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-spec
A 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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