[cisco-voip] SIP CME-as-SRST transfer failing

[Jason Aarons (AM)]

Here is the debug ccsip messages, notice the error down at 17:08:12.228 ? Researching it.

*Nov  8 17:08:12.224: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:7454 at 10.30.2.253:5060 SIP/2.0
Via: SIP/2.0/UDP 10.30.2.253:5060;branch=z9hG4bKBAFA43B
Remote-Party-ID: "7484" <sip:7484 at 10.30.2.253>;party=calling;screen=yes;privacy=off
From: "7484" <sip:7484 at 10.27.16.11>;tag=7C7715E4-81A
To: <sip:7454 at 10.30.2.253>
Date: Fri, 08 Nov 2013 23:08:12 GMT
Call-ID: 79856A55-480111E3-AFB5CA0E-60C774C9 at 10.30.2.253
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 2038626741-1208029667-2947533326-1623684297
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1383952092
Contact: <sip:7484 at 10.30.2.253:5060>
Call-Info: <sip:10.30.2.253:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 409

v=0
o=CiscoSystemsSIP-GW-UserAgent 3729 2253 IN IP4 10.30.2.253
s=SIP Call
c=IN IP4 10.30.2.253
t=0 0
m=audio 17618 RTP/AVP 0 18 100 101 121 19
c=IN IP4 10.30.2.253
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:121 frf-dialed-digit/8000
a=fmtp:121 0-15
a=rtpmap:19 CN/8000

*Nov  8 17:08:12.228: %CALL_CONTROL-6-CALL_LOOP: The incoming call has a global identifier already present in the list of currently handled calls. It is being refused.

*Nov  8 17:08:12.244: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.30.2.253:5060;branch=z9hG4bKBAFA43B
From: "7484" <sip:7484 at 10.27.16.11>;tag=7C7715E4-81A
To: <sip:7454 at 10.30.2.253>;tag=7C7715EC-B31
Date: Fri, 08 Nov 2013 23:08:12 GMT
Call-ID: 79856A55-480111E3-AFB5CA0E-60C774C9 at 10.30.2.253
Timestamp: 1383952092
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (AM)
Sent: Friday, November 08, 2013 6:16 PM
To: Kenneth Hayes
Cc: cisco-voip (cisco-voip at puck.nether.net)
Subject: Re: [cisco-voip] SIP CME-as-SRST transfer failing

This is 100% SIP. The 8961 doesn’t do SCCP.

From: Kenneth Hayes [mailto:kennethwhayes at gmail.com]
Sent: Friday, November 08, 2013 6:13 PM
To: Jason Aarons (AM)
Cc: cisco-voip (cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>)
Subject: Re: [cisco-voip] SIP CME-as-SRST transfer failing


Where is your telephony-service?

Sent from my iPhone

On Nov 8, 2013, at 6:00 PM, "Jason Aarons (AM)" <jason.aarons at dimensiondata.com<mailto:jason.aarons at dimensiondata.com>> wrote:

Ok what am I missing.  SIP 8961s and transfer is failing whilst in SRST mode.


voice service voip
ip address trusted list
  ipv4 192.168.16.10
  ipv4 192.168.16.11
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol none
 modem passthrough nse codec g711ulaw
sip
  bind control source-interface GigabitEthernet0/1.30
  bind media source-interface GigabitEthernet0/1.30
  registrar server
!
voice class codec 99
codec preference 1 g711ulaw
codec preference 2 g729r8
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
voice class h323 1
  h225 timeout tcp establish 3
!
!
voice register global
max-dn 480
max-pool 1
!
voice register pool  1
id network 192.168.2.0 mask 255.255.255.0
preference 1
incoming called-number
 proxy 192.168.2.253 preference 1
dtmf-relay rtp-nte cisco-rtp sip-notify
voice-class codec 1
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itevomcid
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