Well, maybe posting some network capture + script snippet for the
sequential requests will help here.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 10/24/23 6:01 PM, nutxase wrote:
Strangely, when i put a loose_route() or record_route() then it does
not even try transfer
This is a webrtc client going from opensips to asterisk/freeswitch
with mid_registrar
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------- Original Message -------
On Tuesday, October 24th, 2023 at 6:59 AM, Bogdan-Andrei Iancu
<[email protected]> wrote:
:+1:
no lookup for sequential, just loose_route(). Again, you should do
nothing special for REFER. If the BYE's work for you, the REFER
should also.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 10/24/23 7:02 AM, Carlos Eduardo wrote:
It fails because you're sending a sequential request to another
endpoint. As it doesn't have the dialog there, it will fail.
You should route the REFER as any other sequential request and then
the other UA will handle it and transfer.
Em seg., 23 de out. de 2023 às 12:02, nutxase via Users
<[email protected] <mailto:[email protected]>> escreveu:
So when using a webrtc client with mid_registrar it seems the
transfer does nothing
but if i put something like this
if ( has_totag() && is_method("REFER") ) {
mid_registrar_lookup("location","i","$tu:5060");
t_relay();
exit;
}
then a call transfers but doesnt drop the transferer's call
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------- Original Message -------
On Monday, October 23rd, 2023 at 3:39 PM, Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>> wrote:
Hi,
The REFER is an in-dialog request like any other (re-INVITE and
BYE), so no special handling. What transfer scenario are you
currently failing ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com <https://www.opensips-solutions.com>
https://www.siphub.com <https://www.siphub.com>
On 10/16/23 5:49 PM, nutxase via Users wrote:
Hi All
I am using opensips as a mid_registrar for webrtc and
everything is working fine except call transfers, as i
understand they use refer, is there anything specific i need
to change to get them to work?
if you can point me to a module id appreciate it
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