Hello all,<br><br>I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN.<br><br>I configured it like this :<br><br>Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062).<br>
Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy.<br>I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm.<br><br>But the audiocodes returns some errors about SIP headers sent by Kamailio :<br>
<br>
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26]<br>
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected<br><br>Here you have the example of an INVITE from a SIP phone to the PSTN :<br><br>** audiocodes debug **<br>
<br>4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from <a href="http://77.246.81.132:5060">77.246.81.132:5060</a> ----<br><br>INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 <br>Record-Route: <sip:<a href="http://77.246.81.132">77.246.81.132</a>;lr=on;ftag=71078b346a20fb3eo0;nat=yes> <br>
Via: SIP/2.0/UDP <a href="http://77.246.81.132">77.246.81.132</a>;branch=z9hG4bKdace.1ab1d59.0 <br>Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=<a href="http://77.246.81.162">77.246.81.162</a>;branch=z9hG4bK-b432f96 <br>
From: "Sam" <<a href="mailto:sip%3A0123451010@sip.720.fr">sip:0123451010@sip.720.fr</a>>;tag=71078b346a20fb3eo0 <br>To: <<a href="mailto:sip%3A0323719001@sip.720.fr">sip:0323719001@sip.720.fr</a>> <br>
Call-ID: <a href="mailto:944d8aec-27503ee6@192.168.0.113">944d8aec-27503ee6@192.168.0.113</a> <br>CSeq: 102 INVITE <br>Max-Forwards: 49 <br>Contact: "Sam" <<a href="http://sip:0123451010@77.246.81.162:15170">sip:0123451010@77.246.81.162:15170</a>> <br>
Expires: 240 <br>User-Agent: Linksys/SPA941-5.1.8 <br>Content-Length: 281 <br>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER <br>Supported: 100rel, replaces <br>Content-Type: application/sdp <br>P-Asserted-Identity: <0123451010> <br>
Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes <br>v=0 <br>o=- 26933860 26933860 IN IP4 <a href="http://192.168.0.113">192.168.0.113</a> <br>s=- <br>c=IN IP4 <a href="http://77.246.81.133">77.246.81.133</a> <br>
t=0 0 <br>m=audio 35038 RTP/AVP 18 0 8 101 <br>a=rtpmap:18 G729a/8000 <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:8 PCMA/8000 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-15 <br>a=ptime:30 <br>a=sendrecv <br>a=nortpproxy:yes<br>
<br>( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26]<br>( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected<br>( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26]<br>
( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26]<br>( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26]<br><br><br>The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes.<br>
When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations.<br><br>I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ...<br>
Someone can confirm or already met the problem ?<br><br>Many thanks all :)<br><br>.Sam.<br><br><br><br><br>