[OpenSIPS-Users] BYE with different transport

[SamyGo]

Yes record route headers are set just as the default config file has them.

Also do note that the A party and B party are not registered users. The
setup is also behind NAT as well. Regardless of these two the calls work
completely just flawless in case the caller side is on UDP.

I do see BYE reach my opensips and processes inside the In-dialog condition
and sent to the correct IP:Port of the caller just with wrong transport.

I'll post config snippet and traces shortly.

Regards,
Sammy

On Oct 7, 2016 2:05 AM, "Johan De Clercq" <johan at democon.be> wrote:

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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