To get to the bottom of it I'd recommend determining why the ACKs are not getting through to Asterisk rather than trying to work around it. I'm actually suprised Asterisk terminates the call by default when it doesn't get the ACK to it's 200 Ok response that must be new for 1.4.22 as I haven't seen that behaviour in earlier versions. In my opinion it's unwarranted behaviour, if Asterisk is getting RTP then it should leave the call up irrespective of whether it gets an ACK or not.
>From the original SIP trace the ACK does not appear to be arriving at your Asterisk server at all. Try doing a packet trace on the network segment where the calling SIP agent is and see where it's trying to send the ACK to. My guess would be your firewall is incorrectly handling the SIP messages. Generally it's very bad news to use an ALG or firewall to mangle SIP packets as they almost always get it wrong. In your case there is a Record-Route header in the response so the ACK request should be being sent to that address. Perhaps your firewall is not correctly mangling that to allow the request to find its way back to your Asterisk server. Record-Route: <sip:216.82.224.202;lr;ftag=as3ed791f3> Regards, Greyman. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
