4 Dec
2008
4 Dec
'08
2:28 p.m.
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(a)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(a)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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