[cisco-voip] sip trunk problem

Mon Feb 2 17:41:40 EST 2009

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
!

_______________________________________________
cisco-voip mailing list
cisco-voip at 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/20090202/afe38d52/attachment.html>