[OpenSIPS-Users] Attended call transfer in opensips with use of RTPengine

[Simon Gajski]

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
>>
>>
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>> Users at lists.opensips.org
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>
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