[cisco-voip] sip trunk problem

Mon Feb 2 13:58:45 EST 2009

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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