[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
Samuel Muller
sml at 720.fr
Thu Dec 4 16:21:18 CET 2008
Hello guys,
many thanks, you were right :)
I changed the PAI and the RPID stuff and it works ...
-- KAMAILIO --
# flag 9 = clir
if (is_avp_set("$avp(s:caller_cli)/s") && !isflagset(9))
{
if (is_present_hf("P-Asserted-Identity"))
{ remove_hf("P-Asserted-Identity"); }
if (is_present_hf("Remote-Party-ID"))
{ remove_hf("Remote-Party-ID"); }
append_hf("P-Asserted-Identity: $avp(s:caller_cli)
<sip:$avp(s:caller_cli)@$fd>\r\n");
append_hf("Remote-Party-ID: $avp(s:caller_cli)
<sip:$avp(s:caller_cli)@$si>;party=caller;privacy=none;screen=yes\r\n");
}
Do you have a better solution to have the best rpid and pai coding way ?
And, is the P-Preferred-Identity really necessary for PSTN ?
log in the gateway :
-- AUDIOCODES --
4d:15h:33m:43s ( lgr_flow)(51994 ) ---- Incoming SIP Message from
77.246.81.132:5060 ----
INVITE sip:0663128505 at 77.246.81.136:5062;transport=udp SIP/2.0
Record-Route: <sip:77.246.81.132;lr=on;ftag=a4143abfbda0611ao0;nat=yes>
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bK89df.da4cd5e2.0
Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162
;branch=z9hG4bK-8a13206a
From: "Sam" <sip:0123451010 at sip.720.fr <sip%3A0123451010 at sip.720.fr>
>;tag=a4143abfbda0611ao0
To: <sip:0663128505 at sip.720.fr <sip%3A0663128505 at sip.720.fr>>
Call-ID: ced89363-47d540c6 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
<sip:0123451010 at sip.720.fr<sip%3A0123451010 at sip.720.fr>
>
Remote-Party-ID: 0123451010
<sip:0123451010 at 77.246.81.162<sip%3A0123451010 at 77.246.81.162>
>;party=caller;privacy=none;screen=yes
v=0 o=- 28033614 28033614 IN IP4 192.168.0.113
s=-
c=IN IP4 77.246.81.133
t=0 0
m=audio 35056 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 [Time: 15:33:43]
( lgr_flow)(51996 ) | | new GetNewSIPCall created - #357 [Time: 15:33:43]
( sip_stack)(51997 ) new AcSIPCallAPI created - #285 [Time: 15:33:43]
( lgr_stk_mngr)(51998 ) Resource StackSession <#285> Allocated [Time:
15:33:43]
( lgr_flow)(51999 ) | |(SIPTU#357)INVITE State:Idle() [Time: 15:33:43]
( sip_stack)(52000 ) SIPCall(#357) changes state from Idle to Invited [Time:
15:33:43]
( lgr_flow)(52001 ) | | | #285:SIP_SETUP_EV(ced89363-47d540c6 at 192.168.0.113)
[Time: 15:33:43]
( lgr_callf)(52002 ) new Call created - #285 [Time: 15:33:43]
( lgr_stk_ses)(52003 ) SIPStackSession::HandleStackSetupEV - NEWCALL:
SrcPN=0 [Time: 15:33:43]
( lgr_stk_ses)(52004 ) <SESSION #285> SendToCall - event: NEW_CALL_EV m_Call
= 108260848 [Time: 15:33:43]
( lgr_flow)(52033 ) ---- Incoming SIP Message from 77.246.81.132:5060 ----
[Time: 15:33:43]
ACK sip:0663128505 at 77.246.81.136:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bK89df.da4cd5e2.0
From: "Sam" <sip:0123451010 at sip.720.fr <sip%3A0123451010 at sip.720.fr>
>;tag=a4143abfbda0611ao0
Call-ID: ced89363-47d540c6 at 192.168.0.113
To: <sip:0663128505 at sip.720.fr <sip%3A0663128505 at sip.720.fr>
>;tag=1c249703390
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: kamailio 1.4.2 - 720 DEGRES
Content-Length: 0
( sip_stack)(52035 ) UdpRtxMngr::Remove 404 Response 102 INVITE [Time:
15:33:43]
( lgr_flow)(52036 ) | |(SIPTU#357)ACK State:Disconnected(
ced89363-47d540c6 at 192.168.0.113) [Time: 15:33:43]
Again, thanks guys :)
.Sam.
On Thu, Dec 4, 2008 at 1:35 PM, Klaus Darilion <klaus.mailinglists at pernau.at
> wrote:
> 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 <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.
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.kamailio.org
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>
>
--
Samuel MULLER
Ingénieur Reseaux & Telecom
720 DEGRES
+33 (0)663 128 505
sml at 720.fr
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20081204/2c9987be/attachment.htm>
More information about the sr-users
mailing list