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SIP call failing

ronit2k3
Level 1
Level 1

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"

1 Reply 1

Jonathan Schulenberg
Hall of Fame
Hall of Fame

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