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" <[email protected]
<mailto:[email protected]>> 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"
<[email protected] <mailto:[email protected]>> 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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