[asterisk-dev] pjsip seems to send ACK responses to the wrong destination

[Karsten Wemheuer]

Hi *,

i am trying to analyze a problem with pjsip. 

Scenario: Phones are registered to opensips. From there the calls go to
asterisk and then on via the trunk. This works fine. 

In the opposite direction there is sometimes a problem:
A call comes in over the trunk, asterisk sends the INVITE to opensips.
>From there the INVITE goes to the phone. After the call is answered
(200 OK from phone via proxy), asterisk sends the ACK not via the proxy
but directly to the phone. Looking at the debug log it looks like the
destination address of the ACK is obtained from the Contact or RTP data
and not from the Via header.

I would like to check the source code to see if I am doing something
wrong or if there is a bug. Where do I enter to investigate the
construction of the ACK packet?

Thanks for any hints,

Karsten