[Kamailio-Users] Having Problem routing an ACK

[Stagg Shelton]

We've been using this provider for a few weeks now with no issues with  
termination to PSTN.  Below is a sip trace from the PSTN to my interop  
system, with the hangup occuring from the PSTN.  The sip trace appears  
to confirm that the BYE from the provider is formed correctly, and is  
properly relayed by openser.

Thanks
Stagg Shelton

#
U +3.305877 63.209.207.135:5060 -> 8.17.32.184:5060
BYE sip:6783832765 at 8.17.32.19 SIP/2.0*
From:
<sip:8005555555 at 8.17.32.184>;tag=88cfd13f-13c4-48b30a4f- 
e8021926-4d82d20d*
To: "Stagg Test" <sip:6783832765 at 8.17.32.19>;tag=as20591d24*
Call-ID: 0430a1061375ebc12c41bfb4170e854a at 8.17.32.19*
CSeq: 1 BYE*
Via: SIP/2.0/UDP
63.209.207.135:5060;branch=z9hG4bK-9149-48b30a57-e8023819-4169a64b*
Max-Forwards: 15*
Route: <sip:8.17.32.184;lr;did=f9.56eb2673>*
Content-Length: 0*
*

#
U +0.002395 8.17.32.184:5060 -> 8.17.32.19:5060
BYE sip:6783832765 at 8.17.32.19 SIP/2.0*
From:
<sip:8005555555 at 8.17.32.184>;tag=88cfd13f-13c4-48b30a4f- 
e8021926-4d82d20d*
To: "Stagg Test" <sip:6783832765 at 8.17.32.19>;tag=as20591d24*
Call-ID: 0430a1061375ebc12c41bfb4170e854a at 8.17.32.19*
CSeq: 1 BYE*
Via: SIP/2.0/UDP 8.17.32.184;branch=z9hG4bK41a6.c4d351.0*
Via: SIP/2.0/UDP
63.209.207.135:5060;branch=z9hG4bK-9149-48b30a57-e8023819-4169a64b*
Max-Forwards: 14*
Content-Length: 0*
*

#
U +0.003511 8.17.32.19:5060 -> 8.17.32.184:5060
SIP/2.0 200 OK*
Via: SIP/2.0/UDP  
8.17.32.184;branch=z9hG4bK41a6.c4d351.0;received=8.17.32.184*
Via: SIP/2.0/UDP
63.209.207.135:5060;branch=z9hG4bK-9149-48b30a57-e8023819-4169a64b*
From:
<sip:8005555555 at 8.17.32.184>;tag=88cfd13f-13c4-48b30a4f- 
e8021926-4d82d20d*
To: "Stagg Test" <sip:6783832765 at 8.17.32.19>;tag=as20591d24*
Call-ID: 0430a1061375ebc12c41bfb4170e854a at 8.17.32.19*
CSeq: 1 BYE*
User-Agent: Asterisk PBX*
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY*
Supported: replaces*
Contact: <sip:6783832765 at 8.17.32.19>*
Content-Length: 0*
*


On Aug 25, 2008, at 3:30 PM, Iñaki Baz Castillo wrote:

> El Lunes, 25 de Agosto de 2008, Klaus Darilion escribió:
>> Hi!
>>
>> For reference if you want to present some standards to your provider:
>>
>> RFC 3261:
>>
>> 12.2.1.1 Generating the Request
>>
>> ...If the route set is not empty, and the first URI in the route set
>>    contains the lr parameter (see Section 19.1.1), the UAC MUST place
>>    the remote target URI into the Request-URI and MUST include a  
>> Route
>>    header field containing the route set values in order, including  
>> all
>>    parameters.
>>
>> The remote target itself is defined in 12.1.2:
>> ...The remote target MUST be set to the URI
>>    from the Contact header field of the response.
>
>
> One question more:
> If you initiate an outgoing call through that provider, is it  
> correctly
> established?
> In that case try to call to a PSTN phone of you, and hang up the  
> call from the
> PSTN in order to receive a BYE from the provider. Is that BYE  
> correctly
> generated? it should have a "Route: sip:openser_ip" and a RURI like  
> the
> Contact you send in the INVITE.
>
>
> -- 
> Iñaki Baz Castillo
>
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