[OpenSIPS-Users] BYE with different transport

[Bogdan-Andrei Iancu]

Hello Sammy,

Could you check if the BYE you receive does have two RR headers (one UDP 
and one TCP) ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 07.10.2016 15:12, SamyGo wrote:
>
> 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 
> <mailto:johan at democon.be>> wrote:
>
>     Bye is sent directly if not record routed. Do you have record
>     route header?
>
>     Get Outlook for iOS <https://aka.ms/o0ukef>
>
>
>
>
>     On Fri, Oct 7, 2016 at 12:47 AM +0200, "SamyGo"
>     <govoiper at gmail.com <mailto: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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