ā05-29-2017 01:07 AM - edited ā03-17-2019 10:26 AM
I am trying to Integrate Avaya with Cisco Gateway to send calls via SIP trunk
Connectivity to Avaya with Cisco is E1
Calls are failing with error SIP/2.0 400 Invalid Body
Here is failed calls logs
-----------------------------------
INVITE sip:+1937XXXXXXXXXX@32.114.33.70:5060 SIP/2.0
Via: SIP/2.0/UDP 32.253.63.98:5060;branch=z9hG4bK4D5248234A
From: "XXXXXXXXXX,Mir" <sip:79XX@32.253.63.98>;tag=37FFDF46-10B5
To: <sip:+1937XXXXXXXXXX@32.114.33.70>
Date: Fri, 26 May 2017 15:09:07 GMT
Call-ID: 1960B91B-415C11E7-854BEDFE-2BB2A449@32.253.63.98
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0425749255-1096552935-3159552127-0108568832
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6a
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1495811347
Contact: <sip:79XX@32.253.63.98:5060>
Expires: 60
Allow-Events: telephone-event
P-Asserted-Identity: "XXXXXXXXXX,Mir" <sip:79XX@32.253.63.98>
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 881
--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required
v=0
o=CiscoSystemsSIP-GW-UserAgent 8460 2494 IN IP4 32.253.63.98
s=SIP Call
c=IN IP4 32.253.63.98
t=0 0
m=audio 29624 RTP/AVP 0 100 101
c=IN IP4 32.253.63.98
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
a=ptime:30
--uniqueBoundary
Content-Type: application/qsig
Content-Disposition: signal;handling=optional
Content-Length: 58
XXXXXXXXXX,Mirl.............................................................................................................................................................................................................................................................................................................................
2295446: May 26 21:39:07.741: //16008664/19606B07BC52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 400 Invalid Body
Via: SIP/2.0/UDP 32.253.63.98:5060;branch=z9hG4bK4D5248234A
From: "XXXXXXXXXX,Mir" <sip:79XX@32.253.63.98>;tag=37FFDF46-10B5
To: <sip:+1937XXXXXXXXXX@32.114.33.70>;tag=aprqngfrt-4h1grv00000a6
Call-ID: 1960B91B-415C11E7-854BEDFE-2BB2A449@32.253.63.98
CSeq: 101 INVITE
Timestamp: 1495811347
Reason: Q.850;cause=47;text="Call Terminated"
2295447: May 26 21:39:07.743: //-1/xxxxxxxxxxxx/SIP/Msg/sipDisplayBinaryData:
Sending: Binary Message Body
2295448: May 26 21:39:07.743: Content-Type: application/qsig
08 02 AB 76 05 A1 04 03 90 90 A3 18 03 A9 83 91 1E 02 81 83 28 0A 4C 61 73 6B 61 72 2C 4D 69 72 6C 06 00 80 37 39 34 34 70 0E 81 30 30 31 39 33 37 34 34 35 33 34 37 34 0D 0A
2295449: May 26 21:39:07.743: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:+1937XXXXXXXXXX@32.114.33.70:5060 SIP/2.0
Via: SIP/2.0/UDP 32.253.63.98:5060;branch=z9hG4bK4D5248234A
From: "XXXXXXXXXX,Mir" <sip:79XX@32.253.63.98>;tag=37FFDF46-10B5
To: <sip:+1937XXXXXXXXXX@32.114.33.70>;tag=aprqngfrt-4h1grv00000a6
Date: Fri, 26 May 2017 15:09:07 GMT
Call-ID: 1960B91B-415C11E7-854BEDFE-2BB2A449@32.253.63.98
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
2295450: May 26 21:39:07.743: //16008665/19606B07BC52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:+1937XXXXXXXXXX@32.114.35.6:5060 SIP/2.0
Via: SIP/2.0/UDP 32.253.63.98:5060;branch=z9hG4bK4D52491F6D
From: "XXXXXXXXXX,Mir" <sip:79XX@32.253.63.98>;tag=37FFDF94-17A9
To: <sip:+1937XXXXXXXXXX@32.114.35.6>
Date: Fri, 26 May 2017 15:09:07 GMT
Call-ID: 196C9FEF-415C11E7-854DEDFE-2BB2A449@32.253.63.98
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0425749255-1096552935-3159552127-0108568832
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6a
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1495811347
Contact: <sip:79XX@32.253.63.98:5060>
Expires: 60
Allow-Events: telephone-event
P-Asserted-Identity: "XXXXXXXXXX,Mir" <sip:79XX@32.253.63.98>
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 916
--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required
v=0
o=CiscoSystemsSIP-GW-UserAgent 5674 1658 IN IP4 32.253.63.98
s=SIP Call
c=IN IP4 32.253.63.98
t=0 0
m=audio 29230 RTP/AVP 18 0 100 101
c=IN IP4 32.253.63.98
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
--uniqueBoundary
Content-Type: application/qsig
Content-Disposition: signal;handling=optional
Content-Length: 58
,Mirl.............................................................................................................................................................................................................................................................................................................................
2295451: May 26 21:39:07.873: //16008665/19606B07BC52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 400 Invalid Body
Via: SIP/2.0/UDP 32.253.63.98:5060;branch=z9hG4bK4D52491F6D
From: "XXXXXXXXXX,Mir" <sip:79XX@32.253.63.98>;tag=37FFDF94-17A9
To: <sip:+1937XXXXXXXXXX@32.114.35.6>;tag=aprqngfrt-o86och10000a6
Call-ID: 196C9FEF-415C11E7-854DEDFE-2BB2A449@32.253.63.98
CSeq: 101 INVITE
Timestamp: 1495811347
Reason: Q.850;cause=47;text="Call Terminated"
ā05-29-2017 11:34 AM
Just to confirm, you have an Avaya TDM PBX that is connected to a Cisco router using an ISDN E1. What is the Cisco router connected to over SIP?
Your traces aren't complete nor are they annotated to explain the call direction, calling/called numbers, or the IP addresses involved; however, the first thing that stands out at me is the presence of QSIG. That would easily explain the error if the SIP destination isn't expecting QSIG tunneling over SIP.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/pgw/9/feature/module/9-8_1_/QSIG_SIP.html
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