RADIUS:  Called-Station-Id   [30]  18
"7230914428269323"
*Feb  8 20:52:59.171: RADIUS:  Service-Type        [6]   6   Login
[1]



**************** Second Start *******************

*Feb  8 20:52:59.175: RADIUS:  User-Name           [1]   12
"4166657510"
*Feb  8 20:52:59.175: RADIUS:  Acct-Status-Type    [40]  6   Start
[1]
*Feb  8 20:52:59.175: RADIUS:  Calling-Station-Id  [31]  12
"4166657510"
*Feb  8 20:52:59.175: RADIUS:  Called-Station-Id   [30]  18
"7230914428269323"


"gw-final-xlated-cdn=ton:2,npi:1,#:34258914428269323"
"peer-address=34258914428269323"



**************** Seconf Stop *******************

*Feb  8 20:53:49.747: RADIUS:  Acct-Status-Type    [40]  6   Stop
[2]
*Feb  8 20:53:49.747: RADIUS:  Calling-Station-Id  [31]  12
"4166657510"
*Feb  8 20:53:49.747: RADIUS:  Called-Station-Id   [30]  18
"7230914428269323"
*Feb  8 20:53:49.747: RADIUS:  Service-Type        [6]   6   Login
[1]







Here is the dial-peer & result of "" debug translation detail ""

translation-rule 7230011
 Rule 0 ^0111.% 72301
 Rule 1 ^0112.% 72302
 Rule 2 ^0113.% 72303
 Rule 3 ^0114.% 72304
 Rule 4 ^0115.% 72305
 Rule 5 ^0116.% 72306
 Rule 6 ^0117.% 72307
 Rule 7 ^0118.% 72308
 Rule 8 ^0119.% 72309
!

translation-rule 34258
 Rule 0 ^0111.% 342581
 Rule 1 ^0112.% 342582
 Rule 2 ^0113.% 342583
 Rule 3 ^0114.% 342584
 Rule 4 ^0115.% 342585
 Rule 5 ^0116.% 342586
 Rule 6 ^0117.% 342587
 Rule 7 ^0118.% 342588
 Rule 8 ^0119.% 342589
!

dial-peer voice 7230 voip
 preference 1
 destination-pattern 01191T
 translate-outgoing called 7230011
 session target ipv4:198.198.104.1 
 dtmf-relay h245-signal h245-alphanumeric
!

dial-peer voice 34258 voip
 preference 2
 destination-pattern 01191T
 translate-outgoing called 34258
 session target ipv4:200.200.102.2 
 dtmf-relay h245-signal h245-alphanumeric
!



*Feb 10 13:10:37.609: xrule_checking
*Feb 10 13:10:37.609: xrule_checking calling 4166657510, called
011914428269323
*Feb 10 13:10:37.609: xrule_checking peer_tag 723001191, direction 2,
protocol 0
*Feb 10 13:10:37.609: xrule_translation
*Feb 10 13:10:37.609: xrule_translation callednumber 011914428269323,
strlen 15
*Feb 10 13:10:37.609: xrule_translation callednumber 011914428269323
xruleCalledTag=7230011
*Feb 10 13:10:37.609: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.609: xrule_translation Xrule index 0, Numpertype 0x9
*Feb 10 13:10:37.609: dpMatchString, target_number 011914428269323,
match_number ^0111.%
*Feb 10 13:10:37.609: dpMatchString match_tmp 0111, match_len 4
*Feb 10 13:10:37.609: dpMatchString beginning_replace 1, match_tmp
0111,target 011914428269323
*Feb 10 13:10:37.609: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.609: xrule_translation Xrule index 1, Numpertype 0x9
*Feb 10 13:10:37.609: dpMatchString, target_number 011914428269323,
match_number ^0112.%
*Feb 10 13:10:37.609: dpMatchString match_tmp 0112, match_len 4
*Feb 10 13:10:37.609: dpMatchString beginning_replace 1, match_tmp
0112,target 011914428269323
*Feb 10 13:10:37.609: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.609: xrule_translation Xrule index 2, Numpertype 0x9
*Feb 10 13:10:37.609: dpMatchString, target_number 011914428269323,
match_number ^0113.%
*Feb 10 13:10:37.609: dpMatchString match_tmp 0113, match_len 4
*Feb 10 13:10:37.609: dpMatchString beginning_replace 1, match_tmp
0113,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 3, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0114.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0114, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0114,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 4, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0115.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0115, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0115,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 5, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0116.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0116, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0116,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 6, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0117.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0117, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0117,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 7, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0118.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0118, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0118,target 011914428269323
*Feb 10 13:10:37.613: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:37.613: xrule_translation Xrule index 8, Numpertype 0x9
*Feb 10 13:10:37.613: dpMatchString, target_number 011914428269323,
match_number ^0119.%
*Feb 10 13:10:37.613: dpMatchString match_tmp 0119, match_len 4
*Feb 10 13:10:37.613: dpMatchString beginning_replace 1, match_tmp
0119,target 011914428269323
*Feb 10 13:10:37.613: dpMatchString 0. target 011914428269323,match_tmp
0119,match_len 4
*Feb 10 13:10:37.613: replace_string
*Feb 10 13:10:37.613: replace_string match ^0119.%, replace 72309
*Feb 10 13:10:37.613: translation_format replace_rule ^0119.%,
strip_proceeding 0
*Feb 10 13:10:37.613: replace_string match_tmp ^0119.%, strip_proceeding
0
*Feb 10 13:10:37.613: replace_string match_tmp 0119
*Feb 10 13:10:37.613: replace_string direction 2, callparty  2
*Feb 10 13:10:37.613: replace_string direction 2, callparty 2, target
011914428269323
*Feb 10 13:10:37.613: replace_string match_tmp 0119,replace 72309 
*Feb 10 13:10:37.613:
replace_string0.replace10,target11914428269323,current11914428269323,mat
ch_tmp119
*Feb 10 13:10:37.613: replace_string0.1 compare_len 1,match_len 4
*Feb 10 13:10:37.613:
replace_string0.replace101,target1914428269323,current1914428269323,matc
h_tmp19
*Feb 10 13:10:37.613: replace_string0.1 compare_len 2,match_len 4
*Feb 10 13:10:37.613:
replace_string0.replace1011,target914428269323,current914428269323,match
_tmp9
*Feb 10 13:10:37.613: replace_string0.1 compare_len 3,match_len 4
*Feb 10 13:10:37.613:
replace_string0.replace10119,target14428269323,current14428269323,match_
tmp
*Feb 10 13:10:37.613: replace_string0.1 compare_len 4,match_len 4
*Feb 10 13:10:37.613: replace_string 3. replace1 0119, compare_len 4
*Feb 10 13:10:37.613: replace_string 4. replace1 7119,compare_len
3,replace 2309
*Feb 10 13:10:37.613: replace_string 4. replace1 7219,compare_len
2,replace 309
*Feb 10 13:10:37.613: replace_string 4. replace1 7239,compare_len
1,replace 09
*Feb 10 13:10:37.613: replace_string 4. replace1 7230,compare_len
0,replace 9
*Feb 10 13:10:37.613: replace_string 4. replace1 72309,compare_len
-1,replace 
*Feb 10 13:10:37.613: replace_string 5. replace1 72309, compare_len
-1,match_len 4
*Feb 10 13:10:37.613: replace_string 6. replace1 72309,compare_len
-1,current 14428269323
*Feb 10 13:10:37.613: replace_string 7. replace1 723091
*Feb 10 13:10:37.617: replace_string 7. replace1 7230914
*Feb 10 13:10:37.617: replace_string 7. replace1 72309144
*Feb 10 13:10:37.617: replace_string 7. replace1 723091442
*Feb 10 13:10:37.617: replace_string 7. replace1 7230914428
*Feb 10 13:10:37.617: replace_string 7. replace1 72309144282
*Feb 10 13:10:37.617: replace_string 7. replace1 723091442826
*Feb 10 13:10:37.617: replace_string 7. replace1 7230914428269
*Feb 10 13:10:37.617: replace_string 7. replace1 72309144282693
*Feb 10 13:10:37.617: replace_string 7. replace1 723091442826932
*Feb 10 13:10:37.617: replace_string 7. replace1 7230914428269323
*Feb 10 13:10:37.617: replace_string buffer 7230914428269323
*Feb 10 13:10:37.617: xrule_translation index 8,xrule_number
7230914428269323, callparty 2
*Feb 10 13:10:37.617: xrule_translation Return rc = 0
*Feb 10 13:10:37.617: xrule_checking Return rc = 0
*Feb 10 13:10:52.617: xrule_checking
*Feb 10 13:10:52.617: xrule_checking calling 4166657510, called
011914428269323
*Feb 10 13:10:52.617: xrule_checking peer_tag 342581191, direction 2,
protocol 0
*Feb 10 13:10:52.617: xrule_translation
*Feb 10 13:10:52.617: xrule_translation callednumber 011914428269323,
strlen 15
*Feb 10 13:10:52.617: xrule_translation callednumber 011914428269323
xruleCalledTag=34258
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 0, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0111.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0111, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0111,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 1, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0112.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0112, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0112,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 2, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0113.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0113, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0113,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 3, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0114.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0114, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0114,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 4, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0115.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0115, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0115,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 5, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0116.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0116, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0116,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 6, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0117.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0117, match_len 4
*Feb 10 13:10:52.617: dpMatchString beginning_replace 1, match_tmp
0117,target 011914428269323
*Feb 10 13:10:52.617: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.617: xrule_translation Xrule index 7, Numpertype 0x9
*Feb 10 13:10:52.617: dpMatchString, target_number 011914428269323,
match_number ^0118.%
*Feb 10 13:10:52.617: dpMatchString match_tmp 0118, match_len 4
*Feb 10 13:10:52.621: dpMatchString beginning_replace 1, match_tmp
0118,target 011914428269323
*Feb 10 13:10:52.621: xrule_translation called Callparms Numpertype
0xA1, 
	                       match_type 0x2
*Feb 10 13:10:52.621: xrule_translation Xrule index 8, Numpertype 0x9
*Feb 10 13:10:52.621: dpMatchString, target_number 011914428269323,
match_number ^0119.%
*Feb 10 13:10:52.621: dpMatchString match_tmp 0119, match_len 4
*Feb 10 13:10:52.621: dpMatchString beginning_replace 1, match_tmp
0119,target 011914428269323
*Feb 10 13:10:52.621: dpMatchString 0. target 011914428269323,match_tmp
0119,match_len 4
*Feb 10 13:10:52.621: replace_string
*Feb 10 13:10:52.621: replace_string match ^0119.%, replace 342589
*Feb 10 13:10:52.621: translation_format replace_rule ^0119.%,
strip_proceeding 0
*Feb 10 13:10:52.621: replace_string match_tmp ^0119.%, strip_proceeding
0
*Feb 10 13:10:52.621: replace_string match_tmp 0119
*Feb 10 13:10:52.621: replace_string direction 2, callparty  2
*Feb 10 13:10:52.621: replace_string direction 2, callparty 2, target
011914428269323
*Feb 10 13:10:52.621: replace_string match_tmp 0119,replace 342589 
*Feb 10 13:10:52.621:
replace_string0.replace10,target11914428269323,current11914428269323,mat
ch_tmp119
*Feb 10 13:10:52.621: replace_string0.1 compare_len 1,match_len 4
*Feb 10 13:10:52.621:
replace_string0.replace101,target1914428269323,current1914428269323,matc
h_tmp19
*Feb 10 13:10:52.621: replace_string0.1 compare_len 2,match_len 4
*Feb 10 13:10:52.621:
replace_string0.replace1011,target914428269323,current914428269323,match
_tmp9
*Feb 10 13:10:52.621: replace_string0.1 compare_len 3,match_len 4
*Feb 10 13:10:52.621:
replace_string0.replace10119,target14428269323,current14428269323,match_
tmp
*Feb 10 13:10:52.621: replace_string0.1 compare_len 4,match_len 4
*Feb 10 13:10:52.621: replace_string 3. replace1 0119, compare_len 4
*Feb 10 13:10:52.621: replace_string 4. replace1 3119,compare_len
3,replace 42589
*Feb 10 13:10:52.621: replace_string 4. replace1 3419,compare_len
2,replace 2589
*Feb 10 13:10:52.621: replace_string 4. replace1 3429,compare_len
1,replace 589
*Feb 10 13:10:52.621: replace_string 4. replace1 3425,compare_len
0,replace 89
*Feb 10 13:10:52.621: replace_string 4. replace1 34258,compare_len
-1,replace 9
*Feb 10 13:10:52.621: replace_string 4. replace1 342589,compare_len
-2,replace 
*Feb 10 13:10:52.621: replace_string 5. replace1 342589, compare_len
-2,match_len 4
*Feb 10 13:10:52.621: replace_string 6. replace1 342589,compare_len
-2,current 14428269323
*Feb 10 13:10:52.621: replace_string 7. replace1 3425891
*Feb 10 13:10:52.621: replace_string 7. replace1 34258914
*Feb 10 13:10:52.621: replace_string 7. replace1 342589144
*Feb 10 13:10:52.621: replace_string 7. replace1 3425891442
*Feb 10 13:10:52.621: replace_string 7. replace1 34258914428
*Feb 10 13:10:52.621: replace_string 7. replace1 342589144282
*Feb 10 13:10:52.621: replace_string 7. replace1 3425891442826
*Feb 10 13:10:52.621: replace_string 7. replace1 34258914428269
*Feb 10 13:10:52.621: replace_string 7. replace1 342589144282693
*Feb 10 13:10:52.621: replace_string 7. replace1 3425891442826932
*Feb 10 13:10:52.621: replace_string 7. replace1 34258914428269323
*Feb 10 13:10:52.621: replace_string buffer 34258914428269323
*Feb 10 13:10:52.621: xrule_translation index 8,xrule_number
34258914428269323, callparty 2
*Feb 10 13:10:52.621: xrule_translation Return rc = 0
*Feb 10 13:10:52.621: xrule_checking Return rc = 0










-----Original Message-----
From: cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, September 04, 2003 10:52 AM
To: Anderson, Ian
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?

What IOS are you using?

Thanks

Scott

-----Original Message-----
From: Anderson, Ian [mailto:i.anderson@lancaster.ac.uk] 
Sent: Thursday, September 04, 2003 12:54 AM
To: Voll, Scott
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?


I'm running 7905s with build CP79050101SCCP030404A these are connected
through 3550-PWR-SMIs to a 3550-12G-EMI which is serving as the router
and DHCP.
I can provide IOS versions of the two 3550s if required.

No problems with DHCP here.

Ian

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll@wesd.org]
Sent: 03 September 2003 23:24
To: Wes Sisk
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] 7905 and DHCP?


I just tried that load and Its still isn't working?  Any other ideas?
I'm wandering about an IOS UPGrade for the cat 3550 to understand the
cdp so it sets the right Vlan?!?  

---scott

-----Original Message-----
From: Wes Sisk [mailto:wsisk@cisco.com] 
Sent: Wednesday, September 03, 2003 3:00 PM
To: Voll, Scott
Subject: RE: [cisco-voip] 7905 and DHCP?

verified version is 1.1(030404)

there is a package bundle for the low end phones on CCO:
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser

that includes CP7905010101SCCP030818A which I think has the
fix.  Have you tried this load?

/Wes


> -----Original Message-----
> From: Voll, Scott [mailto:Scott.Voll@wesd.org]
> Sent: Wednesday, September 03, 2003 5:50 PM
> To: Wes Sisk; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] 7905 and DHCP?
>
>
> Wes--
>
> Looks like the bug I'm up against. TAC told me to disable cdp.
> When I did that, it put the phone on the data network with the
> DHCP coming from the windows server rather then the IP range from
> the 3640 on the voice vlan.
>
> When is this latest release going to be out???? I could use it now!!!
>
> I did some sniffing and the phone(7905) is sending DHCP request
> and discoveries but the 3640 isn't getting anything from what I
> can tell. I see no dhcp offers.
>
> Tac then had me put the 7905s on a port just set for the voice
> vlan.  Still didn't work.  Any other ideas????  I was supposed to
> have this IP Phone Solution in place a week ago.
>
> Thanks
>
> Scott
>
> PS.  The cat 3550's think the 7905s are HOST as far as cdp knows.
>
> -----Original Message-----
> From: Wes Sisk [mailto:wsisk@cisco.com]
> Sent: Wednesday, September 03, 2003 12:11 PM
> To: Voll, Scott; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] 7905 and DHCP?
>
> possibly:
> CSCea26503 7905 phone failed to get IP Address from DHCP reply packet
> Fixed in the latest version (available whenever we have next
> software release). 79XX will now take DHCP offer packets from
> native VLAN if native VLAN is the same as aux VLAN.
>
> Pls span the switchport the phone is using and get a sniffertrace of
phone
> reboot to confirm dhcpoffer is coming back.
>
> On your router you might run 'deb dhcp detial' if not too busy to see
> if DHCP discover is making it to the router.
>
> /Wes
>
> > -----Original Message-----
> > From: cisco-voip-bounces@puck.nether.net
> > [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> > Sent: Tuesday, September 02, 2003 6:03 PM
> > To: cisco-voip@puck.nether.net
> > Subject: [cisco-voip] 7905 and DHCP?
> >
> >
> > Anyone else having problems with 7905's getting DHCP from a
> > router running DHCP???  The 7940sand 60 don't seem to have any
> > problems.  Will be opening a tac case.
> >
> > Scott Voll
> > Network Analyst, CCNA
> > Willamette ESD
> > Phone: (503) 385-4571
> > scott.voll@wesd.org
> > ?
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


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

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




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

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


End of cisco-voip Digest, Vol 7, Issue 4
****************************************

From Paul_Forbes at Trimble.com  Sat Sep 13 09:26:10 2003
From: Paul_Forbes at Trimble.com (Paul Forbes)
Date: Sat Sep 13 11:26:27 2003
Subject: [ ] - [ ] - [cisco-voip] How do I get a persons name on the phone
	- Email found in subject - Email found in subject
Message-ID: <D7293D0212DA9E409F2E38F6CC815AC4064B4B@uss-am-xch-02.am.trimblecorp.net>

Hi Scott,

When you define a DN on a subscriber's phone, under "Line Settings" you
have the option of entering the user's name information in the "Display
(Internal Caller ID)" field. This can also be added in batch via BAT.

Cheers.

pf

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll@wesd.org] 
Sent: Friday, September 12, 2003 9:47 AM
To: cisco-voip@puck.nether.net
Subject: [ ] - [ ] - [cisco-voip] How do I get a persons name on the
phone - Email found in subject - Email found in subject


OK another newby question.

I have my call manager 3.3.3sr1 up and running, routing and happy.  So
now I'm working on the nice little things.  

I do have corporate directories working and I know how to associate a
user to a phone, but I don't know how to get the persons name on the
phone display.  Any help would be great..... I'm having a hard time
finding any info on Cisco CCO.



Scott Voll
Network Analyst, CCNA 
Willamette ESD


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

From karthi at malagasy.com  Wed Sep 17 19:54:05 2003
From: karthi at malagasy.com (Karthikeyan Arumugam)
Date: Wed Sep 17 11:54:07 2003
Subject: [cisco-voip] voip rural setup
Message-ID: <020901c37d33$ec28d950$cd02a8c0@KARTHI>

dear sir;

Dear lists,

this is my first posting in the list and i hope that i get some suggestion.

I am planning to setup very low cost rural VOIP using VSAT and wiress loop ie with access point 
in point to multipoint mode with IP phones at the CPE.

the problem iam facing is how do I limit just only 4 users or IP conversation/call at a time.

let me explain clearly...i have bandwidth through VSAT  of about 34Kbps and so 4 users can have voice channels at 8 Kbps compression. for instance i have 15 users or subscribers with IP phones connected to wireless.if more users at a time request for IP call then voiceIP call would be degraded.now how do i control them.
could you pls kindly suggest me or help on this.

IPphone > Accesspoint > Router > VSAT <

can you suggest where i can make to limit only 4 users at a time.
i cant limit in the wireless loop as the AP allow all the IP call request
what kind of low end router or access product or VOIP products i can use to limit IP calls at anyone time
this setup is in remote rural areas. all these rural is connected to our NOC through VSAT
if a user picks IP phone he gets IVR from NOC through VSAT:

i would be glad if you could help me in this project.

regards
karthi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030917/804d3196/attachment.htm
From jared at puck.nether.net  Wed Sep 17 16:16:44 2003
From: jared at puck.nether.net (Jared Mauch)
Date: Wed Sep 17 15:16:46 2003
Subject: [cisco-voip] voip rural setup
In-Reply-To: <020901c37d33$ec28d950$cd02a8c0@KARTHI>
References: <020901c37d33$ec28d950$cd02a8c0@KARTHI>
Message-ID: <20030917191644.GF32442@puck.nether.net>

On Wed, Sep 17, 2003 at 06:54:05PM +0300, Karthikeyan Arumugam wrote:
> dear sir;
> 
> Dear lists,
> 
> this is my first posting in the list and i hope that i get some suggestion.
> 
> I am planning to setup very low cost rural VOIP using VSAT and wiress loop ie with access point 
> in point to multipoint mode with IP phones at the CPE.
> 
> the problem iam facing is how do I limit just only 4 users or IP conversation/call at a time.
> 
> let me explain clearly...i have bandwidth through VSAT  of about 34Kbps and so 4 users can have voice channels at 8 Kbps compression. for instance i have 15 users or subscribers with IP phones connected to wireless.if more users at a time request for IP call then voiceIP call would be degraded.now how do i control them.
> could you pls kindly suggest me or help on this.
> 
> IPphone > Accesspoint > Router > VSAT <
> 
> can you suggest where i can make to limit only 4 users at a time.
> i cant limit in the wireless loop as the AP allow all the IP call request
> what kind of low end router or access product or VOIP products i can use to limit IP calls at anyone time
> this setup is in remote rural areas. all these rural is connected to our NOC through VSAT
> if a user picks IP phone he gets IVR from NOC through VSAT:
> 
> i would be glad if you could help me in this project.

	What you probally want to do is do some congestion
managment.  I'm not sure what sort of call manager/sip proxy/whatnot
you are using in this situation, but what you want to do
is track these users in a group of some sort and when the phone attempts
to do a call setup either provide a busy tone back or a congestion/fast busy
tone back to the user.  If you signal back to the VoIP endpoint a
busy tone they will likely retry after a short period when
there are less people using the link.

	I'm not familiar with VSAT (is this a satellite link?  that's my
guess by the name) so can't provide much assistance with ideas
here.

	- Jared

-- 
Jared Mauch  | pgp key available via finger from jared@puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.
From wsisk at cisco.com  Wed Sep 17 23:28:46 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Sep 17 22:30:01 2003
Subject: [cisco-voip] voip rural setup
In-Reply-To: <20030917191644.GF32442@puck.nether.net>
References: <020901c37d33$ec28d950$cd02a8c0@KARTHI>
	<20030917191644.GF32442@puck.nether.net>
Message-ID: <3F69185E.9030100@cisco.com>

Jared,

Not sure what protocol you're using to control the phones at the CPE.
If you're using callmanager+sccp(skinny) you can use the "locations"
feature to do your call admission control (CAC).

If you're running h323, you can use a gatekeeper.
For SIP, sorry no idea, just starting down that path.

/Wes

Jared Mauch wrote:
> On Wed, Sep 17, 2003 at 06:54:05PM +0300, Karthikeyan Arumugam wrote:
> 
>>dear sir;
>>
>>Dear lists,
>>
>>this is my first posting in the list and i hope that i get some suggestion.
>>
>>I am planning to setup very low cost rural VOIP using VSAT and wiress loop ie with access point 
>>in point to multipoint mode with IP phones at the CPE.
>>
>>the problem iam facing is how do I limit just only 4 users or IP conversation/call at a time.
>>
>>let me explain clearly...i have bandwidth through VSAT  of about 34Kbps and so 4 users can have voice channels at 8 Kbps compression. for instance i have 15 users or subscribers with IP phones connected to wireless.if more users at a time request for IP call then voiceIP call would be degraded.now how do i control them.
>>could you pls kindly suggest me or help on this.
>>
>>IPphone > Accesspoint > Router > VSAT <
>>
>>can you suggest where i can make to limit only 4 users at a time.
>>i cant limit in the wireless loop as the AP allow all the IP call request
>>what kind of low end router or access product or VOIP products i can use to limit IP calls at anyone time
>>this setup is in remote rural areas. all these rural is connected to our NOC through VSAT
>>if a user picks IP phone he gets IVR from NOC through VSAT:
>>
>>i would be glad if you could help me in this project.
> 
> 
> 	What you probally want to do is do some congestion
> managment.  I'm not sure what sort of call manager/sip proxy/whatnot
> you are using in this situation, but what you want to do
> is track these users in a group of some sort and when the phone attempts
> to do a call setup either provide a busy tone back or a congestion/fast busy
> tone back to the user.  If you signal back to the VoIP endpoint a
> busy tone they will likely retry after a short period when
> there are less people using the link.
> 
> 	I'm not familiar with VSAT (is this a satellite link?  that's my
> guess by the name) so can't provide much assistance with ideas
> here.
> 
> 	- Jared
> 

From Scott.Voll at wesd.org  Fri Sep 19 09:10:09 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Sep 19 11:10:20 2003
Subject: [cisco-voip] RE: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
	NM-HDV/VWIC-1MFT-T1
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDBE@venus.wesd.org>

What are you running, H323,SIP,MGCP??  Can you use Qsig?  

--Scott

-----Original Message-----
From: tgrace [mailto:tgrace@tgrace.com] 
Sent: Thursday, September 18, 2003 7:51 PM
To: cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
NM-HDV/VWIC-1MFT-T1

I'm having difficulty keepinga PRI TIE line up between our 81C and a
Cisco
2621XM router. The PRI seems to go down shortly after a disconnect from
the
first incoming call to the router from the PBX or PSTN or after a dozen
quick succession calls to PBX extensions. When this happens I can see
the
router TX'ing SABME's but getting no response from the PBX. At other
times
the link will be up but I will get an INFO message back from the PBX
that
the "identified channel does not exist" when I try to place an outbound
call. There also seems to be some issue with disconnects with 5ESS. On a
number of occassions my Nortel phone continued ringing after going
onhook on
the Cisco phone. This isn't consistent and happens perhaps 1 out 10
calls.

Initially tried using the DMS100 protocol with the PBX as net and the
router
as USR. Also did the reverse and then tried configuring both as 5ESS.
Same
type of problems.

Anyone ever run into something like this?


_______________________________________________
cisco-nsp mailing list  cisco-nsp@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-nsp
archive at http://puck.nether.net/pipermail/cisco-nsp/

From Scott.Voll at wesd.org  Fri Sep 19 09:31:24 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Sep 19 11:31:36 2003
Subject: [cisco-voip] RE: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
	NM-HDV/VWIC-1MFT-T1
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDBF@venus.wesd.org>

My suggestion would be to go to H323.  We have been running into
problems with the call progress in MGCP.  TAC said that CM only supports
ISO ver 3 (2000).  Where as the IOS (H3233) supports ECMA, and ISO and
different ver. Of each.  Has fixed our problems.  And really the only
bad thing about H323 is the long config.

Scott

-----Original Message-----
From: Terry Grace [mailto:tgrace@tgrace.com] 
Sent: Friday, September 19, 2003 8:17 AM
To: Voll, Scott; cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: RE: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
NM-HDV/VWIC-1MFT-T1

We're using mgcp. Don't have the qsig package on the PBX.

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll@wesd.org] 
Sent: Friday, September 19, 2003 11:10 AM
To: tgrace; cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: RE: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
NM-HDV/VWIC-1MFT-T1


What are you running, H323,SIP,MGCP??  Can you use Qsig?  

--Scott

-----Original Message-----
From: tgrace [mailto:tgrace@tgrace.com] 
Sent: Thursday, September 18, 2003 7:51 PM
To: cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
NM-HDV/VWIC-1MFT-T1

I'm having difficulty keepinga PRI TIE line up between our 81C and a
Cisco
2621XM router. The PRI seems to go down shortly after a disconnect from
the
first incoming call to the router from the PBX or PSTN or after a dozen
quick succession calls to PBX extensions. When this happens I can see
the
router TX'ing SABME's but getting no response from the PBX. At other
times
the link will be up but I will get an INFO message back from the PBX
that
the "identified channel does not exist" when I try to place an outbound
call. There also seems to be some issue with disconnects with 5ESS. On a
number of occassions my Nortel phone continued ringing after going
onhook on
the Cisco phone. This isn't consistent and happens perhaps 1 out 10
calls.

Initially tried using the DMS100 protocol with the PBX as net and the
router
as USR. Also did the reverse and then tried configuring both as 5ESS.
Same
type of problems.

Anyone ever run into something like this?


_______________________________________________
cisco-nsp mailing list  cisco-nsp@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-nsp
archive at http://puck.nether.net/pipermail/cisco-nsp/


From MPuras at solunet.com  Fri Sep 19 14:52:34 2003
From: MPuras at solunet.com (MPuras@solunet.com)
Date: Fri Sep 19 13:52:51 2003
Subject: [cisco-voip] Dead air but calls shows connected
Message-ID: <719546A93D7DD511BA7800D0B7A87DF001A00548@TRINIDAD>

Hi everyone!  I am experiencing a problem.  I am new to this IP
Telephone/VoIP stuff and really dont know much about how to debug this
issue.   I have 4 FXS interfaces that are interconnected to my PBX here.  I
have configured the PBX such that if I dial the extension or the full
qualified phone number to that remote office the traffic is passed to the
trunks associated with the FXS cards on the router.  Once the router
receives this traffic the call is initiated to the remote office were it
terminates on 4 FXS cards and passed to another PBX from there.  At the
begin of this setup everything was working fine.  The issue that I am having
is that when I dial the remote office I see the call going out
(S_DIGIT_COLLECT, S_ALERTING, S_CONNECT) however I get dead silence.  If I
put that call on hold and I dial the same extension again such that it is
going out another FXS port on the same VWIC the call goes through and I am
able to get the auto attendant at the remote office.  Any help as to what
could be going on and how to fix it?  Here is the config:


<snip>
voice-port 1/0/0
 disconnect-ack
!
voice-port 1/0/1
 disconnect-ack
!
voice-port 1/1/0
 disconnect-ack
!
voice-port 1/1/1
 disconnect-ack
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 1 pots
 destination-pattern 32xxxx7947
 port 1/0/0
!
dial-peer voice 2 voip
 destination-pattern 30xxxx8748
 session target ipv4:192.168.8.2
 req-qos controlled-load
 ip qos dscp ef signaling
!
dial-peer voice 3 pots
 destination-pattern 32xxxx7947
 port 1/0/1
!
dial-peer voice 4 pots
 destination-pattern 32xxxx7947
 port 1/1/0
!
dial-peer voice 5 pots
 destination-pattern 32xxxx7947
 port 1/1/1



I am running a 3620 router with 12.2(15)T7 IP Plus



Thanks, 

Mario Puras 
SoluNet Technical Support
Mailto: mpuras@solunet.com
Direct: (321) 309-1410  
888.449.5766 (USA) / 888.SOLUNET (Canada) 

From mfa at lal.ufl.edu  Fri Sep 19 17:56:21 2003
From: mfa at lal.ufl.edu (Mike Armstrong)
Date: Fri Sep 19 16:56:27 2003
Subject: [cisco-voip] Clicking/popping noise on inbound side of PRI
Message-ID: <01f901c37ef0$7adadb40$6232e60a@lal.ufl.edu>

Warning: I've gone through 4 Cisco engineers at TAC, 3 Verizon technicians,
2 VWICs, and 2 SmartJacks, with no resolution to this problem yet, so this
may not be easy.  I'm looking for out-of-the-box thinking, unencumbered by
reason and conventional troubleshooting procedures.  Have a few beers before
working on this.

I've got a 3640/NM-HDV/VWIC-1MFT-T1 connected to an Adtran HTU-R (Verizon
calls it a "SmrtJack") which is the T1 (DS1) interface to an HDSL circuit
between us and the Verizon CO.  There is an intermittent loud
clicking/popping noise -- like a Geiger counter -- on the inbound side only.
It was first reported in early August around the time we upgraded the router
to IOS version 12.2(15)T.   Reverting to the previous version -- 
12.2(13)T -- did not help.

Cisco has replaced the VWIC, even though they didn't think it was the
problem.  Verizon replaced the HTU-R even though they didn't think it was
the problem.  Both Cisco and Verizon claimed it was my unshielded patch
cable between the VWIC and the HTU-R, so I replaced it with a certified,
shielded cable (10 feet long).  None of these replacements had any effect.
In all this time, looking at the diagnostic data recorded by the HTU-R, I
have seen only a few errors of any kind associated with the Telco circuit,
and most of them were apparently caused by us screwing around with the
circuit.  During the same time, I have recorded 6.7M Line Code Violations
and 800K Path Code Violations (hereafter referred to as "errors") on the
VWIC.

We all believe that layer 1 timing is the issue.  I monitored an open line
for a couple of hours and confirmed that the noise was correlated with T1
controller errors.  I've got "sho controller t1 1/0" data going back to
August 8th, with only a few gaps, in a database keyed to the
15-minute-collection-interval time, and from all this data can tell you:

o We see error rates from 0 to 109,333 per 15-minute interval.
o We can go for hours with 0 errors, then jump to tens of thousands in the
next 15-minute interval.
o Once they start, the errors tend to continue for a while, but it's not
guaranteed.
o There are more errors Saturday, Sunday, and Monday than Tue-Fri.
o There is a broad peak from 2PM-10PM.
o There is no evidence of electrical noise (all our equipment is plugged
into UPS, Telco equipment is line powered).
o There is no evidence of EMF noise in the area, measured crudely with a
broadband AM receiver.
o I can't correlate error-free or error-intensive periods with much of
anything.

We had the Telco run loopbacks and BERT tests.  Cisco has done the same.
They did it together for an hour and a half last week.  I've done the same.
None of us has yet been able to produce similar errors in any test
environment.

Relevant router configs are:

controller T1 1/0
 framing esf
 linecode b8zs
 cablelength short 133
 pri-group timeslots 1-20,24
 no yellow generation
 no yellow detection

...

interface Serial1/0:23
 no ip address
 no logging event link-status
 isdn switch-type primary-ni
 isdn incoming-voice voice
 (encapsulation HDLC by default)

Any AHA!s out there?

Mike Armstrong
University of Florida CREC
Lake Alfred, FL
863.956.5891

From vances at motivity.ca  Fri Sep 19 19:01:23 2003
From: vances at motivity.ca (Vance Shipley)
Date: Fri Sep 19 18:01:42 2003
Subject: [cisco-voip] Clicking/popping noise on inbound side of PRI
In-Reply-To: <01f901c37ef0$7adadb40$6232e60a@lal.ufl.edu>
References: <01f901c37ef0$7adadb40$6232e60a@lal.ufl.edu>
Message-ID: <20030919220123.GE25374@frogman.motivity.ca>

Mike,

I would want to run a 24 BERT test from the CO to the Cisco.  Put the
Cisco into remote loopback mode and have Verizon run a BERT test through
that loopback.  This will test the Cisco LIU, the cable, the SmartJack
and everything else out to the CO.  You need to do this for 24 hours
'cause you don't have a 100% failure rate.  If you could reproduce the
eoors all the time you could draw conclusions earlier but as it is you'll
need to run long enough to know that you would have seen the problem.

If that test shows errors then you would want to restest with the loopback 
being provided by the SmartJack.  If it still shows errors Verizon will
accept responsibility and fix it.  If not Cisco will blame your cable or
take resonsibility.

Now as far as the cable goes, what is a "certified cable"?  Is it a
certified RJ-48C cable?  Or is it an ethernet cable?  I don't believe
a lack of sheilding is going to cause this problem however you need
to make sure you have your signals on twisted pairs.  RJ-48C uses pins
1&2 and pins 4&5.  You must have pins 1&2 use a twisted pair, and the
same for pins 4&5.  A flat "silver satin" cable will not work.  A cable
where the white-blue/blue-white, etc. pairs are not using the right pins
will not work.  A combination of two wires where one is <color> with 
white stripes and the other is white with <color> stripes makes up a 
twisted pair (e.g. white-orange/orange-white).

99% of the time these sort of problems are caused by cabling/connector
issues.

Next question:  where are you deriving timing from?  You must be 
clocking off the incoming T1 from Verizon.  If you are using an internal
clock, or clocking off another span which is not itself timed off
Verizon (or a higher Stratum level) you will have problems.

	-Vance


From WCarter at sentinel.com  Sat Sep 20 01:47:51 2003
From: WCarter at sentinel.com (Carter, Bill)
Date: Sat Sep 20 01:47:57 2003
Subject: [cisco-voip] Clicking/popping noise on inbound side of PRI
Message-ID: <F4059FD8E8F9D141BE8C48C0DCF1FA157942A6@rodan.sentinel.com>

Avaya!!!

-----Original Message-----
From: Mike Armstrong [mailto:mfa@lal.ufl.edu] 
Sent: Friday, September 19, 2003 3:56 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Clicking/popping noise on inbound side of PRI


Warning: I've gone through 4 Cisco engineers at TAC, 3 Verizon
technicians, 2 VWICs, and 2 SmartJacks, with no resolution to this
problem yet, so this may not be easy.  I'm looking for out-of-the-box
thinking, unencumbered by reason and conventional troubleshooting
procedures.  Have a few beers before working on this.

I've got a 3640/NM-HDV/VWIC-1MFT-T1 connected to an Adtran HTU-R
(Verizon calls it a "SmrtJack") which is the T1 (DS1) interface to an
HDSL circuit between us and the Verizon CO.  There is an intermittent
loud clicking/popping noise -- like a Geiger counter -- on the inbound
side only. It was first reported in early August around the time we
upgraded the router
to IOS version 12.2(15)T.   Reverting to the previous version -- 
12.2(13)T -- did not help.

Cisco has replaced the VWIC, even though they didn't think it was the
problem.  Verizon replaced the HTU-R even though they didn't think it
was the problem.  Both Cisco and Verizon claimed it was my unshielded
patch cable between the VWIC and the HTU-R, so I replaced it with a
certified, shielded cable (10 feet long).  None of these replacements
had any effect. In all this time, looking at the diagnostic data
recorded by the HTU-R, I have seen only a few errors of any kind
associated with the Telco circuit, and most of them were apparently
caused by us screwing around with the circuit.  During the same time, I
have recorded 6.7M Line Code Violations and 800K Path Code Violations
(hereafter referred to as "errors") on the VWIC.

We all believe that layer 1 timing is the issue.  I monitored an open
line for a couple of hours and confirmed that the noise was correlated
with T1 controller errors.  I've got "sho controller t1 1/0" data going
back to August 8th, with only a few gaps, in a database keyed to the
15-minute-collection-interval time, and from all this data can tell you:

o We see error rates from 0 to 109,333 per 15-minute interval. o We can
go for hours with 0 errors, then jump to tens of thousands in the next
15-minute interval. o Once they start, the errors tend to continue for a
while, but it's not guaranteed. o There are more errors Saturday,
Sunday, and Monday than Tue-Fri. o There is a broad peak from 2PM-10PM.
o There is no evidence of electrical noise (all our equipment is plugged
into UPS, Telco equipment is line powered). o There is no evidence of
EMF noise in the area, measured crudely with a broadband AM receiver. o
I can't correlate error-free or error-intensive periods with much of
anything.

We had the Telco run loopbacks and BERT tests.  Cisco has done the same.
They did it together for an hour and a half last week.  I've done the
same. None of us has yet been able to produce similar errors in any test
environment.

Relevant router configs are:

controller T1 1/0
 framing esf
 linecode b8zs
 cablelength short 133
 pri-group timeslots 1-20,24
 no yellow generation
 no yellow detection

...

interface Serial1/0:23
 no ip address
 no logging event link-status
 isdn switch-type primary-ni
 isdn incoming-voice voice
 (encapsulation HDLC by default)

Any AHA!s out there?

Mike Armstrong
University of Florida CREC
Lake Alfred, FL
863.956.5891

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

From mfa at lal.ufl.edu  Sat Sep 20 16:31:27 2003
From: mfa at lal.ufl.edu (Mike Armstrong)
Date: Sat Sep 20 15:31:39 2003
Subject: [cisco-voip] Clicking/popping noise on inbound side of PRI
References: <01f901c37ef0$7adadb40$6232e60a@lal.ufl.edu>
	<20030919220123.GE25374@frogman.motivity.ca>
Message-ID: <006301c37fad$c9af8960$1a02a8c0@lal.ufl.edu>

Sorry, I neglected to cc the list when I replied to Scott yesterday; here's
the essence of it:

We use 1-20,24 because we're only paying for 20 channels.  I also questioned
this, but TAC didn't raise any
flags.  Also, everything worked fine for 8 months, so I'm fairly confident
this isn't it.

Late breaking news... Last night I spent about 1/2 hour of quality time with
the
people who built the SmartJack.  It has a control port on it -- an RS-232
connection for a VT-100 to do all sorts of diagnostic stuff.  I used it
yesterday
morning to do some loopback testing with no problems.  However, there were
no errors this morning either.  Tonight, I tried to do the same, with error
rates approaching 100,000 in a 15-minute interval, and the VT-100 stuff was
unusable.  The screens were unstable, the SmartJack acted like it was
seeing random input from my laptop, etc. -- a real GIGO situation, but I
wasn't inputting anything, let alone G.  The Adtran guy was baffled -- said
he couldn't account for that at all.  We disconnected the ground, the
router, and my laptop power supply, leaving the SmartJack connected only to
the Telco line and the laptop, which was now running on battery power
alone.  The problem persisted.  Mr Adtran was even more baffled -- said if
he was trying to design a SmartJack to do that, he wouldn't know where to
start.

The SmartJack derives its power from the HTU-C at the other end of the span.
Suspecting dirty power, we're eyeing the span power now, but don't know how
to measure/diagnose it.  As a
first cut, we decided to ask Verizon to plug in a passive T1 monitor on the
customer side of the SmartJack for a while to see if they see errors there.
They're supposed to show up Monday.
If they see errors, the theory goes, it's their problem.  I like your
approach, though.  Curiously,
Verizon showed up last week to install another T1, which terminates right
next to this one.  Problem is, nobody has any idea who ordered it, or what
it's for.  Maybe I'll make use of it.

Mike Armstrong
University of Florida CREC
Lake Alfred, FL
863.956.5891
----- Original Message ----- 
From: "Vance Shipley" <vances@motivity.ca>
To: "Mike Armstrong" <mfa@lal.ufl.edu>
Cc: <cisco-voip@puck.nether.net>
Sent: Friday, September 19, 2003 6:01 PM
Subject: Re: [cisco-voip] Clicking/popping noise on inbound side of PRI


> Mike,
>
> I would want to run a 24 BERT test from the CO to the Cisco.  Put the
> Cisco into remote loopback mode and have Verizon run a BERT test through
> that loopback.  This will test the Cisco LIU, the cable, the SmartJack
> and everything else out to the CO.  You need to do this for 24 hours
> 'cause you don't have a 100% failure rate.  If you could reproduce the
> eoors all the time you could draw conclusions earlier but as it is you'll
> need to run long enough to know that you would have seen the problem.
>
> If that test shows errors then you would want to restest with the loopback
> being provided by the SmartJack.  If it still shows errors Verizon will
> accept responsibility and fix it.  If not Cisco will blame your cable or
> take resonsibility.
>
> Now as far as the cable goes, what is a "certified cable"?  Is it a
> certified RJ-48C cable?  Or is it an ethernet cable?  I don't believe
> a lack of sheilding is going to cause this problem however you need
> to make sure you have your signals on twisted pairs.  RJ-48C uses pins
> 1&2 and pins 4&5.  You must have pins 1&2 use a twisted pair, and the
> same for pins 4&5.  A flat "silver satin" cable will not work.  A cable
> where the white-blue/blue-white, etc. pairs are not using the right pins
> will not work.  A combination of two wires where one is <color> with
> white stripes and the other is white with <color> stripes makes up a
> twisted pair (e.g. white-orange/orange-white).
>
> 99% of the time these sort of problems are caused by cabling/connector
> issues.
>
> Next question:  where are you deriving timing from?  You must be
> clocking off the incoming T1 from Verizon.  If you are using an internal
> clock, or clocking off another span which is not itself timed off
> Verizon (or a higher Stratum level) you will have problems.
>
> -Vance
>
>

From mfa at lal.ufl.edu  Sat Sep 20 17:14:18 2003
From: mfa at lal.ufl.edu (Mike Armstrong)
Date: Sat Sep 20 16:14:19 2003
Subject: [cisco-voip] Clicking/popping noise on inbound side of PRI
Message-ID: <007501c37fb3$c5c15080$1a02a8c0@lal.ufl.edu>

We can't bring the system down for 24 hours, unfortunately, since that's the
only voice link to the outside world.

Cisco and Verizon worked together for about 2 hours on a Friday afternoon a
week ago doing all sorts of loopbacks and Bert testing -- results were
apparently inconclusive, but Verizon replaced the HTU-R "SmartJack" out of
the goodness of their heart.  At the time, I didn't know squat about how
this circuit worked, so I just listened to the experts doing their thing.
However, now that I do know squat about it, I believe there's a giant
loophole in Verizon's diagnostic capabilities in this particular
installation, which I'll mention later.

The "certified cable" is a Cat5E-certifed Ethernet cable; I think it's good
for up to 350MHz or something like that.  Pairs are 1-2, 3-6, 4-5, 7-8, so
it should work fine in this application.  Hell, given the distance involved,
barbed-wire would probably work.  I have a friend who runs 10Mbps Ethernet
over about a 50' run using zip cord.

But I digress; we're deriving timing from the line.  Interestingly,
switching to clock source internal has little effect.

I talked with the manufacturer of the HTU-R for a while last night, and
looked more closely at how the Telco monitors the span.  Since the circuit
is actually an HDSL link, the main function of the HTU-R is to transform the
HDSL signal (2 loops) into a traditional DS1 for delivery to the customer.
Internal error monitoring is extensive, with logs kept for both loops on the
HDSL side, as well as incoming DS1 data.  See anything missing?  They do not
monitor or log conditions on the outgoing DS1 -- which is where we're seeing
all our errors.

However, the HTU-R has ports on it to allow passive monitoring of the DS1 on
the customer side of things.  Verizon is coming out Monday morning to
install their monitoring equipment and we'll let it cook for however long it
takes.  We've logged just under a million errors on the last 6 Mondays
(Monday is our 2nd-cleanest day), so we should be able to produce enough
events to see if the Verizon equipment sees the same errors the Cisco
equipment does.  If it does, I think Verizon will admit defeat; if it
doesn't, it's back to Cisco.

As I mentioned in a reply to Scott Voll, I've now got reason to suspect the
problem is at the other end of the span, where an HTU-C (CO side of the HDSL
span) supplies power to the HTU-R via the span.  I'll keep you posted.

Mike Armstrong
University of Florida CREC
Lake Alfred, FL
863.956.5891
----- Original Message ----- 
From: "Vance Shipley" <vances@motivity.ca>
To: "Mike Armstrong" <mfa@lal.ufl.edu>
Cc: <cisco-voip@puck.nether.net>
Sent: Friday, September 19, 2003 6:01 PM
Subject: Re: [cisco-voip] Clicking/popping noise on inbound side of PRI


> Mike,
>
> I would want to run a 24 BERT test from the CO to the Cisco.  Put the
> Cisco into remote loopback mode and have Verizon run a BERT test through
> that loopback.  This will test the Cisco LIU, the cable, the SmartJack
> and everything else out to the CO.  You need to do this for 24 hours
> 'cause you don't have a 100% failure rate.  If you could reproduce the
> eoors all the time you could draw conclusions earlier but as it is you'll
> need to run long enough to know that you would have seen the problem.
>
> If that test shows errors then you would want to restest with the loopback
> being provided by the SmartJack.  If it still shows errors Verizon will
> accept responsibility and fix it.  If not Cisco will blame your cable or
> take resonsibility.
>
> Now as far as the cable goes, what is a "certified cable"?  Is it a
> certified RJ-48C cable?  Or is it an ethernet cable?  I don't believe
> a lack of sheilding is going to cause this problem however you need
> to make sure you have your signals on twisted pairs.  RJ-48C uses pins
> 1&2 and pins 4&5.  You must have pins 1&2 use a twisted pair, and the
> same for pins 4&5.  A flat "silver satin" cable will not work.  A cable
> where the white-blue/blue-white, etc. pairs are not using the right pins
> will not work.  A combination of two wires where one is <color> with
> white stripes and the other is white with <color> stripes makes up a
> twisted pair (e.g. white-orange/orange-white).
>
> 99% of the time these sort of problems are caused by cabling/connector
> issues.
>
> Next question:  where are you deriving timing from?  You must be
> clocking off the incoming T1 from Verizon.  If you are using an internal
> clock, or clocking off another span which is not itself timed off
> Verizon (or a higher Stratum level) you will have problems.
>
> -Vance
>
>

From Scott.Voll at wesd.org  Mon Sep 22 10:12:37 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Sep 22 12:12:43 2003
Subject: [cisco-voip] Help with attendant console
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDC2@venus.wesd.org>

I'm trying to get attendant console working(1.2.1).  Have followed the directions and I'm getting an error, "Initialization of call control failed. Retrying"  when I log onto the console on the PC.  Any Idea's?  I have deleted and remade all console users, Pilot points, Hunt groups, and the AC user.  I have also stopped and restarted the TCD and CTI services.  Nothing will get rid of the error.  Anyone come up against this before?  Any help would be appreciated.  FYI I'm running CM 3.3.3sr1.

Scott

Scott Voll
Network Analyst, CCNA 
Willamette ESD
Phone: (503) 385-4571
scott.voll@wesd.org
?


From wsisk at cisco.com  Mon Sep 22 13:17:17 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Mon Sep 22 12:17:29 2003
Subject: [cisco-voip] Help with attendant console
In-Reply-To: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDC2@venus.wesd.org>
Message-ID: <ELEMLGDGDBLINAJEIOCLOEDEMOAA.wsisk@cisco.com>

Scott,

Have you looked at the Detailed TCDSrv traces?  I'm guessing that
your AC user might now have CTI Use Enabled checked in the user config.

/Wes

> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> Sent: Monday, September 22, 2003 12:13 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Help with attendant console
>
>
> I'm trying to get attendant console working(1.2.1).  Have
> followed the directions and I'm getting an error, "Initialization
> of call control failed. Retrying"  when I log onto the console on
> the PC.  Any Idea's?  I have deleted and remade all console
> users, Pilot points, Hunt groups, and the AC user.  I have also
> stopped and restarted the TCD and CTI services.  Nothing will get
> rid of the error.  Anyone come up against this before?  Any help
> would be appreciated.  FYI I'm running CM 3.3.3sr1.
>
> Scott
>
> Scott Voll
> Network Analyst, CCNA
> Willamette ESD
> Phone: (503) 385-4571
> scott.voll@wesd.org
> ?
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From WCarter at sentinel.com  Mon Sep 22 12:27:39 2003
From: WCarter at sentinel.com (Carter, Bill)
Date: Mon Sep 22 12:27:50 2003
Subject: [cisco-voip] Help with attendant console
Message-ID: <F4059FD8E8F9D141BE8C48C0DCF1FA157942B4@rodan.sentinel.com>

Make sure the device is associated with the ac user.

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll@wesd.org]
Sent: Monday, September 22, 2003 11:13 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Help with attendant console


I'm trying to get attendant console working(1.2.1).  Have followed the directions and I'm getting an error, "Initialization of call control failed. Retrying"  when I log onto the console on the PC.  Any Idea's?  I have deleted and remade all console users, Pilot points, Hunt groups, and the AC user.  I have also stopped and restarted the TCD and CTI services.  Nothing will get rid of the error.  Anyone come up against this before?  Any help would be appreciated.  FYI I'm running CM 3.3.3sr1.

Scott

Scott Voll
Network Analyst, CCNA 
Willamette ESD
Phone: (503) 385-4571
scott.voll@wesd.org
?


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

From rratliff at cisco.com  Mon Sep 22 13:29:24 2003
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Sep 22 12:29:29 2003
Subject: [cisco-voip] Help with attendant console
References: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDC2@venus.wesd.org>
Message-ID: <025d01c38126$af474610$4d3a6640@amer.cisco.com>

Have you turned on client and TCD traces?  There should at least be some
more useful information in them.

-Ryan
----- Original Message ----- 
From: "Voll, Scott" <Scott.Voll@wesd.org>
To: <cisco-voip@puck.nether.net>
Sent: Monday, September 22, 2003 12:12 PM
Subject: [cisco-voip] Help with attendant console


> I'm trying to get attendant console working(1.2.1).  Have followed the
directions and I'm getting an error, "Initialization of call control failed.
Retrying"  when I log onto the console on the PC.  Any Idea's?  I have
deleted and remade all console users, Pilot points, Hunt groups, and the AC
user.  I have also stopped and restarted the TCD and CTI services.  Nothing
will get rid of the error.  Anyone come up against this before?  Any help
would be appreciated.  FYI I'm running CM 3.3.3sr1.
>
> Scott
>
> Scott Voll
> Network Analyst, CCNA
> Willamette ESD
> Phone: (503) 385-4571
> scott.voll@wesd.org
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From Scott.Voll at wesd.org  Mon Sep 22 10:57:18 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Sep 22 12:57:24 2003
Subject: [cisco-voip] Help with attendant console
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D4A2@venus.wesd.org>

Wes-- I'm assuming that you meant Might Not have CTI Use Enabled.  I do have it checked.  

Here is the trace of TCD Detailed.  I do see some errors but not sure what it means.  See below.

Thanks

Scott

PS.  The corporate directory works and so does the speed dials and also shows on hook and off.  

    1 09/22 08:27:22.314  INFO 
    2 09/22 08:27:22.324  INFO **************************************
    3 09/22 08:27:22.324  INFO ****** Attendant Console Server ******
    4 09/22 08:27:22.324  INFO * Version  : 1.2(1)
    5 09/22 08:27:22.324  INFO * Built on : Fri May  2 15:51:53 PDT 2003
    6 09/22 08:27:22.324  INFO **************************************
    7 09/22 08:27:22.324  INFO 
    8 09/22 08:27:22.384 DEBUG ACBootstrap: Initialization started
    9 09/22 08:27:22.404 DEBUG _ldapUrl:ldap://neptune.wesd.org:389
   10 09/22 08:27:22.404 DEBUG _directoryManagerDN:cn=Administrator, cn=Users, dc=wesd,dc=org
   11 09/22 08:27:22.484 DEBUG _directoryManagerPassword:******
   12 09/22 08:27:22.484 DEBUG _ciscoBase:ou=Cisco, dc=wesd,dc=org
   13 09/22 08:27:22.484 DEBUG _ccnBase:ou=CCN,ou=Cisco, dc=wesd,dc=org
   14 09/22 08:27:22.484 DEBUG _directoryType:ADS
   15 09/22 08:27:22.985 DEBUG _userBase:cn=Users, dc=wesd,dc=org
   16 09/22 08:27:22.985 DEBUG _userOC:User
   17 09/22 08:27:22.985 DEBUG _profilesBase:ou=profiles, ou=CCN
   18 09/22 08:27:22.985 DEBUG _internalUniqueAttributeName:cn
   19 09/22 08:27:23.075 DEBUG ACBootstrap: Computing jtapi.jar's checksum...
   20 09/22 08:27:23.075 DEBUG ACBootstrap: jtapi.jar's checksum: 354288330
   21 09/22 08:27:23.075 DEBUG ACBootstrap: Initialization completed
   22 09/22 08:27:23.075 DEBUG ACServer.properties exists. Skipping creation.
   23 09/22 08:27:23.075  INFO Checking connectivity with database...
   24 09/22 08:27:23.085  INFO ACCallManagerDB dbConection: DSN=CiscoCallManager;SERVER=WESDCM1;DATABASE=CCM0301;Trusted_Connection=yes
   25 09/22 08:27:23.085 DEBUG ACCallManagerDB dbConection: 0
   26 09/22 08:27:23.085 DEBUG      SERVER=WESDCM1
   27 09/22 08:27:23.085 DEBUG    DATABASE=CCM0301
   28 09/22 08:27:23.085  INFO ACCallManagerDB dbConection: DSN=CiscoCallManager;SERVER=wesdcm2;DATABASE=CCM0301;Trusted_Connection=yes
   29 09/22 08:27:23.085 DEBUG ACCallManagerDB dbConection: 1
   30 09/22 08:27:23.085 DEBUG      SERVER=wesdcm2
   31 09/22 08:27:23.095 DEBUG    DATABASE=CCM0301
   32 09/22 08:27:23.095 DEBUG Checking if primary database is up and running...
   33 09/22 08:27:23.095  INFO Attempting to access (publisher) db on: WESDCM1
   34 09/22 08:27:23.335  INFO Successfully opened (publisher) db on: WESDCM1
   35 09/22 08:27:23.335  INFO Checking connectivity with database...done
   36 09/22 08:27:23.836 DEBUG ACCallManagers: CM_wesdcm1 is thisServer
   37 09/22 08:27:24.828 DEBUG New trace level: ERROR
   38 09/22 08:27:25.859 ERROR java.lang.NullPointerException
	at java.text.CollationElementIterator.<init>(CollationElementIterator.java:102)
	at java.text.RuleBasedCollator.getCollationElementIterator(RuleBasedCollator.java:358)
	at java.text.RuleBasedCollator.compare(RuleBasedCollator.java:405)
	at java.text.Collator.equals(Collator.java:322)
	at com.cisco.ac.server.DBL.ACHuntGroup.setOwner(ACHuntGroup.java:164)
	at com.cisco.ac.server.DBL.ACHuntGroup.<init>(ACHuntGroup.java:64)
	at com.cisco.ac.server.DBL.ACDatabaseParameters.loadPilotTable(ACDatabaseParameters.java:144)
	at com.cisco.ac.server.DBL.ACDatabaseParameters.getHuntGroups(ACDatabaseParameters.java:77)
	at com.cisco.ac.server.telephony.ACServerTelephony.init(ACServerTelephony.java:51)
	at com.cisco.ac.server.ACMain.startACServer(ACMain.java:123)
	at com.cisco.ac.server.ACMain.main(ACMain.java:40)

   39 09/22 08:27:25.869 ERROR HuntGroups:
HuntGroupsUsers:
pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
Sessions:
Active Sessions:
Deactive Sessions:

   40 09/22 08:27:58.867 ERROR java.lang.NullPointerException
	at java.text.CollationElementIterator.<init>(CollationElementIterator.java:102)
	at java.text.RuleBasedCollator.getCollationElementIterator(RuleBasedCollator.java:358)
	at java.text.RuleBasedCollator.compare(RuleBasedCollator.java:405)
	at java.text.Collator.equals(Collator.java:322)
	at com.cisco.ac.server.DBL.ACHuntGroup.setOwner(ACHuntGroup.java:164)
	at com.cisco.ac.server.DBL.ACHuntGroup.<init>(ACHuntGroup.java:64)
	at com.cisco.ac.server.DBL.ACDatabaseParameters.loadPilotTable(ACDatabaseParameters.java:144)
	at com.cisco.ac.server.DBL.ACDatabaseParameters.getHuntGroups(ACDatabaseParameters.java:77)
	at com.cisco.ac.server.telephony.ACServerTelephony.pilotChanged(ACServerTelephony.java:257)
	at com.cisco.ac.server.telephony.ACServerTelephony.databaseChanged(ACServerTelephony.java:195)
	at com.cisco.ac.server.DBL.ACDatabaseParameters.informDBCNListerners(ACDatabaseParameters.java:911)
	at com.cisco.ac.server.DBL.ACDatabaseParameters.pilotPointChanged(ACDatabaseParameters.java:951)
	at com.cisco.ac.server.DBL.ACDatabaseParameters.pilotPointChanged(ACDatabaseParameters.java:947)
	at com.cisco.ac.server.DBL.DBCN.ACDBCNAction.run(ACDBCNAction.java:26)

   41 09/22 08:27:58.867 ERROR HuntGroups:
HuntGroupsUsers:
pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
Sessions:
Active Sessions:
Deactive Sessions:

   42 09/22 08:28:17.975 ERROR HuntGroups:
HuntGroupsUsers:
pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
Sessions:
Active Sessions:
    SessionID: 1, CMName: CM_wesdcm1, IsLocal: true, UserName: svoll, Device: SEP000DBD4B3289, DeactivateReason: 0, Active: true, OnLine: false, QueryDirectory: false, QueryUsers: false, KeepAliveTimerCount: 0, CreationTime: 1064244482483
Deactive Sessions:

   43 09/22 09:47:13.831  INFO **** Switching log output to file: c:\Program Files\Cisco\Trace\TCD\tcdsrv.txt ****
   44 09/22 09:47:14.633  INFO ServerClient get telephony parameters
   45 09/22 09:47:14.643  INFO ServerClient get CM group: device: SEP000DBD4B3289
   46 09/22 09:48:14.771  INFO ServerClient get telephony parameters
   47 09/22 09:48:14.771  INFO ServerClient get CM group: device: SEP000DBD4B3289
   48 09/22 09:49:12.595 DEBUG Writing user (svoll) data to directory...
   49 09/22 09:49:12.876 DEBUG Writing user (svoll) data to directory done.
   50 09/22 09:49:12.886  INFO ServerClient Logout SessionNumber: 6 RemoveSession: true
   51 09/22 09:49:12.886 DEBUG ServerClient Logout SessionNumber: 6 exists
   52 09/22 09:49:12.886  INFO ServerClient Go Offline SessionNumber: 6
   53 09/22 09:49:12.886 DEBUG ServerClient goOffline SessionNumber: 6 exists
   54 09/22 09:49:12.896  INFO ACServerTelephony user offline : svoll
   55 09/22 09:49:12.896  INFO ACServerTelephony subscribeLineState 
   56 09/22 09:49:12.896 DEBUG Added server subscription: server: CM_wesdcm1, count: 0
   57 09/22 09:49:12.896  INFO ACInterServerConnections informSubscribeLSS server: CM_wesdcm1 Servers: CM_wesdcm2, 
   58 09/22 09:49:12.896  INFO ACInterServerConnections informUserOffline username: svoll Servers: CM_wesdcm2, 
   59 09/22 09:49:12.896  INFO ACInterServerComm: server : CM_wesdcm2
   60 09/22 09:49:12.896 DEBUG Added server subscription: server: CM_wesdcm2, count: 0
   61 09/22 09:49:12.896  INFO ACServerClientKeepAlive stop the timer
   62 09/22 09:49:12.906  INFO ACServerTelephony user logout : svoll
   63 09/22 09:49:12.906  INFO ACInterServerConnections informUserLogout username: svoll Servers: CM_wesdcm2, 
   64 09/22 09:49:12.966 DEBUG ACInterServerComm HGLineStateChanged: numEntries: 0
   65 09/22 09:49:12.996 DEBUG Sending 0 HG linestate(s) sent to server: CM_wesdcm2
   66 09/22 09:49:12.996  INFO ACInterServerConnections: sendLineStateToServer(): CM_wesdcm2
   67 09/22 09:49:12.996 DEBUG Updating 0 HG linestate(s) sent locally
   68 09/22 09:49:15.499  INFO ServerClient GetDevices DN: 5961
   69 09/22 09:49:15.590  INFO ServerClient getCMList
   70 09/22 09:49:15.600 DEBUG Exception: java.io.FileNotFoundException: etc\DialRules.xml (The system cannot find the file specified): Reading file: etc\DialRules.xml, offset: 0, len: 100000
   71 09/22 09:49:15.660 DEBUG Exception: java.io.FileNotFoundException: etc\VoiceMailProfiles.xml (The system cannot find the file specified): Reading file: etc\VoiceMailProfiles.xml, offset: 0, len: 100000
   72 09/22 09:49:19.685  INFO ServerClient Login: Username: svoll Device: SEP000DBD4B3289 ForceLogout: false
   73 09/22 09:49:19.866  INFO ServerClient username: svoll creating new session
   74 09/22 09:49:19.866  INFO ACServerClientSessionManager create new session for: svoll CM: CM_wesdcm1 snum: 7
   75 09/22 09:49:19.866  INFO ACServerClientKeepAlive start the timer
   76 09/22 09:49:19.876  INFO ACServerTelephony user login : svoll
   77 09/22 09:49:19.876  INFO ACInterServerConnections informUserLogin username: svoll Servers: CM_wesdcm2, 
   78 09/22 09:49:20.086 DEBUG Reading user (svoll) attributes from directory...
   79 09/22 09:49:20.176 DEBUG Reading user (svoll) attributes from directory done
   80 09/22 09:49:20.797  INFO ServerClient Get Lines Device : SEP000DBD4B3289
   81 09/22 09:49:21.128  INFO ServerClient getSupportPark().
   82 09/22 09:49:21.338  INFO ServerClient getSupportPark().
   83 09/22 09:49:21.348  INFO ServerClient getSupportPark().
   84 09/22 09:49:21.678 DEBUG ServerClient subscribelinestate: inet address: 198.237.183.2 Port: 4176
   85 09/22 09:49:21.678 DEBUG Added client subscription: 198.237.183.2, count: 2
   86 09/22 09:49:21.678  INFO ACInterServerConnections informSubscribeLSS inet address: 198.237.183.2 Port: 4176 Servers: CM_wesdcm2, 
   87 09/22 09:49:21.708 DEBUG Sending 0 linestate(s) sent to client: 198.237.183.2
   88 09/22 09:49:21.799  INFO ServerClient getSupportPark().
   89 09/22 09:49:21.799  INFO ServerClient getSupportPark().
   90 09/22 09:49:21.919  INFO ServerClient get server version
   91 09/22 09:49:21.919  INFO ServerClient getUserInfo SessionNumber: 7
   92 09/22 09:49:21.929  INFO ServerClient getSupportPark().
   93 09/22 09:49:22.309  INFO ServerClient get telephony parameters
   94 09/22 09:49:22.319  INFO ServerClient get CM group: device: SEP000DBD4B3289
   95 09/22 09:49:22.680 DEBUG Exception: java.io.FileNotFoundException: UserLists\CorporateDirectory.txt (The system cannot find the file specified): Reading file: UserLists\CorporateDirectory.txt, offset: 0, len: 100000
   96 09/22 09:49:26.185  INFO ServerClient GoOnline SessionNumber: 7
   97 09/22 09:49:26.185 DEBUG ServerClient goOnline SessionNumber: 7 exists
   98 09/22 09:49:26.185  INFO ACServerTelephony user online : svoll
   99 09/22 09:49:26.185  INFO ACServerTelephony subscribeLineState 
  100 09/22 09:49:26.185 DEBUG Added server subscription: server: CM_wesdcm1, count: 0
  101 09/22 09:49:26.185  INFO ACInterServerConnections informSubscribeLSS server: CM_wesdcm1 Servers: CM_wesdcm2, 
  102 09/22 09:49:26.185  INFO ACInterServerConnections informUserOnline username: svoll Servers: CM_wesdcm2, 
  103 09/22 09:49:26.185  INFO ACInterServerComm: server : CM_wesdcm2
  104 09/22 09:49:26.185 DEBUG Added server subscription: server: CM_wesdcm2, count: 0
  105 09/22 09:49:26.285 DEBUG Sending 0 HG linestate(s) sent to server: CM_wesdcm2
  106 09/22 09:49:26.285  INFO ACInterServerConnections: sendLineStateToServer(): CM_wesdcm2
  107 09/22 09:49:26.285 DEBUG Updating 0 HG linestate(s) sent locally
  108 09/22 09:49:26.305 DEBUG ACInterServerComm HGLineStateChanged: numEntries: 0

-----Original Message-----
From: Wes Sisk [mailto:wsisk@cisco.com] 
Sent: Monday, September 22, 2003 9:17 AM
To: Voll, Scott; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Help with attendant console

Scott,

Have you looked at the Detailed TCDSrv traces?  I'm guessing that
your AC user might now have CTI Use Enabled checked in the user config.

/Wes

> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> Sent: Monday, September 22, 2003 12:13 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Help with attendant console
>
>
> I'm trying to get attendant console working(1.2.1).  Have
> followed the directions and I'm getting an error, "Initialization
> of call control failed. Retrying"  when I log onto the console on
> the PC.  Any Idea's?  I have deleted and remade all console
> users, Pilot points, Hunt groups, and the AC user.  I have also
> stopped and restarted the TCD and CTI services.  Nothing will get
> rid of the error.  Anyone come up against this before?  Any help
> would be appreciated.  FYI I'm running CM 3.3.3sr1.
>
> Scott
>
> Scott Voll
> Network Analyst, CCNA
> Willamette ESD
> Phone: (503) 385-4571
> scott.voll@wesd.org
> ?
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From gea at ufl.edu  Mon Sep 22 15:05:36 2003
From: gea at ufl.edu (Greg Ayoub)
Date: Mon Sep 22 14:05:43 2003
Subject: [cisco-voip] Help with attendant console
References: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDC2@venus.wesd.org>
	<025d01c38126$af474610$4d3a6640@amer.cisco.com>
Message-ID: <000f01c38134$20a2ab80$f74fb29f@10CH311>

There is a bug in the phone load for 3.33sp1.  Try upgrading to the latest
7960 phone load found here -



----- Original Message ----- 
From: "Ryan Ratliff" <rratliff@cisco.com>
To: "Voll, Scott" <Scott.Voll@wesd.org>; <cisco-voip@puck.nether.net>
Sent: Monday, September 22, 2003 12:29 PM
Subject: Re: [cisco-voip] Help with attendant console


> Have you turned on client and TCD traces?  There should at least be some
> more useful information in them.
>
> -Ryan
> ----- Original Message ----- 
> From: "Voll, Scott" <Scott.Voll@wesd.org>
> To: <cisco-voip@puck.nether.net>
> Sent: Monday, September 22, 2003 12:12 PM
> Subject: [cisco-voip] Help with attendant console
>
>
> > I'm trying to get attendant console working(1.2.1).  Have followed the
> directions and I'm getting an error, "Initialization of call control
failed.
> Retrying"  when I log onto the console on the PC.  Any Idea's?  I have
> deleted and remade all console users, Pilot points, Hunt groups, and the
AC
> user.  I have also stopped and restarted the TCD and CTI services.
Nothing
> will get rid of the error.  Anyone come up against this before?  Any help
> would be appreciated.  FYI I'm running CM 3.3.3sr1.
> >
> > Scott
> >
> > Scott Voll
> > Network Analyst, CCNA
> > Willamette ESD
> > Phone: (503) 385-4571
> > scott.voll@wesd.org
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From gea at ufl.edu  Mon Sep 22 15:06:24 2003
From: gea at ufl.edu (Greg Ayoub)
Date: Mon Sep 22 14:06:34 2003
Subject: [cisco-voip] Help with attendant console
References: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDC2@venus.wesd.org>
	<025d01c38126$af474610$4d3a6640@amer.cisco.com>
Message-ID: <001a01c38134$3cb6eed0$f74fb29f@10CH311>

Sorry,

Here is the link...

http://www.cisco.com/cgi-bin/tablebuild.pl/cmva-3des


----- Original Message ----- 
From: "Ryan Ratliff" <rratliff@cisco.com>
To: "Voll, Scott" <Scott.Voll@wesd.org>; <cisco-voip@puck.nether.net>
Sent: Monday, September 22, 2003 12:29 PM
Subject: Re: [cisco-voip] Help with attendant console


> Have you turned on client and TCD traces?  There should at least be some
> more useful information in them.
>
> -Ryan
> ----- Original Message ----- 
> From: "Voll, Scott" <Scott.Voll@wesd.org>
> To: <cisco-voip@puck.nether.net>
> Sent: Monday, September 22, 2003 12:12 PM
> Subject: [cisco-voip] Help with attendant console
>
>
> > I'm trying to get attendant console working(1.2.1).  Have followed the
> directions and I'm getting an error, "Initialization of call control
failed.
> Retrying"  when I log onto the console on the PC.  Any Idea's?  I have
> deleted and remade all console users, Pilot points, Hunt groups, and the
AC
> user.  I have also stopped and restarted the TCD and CTI services.
Nothing
> will get rid of the error.  Anyone come up against this before?  Any help
> would be appreciated.  FYI I'm running CM 3.3.3sr1.
> >
> > Scott
> >
> > Scott Voll
> > Network Analyst, CCNA
> > Willamette ESD
> > Phone: (503) 385-4571
> > scott.voll@wesd.org
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From wsisk at cisco.com  Mon Sep 22 17:35:41 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Mon Sep 22 16:37:20 2003
Subject: [cisco-voip] Help with attendant console
In-Reply-To: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D4A2@venus.wesd.org>
References: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D4A2@venus.wesd.org>
Message-ID: <3F6F5D1D.8030600@cisco.com>

Scott,

Can you access the trace configuration under http://WESDCM1/ccmservice 
and set TCD trace level to detailed, enable high level and low level 
trace, then restart the Cisco Telephony Call Dispatch service.  Then 
please re-send the log.

/Wes

Voll, Scott wrote:
> Wes-- I'm assuming that you meant Might Not have CTI Use Enabled.  I do have it checked.  
> 
> Here is the trace of TCD Detailed.  I do see some errors but not sure what it means.  See below.
> 
> Thanks
> 
> Scott
> 
> PS.  The corporate directory works and so does the speed dials and also shows on hook and off.  
> 
>     1 09/22 08:27:22.314  INFO 
>     2 09/22 08:27:22.324  INFO **************************************
>     3 09/22 08:27:22.324  INFO ****** Attendant Console Server ******
>     4 09/22 08:27:22.324  INFO * Version  : 1.2(1)
>     5 09/22 08:27:22.324  INFO * Built on : Fri May  2 15:51:53 PDT 2003
>     6 09/22 08:27:22.324  INFO **************************************
>     7 09/22 08:27:22.324  INFO 
>     8 09/22 08:27:22.384 DEBUG ACBootstrap: Initialization started
>     9 09/22 08:27:22.404 DEBUG _ldapUrl:ldap://neptune.wesd.org:389
>    10 09/22 08:27:22.404 DEBUG _directoryManagerDN:cn=Administrator, cn=Users, dc=wesd,dc=org
>    11 09/22 08:27:22.484 DEBUG _directoryManagerPassword:******
>    12 09/22 08:27:22.484 DEBUG _ciscoBase:ou=Cisco, dc=wesd,dc=org
>    13 09/22 08:27:22.484 DEBUG _ccnBase:ou=CCN,ou=Cisco, dc=wesd,dc=org
>    14 09/22 08:27:22.484 DEBUG _directoryType:ADS
>    15 09/22 08:27:22.985 DEBUG _userBase:cn=Users, dc=wesd,dc=org
>    16 09/22 08:27:22.985 DEBUG _userOC:User
>    17 09/22 08:27:22.985 DEBUG _profilesBase:ou=profiles, ou=CCN
>    18 09/22 08:27:22.985 DEBUG _internalUniqueAttributeName:cn
>    19 09/22 08:27:23.075 DEBUG ACBootstrap: Computing jtapi.jar's checksum...
>    20 09/22 08:27:23.075 DEBUG ACBootstrap: jtapi.jar's checksum: 354288330
>    21 09/22 08:27:23.075 DEBUG ACBootstrap: Initialization completed
>    22 09/22 08:27:23.075 DEBUG ACServer.properties exists. Skipping creation.
>    23 09/22 08:27:23.075  INFO Checking connectivity with database...
>    24 09/22 08:27:23.085  INFO ACCallManagerDB dbConection: DSN=CiscoCallManager;SERVER=WESDCM1;DATABASE=CCM0301;Trusted_Connection=yes
>    25 09/22 08:27:23.085 DEBUG ACCallManagerDB dbConection: 0
>    26 09/22 08:27:23.085 DEBUG      SERVER=WESDCM1
>    27 09/22 08:27:23.085 DEBUG    DATABASE=CCM0301
>    28 09/22 08:27:23.085  INFO ACCallManagerDB dbConection: DSN=CiscoCallManager;SERVER=wesdcm2;DATABASE=CCM0301;Trusted_Connection=yes
>    29 09/22 08:27:23.085 DEBUG ACCallManagerDB dbConection: 1
>    30 09/22 08:27:23.085 DEBUG      SERVER=wesdcm2
>    31 09/22 08:27:23.095 DEBUG    DATABASE=CCM0301
>    32 09/22 08:27:23.095 DEBUG Checking if primary database is up and running...
>    33 09/22 08:27:23.095  INFO Attempting to access (publisher) db on: WESDCM1
>    34 09/22 08:27:23.335  INFO Successfully opened (publisher) db on: WESDCM1
>    35 09/22 08:27:23.335  INFO Checking connectivity with database...done
>    36 09/22 08:27:23.836 DEBUG ACCallManagers: CM_wesdcm1 is thisServer
>    37 09/22 08:27:24.828 DEBUG New trace level: ERROR
>    38 09/22 08:27:25.859 ERROR java.lang.NullPointerException
> 	at java.text.CollationElementIterator.<init>(CollationElementIterator.java:102)
> 	at java.text.RuleBasedCollator.getCollationElementIterator(RuleBasedCollator.java:358)
> 	at java.text.RuleBasedCollator.compare(RuleBasedCollator.java:405)
> 	at java.text.Collator.equals(Collator.java:322)
> 	at com.cisco.ac.server.DBL.ACHuntGroup.setOwner(ACHuntGroup.java:164)
> 	at com.cisco.ac.server.DBL.ACHuntGroup.<init>(ACHuntGroup.java:64)
> 	at com.cisco.ac.server.DBL.ACDatabaseParameters.loadPilotTable(ACDatabaseParameters.java:144)
> 	at com.cisco.ac.server.DBL.ACDatabaseParameters.getHuntGroups(ACDatabaseParameters.java:77)
> 	at com.cisco.ac.server.telephony.ACServerTelephony.init(ACServerTelephony.java:51)
> 	at com.cisco.ac.server.ACMain.startACServer(ACMain.java:123)
> 	at com.cisco.ac.server.ACMain.main(ACMain.java:40)
> 
>    39 09/22 08:27:25.869 ERROR HuntGroups:
> HuntGroupsUsers:
> pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
> Sessions:
> Active Sessions:
> Deactive Sessions:
> 
>    40 09/22 08:27:58.867 ERROR java.lang.NullPointerException
> 	at java.text.CollationElementIterator.<init>(CollationElementIterator.java:102)
> 	at java.text.RuleBasedCollator.getCollationElementIterator(RuleBasedCollator.java:358)
> 	at java.text.RuleBasedCollator.compare(RuleBasedCollator.java:405)
> 	at java.text.Collator.equals(Collator.java:322)
> 	at com.cisco.ac.server.DBL.ACHuntGroup.setOwner(ACHuntGroup.java:164)
> 	at com.cisco.ac.server.DBL.ACHuntGroup.<init>(ACHuntGroup.java:64)
> 	at com.cisco.ac.server.DBL.ACDatabaseParameters.loadPilotTable(ACDatabaseParameters.java:144)
> 	at com.cisco.ac.server.DBL.ACDatabaseParameters.getHuntGroups(ACDatabaseParameters.java:77)
> 	at com.cisco.ac.server.telephony.ACServerTelephony.pilotChanged(ACServerTelephony.java:257)
> 	at com.cisco.ac.server.telephony.ACServerTelephony.databaseChanged(ACServerTelephony.java:195)
> 	at com.cisco.ac.server.DBL.ACDatabaseParameters.informDBCNListerners(ACDatabaseParameters.java:911)
> 	at com.cisco.ac.server.DBL.ACDatabaseParameters.pilotPointChanged(ACDatabaseParameters.java:951)
> 	at com.cisco.ac.server.DBL.ACDatabaseParameters.pilotPointChanged(ACDatabaseParameters.java:947)
> 	at com.cisco.ac.server.DBL.DBCN.ACDBCNAction.run(ACDBCNAction.java:26)
> 
>    41 09/22 08:27:58.867 ERROR HuntGroups:
> HuntGroupsUsers:
> pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
> Sessions:
> Active Sessions:
> Deactive Sessions:
> 
>    42 09/22 08:28:17.975 ERROR HuntGroups:
> HuntGroupsUsers:
> pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
> Sessions:
> Active Sessions:
>     SessionID: 1, CMName: CM_wesdcm1, IsLocal: true, UserName: svoll, Device: SEP000DBD4B3289, DeactivateReason: 0, Active: true, OnLine: false, QueryDirectory: false, QueryUsers: false, KeepAliveTimerCount: 0, CreationTime: 1064244482483
> Deactive Sessions:
> 
>    43 09/22 09:47:13.831  INFO **** Switching log output to file: c:\Program Files\Cisco\Trace\TCD\tcdsrv.txt ****
>    44 09/22 09:47:14.633  INFO ServerClient get telephony parameters
>    45 09/22 09:47:14.643  INFO ServerClient get CM group: device: SEP000DBD4B3289
>    46 09/22 09:48:14.771  INFO ServerClient get telephony parameters
>    47 09/22 09:48:14.771  INFO ServerClient get CM group: device: SEP000DBD4B3289
>    48 09/22 09:49:12.595 DEBUG Writing user (svoll) data to directory...
>    49 09/22 09:49:12.876 DEBUG Writing user (svoll) data to directory done.
>    50 09/22 09:49:12.886  INFO ServerClient Logout SessionNumber: 6 RemoveSession: true
>    51 09/22 09:49:12.886 DEBUG ServerClient Logout SessionNumber: 6 exists
>    52 09/22 09:49:12.886  INFO ServerClient Go Offline SessionNumber: 6
>    53 09/22 09:49:12.886 DEBUG ServerClient goOffline SessionNumber: 6 exists
>    54 09/22 09:49:12.896  INFO ACServerTelephony user offline : svoll
>    55 09/22 09:49:12.896  INFO ACServerTelephony subscribeLineState 
>    56 09/22 09:49:12.896 DEBUG Added server subscription: server: CM_wesdcm1, count: 0
>    57 09/22 09:49:12.896  INFO ACInterServerConnections informSubscribeLSS server: CM_wesdcm1 Servers: CM_wesdcm2, 
>    58 09/22 09:49:12.896  INFO ACInterServerConnections informUserOffline username: svoll Servers: CM_wesdcm2, 
>    59 09/22 09:49:12.896  INFO ACInterServerComm: server : CM_wesdcm2
>    60 09/22 09:49:12.896 DEBUG Added server subscription: server: CM_wesdcm2, count: 0
>    61 09/22 09:49:12.896  INFO ACServerClientKeepAlive stop the timer
>    62 09/22 09:49:12.906  INFO ACServerTelephony user logout : svoll
>    63 09/22 09:49:12.906  INFO ACInterServerConnections informUserLogout username: svoll Servers: CM_wesdcm2, 
>    64 09/22 09:49:12.966 DEBUG ACInterServerComm HGLineStateChanged: numEntries: 0
>    65 09/22 09:49:12.996 DEBUG Sending 0 HG linestate(s) sent to server: CM_wesdcm2
>    66 09/22 09:49:12.996  INFO ACInterServerConnections: sendLineStateToServer(): CM_wesdcm2
>    67 09/22 09:49:12.996 DEBUG Updating 0 HG linestate(s) sent locally
>    68 09/22 09:49:15.499  INFO ServerClient GetDevices DN: 5961
>    69 09/22 09:49:15.590  INFO ServerClient getCMList
>    70 09/22 09:49:15.600 DEBUG Exception: java.io.FileNotFoundException: etc\DialRules.xml (The system cannot find the file specified): Reading file: etc\DialRules.xml, offset: 0, len: 100000
>    71 09/22 09:49:15.660 DEBUG Exception: java.io.FileNotFoundException: etc\VoiceMailProfiles.xml (The system cannot find the file specified): Reading file: etc\VoiceMailProfiles.xml, offset: 0, len: 100000
>    72 09/22 09:49:19.685  INFO ServerClient Login: Username: svoll Device: SEP000DBD4B3289 ForceLogout: false
>    73 09/22 09:49:19.866  INFO ServerClient username: svoll creating new session
>    74 09/22 09:49:19.866  INFO ACServerClientSessionManager create new session for: svoll CM: CM_wesdcm1 snum: 7
>    75 09/22 09:49:19.866  INFO ACServerClientKeepAlive start the timer
>    76 09/22 09:49:19.876  INFO ACServerTelephony user login : svoll
>    77 09/22 09:49:19.876  INFO ACInterServerConnections informUserLogin username: svoll Servers: CM_wesdcm2, 
>    78 09/22 09:49:20.086 DEBUG Reading user (svoll) attributes from directory...
>    79 09/22 09:49:20.176 DEBUG Reading user (svoll) attributes from directory done
>    80 09/22 09:49:20.797  INFO ServerClient Get Lines Device : SEP000DBD4B3289
>    81 09/22 09:49:21.128  INFO ServerClient getSupportPark().
>    82 09/22 09:49:21.338  INFO ServerClient getSupportPark().
>    83 09/22 09:49:21.348  INFO ServerClient getSupportPark().
>    84 09/22 09:49:21.678 DEBUG ServerClient subscribelinestate: inet address: 198.237.183.2 Port: 4176
>    85 09/22 09:49:21.678 DEBUG Added client subscription: 198.237.183.2, count: 2
>    86 09/22 09:49:21.678  INFO ACInterServerConnections informSubscribeLSS inet address: 198.237.183.2 Port: 4176 Servers: CM_wesdcm2, 
>    87 09/22 09:49:21.708 DEBUG Sending 0 linestate(s) sent to client: 198.237.183.2
>    88 09/22 09:49:21.799  INFO ServerClient getSupportPark().
>    89 09/22 09:49:21.799  INFO ServerClient getSupportPark().
>    90 09/22 09:49:21.919  INFO ServerClient get server version
>    91 09/22 09:49:21.919  INFO ServerClient getUserInfo SessionNumber: 7
>    92 09/22 09:49:21.929  INFO ServerClient getSupportPark().
>    93 09/22 09:49:22.309  INFO ServerClient get telephony parameters
>    94 09/22 09:49:22.319  INFO ServerClient get CM group: device: SEP000DBD4B3289
>    95 09/22 09:49:22.680 DEBUG Exception: java.io.FileNotFoundException: UserLists\CorporateDirectory.txt (The system cannot find the file specified): Reading file: UserLists\CorporateDirectory.txt, offset: 0, len: 100000
>    96 09/22 09:49:26.185  INFO ServerClient GoOnline SessionNumber: 7
>    97 09/22 09:49:26.185 DEBUG ServerClient goOnline SessionNumber: 7 exists
>    98 09/22 09:49:26.185  INFO ACServerTelephony user online : svoll
>    99 09/22 09:49:26.185  INFO ACServerTelephony subscribeLineState 
>   100 09/22 09:49:26.185 DEBUG Added server subscription: server: CM_wesdcm1, count: 0
>   101 09/22 09:49:26.185  INFO ACInterServerConnections informSubscribeLSS server: CM_wesdcm1 Servers: CM_wesdcm2, 
>   102 09/22 09:49:26.185  INFO ACInterServerConnections informUserOnline username: svoll Servers: CM_wesdcm2, 
>   103 09/22 09:49:26.185  INFO ACInterServerComm: server : CM_wesdcm2
>   104 09/22 09:49:26.185 DEBUG Added server subscription: server: CM_wesdcm2, count: 0
>   105 09/22 09:49:26.285 DEBUG Sending 0 HG linestate(s) sent to server: CM_wesdcm2
>   106 09/22 09:49:26.285  INFO ACInterServerConnections: sendLineStateToServer(): CM_wesdcm2
>   107 09/22 09:49:26.285 DEBUG Updating 0 HG linestate(s) sent locally
>   108 09/22 09:49:26.305 DEBUG ACInterServerComm HGLineStateChanged: numEntries: 0
> 
> -----Original Message-----
> From: Wes Sisk [mailto:wsisk@cisco.com] 
> Sent: Monday, September 22, 2003 9:17 AM
> To: Voll, Scott; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] Help with attendant console
> 
> Scott,
> 
> Have you looked at the Detailed TCDSrv traces?  I'm guessing that
> your AC user might now have CTI Use Enabled checked in the user config.
> 
> /Wes
> 
> 
>>-----Original Message-----
>>From: cisco-voip-bounces@puck.nether.net
>>[mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
>>Sent: Monday, September 22, 2003 12:13 PM
>>To: cisco-voip@puck.nether.net
>>Subject: [cisco-voip] Help with attendant console
>>
>>
>>I'm trying to get attendant console working(1.2.1).  Have
>>followed the directions and I'm getting an error, "Initialization
>>of call control failed. Retrying"  when I log onto the console on
>>the PC.  Any Idea's?  I have deleted and remade all console
>>users, Pilot points, Hunt groups, and the AC user.  I have also
>>stopped and restarted the TCD and CTI services.  Nothing will get
>>rid of the error.  Anyone come up against this before?  Any help
>>would be appreciated.  FYI I'm running CM 3.3.3sr1.
>>
>>Scott
>>
>>Scott Voll
>>Network Analyst, CCNA
>>Willamette ESD
>>Phone: (503) 385-4571
>>scott.voll@wesd.org
>> 
>>
>>
>>_______________________________________________
>>cisco-voip mailing list
>>cisco-voip@puck.nether.net
>>https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 

From rowdy at zenon.net  Tue Sep 23 14:49:33 2003
From: rowdy at zenon.net (Vadim Shesterin)
Date: Tue Sep 23 05:49:39 2003
Subject: [cisco-voip] Fax over Cisco VoIP
In-Reply-To: <27BBC8307A2C994092D90693E15CDA94E639DD@irvexchange.datum.com>
References: <27BBC8307A2C994092D90693E15CDA94E639DD@irvexchange.datum.com>
Message-ID: <3F70172D.1080502@zenon.net>

Hi, Steve,

As I knew Callmanager 3.xx can work only with Fax Pass-Through mode.
It don't support T.38

For configuring Fax Pass-Through mode on dial-peers see fillowing links:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/pt.htm

Steve Covin wrote:
> I'm very new to asking questions on the list.
> Have there been any issues with the Fax and VoIP with the Call manager 3.xx?
> thanks
> 
> Steve Covin
> Symmetricom Austin
> Office: 512-721-4527 
> Cell :   512-694-0981

Regards,
-- 
Vadim Shesterin, IPNOC, Zenon N.S.P.
phone: +7 (095) 956-4035

From Scott.Voll at wesd.org  Tue Sep 23 08:54:47 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Sep 23 10:54:53 2003
Subject: [cisco-voip] Help with attendant console
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F0120D4AB@venus.wesd.org>

Wes---

YOU ARE THE MAN!!!!!!!!!!!!!

YOU ROCK.   THANKS SO MUCH!!!!!!!!!!

The subscriber was not in the CM group.  Soon as I put it in the CM
Group it works.  

Thanks for you time and Help!

Scott Voll
Network Analyst, CCNA
Willamette ESD
Scott.voll@wesd.org


-----Original Message----- From: Wes Sisk [mailto:wsisk@cisco.com] 

Scott,

Doing some checking, found 1 prior case with all the same symptoms.
Here was their resolution:

Overnight, the customer discovered that the primary call manager 
(subscriber) had been removed from the CM group used by the phones.   It

was still in the cluster but devices were not receiving the address of 
the primary in the tftp load.   When the customer added the primary ccm 
back into the group, AC 1.1(3) started working again.

This just might be the case based on notes of my previous testing with
AC:

This information is applicable to AC 1.1(3).

The first time ACClient connects to an instance of ACserver
it downloads the servers listed in the device pool of the attendant
phone, uses that as the list of call processing servers (where
the client looks for CTIManager to be running), and caches the list
in the file
C:\Program Files\Cisco\Call Manager Attendant
Console\data\GlobalSettings.xml

The ACServer inspects the Callmanager table in the current CCM database
and uses the list of CCM servers ad the list of servers where the
ACServer process (tcdsrv.exe) should be active.

In default configurations where CTIManager and TCDSrv are running
on all nodes in the CCM cluster, the only client configuration required
to ensure redundancy for the ACClient is configuring the ip address of
1 server running TCDSrv.exe in the ACclient under settings->Basic->
"Attendant Server".

If for some reason TCDSrv and CTIManager are not running on all nodes
in the cluster, it will be necessary to manually populate the
"Call Processing Server", Adanced Tab, Attendant settings with a
comma separated list of the servers in the cluster with an active
CTIManager process.

If the IPAddresses of the CMServers change or
If the device pool of the Attendant phone changes.
It is necessary to close and open the ACClient so ACclient can
re-download the list of servers.  Note it may be necessary to change
the ipaddress in the settings->Basic->"Attendant Server" if the
IP's of all nodes in cluster changed.

/Wes





Voll, Scott wrote:
> Wes-- I'm assuming that you meant Might Not have CTI Use Enabled.  I
do have it checked.  
> 
> Here is the trace of TCD Detailed.  I do see some errors but not sure
what it means.  See below.
> 
> Thanks
> 
> Scott
> 
> PS.  The corporate directory works and so does the speed dials and
also shows on hook and off.  
> 
>     1 09/22 08:27:22.314  INFO 
>     2 09/22 08:27:22.324  INFO **************************************
>     3 09/22 08:27:22.324  INFO ****** Attendant Console Server ******
>     4 09/22 08:27:22.324  INFO * Version  : 1.2(1)
>     5 09/22 08:27:22.324  INFO * Built on : Fri May  2 15:51:53 PDT
2003
>     6 09/22 08:27:22.324  INFO **************************************
>     7 09/22 08:27:22.324  INFO 
>     8 09/22 08:27:22.384 DEBUG ACBootstrap: Initialization started
>     9 09/22 08:27:22.404 DEBUG _ldapUrl:ldap://neptune.wesd.org:389
>    10 09/22 08:27:22.404 DEBUG _directoryManagerDN:cn=Administrator,
cn=Users, dc=wesd,dc=org
>    11 09/22 08:27:22.484 DEBUG _directoryManagerPassword:******
>    12 09/22 08:27:22.484 DEBUG _ciscoBase:ou=Cisco, dc=wesd,dc=org
>    13 09/22 08:27:22.484 DEBUG _ccnBase:ou=CCN,ou=Cisco,
dc=wesd,dc=org
>    14 09/22 08:27:22.484 DEBUG _directoryType:ADS
>    15 09/22 08:27:22.985 DEBUG _userBase:cn=Users, dc=wesd,dc=org
>    16 09/22 08:27:22.985 DEBUG _userOC:User
>    17 09/22 08:27:22.985 DEBUG _profilesBase:ou=profiles, ou=CCN
>    18 09/22 08:27:22.985 DEBUG _internalUniqueAttributeName:cn
>    19 09/22 08:27:23.075 DEBUG ACBootstrap: Computing jtapi.jar's
checksum...
>    20 09/22 08:27:23.075 DEBUG ACBootstrap: jtapi.jar's checksum:
354288330
>    21 09/22 08:27:23.075 DEBUG ACBootstrap: Initialization completed
>    22 09/22 08:27:23.075 DEBUG ACServer.properties exists. Skipping
creation.
>    23 09/22 08:27:23.075  INFO Checking connectivity with database...
>    24 09/22 08:27:23.085  INFO ACCallManagerDB dbConection:
DSN=CiscoCallManager;SERVER=WESDCM1;DATABASE=CCM0301;Trusted_Connection=
yes
>    25 09/22 08:27:23.085 DEBUG ACCallManagerDB dbConection: 0
>    26 09/22 08:27:23.085 DEBUG      SERVER=WESDCM1
>    27 09/22 08:27:23.085 DEBUG    DATABASE=CCM0301
>    28 09/22 08:27:23.085  INFO ACCallManagerDB dbConection:
DSN=CiscoCallManager;SERVER=wesdcm2;DATABASE=CCM0301;Trusted_Connection=
yes
>    29 09/22 08:27:23.085 DEBUG ACCallManagerDB dbConection: 1
>    30 09/22 08:27:23.085 DEBUG      SERVER=wesdcm2
>    31 09/22 08:27:23.095 DEBUG    DATABASE=CCM0301
>    32 09/22 08:27:23.095 DEBUG Checking if primary database is up and
running...
>    33 09/22 08:27:23.095  INFO Attempting to access (publisher) db on:
WESDCM1
>    34 09/22 08:27:23.335  INFO Successfully opened (publisher) db on:
WESDCM1
>    35 09/22 08:27:23.335  INFO Checking connectivity with
database...done
>    36 09/22 08:27:23.836 DEBUG ACCallManagers: CM_wesdcm1 is
thisServer
>    37 09/22 08:27:24.828 DEBUG New trace level: ERROR
>    38 09/22 08:27:25.859 ERROR java.lang.NullPointerException
> 	at
java.text.CollationElementIterator.<init>(CollationElementIterator.java:
102)
> 	at
java.text.RuleBasedCollator.getCollationElementIterator(RuleBasedCollato
r.java:358)
> 	at
java.text.RuleBasedCollator.compare(RuleBasedCollator.java:405)
> 	at java.text.Collator.equals(Collator.java:322)
> 	at
com.cisco.ac.server.DBL.ACHuntGroup.setOwner(ACHuntGroup.java:164)
> 	at
com.cisco.ac.server.DBL.ACHuntGroup.<init>(ACHuntGroup.java:64)
> 	at
com.cisco.ac.server.DBL.ACDatabaseParameters.loadPilotTable(ACDatabasePa
rameters.java:144)
> 	at
com.cisco.ac.server.DBL.ACDatabaseParameters.getHuntGroups(ACDatabasePar
ameters.java:77)
> 	at
com.cisco.ac.server.telephony.ACServerTelephony.init(ACServerTelephony.j
ava:51)
> 	at com.cisco.ac.server.ACMain.startACServer(ACMain.java:123)
> 	at com.cisco.ac.server.ACMain.main(ACMain.java:40)
> 
>    39 09/22 08:27:25.869 ERROR HuntGroups:
> HuntGroupsUsers:
> pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
> Sessions:
> Active Sessions:
> Deactive Sessions:
> 
>    40 09/22 08:27:58.867 ERROR java.lang.NullPointerException
> 	at
java.text.CollationElementIterator.<init>(CollationElementIterator.java:
102)
> 	at
java.text.RuleBasedCollator.getCollationElementIterator(RuleBasedCollato
r.java:358)
> 	at
java.text.RuleBasedCollator.compare(RuleBasedCollator.java:405)
> 	at java.text.Collator.equals(Collator.java:322)
> 	at
com.cisco.ac.server.DBL.ACHuntGroup.setOwner(ACHuntGroup.java:164)
> 	at
com.cisco.ac.server.DBL.ACHuntGroup.<init>(ACHuntGroup.java:64)
> 	at
com.cisco.ac.server.DBL.ACDatabaseParameters.loadPilotTable(ACDatabasePa
rameters.java:144)
> 	at
com.cisco.ac.server.DBL.ACDatabaseParameters.getHuntGroups(ACDatabasePar
ameters.java:77)
> 	at
com.cisco.ac.server.telephony.ACServerTelephony.pilotChanged(ACServerTel
ephony.java:257)
> 	at
com.cisco.ac.server.telephony.ACServerTelephony.databaseChanged(ACServer
Telephony.java:195)
> 	at
com.cisco.ac.server.DBL.ACDatabaseParameters.informDBCNListerners(ACData
baseParameters.java:911)
> 	at
com.cisco.ac.server.DBL.ACDatabaseParameters.pilotPointChanged(ACDatabas
eParameters.java:951)
> 	at
com.cisco.ac.server.DBL.ACDatabaseParameters.pilotPointChanged(ACDatabas
eParameters.java:947)
> 	at
com.cisco.ac.server.DBL.DBCN.ACDBCNAction.run(ACDBCNAction.java:26)
> 
>    41 09/22 08:27:58.867 ERROR HuntGroups:
> HuntGroupsUsers:
> pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
> Sessions:
> Active Sessions:
> Deactive Sessions:
> 
>    42 09/22 08:28:17.975 ERROR HuntGroups:
> HuntGroupsUsers:
> pkid: 0x849B66247FFFD542B1805B3FAAB62AAE, userid: svoll, passwd: *****
> Sessions:
> Active Sessions:
>     SessionID: 1, CMName: CM_wesdcm1, IsLocal: true, UserName: svoll,
Device: SEP000DBD4B3289, DeactivateReason: 0, Active: true, OnLine:
false, QueryDirectory: false, QueryUsers: false, KeepAliveTimerCount: 0,
CreationTime: 1064244482483
> Deactive Sessions:
> 
>    43 09/22 09:47:13.831  INFO **** Switching log output to file:
c:\Program Files\Cisco\Trace\TCD\tcdsrv.txt ****
>    44 09/22 09:47:14.633  INFO ServerClient get telephony parameters
>    45 09/22 09:47:14.643  INFO ServerClient get CM group: device:
SEP000DBD4B3289
>    46 09/22 09:48:14.771  INFO ServerClient get telephony parameters
>    47 09/22 09:48:14.771  INFO ServerClient get CM group: device:
SEP000DBD4B3289
>    48 09/22 09:49:12.595 DEBUG Writing user (svoll) data to
directory...
>    49 09/22 09:49:12.876 DEBUG Writing user (svoll) data to directory
done.
>    50 09/22 09:49:12.886  INFO ServerClient Logout SessionNumber: 6
RemoveSession: true
>    51 09/22 09:49:12.886 DEBUG ServerClient Logout SessionNumber: 6
exists
>    52 09/22 09:49:12.886  INFO ServerClient Go Offline SessionNumber:
6
>    53 09/22 09:49:12.886 DEBUG ServerClient goOffline SessionNumber: 6
exists
>    54 09/22 09:49:12.896  INFO ACServerTelephony user offline : svoll
>    55 09/22 09:49:12.896  INFO ACServerTelephony subscribeLineState 
>    56 09/22 09:49:12.896 DEBUG Added server subscription: server:
CM_wesdcm1, count: 0
>    57 09/22 09:49:12.896  INFO ACInterServerConnections
informSubscribeLSS server: CM_wesdcm1 Servers: CM_wesdcm2, 
>    58 09/22 09:49:12.896  INFO ACInterServerConnections
informUserOffline username: svoll Servers: CM_wesdcm2, 
>    59 09/22 09:49:12.896  INFO ACInterServerComm: server : CM_wesdcm2
>    60 09/22 09:49:12.896 DEBUG Added server subscription: server:
CM_wesdcm2, count: 0
>    61 09/22 09:49:12.896  INFO ACServerClientKeepAlive stop the timer
>    62 09/22 09:49:12.906  INFO ACServerTelephony user logout : svoll
>    63 09/22 09:49:12.906  INFO ACInterServerConnections
informUserLogout username: svoll Servers: CM_wesdcm2, 
>    64 09/22 09:49:12.966 DEBUG ACInterServerComm HGLineStateChanged:
numEntries: 0
>    65 09/22 09:49:12.996 DEBUG Sending 0 HG linestate(s) sent to
server: CM_wesdcm2
>    66 09/22 09:49:12.996  INFO ACInterServerConnections:
sendLineStateToServer(): CM_wesdcm2
>    67 09/22 09:49:12.996 DEBUG Updating 0 HG linestate(s) sent locally
>    68 09/22 09:49:15.499  INFO ServerClient GetDevices DN: 5961
>    69 09/22 09:49:15.590  INFO ServerClient getCMList
>    70 09/22 09:49:15.600 DEBUG Exception:
java.io.FileNotFoundException: etc\DialRules.xml (The system cannot find
the file specified): Reading file: etc\DialRules.xml, offset: 0, len:
100000
>    71 09/22 09:49:15.660 DEBUG Exception:
java.io.FileNotFoundException: etc\VoiceMailProfiles.xml (The system
cannot find the file specified): Reading file:
etc\VoiceMailProfiles.xml, offset: 0, len: 100000
>    72 09/22 09:49:19.685  INFO ServerClient Login: Username: svoll
Device: SEP000DBD4B3289 ForceLogout: false
>    73 09/22 09:49:19.866  INFO ServerClient username: svoll creating
new session
>    74 09/22 09:49:19.866  INFO ACServerClientSessionManager create new
session for: svoll CM: CM_wesdcm1 snum: 7
>    75 09/22 09:49:19.866  INFO ACServerClientKeepAlive start the timer
>    76 09/22 09:49:19.876  INFO ACServerTelephony user login : svoll
>    77 09/22 09:49:19.876  INFO ACInterServerConnections
informUserLogin username: svoll Servers: CM_wesdcm2, 
>    78 09/22 09:49:20.086 DEBUG Reading user (svoll) attributes from
directory...
>    79 09/22 09:49:20.176 DEBUG Reading user (svoll) attributes from
directory done
>    80 09/22 09:49:20.797  INFO ServerClient Get Lines Device :
SEP000DBD4B3289
>    81 09/22 09:49:21.128  INFO ServerClient getSupportPark().
>    82 09/22 09:49:21.338  INFO ServerClient getSupportPark().
>    83 09/22 09:49:21.348  INFO ServerClient getSupportPark().
>    84 09/22 09:49:21.678 DEBUG ServerClient subscribelinestate: inet
address: 198.237.183.2 Port: 4176
>    85 09/22 09:49:21.678 DEBUG Added client subscription:
198.237.183.2, count: 2
>    86 09/22 09:49:21.678  INFO ACInterServerConnections
informSubscribeLSS inet address: 198.237.183.2 Port: 4176 Servers:
CM_wesdcm2, 
>    87 09/22 09:49:21.708 DEBUG Sending 0 linestate(s) sent to client:
198.237.183.2
>    88 09/22 09:49:21.799  INFO ServerClient getSupportPark().
>    89 09/22 09:49:21.799  INFO ServerClient getSupportPark().
>    90 09/22 09:49:21.919  INFO ServerClient get server version
>    91 09/22 09:49:21.919  INFO ServerClient getUserInfo SessionNumber:
7
>    92 09/22 09:49:21.929  INFO ServerClient getSupportPark().
>    93 09/22 09:49:22.309  INFO ServerClient get telephony parameters
>    94 09/22 09:49:22.319  INFO ServerClient get CM group: device:
SEP000DBD4B3289
>    95 09/22 09:49:22.680 DEBUG Exception:
java.io.FileNotFoundException: UserLists\CorporateDirectory.txt (The
system cannot find the file specified): Reading file:
UserLists\CorporateDirectory.txt, offset: 0, len: 100000
>    96 09/22 09:49:26.185  INFO ServerClient GoOnline SessionNumber: 7
>    97 09/22 09:49:26.185 DEBUG ServerClient goOnline SessionNumber: 7
exists
>    98 09/22 09:49:26.185  INFO ACServerTelephony user online : svoll
>    99 09/22 09:49:26.185  INFO ACServerTelephony subscribeLineState 
>   100 09/22 09:49:26.185 DEBUG Added server subscription: server:
CM_wesdcm1, count: 0
>   101 09/22 09:49:26.185  INFO ACInterServerConnections
informSubscribeLSS server: CM_wesdcm1 Servers: CM_wesdcm2, 
>   102 09/22 09:49:26.185  INFO ACInterServerConnections
informUserOnline username: svoll Servers: CM_wesdcm2, 
>   103 09/22 09:49:26.185  INFO ACInterServerComm: server : CM_wesdcm2
>   104 09/22 09:49:26.185 DEBUG Added server subscription: server:
CM_wesdcm2, count: 0
>   105 09/22 09:49:26.285 DEBUG Sending 0 HG linestate(s) sent to
server: CM_wesdcm2
>   106 09/22 09:49:26.285  INFO ACInterServerConnections:
sendLineStateToServer(): CM_wesdcm2
>   107 09/22 09:49:26.285 DEBUG Updating 0 HG linestate(s) sent locally
>   108 09/22 09:49:26.305 DEBUG ACInterServerComm HGLineStateChanged:
numEntries: 0
> 
> -----Original Message-----
> From: Wes Sisk [mailto:wsisk@cisco.com] 
> Sent: Monday, September 22, 2003 9:17 AM
> To: Voll, Scott; cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] Help with attendant console
> 
> Scott,
> 
> Have you looked at the Detailed TCDSrv traces?  I'm guessing that
> your AC user might now have CTI Use Enabled checked in the user
config.
> 
> /Wes
> 
> 
>>-----Original Message-----
>>From: cisco-voip-bounces@puck.nether.net
>>[mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
>>Sent: Monday, September 22, 2003 12:13 PM
>>To: cisco-voip@puck.nether.net
>>Subject: [cisco-voip] Help with attendant console
>>
>>
>>I'm trying to get attendant console working(1.2.1).  Have
>>followed the directions and I'm getting an error, "Initialization
>>of call control failed. Retrying"  when I log onto the console on
>>the PC.  Any Idea's?  I have deleted and remade all console
>>users, Pilot points, Hunt groups, and the AC user.  I have also
>>stopped and restarted the TCD and CTI services.  Nothing will get
>>rid of the error.  Anyone come up against this before?  Any help
>>would be appreciated.  FYI I'm running CM 3.3.3sr1.
>>
>>Scott
>>
>>Scott Voll
>>Network Analyst, CCNA
>>Willamette ESD
>>Phone: (503) 385-4571
>>scott.voll@wesd.org
>> 
>>
>>
>>_______________________________________________
>>cisco-voip mailing list
>>cisco-voip@puck.nether.net
>>https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 


From jeff at expertcity.com  Tue Sep 23 13:07:14 2003
From: jeff at expertcity.com (Jeff Behl)
Date: Tue Sep 23 15:07:10 2003
Subject: [cisco-voip] simple q
Message-ID: <3F7099E2.6040309@expertcity.com>

A branch office has a 2651XM w/PRI as their voicegw, controlled by a cm 
3.3 machine via MGCP.  I think i'm doing this the wrong way...

Their old analog line is going away but the Telco has agreed to route 
the old 4 digit extension down the PRI (they already have a new number 
which is being handled by the CM).  what's the easiest way to force the 
new DN to go to an existing DN?  I tried creating a simple route pattern 
with the old analog 4 digit DN as the pattern that has a Called Party 
Transform mask of the DN i want the call to go to, but this results in a 
fast busy.  I would rather not just make this another line on their phone.

tia



From wsisk at cisco.com  Tue Sep 23 17:49:00 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Tue Sep 23 16:49:07 2003
Subject: [cisco-voip] simple q
In-Reply-To: <3F7099E2.6040309@expertcity.com>
Message-ID: <ELEMLGDGDBLINAJEIOCLIEFIMOAA.wsisk@cisco.com>

Use a translation pattern to trasnform 1000 to 2000 and use
the correct CSS and partitions.

/Wes

> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Jeff Behl
> Sent: Tuesday, September 23, 2003 3:07 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] simple q
>
>
> A branch office has a 2651XM w/PRI as their voicegw, controlled by a cm
> 3.3 machine via MGCP.  I think i'm doing this the wrong way...
>
> Their old analog line is going away but the Telco has agreed to route
> the old 4 digit extension down the PRI (they already have a new number
> which is being handled by the CM).  what's the easiest way to force the
> new DN to go to an existing DN?  I tried creating a simple route pattern
> with the old analog 4 digit DN as the pattern that has a Called Party
> Transform mask of the DN i want the call to go to, but this results in a
> fast busy.  I would rather not just make this another line on their phone.
>
> tia
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From medley at mac.com  Tue Sep 23 17:51:37 2003
From: medley at mac.com (Tim Medley)
Date: Tue Sep 23 16:52:51 2003
Subject: [cisco-voip] simple q
References: <3F7099E2.6040309@expertcity.com>
Message-ID: <007b01c38214$7d2b1c80$4718c60a@medley>

Jeff,

There's a couple of ways to do this:

1) Create a translation pattern in CCM that matches the old number and
transforms it to the new number
2) Create a Device Profile in CCM with a line configured for the old number
and use call forward all to direct the call to the new number

tm


Tim Medley, CCDP, CCNP
IPT Consultant

http://www.iptnetworkers.com

----- Original Message ----- 
From: "Jeff Behl" <jeff@expertcity.com>
To: <cisco-voip@puck.nether.net>
Sent: Tuesday, September 23, 2003 3:07 PM
Subject: [cisco-voip] simple q


> A branch office has a 2651XM w/PRI as their voicegw, controlled by a cm
> 3.3 machine via MGCP.  I think i'm doing this the wrong way...
>
> Their old analog line is going away but the Telco has agreed to route
> the old 4 digit extension down the PRI (they already have a new number
> which is being handled by the CM).  what's the easiest way to force the
> new DN to go to an existing DN?  I tried creating a simple route pattern
> with the old analog 4 digit DN as the pattern that has a Called Party
> Transform mask of the DN i want the call to go to, but this results in a
> fast busy.  I would rather not just make this another line on their phone.
>
> tia
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From Scott.Voll at wesd.org  Tue Sep 23 15:25:55 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Sep 23 17:25:59 2003
Subject: [cisco-voip] Attendant console with 7960 and 6 lines
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDC6@venus.wesd.org>

What am I doing wrong?

CM 3.3.3sr1
Attendant console 1.2.1
Phone 7960 with 6 lines.  (lines 1-4 the main number 5900, line 5 the old number 1431, and line 6 personal extension for receptionist 5970)

Have followed the config in this link, http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_tech_note09186a008009409c.shtml and I no longer have the 5900 extension available.  It looks like this is an older document, so I'm not sure if something changed in CM 3.3.3sr1.  

Basicly I want the receptionist to have 5 lines to answer.  And be able to use attendant console.  What am I doing wrong.  

Scott Voll
Network Analyst, CCNA 
Willamette ESD
Phone: (503) 385-4571
scott.voll@wesd.org
?

From wsisk at cisco.com  Wed Sep 24 11:26:24 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Sep 24 11:26:33 2003
Subject: [cisco-voip] Attendant console with 7960 and 6 lines
In-Reply-To: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDC6@venus.wesd.org>
Message-ID: <ELEMLGDGDBLINAJEIOCLCEGOMOAA.wsisk@cisco.com>

Scott,

AC (specifically CTI or QBE, the protocol AC uses to communicate with CCM)
does not understand partitions, so it sees all 5 of the lines with the
number
"5900" as the same line which causes a great amount of confusion to users,
to
CCM, to AC, and everyone else I've spoken with.

You will need to setup the phone to have 6 unique DN's, then it will work.

/Wes

> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> Sent: Tuesday, September 23, 2003 5:26 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Attendant console with 7960 and 6 lines
>
>
> What am I doing wrong?
>
> CM 3.3.3sr1
> Attendant console 1.2.1
> Phone 7960 with 6 lines.  (lines 1-4 the main number 5900, line 5
> the old number 1431, and line 6 personal extension for receptionist 5970)
>
> Have followed the config in this link,
> http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/prod
ucts_tech_note09186a008009409c.shtml and I no longer have the 5900 extension
available.  It looks like this is an older document, so I'm not sure if
something changed in CM 3.3.3sr1.

Basicly I want the receptionist to have 5 lines to answer.  And be able to
use attendant console.  What am I doing wrong.

Scott Voll
Network Analyst, CCNA
Willamette ESD
Phone: (503) 385-4571
scott.voll@wesd.org
?

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

From WCarter at sentinel.com  Wed Sep 24 11:50:39 2003
From: WCarter at sentinel.com (Carter, Bill)
Date: Wed Sep 24 11:50:45 2003
Subject: [cisco-voip] Attendant console with 7960 and 6 lines
Message-ID: <F4059FD8E8F9D141BE8C48C0DCF1FA157942C7@rodan.sentinel.com>

Typically we assign

5900 line 1
15900 line 2
25900 line 3
35900 line 4

all line are in Line1PT

-----Original Message-----
From: Wes Sisk [mailto:wsisk@cisco.com]
Sent: Wednesday, September 24, 2003 10:26 AM
To: Voll, Scott; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Attendant console with 7960 and 6 lines


Scott,

AC (specifically CTI or QBE, the protocol AC uses to communicate with CCM)
does not understand partitions, so it sees all 5 of the lines with the
number
"5900" as the same line which causes a great amount of confusion to users,
to
CCM, to AC, and everyone else I've spoken with.

You will need to setup the phone to have 6 unique DN's, then it will work.

/Wes

> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> Sent: Tuesday, September 23, 2003 5:26 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Attendant console with 7960 and 6 lines
>
>
> What am I doing wrong?
>
> CM 3.3.3sr1
> Attendant console 1.2.1
> Phone 7960 with 6 lines.  (lines 1-4 the main number 5900, line 5
> the old number 1431, and line 6 personal extension for receptionist 5970)
>
> Have followed the config in this link,
> http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/prod
ucts_tech_note09186a008009409c.shtml and I no longer have the 5900 extension
available.  It looks like this is an older document, so I'm not sure if
something changed in CM 3.3.3sr1.

Basicly I want the receptionist to have 5 lines to answer.  And be able to
use attendant console.  What am I doing wrong.

Scott Voll
Network Analyst, CCNA
Willamette ESD
Phone: (503) 385-4571
scott.voll@wesd.org
?

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

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

From WCarter at sentinel.com  Wed Sep 24 11:53:52 2003
From: WCarter at sentinel.com (Carter, Bill)
Date: Wed Sep 24 11:53:57 2003
Subject: [cisco-voip] 7920's with Orinoco AP's
Message-ID: <F4059FD8E8F9D141BE8C48C0DCF1FA157942C8@rodan.sentinel.com>

Has anyone be able to make the 7920's work with Orinoco AP's.  Officially 7920 are not supported on any AP's other than Cisco, but I can't find anything that says 7920/Aironet uses anything no standard.  I know I can't create a separate SSID/Voice VLAN.  I don't know if I have any QoS available.



Bill Carter
CCIE 5022
Senior Network Analyst
Sentinel Technologies, Inc.
bcarter@sentinel.com



From mtkalec at cs.hr  Thu Sep 25 05:11:11 2003
From: mtkalec at cs.hr (Miljenko Tkalec)
Date: Thu Sep 25 05:08:52 2003
Subject: [cisco-voip] external caller's name 
Message-ID: <B099C6B93A3D2E40B21BD1407C9D639134C1@houston.intranet.cs.hr>

I was wondering how to have a IP phone look up external caller's number
and display the associated name info, much in the same way it is done
for internal extension numbers. Is there a way this can be done? 
What I mean is: when you receive a call from some local extension, the
name of the corresponding user is displayed. Not so when you are called
from the outside PSTN. What I'd like to do is look up caller numbers `on
the fly' from Cisco Personal Address Book (or other directory) and show
the corresponding relevant info (name, usually) in place of the number.

Miljenko Tkalec, CCNP
CS Computer Systems

From wsisk at cisco.com  Thu Sep 25 17:35:25 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Sep 25 17:35:33 2003
Subject: [cisco-voip] external caller's name
In-Reply-To: <B099C6B93A3D2E40B21BD1407C9D639134C1@houston.intranet.cs.hr>
References: <B099C6B93A3D2E40B21BD1407C9D639134C1@houston.intranet.cs.hr>
Message-ID: <3F735F9D.7020203@cisco.com>

Unfortunately this is not currently possible.  It is a feature request 
that is being roadmapped.  If you can ping your Cisco account team and 
express your interest in this feature we can work to get it implemented 
more quickly.

/Wes

Miljenko Tkalec wrote:

>I was wondering how to have a IP phone look up external caller's number
>and display the associated name info, much in the same way it is done
>for internal extension numbers. Is there a way this can be done? 
>What I mean is: when you receive a call from some local extension, the
>name of the corresponding user is displayed. Not so when you are called
>from the outside PSTN. What I'd like to do is look up caller numbers `on
>the fly' from Cisco Personal Address Book (or other directory) and show
>the corresponding relevant info (name, usually) in place of the number.
>
>Miljenko Tkalec, CCNP
>CS Computer Systems
>
>_______________________________________________
>cisco-voip mailing list
>cisco-voip@puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
>  
>

From ward_knutson at educ8.org  Thu Sep 25 17:46:33 2003
From: ward_knutson at educ8.org (Ward Knutson)
Date: Thu Sep 25 17:46:30 2003
Subject: [cisco-voip] MOH
Message-ID: <DF595BF2E0B4CB4F893BFFD9A0B1BF3A0580C5@701-mail.bismarck.k12.nd.us>

I need some help with my moh feature.  I am running CallManager 3.3.  I
want to turn the moh feature off.  The moh audio source configuration
has the sample audio source and the fixed sources available.  I went
into my device pool settings and set the user hold moh audio source to
none and the network hold moh audio source to none and the blasted music
continues!!!  What am I missing?

 

 

 

Ward Knutson

Network Coordinator

Bismarck Public Schools

Bismarck, ND 58501

701-355-3037

To err is human, to really foul things up requires a computer.

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030925/aa545cc7/attachment.htm
From medley at mac.com  Thu Sep 25 18:45:35 2003
From: medley at mac.com (Tim Medley)
Date: Thu Sep 25 18:47:14 2003
Subject: [cisco-voip] MOH
References: <DF595BF2E0B4CB4F893BFFD9A0B1BF3A0580C5@701-mail.bismarck.k12.nd.us>
Message-ID: <010e01c383b6$c0162870$5100a8c0@medley>

make sure that you have the moh audio sources set to none at the device level and the line level. Also check ans see that your gateways have the moh source set to none.

You may need to reboot the phones for the change to take effect.

You could also stop the MOH Server and/or remove the moh sources.

tm

Tim Medley, CCDP, CCNP
IPT Consultant

http://www.iptnetworkers.com
  ----- Original Message ----- 
  From: Ward Knutson 
  To: cisco-voip@puck.nether.net 
  Sent: Thursday, September 25, 2003 5:46 PM
  Subject: [cisco-voip] MOH


  I need some help with my moh feature.  I am running CallManager 3.3.  I want to turn the moh feature off.  The moh audio source configuration has the sample audio source and the fixed sources available.  I went into my device pool settings and set the user hold moh audio source to none and the network hold moh audio source to none and the blasted music continues!!!  What am I missing?







  Ward Knutson

  Network Coordinator

  Bismarck Public Schools

  Bismarck, ND 58501

  701-355-3037

  To err is human, to really foul things up requires a computer.







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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip@puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030925/046fee2c/attachment.htm
From wsisk at cisco.com  Thu Sep 25 20:27:08 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Sep 25 20:27:16 2003
Subject: [cisco-voip] MOH
In-Reply-To: <DF595BF2E0B4CB4F893BFFD9A0B1BF3A0580C5@701-mail.bismarck.k12.nd.us>
Message-ID: <ELEMLGDGDBLINAJEIOCLIEJHMOAA.wsisk@cisco.com>

To disable MOH, just take your music on hold servers and put them into
MediaResourceGroups the phones and gateways do not have access to.

1. create MRG called NO_MOH
2. add all MOH servers to this MRG
3. DO not add this MRG to any MRGL.

This will make the MOH servers not available to endpoints so tone on hold
will kick in.

/Wes
  -----Original Message-----
  From: cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Ward Knutson
  Sent: Thursday, September 25, 2003 5:47 PM
  To: cisco-voip@puck.nether.net
  Subject: [cisco-voip] MOH


  I need some help with my moh feature.  I am running CallManager 3.3.  I
want to turn the moh feature off.  The moh audio source configuration has
the sample audio source and the fixed sources available.  I went into my
device pool settings and set the user hold moh audio source to none and the
network hold moh audio source to none and the blasted music continues!!!
What am I missing?







  Ward Knutson

  Network Coordinator

  Bismarck Public Schools

  Bismarck, ND 58501

  701-355-3037

  To err is human, to really foul things up requires a computer.




-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030925/c44b99c2/attachment.htm
From tgrace at tgrace.com  Thu Sep 18 22:51:24 2003
From: tgrace at tgrace.com (tgrace)
Date: Fri Sep 26 10:51:22 2003
Subject: [cisco-voip] PRI using a Nortel QPC757 DCHI and Cisco
	NM-HDV/VWIC-1MFT-T1
Message-ID: <8B65FD9AD3BE7A4791D3047AD63FB9700834F1D2@tmg-exch2>

I'm having difficulty keepinga PRI TIE line up between our 81C and a Cisco
2621XM router. The PRI seems to go down shortly after a disconnect from the
first incoming call to the router from the PBX or PSTN or after a dozen
quick succession calls to PBX extensions. When this happens I can see the
router TX'ing SABME's but getting no response from the PBX. At other times
the link will be up but I will get an INFO message back from the PBX that
the "identified channel does not exist" when I try to place an outbound
call. There also seems to be some issue with disconnects with 5ESS. On a
number of occassions my Nortel phone continued ringing after going onhook on
the Cisco phone. This isn't consistent and happens perhaps 1 out 10 calls.

Initially tried using the DMS100 protocol with the PBX as net and the router
as USR. Also did the reverse and then tried configuring both as 5ESS. Same
type of problems.

Anyone ever run into something like this?


From tgrace at tgrace.com  Fri Sep 19 11:16:45 2003
From: tgrace at tgrace.com (Terry Grace)
Date: Fri Sep 26 10:51:24 2003
Subject: [cisco-voip] RE: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
	NM-HDV/VWIC-1MFT-T1
In-Reply-To: <8B65FD9AD3BE7A4791D3047AD63FB970089A1DC9@tmg-exch2>
Message-ID: <8B65FD9AD3BE7A4791D3047AD63FB9700834F1D8@tmg-exch2>

We're using mgcp. Don't have the qsig package on the PBX.

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll@wesd.org] 
Sent: Friday, September 19, 2003 11:10 AM
To: tgrace; cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: RE: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
NM-HDV/VWIC-1MFT-T1


What are you running, H323,SIP,MGCP??  Can you use Qsig?  

--Scott

-----Original Message-----
From: tgrace [mailto:tgrace@tgrace.com] 
Sent: Thursday, September 18, 2003 7:51 PM
To: cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: [nsp] PRI using a Nortel QPC757 DCHI and Cisco NM-HDV/VWIC-1MFT-T1

I'm having difficulty keepinga PRI TIE line up between our 81C and a Cisco
2621XM router. The PRI seems to go down shortly after a disconnect from the
first incoming call to the router from the PBX or PSTN or after a dozen
quick succession calls to PBX extensions. When this happens I can see the
router TX'ing SABME's but getting no response from the PBX. At other times
the link will be up but I will get an INFO message back from the PBX that
the "identified channel does not exist" when I try to place an outbound
call. There also seems to be some issue with disconnects with 5ESS. On a
number of occassions my Nortel phone continued ringing after going onhook on
the Cisco phone. This isn't consistent and happens perhaps 1 out 10 calls.

Initially tried using the DMS100 protocol with the PBX as net and the router
as USR. Also did the reverse and then tried configuring both as 5ESS. Same
type of problems.

Anyone ever run into something like this?


_______________________________________________
cisco-nsp mailing list  cisco-nsp@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-nsp
archive at http://puck.nether.net/pipermail/cisco-nsp/


From cwaltzer at aventuranetworks.com  Mon Sep 22 10:11:10 2003
From: cwaltzer at aventuranetworks.com (Craig A. Waltzer)
Date: Fri Sep 26 10:51:26 2003
Subject: [cisco-voip] ATA 186 Firmware
Message-ID: <00c101c38113$5ff96f10$0201a8c0@oemcomputer>

Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 4890 bytes
Desc: not available
Url : https://puck.nether.net/pipermail/cisco-voip/attachments/20030922/5ce087ae/attachment.jpg
From tomas at tronet.com  Fri Sep 26 11:15:31 2003
From: tomas at tronet.com (Tomas Daniska)
Date: Fri Sep 26 11:15:36 2003
Subject: [cisco-voip] external caller's name
Message-ID: <A44DA7EDD8262343B02C64AF7E063A07BCB407@kenya.ba.tronet.sk>


mr. Tkalec,


please contact me off-list - i believe we do have exactly what you need


--
 
Tomas Daniska
systems engineer
Tronet Computer Networks
Plynarenska 5, 829 75 Bratislava, Slovakia
tel: +421 2 58224111, fax: +421 2 58224199
 
A transistor protected by a fast-acting fuse will protect the fuse by
blowing first.




> 
> Message: 1
> Date: Thu, 25 Sep 2003 17:35:25 -0400
> From: Wes Sisk <wsisk@cisco.com>
> Subject: Re: [cisco-voip] external caller's name
> To: Miljenko Tkalec <mtkalec@cs.hr>
> Cc: cisco-voip@puck.nether.net
> Message-ID: <3F735F9D.7020203@cisco.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> Unfortunately this is not currently possible.  It is a 
> feature request 
> that is being roadmapped.  If you can ping your Cisco account 
> team and 
> express your interest in this feature we can work to get it 
> implemented 
> more quickly.
> 
> /Wes
> 
> Miljenko Tkalec wrote:
> 
> >I was wondering how to have a IP phone look up external 
> caller's number 
> >and display the associated name info, much in the same way 
> it is done 
> >for internal extension numbers. Is there a way this can be 
> done? What I 
> >mean is: when you receive a call from some local extension, 
> the name of 
> >the corresponding user is displayed. Not so when you are called from 
> >the outside PSTN. What I'd like to do is look up caller 
> numbers `on the 
> >fly' from Cisco Personal Address Book (or other directory) 
> and show the 
> >corresponding relevant info (name, usually) in place of the number.
> >
> >Miljenko Tkalec, CCNP
> >CS Computer Systems

From Scott.Voll at wesd.org  Fri Sep 26 12:18:25 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Sep 26 12:18:35 2003
Subject: [cisco-voip] RE: [nsp] PRI using a Nortel QPC757 DCHI and
	CiscoNM-HDV/VWIC-1MFT-T1
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDC8@venus.wesd.org>

We ran into a problem with MGCP not working with our PBX.  The call
Progess protocol has a bug (CSCdt68985CSCec33755).  We had to move to
H323.  Now it's working.  

Scott

-----Original Message-----
From: Terry Grace [mailto:tgrace@tgrace.com] 
Sent: Friday, September 19, 2003 8:17 AM
To: Voll, Scott; cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: [cisco-voip] RE: [nsp] PRI using a Nortel QPC757 DCHI and
CiscoNM-HDV/VWIC-1MFT-T1

We're using mgcp. Don't have the qsig package on the PBX.

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll@wesd.org] 
Sent: Friday, September 19, 2003 11:10 AM
To: tgrace; cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: RE: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
NM-HDV/VWIC-1MFT-T1


What are you running, H323,SIP,MGCP??  Can you use Qsig?  

--Scott

-----Original Message-----
From: tgrace [mailto:tgrace@tgrace.com] 
Sent: Thursday, September 18, 2003 7:51 PM
To: cisco-nsp@puck.nether.net; cisco-voip@puck.nether.net
Subject: [nsp] PRI using a Nortel QPC757 DCHI and Cisco
NM-HDV/VWIC-1MFT-T1

I'm having difficulty keepinga PRI TIE line up between our 81C and a
Cisco
2621XM router. The PRI seems to go down shortly after a disconnect from
the
first incoming call to the router from the PBX or PSTN or after a dozen
quick succession calls to PBX extensions. When this happens I can see
the
router TX'ing SABME's but getting no response from the PBX. At other
times
the link will be up but I will get an INFO message back from the PBX
that
the "identified channel does not exist" when I try to place an outbound
call. There also seems to be some issue with disconnects with 5ESS. On a
number of occassions my Nortel phone continued ringing after going
onhook on
the Cisco phone. This isn't consistent and happens perhaps 1 out 10
calls.

Initially tried using the DMS100 protocol with the PBX as net and the
router
as USR. Also did the reverse and then tried configuring both as 5ESS.
Same
type of problems.

Anyone ever run into something like this?


_______________________________________________
cisco-nsp mailing list  cisco-nsp@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-nsp
archive at http://puck.nether.net/pipermail/cisco-nsp/


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

From jpablo.romero at ifxnw.com.ar  Fri Sep 26 12:30:58 2003
From: jpablo.romero at ifxnw.com.ar (Juan Pablo Romero)
Date: Fri Sep 26 12:30:56 2003
Subject: [cisco-voip] ATA 186 Firmware
In-Reply-To: <00c101c38113$5ff96f10$0201a8c0@oemcomputer>
Message-ID: <000d01c3844b$91a61460$e32850c8@rupe>

Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 4890 bytes
Desc: not available
Url : https://puck.nether.net/pipermail/cisco-voip/attachments/20030926/d888ff71/attachment.jpg
From karthi at malagasy.com  Mon Sep 29 03:00:31 2003
From: karthi at malagasy.com (Karthikeyan Arumugam)
Date: Mon Sep 29 03:00:07 2003
Subject: [cisco-voip] h323 problem
Message-ID: <01d101c38657$5f9cbc90$12a2383e@KARTHI>

Dear list,

Iam trying to configure Audiocode gateway in FXS mode to connect to my gatekeeper.
i get dial tone and could see the number dialed...in my gatekeepr call monitor.
but the call is not through. but i dont see the destination IP(route) in my gatekker call monitor 
it shows 0.0.0.0 and i see source IP(my gateway)

when i checked through syslog :::
i get this error 
09/29 09:30:56.55 : 62.56.162.24 : WARNING :(     lgr_stack)(343       ) ?? [WARNING] Arrived Release contains no h.225 Reason neither q.931 Cause information stateMode:3
after this warning message the call is disconnected(ie gatekeeper relaeses the call)

I use h323 signalling.

pls advice any suggestion or where could be the problem

thanks by advance

karthi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030929/1696e694/attachment.htm
From wsisk at cisco.com  Mon Sep 29 09:36:14 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Mon Sep 29 09:36:30 2003
Subject: [cisco-voip] h323 problem
In-Reply-To: <01d101c38657$5f9cbc90$12a2383e@KARTHI>
Message-ID: <ELEMLGDGDBLINAJEIOCLAELOMOAA.wsisk@cisco.com>

Greetings Karthi,

Please enable "debug ras", "deb h225 asn1", and "debug gate main 5" on your
gatekeeper.
Then place a successfull call between 2 GK endpoints and place a failed call
between 2 GK endpoints.

Compare the debugs and find the difference.
Your syslog message below simply says that a reason code was not provided
when the call terminated.  While this is not necessarily good, the call is
already being torn down so you need to look earlier in time to figure out
why the call was disconnected.

/Wes
  -----Original Message-----
  From: cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Karthikeyan Arumugam
  Sent: Monday, September 29, 2003 3:01 AM
  To: cisco-voip@puck.nether.net
  Subject: [cisco-voip] h323 problem


  Dear list,

  Iam trying to configure Audiocode gateway in FXS mode to connect to my
gatekeeper.
  i get dial tone and could see the number dialed...in my gatekeepr call
monitor.
  but the call is not through. but i dont see the destination IP(route) in
my gatekker call monitor
  it shows 0.0.0.0 and i see source IP(my gateway)

  when i checked through syslog :::
  i get this error
  09/29 09:30:56.55 : 62.56.162.24 : WARNING :(     lgr_stack)(343       )
?? [WARNING] Arrived Release contains no h.225 Reason neither q.931 Cause
information stateMode:3
  after this warning message the call is disconnected(ie gatekeeper relaeses
the call)

  I use h323 signalling.

  pls advice any suggestion or where could be the problem

  thanks by advance

  karthi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030929/bbb42d6e/attachment.htm
From Scott.Voll at wesd.org  Mon Sep 29 20:05:13 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Sep 29 20:05:21 2003
Subject: [cisco-voip] Extension mobility CM 3.3
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDCB@venus.wesd.org>

I have followed the instructions on how to set up Extension Mobility and
Now I'm getting an "login Unsuccessful" "Error[6]"  I have found on CCO
error 6 means---> 

6

When a user tries to log in to a phone that is configured for Cisco
CallManager Extension Mobility and enters UserID and PIN, the phone
displays "6."

The service is not authenticating the user.

A problem may exist with Virtual Directory. Verify that the Virtual
Directory Login Password is correct.

 

 

What is the Virtual Directory and how do I check to see if my password
is correct?

 

Scott Voll

Network Analyst, CCNA 

Willamette ESD

Phone: (503) 385-4571

scott.voll@wesd.org <mailto:scott.voll@wesd.org> 

 

Confidential

This E-Mail and any attachments transmitted with it are property of
Willamette ESD and/or its affiliates, are confidential, and are intended
solely for the use of the individual or entity to which this E-Mail is
addressed.  If you are not one of the named recipient(s) or otherwise
have reason to believe that you have received this E-Mail in error,
please notify me at scott.voll@wesd.org  and delete this E-Mail and any
attachments.  Any other use, retention, forwarding, printing, or copying
of this E-Mail is strictly prohibited.

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030929/f6e28e96/attachment.htm
From wsisk at cisco.com  Tue Sep 30 09:10:57 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Tue Sep 30 09:11:27 2003
Subject: [cisco-voip] Extension mobility CM 3.3
In-Reply-To: <A85EF8C2C10DA6499807D44B7FDFEB4F17DDCB@venus.wesd.org>
Message-ID: <ELEMLGDGDBLINAJEIOCLMEAFMPAA.wsisk@cisco.com>

The "virtual directory" is the IIS virtual directory as viewed in
Administrative Tools ->Internet Services Manager from the start menu.  There
are a few security configurations there.  This may be referring to the
"password" for the "application" which
can be confiugred under start->programs->administrative tools->component
services.  Then under console root->component services->computers->my
computer->com+ applications.  Right click on your application/virtual
directory and enter the correct password on the Identity tab.

To correct the password:
1. query the IIS metabase for the current password
    (http://www.iisfaq.com/default.aspx?View=A128)
2. set the user password to match what IIS is using
    (in computer management)
3. configure your applications to use that password
    (in component services)

/Wes
  -----Original Message-----
  From: cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
  Sent: Monday, September 29, 2003 8:05 PM
  To: pacug@patch.com; cisco-voip@puck.nether.net
  Subject: [cisco-voip] Extension mobility CM 3.3


  I have followed the instructions on how to set up Extension Mobility and
Now I'm getting an "login Unsuccessful" "Error[6]"  I have found on CCO
error 6 means-a

        6
       When a user tries to log in to a phone that is configured for Cisco
CallManager Extension Mobility and enters UserID and PIN, the phone displays
"6."

        The service is not authenticating the user.

        A problem may exist with Virtual Directory. Verify that the Virtual
Directory Login Password is correct.






  What is the Virtual Directory and how do I check to see if my password is
correct?



  Scott Voll

  Network Analyst, CCNA

  Willamette ESD

  Phone: (503) 385-4571

  scott.voll@wesd.org



  Confidential

  This E-Mail and any attachments transmitted with it are property of
Willamette ESD and/or its affiliates, are confidential, and are intended
solely for the use of the individual or entity to which this E-Mail is
addressed.  If you are not one of the named recipient(s) or otherwise have
reason to believe that you have received this E-Mail in error, please notify
me at scott.voll@wesd.org  and delete this E-Mail and any attachments.  Any
other use, retention, forwarding, printing, or copying of this E-Mail is
strictly prohibited.




-------------- next part --------------
An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20030930/3d1c1fd5/attachment.htm
