Hi All, I have a opensips 2.2 with residential script loaded. A TCP client makes a call and that call gets forwarded to FreeSWITCH over UDP. The call establishes just fine and everything works smooth untill the B party sends the BYE. That BYE comes over UDP and hence opensips tries to send the BYE to the A side over UDP. Hence as a result A party's phone stays oncall.
I have to manually go to the loose_route's BYE section and set the force_send_socket ($fs) to use TCP. Is there something that tells opensips to use same transport as the INITIAL invite? Regards, Sammy
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