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

[Klaus Darilion]

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-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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>>
>>
> 
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