[Kamailio-Users] Having Problem routing an ACK

[Klaus Darilion]

Hi!

AFAIS the client is buggy (or is there a NAT ALG/Firewall between client 
and SIP proxy?). Compare the Contact header in the 200 OK and the 
request URI in the ACK. They MUST be the same!!!

regards
klaus


U +0.000315 8.17.32.184:5060 -> 63.209.207.135:5060
SIP/2.0 200 OK*
Via: SIP/2.0/UDP 
63.209.207.135:5060;branch=z9hG4bK-8921-48b022df-dcaa3e6a-2f5ec169*
Record-Route: <sip:8.17.32.184;lr=on;did=952.4d684275>*
From: Anonymous 
<sip:restricted at 63.209.207.135>;tag=88cfd13f-13c4-48b022df-dcaa3e6a-b4657f0*
To: <sip:+16783832765 at 8.17.32.184:5060>;tag=as40da5b97*
Call-ID: ATLMGC0720080823144655027771 at 209.244.63.15*
CSeq: 1 INVITE*
User-Agent: Asterisk PBX*
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY*
Supported: replaces*
Contact: <sip:+16783832765 at 8.17.32.19>*
Content-Type: application/sdp*
Content-Length: 180*



U +0.072541 63.209.207.135:5060 -> 8.17.32.184:5060
ACK sip:+16783832765 at 8.17.32.184 SIP/2.0*
From: Anonymous 
<sip:restricted at 63.209.207.135>;tag=88cfd13f-13c4-48b022df-dcaa3e6a-b4657f0*
To: <sip:+16783832765 at 8.17.32.184:5060>;tag=as40da5b97*
Call-ID: ATLMGC0720080823144655027771 at 209.244.63.15*
CSeq: 1 ACK*
Via: SIP/2.0/UDP 
63.209.207.135:5060;branch=z9hG4bK-8922-48b022e5-dcaa5757-3884948f*
Max-Forwards: 15*
Contact: <sip:restricted at 63.209.207.135:5060;transport=udp>*
Route: <sip:8.17.32.184;lr;did=952.4d684275>*
Content-Length: 0*




Stagg Shelton schrieb:
> Thanks again Iñaki.  I am attaching siptrace.txt file.  I can see that 
> there appears to be something odd with the ACKs in that they appear to 
> be sent from my openser back to my openser in a loop until the max 
> forwards is reached.
> 
> 
> ------------------------------------------------------------------------
> 
> 
> 
> 
> 
> 
> 
> Thank you for your help.
> 
> 
> 
> Stagg Shelton.
> 
> 
> 
> 
> 
> On Aug 23, 2008, at 10:08 AM, Iñaki Baz Castillo wrote:
> 
> 
> 
>> El Sábado, 23 de Agosto de 2008, Stagg Shelton escribió:
>>> Iñaki,
>>>
>>> Thank you for your response.  I have enabled the siptrace module in
>>> openser.  The data in the mysql table only shows the trace between the
>>> carrier and openser.  Can I submit a pcap file that shows all of the
>>> SIP communication that occured during the call.
>>
>> Hi, you don't need to enable siptrace. Just install "ngrep" and do:
>>
>>  ngrep -d any -P '*' -W byline -T port 5060
>>
>>
>> -- 
>> Iñaki Baz Castillo
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> 
> 
> 
> ------------------------------------------------------------------------
> 
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