Possible codec issue on CUBE

[alexis.katsavras]

Hi there,

I have the following call flow set-up:

CUCM 10.5 <==> SIP <==> CUBE 15.3 <==> SIP <==> Telco

Incoming calls work fine.

Outgoing calls to mobile only are failing. When the call is made the mobile phone will ring but one's its answered the call is drop. To me this would suggest that this a codec issue

Below are the debugs but the only thing I see is "SIP/2.0 500 Internal Server Error"

From: "Alexis"<sip:0131306XXXX@ippbx2.XXXXX.co.uk>;tag=E9322CC-734
To: <sip:07759134XXXX@ippbx2.XXXX.co.uk>;tag=94040
Call-ID: E1C23D57-F49011E4-8595C68F-103E8103@135.X.X.X
CSeq: 102 INVITE
Via: SIP/2.0/UDP 135.X.X.X:5060;branch=z9hG4bK12C8431009
Content-Type: application/sdp
Contact: <sip:077591XXXX@135.X.X.X:5060>
User-Agent:  Nortel SESM 14.0.16.3
Supported: replaces,tdialog
P-Asserted-Identity: <sip:anonymous@135.X.X.X>
Allow: INVITE,BYE,CANCEL,ACK,REGISTER,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,INFO,REFER,OPTIONS,PUBLISH,PRACK
x-nt-location: -1
Require: timer
Timestamp: 1431072782
Session-Expires: 1800;refresher=uac
Content-Length: 260

v=0
o=sbc-uk-wv-g001a 16215776 16215777 IN IP4 135.X.X.X
s=sip call
e=unknown@invalid.net
c=IN IP4 135.X.X.X
t=0 0
a=sendrecv
m=audio 40428 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

008852: May  8 08:13:09.754: //1519569/0A7980800001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/TCP 10.X.X.X:5060;branch=z9hG4bK517066a6d6e026f
From: "Alexis Katsavras" <sip:44131306XXXX@10.X.X.X>;tag=15171831~5ef6baea-c4ed-5f9e-769a-02a480f7b0f7-35096096
To: <sip:90775913XXX@10.X.X.X>;tag=E932D04-15FA
Call-ID: a798080-54c1700e-29ad51-a0c640a@10.X.X.X
CSeq: 101 INVITE
Content-Length: 0

008853: May  8 08:13:09.754: //1519570/0A7980800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:0775913XXXX@135.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 135.X.X.X:5060;branch=z9hG4bK12C8461859
From: "Alexis Katsavras" <sip:44131306XXXX@ippbx2.XXXX.co.uk>;tag=E9322CC-734
To: <sip:0775913XXXX@ippbx2.XXXX.co.uk>;tag=94040
Date: Fri, 08 May 2015 08:13:02 GMT
Call-ID: E1C23D57-F49011E4-8595C68F-103E8103@135.X.X.X
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0

008854: May  8 08:13:09.754: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:90775913XXX@10.X.X.X:5060 SIP/2.0
Via: SIP/2.0/TCP 10.X.X.X:5060;branch=z9hG4bK517066a6d6e026f
From: "Alexis Katsavras" <sip:44131306XXXX@10.X.X.X>;tag=15171831~5ef6baea-c4ed-5f9e-769a-02a480f7b0f7-35096096
To: <sip:90775913XXX@10.X.X.X>;tag=E932D04-15FA
Date: Fri, 08 May 2015 08:13:02 GMT
Call-ID: a798080-54c1700e-29ad51-a0c640a@10.X.X.X
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

008855: May  8 08:13:09.758: //1519570/0A7980800000/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:0775913XXXX@135.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 135.X.X.X:5060;branch=z9hG4bK12C847BC8
From: "Alexis Katsavras" <sip:44131306XXXX@ippbx2.XXXXX.co.uk>;tag=E9322CC-734
To: <sip:0775913XXXX@ippbx2.XXXXX.co.uk>;tag=94040
Date: Fri, 08 May 2015 08:13:02 GMT
Call-ID: E1C23D57-F49011E4-8595C68F-103E8103@135.X.X.X
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Max-Forwards: 70
Timestamp: 1431072789
CSeq: 103 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=216,OS=34560,PR=164,OR=26240,PL=0,JI=0,LA=0,DU=0
Content-Length: 0

Any help would be great

Thanks

alexis

[Chris Deren]

Change your outbound dial peer to use codec-class that has both G711 and G729 listed, based on the SDP content in the original INVITE you are only sending G711.

[alexis.katsavras]

Found the resolution to this. Change made under the SIP Profile for the SIP Trunk

[Pointbreak]

Hi ,

What above sniped file implies. Thanks in advance