[cisco-voip] sip trunk problem

Mon Feb 2 16:00:46 EST 2009

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 ³debug ccsip message²

*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 doesn¹t have any G711 in it. Are we sure your provider
isn¹t looking for G711?

It may be helpful to get a ³debug ccsip message².

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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