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

[Mikael Magnusson]

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