[cisco-voip] sip trunk problem

Wed Feb 4 04:12:18 EST 2009

Hi all. Thats was is bug.  I must upgrade CUCM 6.1.2 .

Thanks all.

http://www.gossamer-threads.com/lists/cisco/voip/96327

The issue actually turned out to be a CCM Bug. We worked with Mike on Friday and have him up and working via a hardware MTP on his router until we can implement a code that has the fix for the bug below. Ideally we want to have a setup at the end of the day without MTP. 

CSCsu72615 Bug Details 

CUCM fails to prack second 18X 

Symptom: 

CUCM 6.1.2 

When working with SIP provider we can get a 180 ringing contains an SDP. 

If you get a SDP in a 180 ringing, and it arrives before CUCM sends a 200 OK for a previous PRACK you can run into issues where your call gets disconnected. 

CUCM needs to respond to the PRACK, but we don't. 

Conditions: 

6.1.2 

SIIP trunk to provider with out MTP. 

1st Found-In 

6.1 

Fixed-In 

7.1(0.39000.81) 

6.1(2.9901.189) 

6.1(2.1119.1) 

7.0(1.12003.1) 

7.0(1.12003.2) 

Thanks, 

Jim Pender 

Senior Data Network Engineer 

PAETEC Communications

From: Barış Gülten [mailto:barisgulten at gmail.com] 
Sent: Tuesday, February 03, 2009 10:04 PM
To: 'Chris Ward'; 'cisco-voip at puck.nether.net'
Subject: RE: [cisco-voip] sip trunk problem

Hi Chris, 

Here is the below debug for h225 event and q931, h245 event. Is there any idea about this? 

Thanks.

*Feb  3 21:56:25.219: h323chan_chn_process_read_socket: fd=0 of type LISTENING has data

*Feb  3 21:56:25.219: Changing to new event: ACCEPT

h323chan_chn_accept: fd=0

*Feb  3 21:56:25.219: h323chan_gw_accept: TCP connection accepted from 172.16.16.10:54297 on fd=3

*Feb  3 21:56:25.219: h323chan_chn_accept: Local(0x0) accepts TCP conn from 172.16.16.10(0xAC10100A) port (54297); fd=3changing from LISTENING state to ACCEPTED state

*Feb  3 21:56:25.219: h323chan_chn_process_read_socket: fd=3 of type ACCEPTED has data

Hex representation of the SETUP TPKT received: 080200370504038090A26C06008131303533700C8030353330353137393936357E00BF0520B0060008914A00050201804386401F0031003000350033000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000022C0B50000120F436973636F43616C6C4D616E61676572003100010500386384ACC98000F5DD693BA481983700BE01AC100A0A00D50D80000700AC10100A06B8110000F5DD693BA481983700BE01AC100A0A010001000100010010A001000F0140B5000012088144000400010300

  Q931 Message IE Decodes

Protocol Discriminator : 0x08

CRV Length             : 2

CRV Value              : 0x0037

Message Type           : 0x05: SETUP

 Bearer Capability: Length Of IE=3

 Data 8090A2

 Calling Party Number: Length Of IE=6

 Data 008131303533

 Called Party Number: Length Of IE=12

 Data 803035333035313739393635

 User-User: Length Of IE=191

 Data 0520B0060008914A00050201804386401F0031003000350033000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000022C0B50000120F436973636F43616C6C4D616E61676572003100010500386384ACC98000F5DD693BA481983700BE01AC100A0A00D50D80000700AC10100A06B8110000F5DD693BA481983700BE01AC100A0A010001000100010010A001000F0140B5000012088144000400010300h225ParseData: Q.931 SETUP received on fd=3

*Feb  3 21:56:25.227: compose_TunnelledSignallingMessage_ciscoNo tunnelled content.

  Q931 Message IE Decodes

Protocol Discriminator : 0x08

CRV Length             : 2

CRV Value              : 0x8037

Message Type           : 0x02: CALL_PROC

 User-User: Length Of IE=51

 Data 052180060008914A00042800B500001240013C050100044300110000F5DD693BA481983700BE01AC100A0A0100010010800100

Hex representation of the CALL PROCEEDING TPKT to send.: 08028037027E0033052180060008914A00042800B500001240013C050100044300110000F5DD693BA481983700BE01AC100A0A0100010010800100

h225CallProcRequest: Q.931 CALL PROCEEDING sent fromfd=3.

*Feb  3 21:56:27.727: compose_TunnelledSignallingMessage_ciscoNo tunnelled content.

  Q931 Message IE Decodes

Protocol Discriminator : 0x08

CRV Length             : 2

CRV Value              : 0x8037

Message Type           : 0x03: PROGRESS

 Facility: Length Of IE=0

 Data 

 Progress Ind: Length Of IE=2

 Data 8088

 User-User: Length Of IE=87

 Data 0528002B80060008914A00042800B500001240013C0501000000F5DD693BA481983700BE01AC100A0A05800100010011A010010E60000110007B00015504430000000100120140B50000120B60011000011E041E028188

Hex representation of the PROGRESS TPKT to send.: 08028037031C001E0280887E00570528002B80060008914A00042800B500001240013C0501000000F5DD693BA481983700BE01AC100A0A05800100010011A010010E60000110007B00015504430000000100120140B50000120B60011000011E041E028188

h225ProgressRequest: Q.931 PROGRESS sent from fd=3

*Feb  3 21:56:29.327: compose_TunnelledSignallingMessage_ciscoNo tunnelled content.

  Q931 Message IE Decodes

Protocol Discriminator : 0x08

CRV Length             : 2

CRV Value              : 0x8037

Message Type           : 0x01: ALERTING

 Signal: Length Of IE=1

 Data 01

 User-User: Length Of IE=68

 Data 052380060008914A00042800B500001240013C05010006C300110000F5DD693BA481983700BE01AC100A0A01000100118010010E60000110007C00015504430000000100

Hex representation of the ALERTING TPKT to send.: 08028037013401017E0044052380060008914A00042800B500001240013C05010006C300110000F5DD693BA481983700BE01AC100A0A01000100118010010E60000110007C00015504430000000100

h225AlertRequest: Q.931 ALERTING sent from fd=3. Call state changed to [Call Received].

*Feb  3 21:56:31.827: compose_TunnelledSignallingMessage_ciscoNo tunnelled content.

  Q931 Message IE Decodes

Protocol Discriminator : 0x08

CRV Length             : 2

CRV Value              : 0x8037

Message Type           : 0x5A: RELEASE_COMP

 Cause: Length Of IE=2

 Data 80FF

 User-User: Length Of IE=34

 Data 052580060008914A00041100110000F5DD693BA481983700BE01AC100A0A10800100

Hex representation of the RELEASE COMPLETE TPKT to send.: 080280375A080280FF7E0022052580060008914A00041100110000F5DD693BA481983700BE01AC100A0A10800100

h225TerminateRequest: Q.931 RELEASE COMPLETE sent from fd=3. Call state changed to [Null].

*Feb  3 21:56:31.831: compose_TunnelledSignallingMessage_ciscoNo tunnelled content.

  Q931 Message IE Decodes

Protocol Discriminator : 0x08

CRV Length             : 2

CRV Value              : 0x8037

Message Type           : 0x07: CONNECT

 Bearer Capability: Length Of IE=3

 Data 8090A2

 User-User: Length Of IE=85

 Data 052280060008914A00042800B500001240013C0501000000F5DD693BA481983700BE01AC100A0A1D0C00110000F5DD693BA481983700BE01AC100A0A01000100118010010E60000110007D00015504430000000100

Hex representation of the CONNECT TPKT to send.: 080280370704038090A27E0055052280060008914A00042800B500001240013C0501000000F5DD693BA481983700BE01AC100A0A1D0C00110000F5DD693BA481983700BE01AC100A0A01000100118010010E60000110007D00015504430000000100

h225SetupResponse: Q.931 CONNECT sent from fd=3

*Feb  3 21:56:31.831: compose_TunnelledSignallingMessage_ciscoNo tunnelled content.

  Q931 Message IE Decodes

Protocol Discriminator : 0x08

CRV Length             : 2

CRV Value              : 0x8037

Message Type           : 0x6E: NOTIFY

 Notification Ind: Length Of IE=1

 Data F1

 Display: Length Of IE=0

 Data 

 Connected Number: Length Of IE=13

 Data 00003035333035313739393635a

 User-User: Length Of IE=33

 Data 0528501900060008914A00040000F5DD693BA481983700BE01AC100A0A10800100

Hex representation of the NOTIFY TPKT to send.: 080280376E2701F128004C0D000030353330353137393936357E00210528501900060008914A00040000F5DD693BA481983700BE01AC100A0A10800100

h225NotifyRequest: Q.931 NOTIFY sent from fd=3

*Feb  3 21:56:31.831: h323chan_chn_process_read_socket: fd=3 of type ACCEPTED has data

*Feb  3 21:56:31.831: h323chan_recvdata: recv failure on fd=3: errno=254 errstr=Connection reset by peerh323chan_chn_close: Calls[1] Exist on socketfd=3 Owner[2]

*Feb  3 21:56:31.835: h323chan_close: TCP connection from fd=3 closed

From: Chris Ward [mailto:chrward at cisco.com] 
Sent: Tuesday, February 03, 2009 12:42 AM
To: Barış Gülten; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] sip trunk problem

It looks like a media negotiation issue. Sounds like the media is failing on the H323 leg from the GW to CUCM.

You might need to turn on some H225 and H245 debugging from the GW.

Chris Ward 

  _____  

From: Barış Gülten <barisgulten at gmail.com>
Date: Mon, 2 Feb 2009 23:54:31 +0200
To: Chris Ward <chrward at cisco.com>, <cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] sip trunk problem

Chris,
Are there any suggestions about this sip trunk ? Provider gave us real ip and they expecting calls from  these real ip s. 
Thanks.

From: Barış Gülten [mailto:barisgulten at gmail.com] 
Sent: Monday, February 02, 2009 11:08 PM
To: 'Chris Ward'; 'cisco-voip at puck.nether.net'
Subject: RE: [cisco-voip] sip trunk problem

Chris,
Sccp Phones >CUCM >h323 gateway> router > sip trunk(outside providers).
Thanks.

From: Chris Ward [mailto:chrward at cisco.com] 
Sent: Monday, February 02, 2009 11:01 PM
To: Baris Gulten; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] sip trunk problem

How are calls being sent from the CUCM to the GW?

Chris Ward 
Cisco Systems Inc. 
Customer Support Engineer
Unified Communication Infrastructure
Boxborough, MA 
9:00am - 6:00pm Eastern 
978-936-0217
chrward at cisco.com 

  _____  

From: Baris Gulten <barisgulten at gmail.com>
Date: Mon, 2 Feb 2009 22:22:24 +0200
To: Chris Ward <chrward at cisco.com>, <cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] sip trunk problem

Chris,
Call starting from CUCM sccp iphones.  I am not sure exactly what is my providers demand.
Thanks.

From: Chris Ward [mailto:chrward at cisco.com] 
Sent: Monday, February 02, 2009 10:14 PM
To: Baris Gulten; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] sip trunk problem

Baris,

Does your provider require early media? I do see that you are not sending SDP in the initial INVITE.

Also, how is this call being sent from CUCM? H323?

Chris Ward 

  _____  

From: Baris Gulten <barisgulten at gmail.com>
Date: Mon, 2 Feb 2009 21:45:55 +0200
To: Chris Ward <chrward at cisco.com>, <cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] sip trunk problem

Hi Chris,
I set up g729br8 codec also ringing working but when i  off-hook after then calls dropping.
Here is the below debug result.(ccsip message), thanks.

*Feb  2 21:29:51.075: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:0xxx at xxx0:5060 SIP/2.0
Via: SIP/2.0/UDP  xxx:5060;branch=z9hG4bK2B1562
From: <sip: 0xxx at xxx>;tag=619AD0-1324
To: <sip:0xxx at xxx>
Date: Mon, 02 Feb 2009 21:29:51 GMT
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5 at xxx
Supported: 100rel,timer,replaces
Min-SE:  1800
Cisco-Guid: 2153288071-2118939032-437513217-2886732291
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:0xxx at xxx2>;party=calling;screen=yes;privacy=off
Timestamp: 1233610191
Contact: <sip:0xxx at xxx2:5060>
Expires: 180
Allow-Events: telephone-event

*Feb  2 21:29:51.147: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5 at xxx
From: <sip:0xxx at xxx>;tag=619AD0-1324
To: <sip:0xxx at xxx>;tag=3367
CSeq: 101 INVITE
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562
Supported: timer,100rel
Content-Length: 0

*Feb  2 21:29:53.211: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5 at xxx
From: <sip:0xxx at xxx>;tag=619AD0-1324
To: <sip:0xxx at xxx>;tag=3367
CSeq: 101 INVITE
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562
Supported: timer,100rel
Content-Length: 0

*Feb  2 21:29:53.243: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5 at xxx
From: <sip:0xxx at xxx>;tag=619AD0-1324
To: <sip:0xxx at xxx>;tag=3367
Content-Type: application/sdp
CSeq: 101 INVITE
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562
Supported: timer,100rel
Content-Length: 258

v=0
o=MG4000|2.0 56404 56404 IN IP4 xxx
s=-
c=IN IP4 xxx
t=0 0
m=audio 48180 RTP/AVP 18 97 101 13
a=rtpmap:97 G.729b/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=yes
a=ptime:10
a=rtpmap:13 CN/8000

*Feb  2 21:30:02.819: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
CANCEL sip:0xxx at xxx:5060 SIP/2.0
Via: SIP/2.0/UDP  xxx:5060;branch=z9hG4bK2B1562
From: <sip:0xxx at xxx>;tag=619AD0-1324
To: <sip:0xxx at xxx>
Date: Mon, 02 Feb 2009 21:29:51 GMT
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5 at xxx
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1233610202
Content-Length: 0

*Feb  2 21:30:02.879: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5 at xxx
From: <sip:0xxx at xxx>;tag=619AD0-1324
To: <sip:0xxx at xxx>;tag=8819
CSeq: 101 CANCEL
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562
Contact: sip:0xxx at xxx:5060;user=phone
Supported: timer,100rel
Content-Length: 0

*Feb  2 21:30:02.887: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 487 Request Terminated
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5 at xxx
From: <sip:0xxx at xxx2>;tag=619AD0-1324
To: <sip:0xxx at xxx>;tag=3367
CSeq: 101 INVITE
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562
Supported: timer,100rel
Content-Length: 0

*Feb  2 21:30:02.887: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:0xxx at xxx:5060 SIP/2.0
Via: SIP/2.0/UDP  xxx:5060;branch=z9hG4bK2B1562
From: <sip:0xxx at xxx>;tag=619AD0-1324
To: <sip:0xxx at xxx>;tag=3367
Date: Mon, 02 Feb 2009 21:29:51 GMT
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5 at xxx
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

From: Chris Ward [mailto:chrward at cisco.com] 
Sent: Monday, February 02, 2009 9:03 PM
To: Baris Gulten; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] sip trunk problem

Hi Baris,

As a test, can you try and remove the voice-class codec from the dial-peer and add a specific codec?

Try this:

Codec g729br8

Looks like this is the codec the provider is wanting. I know its in your voice-class codec list, but I would still try it.

Chris Ward 

  _____  

From: Baris Gulten <barisgulten at gmail.com>
Date: Mon, 2 Feb 2009 20:58:45 +0200
To: Chris Ward <chrward at cisco.com>, <cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] sip trunk problem

Call working one time ring after then dropping. 
Here is the below “debug ccsip message”

*Feb  2 20:52:18.951: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:0xxxxxx at xxx:5060 SIP/2.0
Via: SIP/2.0/UDP  xxxx:5060;branch=z9hG4bK151BC
From: <sip: 0xxxxxx at xxx>;tag=3F3D70-212E
To: <sip: 0xxxxxx at xxx>
Date: Mon, 02 Feb 2009 20:52:18 GMT
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5 at 84.44.99.162
Supported: 100rel,timer,replaces
Min-SE:  1800
Cisco-Guid: 2154448201-2990764440-101963777-2886732291
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip: 0xxxxxx at xxx>;party=calling;screen=yes;privacy=off
Timestamp: 1233607938
Contact: <sip: 0xxxxxx at xxx:5060>
Expires: 180
Allow-Events: telephone-event

*Feb  2 20:52:19.031: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5 at xxx
From: <sip: 0xxxxxx at xxx>;tag=3F3D70-212E
To: <sip: 0xxxxxx at xxx>;tag=28846
CSeq: 101 INVITE
Via: SIP/2.0/UDP xxxx:5060;branch=z9hG4bK151BC
Supported: timer,100rel
Content-Length: 0

*Feb  2 20:52:21.207: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5 at xxx
From: <sip: 0xxxxxx at xxx>;tag=3F3D70-212E
To: <sip: 0xxxxxx at xxx>;tag=28846
CSeq: 101 INVITE
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC
Supported: timer,100rel
Content-Length: 0

*Feb  2 20:52:21.243: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5 at xxx
From: <sip: 0xxxxxx at xxx>;tag=3F3D70-212E
To: <sip: 0xxxxxx at xxx>;tag=28846
Content-Type: application/sdp
CSeq: 101 INVITE
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC
Supported: timer,100rel
Content-Length: 258

v=0
o=MG4000|2.0 99854 99854 IN IP4 xxx
s=-
c=IN IP4 62.244.254.131
t=0 0
m=audio 48296 RTP/AVP 18 97 101 13
a=rtpmap:97 G.729b/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=yes
a=ptime:10
a=rtpmap:13 CN/8000

*Feb  2 20:52:21.247: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
CANCEL sip: 0xxxxxx at xxx:5060 SIP/2.0
Via: SIP/2.0/UDP  xxx:5060;branch=z9hG4bK151BC
From: <sip: 0xxxxxx at xxx>;tag=3F3D70-212E
To: <sip: 0xxxxxx at xxx>
Date: Mon, 02 Feb 2009 20:52:18 GMT
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5 at xxx
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1233607941
Content-Length: 0

*Feb  2 20:52:21.307: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5 at xxx
From: <sip: 0xxxxxx at xxx>;tag=3F3D70-212E
To: <sip: 0xxxxxx at xxx>;tag=29754
CSeq: 101 CANCEL
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC
Contact: sip: 0xxxxxx at xxx:5060;user=phone
Supported: timer,100rel
Content-Length: 0

*Feb  2 20:52:21.315: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 487 Request Terminated
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5 at xxx
From: <sip: 0xxxxxx at xxx>;tag=3F3D70-212E
To: <sip: 0xxxxxx at xxx>;tag=28846
CSeq: 101 INVITE
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC
Supported: timer,100rel
Content-Length: 0

*Feb  2 20:52:21.319: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip: 0xxxxxx at xxx:5060 SIP/2.0
Via: SIP/2.0/UDP  xxx:5060;branch=z9hG4bK151BC
From: <sip: 0xxxxxx at xxx >;tag=3F3D70-212E
To: <sip: 0xxxxxx at xxx >;tag=28846
Date: Mon, 02 Feb 2009 20:52:18 GMT
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5 at xxx
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

From: Chris Ward [mailto:chrward at cisco.com] 
Sent: Monday, February 02, 2009 8:44 PM
To: Baris Gulten; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] sip trunk problem

Looks like a media negotiation failure. The only thing that I can see is that your voice-class doesn’t have any G711 in it. Are we sure your provider isn’t looking for G711?

It may be helpful to get a “debug ccsip message”.

Chris Ward 

  _____  

From: Baris Gulten <barisgulten at gmail.com>
Date: Mon, 2 Feb 2009 20:39:01 +0200
To: <cisco-voip at puck.nether.net>
Subject: [cisco-voip] sip trunk problem

Hi everybody,
I have ccm 6.1.1, also 2851 router. 
I define sip trunk on 2851 router. I have trouble when i decide make a call. Lets hold my hand J

Here is the debug result
vgw#sh debug
CCSIP SPI: SIP Call Events tracing is enabled   (filter is OFF)
CCSIP SPI: SIP error debug tracing is enabled   (filter is OFF
-------------
#
*Feb  2 20:40:30.779: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
*Feb  2 20:40:30.783: //109/803045A3FD12/SIP/Event/sipSPICreateRpid: Received Octet3A=0x83 -> Setting ;screen=yes ;privacy=off
*Feb  2 20:40:33.295: //109/803045A3FD12/SIP/Error/sipSPIDoAudioNegotiation: Media negotiation failed for m-line 1
*Feb  2 20:40:33.295: //109/803045A3FD12/SIP/Error/sipSPIDoMediaNegotiation: 
no valid fax or audio streams
*Feb  2 20:40:33.295: //109/803045A3FD12/SIP/Error/ccsip_api_call_cut_progress: MediaNegotiation Failure - Send Cancel
*Feb  2 20:40:33.295: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

Here is the config below (i did before cme router with this config)
!
voice rtp send-recv
!
voice service voip 
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 h323
  session transport tcp calls-per-connection 200
 sip
  bind control source-interface GigabitEthernet0/1
  bind media source-interface GigabitEthernet0/1
  registrar server expires max 3600 min 3600
  no call service stop
!
voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g729br8
 codec preference 3 g723r63
 codec preference 4 g723r53
 codec preference 5 g726r24
 codec preference 6 g726r16
 codec preference 7 g726r32
 codec preference 8 g723ar53
 codec preference 9 g723ar63
!  
interface GigabitEthernet0/1
 ip address xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx (defined real ip by sip provider)
 duplex auto
 speed auto
!

!
dial-peer voice 100 voip
 preference 1
 voice-class codec 1
 destination-pattern 053T
 session protocol sipv2
 session target ipv4:xxx.xxx.xxx.xxx (sip server ip)
 dtmf-relay rtp-nte
 clid network-number xxxxxxxxx
!

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