[OpenSIPS-Users] Handling REFER

[Bogdan-Andrei Iancu]

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
>
>
> Sent with Proton Mail <https://proton.me/> secure email.
>
> ------- Original Message -------
> On Tuesday, October 24th, 2023 at 6:59 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org> 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 
>>> <users at lists.opensips.org <mailto:users at lists.opensips.org>> 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
>>>
>>>
>>>     Sent with Proton Mail <https://proton.me/> secure email.
>>>
>>>     ------- Original Message -------
>>>     On Monday, October 23rd, 2023 at 3:39 PM, Bogdan-Andrei Iancu
>>>     <bogdan at opensips.org <mailto:bogdan at opensips.org>> 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
>>>>>
>>>>>
>>>>>
>>>>>     Sent with Proton Mail <https://proton.me/> secure email.
>>>>>
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>>>>
>>>
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>>>
>>>
>>>
>>> -- 
>>> *Carlos E. Wagner*
>>> /Tecnólogo em Telecomunicações, Opensips Certified Professional/****
>>>
>>> *
>>> *Fone:*+55 48 99981-0894
>>> *
>>> *E-mail:*kaduww at gmail.com <mailto:kaduww at gmail.com>_
>>> _
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>>>
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>>
>

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