Samuel Muller-2 wrote: > > Hello all, > > I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the > purpose > is to have several interconnections with PSTN. > > I configured it like this : > > Audiocodes registers as a gateway to the Kamailio, using a dedicated port > (5062). > Registration seems to be OK, and the pstn gw uses OPTIONS method to ping > the > proxy. > I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. > > But the audiocodes returns some errors about SIP headers sent by Kamailio > : > > ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: > 12:30:26] > ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected > symbol > '0' in scheme. ALPHA expected > > Here you have the example of an INVITE from a SIP phone to the PSTN : > > ** audiocodes debug ** > > 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from > 77.246.81.132:5060 ---- > > INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 > Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> > Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 > Via: SIP/2.0/UDP > 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 > > From: "Sam" <sip:0123451...@sip.720.fr > <sip%3a0123451...@sip.720.fr>>;tag=71078b346a20fb3eo0 > > To: <sip:0323719...@sip.720.fr <sip%3a0323719...@sip.720.fr>> > Call-ID: 944d8aec-27503ee6@192.168.0.113 > CSeq: 102 INVITE > Max-Forwards: 49 > Contact: "Sam" <sip:0123451010@77.246.81.162:15170> > Expires: 240 > User-Agent: Linksys/SPA941-5.1.8 > Content-Length: 281 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER > Supported: 100rel, replaces > Content-Type: application/sdp > P-Asserted-Identity: <0123451010> > Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes > v=0 > o=- 26933860 26933860 IN IP4 192.168.0.113 > s=- > c=IN IP4 77.246.81.133 > t=0 0 > m=audio 35038 RTP/AVP 18 0 8 101 > a=rtpmap:18 G729a/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:30 > a=sendrecv > a=nortpproxy:yes > > ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: > 12:30:26] > ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected > symbol > '0' in scheme. ALPHA expected > ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] > ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] > ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] > > > The outgoing INVITE from Kamailio is exactly the same received by the > AudioCodes. > When I searched over Google, I just found 2 answers about Asterisk / > Audiocodes unsolved problem, but no more informations. > > I supposed that the problem is as indicated : " s=- " where source is > empty > in place of "NULL" / "0" or something like this ... > Someone can confirm or already met the problem ? > > Many thanks all :) > > .Sam. > > _______________________________________________ > Users mailing list > us...@lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > -- View this message in context: http://old.nabble.com/AudioCodes-%2B-Kamailio-%3A-Problem-in-SIP-Message-Headers-tp20831861p34481128.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
