Further, the log message does not have an empty line between SIP headers and the body. Either you have forgotten to add \r\n when adding the header or this is just not diplays correctly in the logfile.
klaus
Raj Jain schrieb:
It seems that the P-Asserted-Identity header is not correctly formatted in the INVITE. It must be a sip, sips, or tel URI. This would be something that your proxy is adding to the INVITE. Here is a quote from section RFC 3325.
9.1 The P-Asserted-Identity Header
The P-Asserted-Identity header field is used among trusted SIP entities (typically intermediaries) to carry the identity of the user sending a SIP message as it was verified by authentication.
PAssertedID = "P-Asserted-Identity" HCOLON PAssertedID-value *(COMMA PAssertedID-value) PAssertedID-value = name-addr / addr-specA P-Asserted-Identity header field value MUST consist of exactly one name-addr or addr-spec. There may be one or two P-Asserted-Identity values. If there is one value, it MUST be a sip, sips, or tel URI.
-- Raj Jain
On Thu, Dec 4, 2008 at 6:41 AM, Samuel Muller sml@720.fr 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:0123451010@sip.720.fr;tag=71078b346a20fb3eo0 To: sip:0323719001@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.
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