Hi
no, we don't use B2B on OpenSIPS side. Is this the correctway to do it?
The thing is that I don't know what would be the best way to implement
this with use of RTPengine.
I found very little info available online.
Call forwards that I manually set in DB (cfu, cfnr, cfb.....like we were
doing on last bootcamp) are working fine.
Only issue is with answered call and then attempting to transfer it.
BR
Simon
Bogdan-Andrei Iancu je 01.03.2022 ob 15:30 napisal:
Hi Simon,
Do you use B2B on the OpenSIPS side ? Which entity is actually
performing the transfer ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS eBootcamp
https://www.opensips.org/Training/Bootcamp
On 2/24/22 1:54 PM, Simon Gajski via Users wrote:
Hi
I am using opensips 3.2 with rtpengine on same server and trying to
achieve attended call transfer.
In theory, I'm trying to do:
1. A calls B...and B answers
2. B puts A on hold (MOH is played from RTPengine)
3. B calls C...and C answers
Now the funny part:
B tries to transfer A to C and sends REFER to opensips
In opensips I responds with 202 Accepted and B gets disconnected.
However A and C don't get connected together
A still receives MOH and C has no voice
We have another installation of opensips where REFER handles
Freeswitch, and there such type of transfer is working fine.
Can someone help me how to handle such call behaviour in opensips
with RTPengine?
relevant part of code:
route[handle_sequential]{
...
if(is_method("REFER")) {
xlog("[IN_DIALOG] [$rm] Transfer from $fu to $tu");
send_reply(202, "Accepted");
#what next?
exit;
}
...
}
Thank you!
Simon
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
