[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers

[Raj Jain]

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 at 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 at 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 at sip.720.fr>;tag=71078b346a20fb3eo0
> To: <sip:0323719001 at sip.720.fr>
> Call-ID: 944d8aec-27503ee6 at 192.168.0.113
> CSeq: 102 INVITE
> Max-Forwards: 49
> Contact: "Sam" <sip:0123451010 at 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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