Re: [Asterisk-Users] Error in Incoming SIP call
Lal, Deepak (Contractor) wrote: When I get a SIP call, I get the following error: *CLI NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp' Which client is used to place the call? I haven't seen multipart/mixed used, even though it is not incorrect at all. Could you do a SIP debug and capture the whole SIP invite? This might not be related to your problem, I'm just curious of what the other part of the payload can be. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Error in Incoming SIP call
It works now - It seems I had a space after extension# and that was causing a problem. The client is a Cirpack (www.cirpack.com) softswitch. The sip debug output (AS REQUESTED) is: SIP debug output *CLI sip debug SIP Debugging Enabled *CLI Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Allow: UPDATE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061;user=phone Content-Type: multipart/mixed;boundary=unique-boundary-1 CSeq: 220 INVITE From: sip:[EMAIL PROTECTED];user=phone;tag=4aa44670Max-Fo rwards: 31 MIME-Version: 1.0 To: sip:[EMAIL PROTECTED];user=phone User-Agent: Cirpack/v4.3o (gw_sip) Via: SIP/2.0/UDP 172.31.128.11:5061;branch=z9hG4bK-DAA Content-Length: 520 --unique-boundary-1 Content-Type: application/ISUP;version=cp10isup;base=etsi121 Content-Disposition: signal;handling=optional 01 07 02 70 00 02 01 03 09 02 0a 00 0a 07 03 13 15 44 12 01 20 04 08 83 10 15 74 77 11 11 0f 06 01 10 00 --unique-boundary-1 Content-Type: application/SDP v=0 o=cp10 1068206724 1068206724 IN IP4 172.31.128.12 s=SIP Call c=IN IP4 172.31.128.12 t=0 0 m=audio 16636 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=maxptime:30 --unique-boundary-1-- 13 headers, 21 lines Using latest request as basis request Sending to 172.31.128.11 : 5061 (non-NAT) NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp' Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Allow: UPDATE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061;user=phone Content-Type: multipart/mixed;boundary=unique-boundary-1 CSeq: 220 INVITE From: sip:[EMAIL PROTECTED];user=phone;tag=4aa44670Max-Fo rwards: 31 MIME-Version: 1.0 To: sip:[EMAIL PROTECTED];user=phone User-Agent: Cirpack/v4.3o (gw_sip) Via: SIP/2.0/UDP 172.31.128.11:5061;branch=z9hG4bK-DAA Content-Length: 520 --unique-boundary-1 Content-Type: application/ISUP;version=cp10isup;base=etsi121 Content-Disposition: signal;handling=optional 01 07 02 70 00 02 01 03 09 02 0a 00 0a 07 03 13 15 44 12 01 20 04 08 83 10 15 74 77 11 11 0f 06 01 10 00 --unique-boundary-1 Content-Type: application/SDP v=0 o=cp10 1068206724 1068206724 IN IP4 172.31.128.12 s=SIP Call c=IN IP4 172.31.128.12 t=0 0 m=audio 16636 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=maxptime:30 --unique-boundary-1-- 13 headers, 21 lines Ignoring this request Looking for 514777 in incoming list_route: hop: sip:[EMAIL PROTECTED]:5061;user=phone Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.31.128.11:5061;branch=z9hG4bK-DAA From: sip:[EMAIL PROTECTED];user=phone;tag=4aa44670To: sip:[EMAIL PROTECTED];user=phone;tag=as56dff219 Call-ID: [EMAIL PROTECTED] CSeq: 220 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 172.31.128.11:5061 -- Executing Dial(SIP/-08114370, Zap/2|10) in new stack -- Called 2 -- Zap/2-1 is ringing Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.31.128.11:5061;branch=z9hG4bK-DAA From: sip:[EMAIL PROTECTED];user=phone;tag=4aa44670To: sip:[EMAIL PROTECTED];user=phone;tag=as56dff219 Call-ID: [EMAIL PROTECTED] CSeq: 220 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 172.31.128.11:5061 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Nobody picked up in 1 ms -- Hungup 'Zap/2-1' - end of SIP debug -- -Original Message- From: Olle E. Johansson To: [EMAIL PROTECTED] Sent: 11/7/03 7:02 AM Subject: Re: [Asterisk-Users] Error in Incoming SIP call Lal, Deepak (Contractor) wrote: When I get a SIP call, I get the following error: *CLI NOTICE[1133718080]: File
Re: [Asterisk-Users] Error in Incoming SIP call
--unique-boundary-1 Content-Type: application/ISUP;version=cp10isup;base=etsi121 Content-Disposition: signal;handling=optional 01 07 02 70 00 02 01 03 09 02 0a 00 0a 07 03 13 15 44 12 01 20 04 08 83 10 15 74 77 11 11 0f 06 01 10 00 --unique-boundary-1 Hi! Content-type: application/ISUP --- http://www.ietf.org/rfc/rfc3204.txt This document describes MIME types for application/ISUP and application/QSIG objects for use in SIP applications, according to the rules defined in RFC 2048. These types can be used to identify ISUP and QSIG objects within a SIP message such as INVITE or INFO, as might be implemented when using SIP in an environment where part of the call involves interworking to the PSTN. --- Interesting, but understandable if it's a softswitch. Haven't seen that one before. Seems like we need a small change in the MIME handling of the chan_sip to handle MIME multipart message bodies. Right now, the message body is parses from top down, regardless of what's in it (as far as I can understand the source). Thank your for the debug output! There's always something new to learn, isn't it? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error in Incoming SIP call
John Todd wrote: exten = 514777,1,Dial,Zap/2|10 Try: exten = 514777,1,Dial(Zap/2,10) I think these two versions of giving arguments are confusing. Reading docs and show application texts, both variants are used, sometimes even in the same text. Is the first syntax old, to be replaced by the more easy to understand second syntax? If so, we have to remove the confusing old syntax from documentation. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error in Incoming SIP call
When I get a SIP call, I get the following error: *CLI NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 514777, 1) == Spawn extension (incoming, 514777, 1) exited non-zero on 'SIP/-08114358' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, h, 1) == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' In my extensions file, I have the following defined: [incoming] exten = 514777,1,Dial,Zap/2|10 Any suggestions will be appreciated! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error in Incoming SIP call
what does show dialplan incoming show ? Also try using Dial(Zap/bla,10) instead Maritn On Thu, 6 Nov 2003, Lal, Deepak (Contractor) wrote: When I get a SIP call, I get the following error: *CLI NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 514777, 1) == Spawn extension (incoming, 514777, 1) exited non-zero on 'SIP/-08114358' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, h, 1) == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' In my extensions file, I have the following defined: [incoming] exten = 514777,1,Dial,Zap/2|10 Any suggestions will be appreciated! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error in Incoming SIP call
When I get a SIP call, I get the following error: *CLI NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 514777, 1) == Spawn extension (incoming, 514777, 1) exited non-zero on 'SIP/-08114358' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, h, 1) == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' In my extensions file, I have the following defined: [incoming] exten = 514777,1,Dial,Zap/2|10 Any suggestions will be appreciated! Try: exten = 514777,1,Dial(Zap/2,10) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error in Incoming SIP call
On Thu, 6 Nov 2003, Lal, Deepak (Contractor) wrote: When I get a SIP call, I get the following error: WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 514777, 1) == Spawn extension (incoming, 514777, 1) exited non-zero on 'SIP/-08114358' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, h, 1) == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' In my extensions file, I have the following defined: [incoming] exten = 514777,1,Dial,Zap/2|10 Is this copied and pasted or retyped? I've seen this before when you have a space in the extension line. There shouldn't be any. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
