Merge 2 calls

Hi, I have the following problem to solve:
the operator calls external number A, after the answer put in hold the call A and make another call to exten B. After the answer the goal is to join A and B and the operator close his phone.
Is it possibile this usign Freepbx ?

Don’t use Hold, transfer both calls to the same conference room (have here keep track of the rooms :wink: )

so I should set up a conference and transfer the call to the same room and after I can call my phone ?

you can transfer lots of people into a conference

Thanks !

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With most IP phones, it’s much simpler. While talking to A, press Conf (depending on the phone, a physical key or softkey). A is automatically put on hold. Dial B, you can talk privately with B. Then press Conf again and all three parties are connected. Hang up and A remains connected to B. (With some phones, you need to configure 'transfer on hangup conference for this to work.)

If the operator knows ahead of time that they won’t need to talk with A and B at the same time, it’s even easier. While talking to A, press Tran, dial B. If desired, tell B that you are transferring A to him.
Hang up and A is connected to B.

Thanks !
I have to record the call. May I do it ?

If you record all calls there is no problem.

If you transfer both parties to a conference with Record Conference set to Yes, there is also no problem.

However, I don’t know whether on-demand recording initiated by the operator will continue recording after a transfer, or whether it matters if the transfer is done via SIP (Tran key on phone) or DTMF (dialing *2).

Thanks for your prompt reply.
Caller A call the pbx. I answer and transfer the caller A to 888 conference room, then I call B on mobile and transfer B to 888 conference room, then I have to call 888 to put myself in conference room, I talk to both A and B and then I hang up, while A nd B are talking.
I set record conference to yes.
When caller A hangup and the caller B hang up, the B channel remain in ACK status for 31 sec, after I see on cli:
slight_smile:chan_sip.c:29968 check_rtp_timeout: Disconnecting call ‘SIP/out_01212120992-0000003e’ for lack of RTP activity in 31 seconds
Is there any way to avoid 30 sec of empty recording ?

I am guessing that this has nothing to do with conferences or recording, but that remote BYE on outgoing calls is not working properly. You can confirm or refute this hypothesis by calling your mobile, answering, then hanging up the mobile. Your IP phone should show a disconnect within one or two seconds.

If BYE is indeed not being seen (it takes 30 seconds to disconnect), there could be a problem with the trunk configuration (sip debug should show what’s wrong), or with a hardware firewall between PBX and the internet. Please provide some details about your network.

Conceptually , an asterisk conference bridge is immediately ‘stateless’ unless it has ‘state’, you can kill the conferance immediately if you set ‘leader leaves’ but otherwise the state of the conferance is if there are active participants +0 but if they all leave, you are left with having to poll the connections more often than 30 seconds, which is expensive, or wait for the individual 30s timeouts as it is written right now.

Perhaps the first guy in is a pinless ‘leader’

On my system, even if Leader Leave is set to No, the conference is killed as soon as the last participant leaves. I’m fairly sure that in the OP’s case, a BYE is not received when B hangs up.

I am using a raspberry PI4. Surfing on the web I found that conference needs dahdi driver. I make asterisk from source without dahdi driver, I will try to recompile including dahdi

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