[asterisk-dev] Re: ACK during reinvite not sent through SIP Proxy

[H Quintana]

Thanks Mikael,

I have it working now by configuring the SIP proxy to
set the record-route in the 200 reply.

* stores and doesnt modify the Route set when receives
the initial 200 coming from the Called party.  In the
case of the Caller, it seems  * is not storing the
Route set after the first INVITE coming from the
Caller.  Is this a bug?

Best regards,


Humberto


====================================================
H Quintana wrote:
> Hi,
> 
> I'm using 1.2.12.1 with canreinvite=yes and a SIP
> proxy that sets the Record-Route header.  
> 
> After the call is established, Asterisk sends the
> reINVITE to the Caller party through the SIP Proxy,
> once * receives the 200, the ACK is not going
through
> the SIP proxy. In the case of the Called party the
ACK
> goes through the SIP Proxy.
> 
> I also found that further reINVITEs to the Caller
are
> sent directly to its IP and port whithout using the
> SIP proxy.
> 
> The packet capture with the reInvite, the 200 and
the
> ACK, and the second reInvite are at:
> 
> http://pastebin.com/800980
> 
> 
> Any ideas? 
> 

Apparently * cleared the route set when it received
200 Ok for reINVITE 
without Record-Route header, as a result the
subsequent requests within 
the dialog were sent directly to the address in the
remote target uri.

But the route set should not be modified after the 2xx
for the initial 
INVITE is received.

Mikael

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