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

[Samuel Muller]

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
<sip%3A0123451010 at sip.720.fr>>;tag=71078b346a20fb3eo0

To: <sip:0323719001 at sip.720.fr <sip%3A0323719001 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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