Before I start debugging I thought I'd ask. I have a sip message from a gateway (Cisco IOS) that is being sliently dropped. Should ser be able to handle a message that looks like this? I haven't done a multipart message before.
Thanks, ---greg
INVITE sip:+19194724170@sn-sip-in.ca-sn1.cisco.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.109.91:5060;branch=z9hG4bK4F0109C Remote-Party-ID: sip: +19199915651@172.18.109.91;party=calling;screen=yes;privacy=off From: sip:+19199915651@172.18.109.91;tag=249E4980-FFC To: sip:+19194724170@sn-sip-in.ca-sn1.cisco.com Date: Tue, 13 Jun 2006 19:37:55 GMT Call-ID: F241878C-FA4A11DA-81EFC5C8-2F1D7951@172.18.109.91 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 4064260884-4199158234-2157445126-1397050494 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 10 Timestamp: 1150227475 Contact: sip:+19199915651@172.18.109.91:5060 Expires: 180 Allow-Events: telephone-event Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 797
--uniqueBoundary Content-Type: application/sdp Content-Disposition: session;handling=required
v=0 o=CiscoSystemsSIP-GW-UserAgent 8340 2382 IN IP4 172.18.109.91 s=SIP Call c=IN IP4 172.18.109.91 t=0 0 m=audio 19122 RTP/AVP 18 3 0 4 100 101 c=IN IP4 172.18.109.91 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16
--uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional
IAM, PRN,isdn*,,NI***, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,02,,1,4724170 CGN,04,,1,y,2,9199915651 CPC,09 FCI,,,,,,,y, GCI,f23fb314fa4a11da8098000653454c7e
--uniqueBoundary--
What does your log say?
On 6/14/06, Greg Fausak lgfausak@gmail.com wrote:
Before I start debugging I thought I'd ask. I have a sip message from a gateway (Cisco IOS) that is being sliently dropped. Should ser be able to handle a message that looks like this? I haven't done a multipart message before.
Thanks, ---greg
INVITE sip:+19194724170@sn-sip-in.ca-sn1.cisco.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.109.91:5060;branch=z9hG4bK4F0109C Remote-Party-ID: sip: +19199915651@172.18.109.91;party=calling;screen=yes;privacy=off From: sip:+19199915651@172.18.109.91;tag=249E4980-FFC To: sip:+19194724170@sn-sip-in.ca-sn1.cisco.com Date: Tue, 13 Jun 2006 19:37:55 GMT Call-ID: F241878C-FA4A11DA-81EFC5C8-2F1D7951@172.18.109.91 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 4064260884-4199158234-2157445126-1397050494 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 10 Timestamp: 1150227475 Contact: sip:+19199915651@172.18.109.91:5060 Expires: 180 Allow-Events: telephone-event Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 797
--uniqueBoundary Content-Type: application/sdp Content-Disposition: session;handling=required
v=0 o=CiscoSystemsSIP-GW-UserAgent 8340 2382 IN IP4 172.18.109.91 s=SIP Call c=IN IP4 172.18.109.91 t=0 0 m=audio 19122 RTP/AVP 18 3 0 4 100 101 c=IN IP4 172.18.109.91 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16
--uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional
IAM, PRN,isdn*,,NI***, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,02,,1,4724170 CGN,04,,1,y,2,9199915651 CPC,09 FCI,,,,,,,y, GCI,f23fb314fa4a11da8098000653454c7e
--uniqueBoundary--
-- Greg Fausak greg@thursday.com _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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Greg Fausak