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=000000000000000000004aa44670Max-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=000000000000000000004aa44670Max-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 5147771111 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=000000000000000000004aa44670To: <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=000000000000000000004aa44670To: <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 10000 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 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users