[OpenSIPS-Users] Handling REFER
nutxase
nutxase at proton.me
Tue Oct 24 15:01:52 UTC 2023
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 <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> 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> 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.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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>> --
>>
>> Carlos E. Wagner
>> Tecnólogo em Telecomunicações, Opensips Certified Professional
>>
>> Fone: +55 48 99981-0894
>> E-mail: kaduww at gmail.com
>>
>> LinkedIn: https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/
>>
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