Re: [SR-Users] How do deal with a lost ACK and TM

I am not 100% sure I fully understand my issue, but I think I'm on the right track. I have a situation where Asterisk will drop calls a few seconds after they are set up. What I believe is happening: a=asterisk k=kamailio k --> INVITE --> a a --> 100 TRYING --> k a --> 183 RINGING --> k a --> 200 OK --> k k --> ACK -->a (this packet is never received by asterisk) a --> 200 OK (retransmit) -->k a --> 200 OK (retransmit) -->k a --> 200 OK (retransmit) -->k a --> no response to critical packet - terminating call It doesn't appear that Kamailio is retransmitting the ACK. I would think that would be part of the TM module, but perhaps I am not using it properly. Is this possible or is there something else going on? Thanks. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users