[OpenSIPS-Users] BYE with different transport

[Johan De Clercq]

Bye is sent directly if not record routed. Do you have record route header?

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On Fri, Oct 7, 2016 at 12:47 AM +0200, "SamyGo" <govoiper at gmail.com> wrote:












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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