How to Keep Call Recording Active After Assisted Transfer Bridge in Asterisk?

[guilerezende]

Hello, my name is Guilherme, and I have a question regarding Asterisk’s bridge behavior.

I’m facing the following issue:

When I place an outbound call and then put the channel on hold to start an internal call for an assisted transfer, everything works normally until the moment of the transfer. After the second person answers on their phone or softphone, I complete the transfer. At this point, a new bridge is created, but it seems that this bridge is not handled by Asterisk itself. Instead, it looks like the phone/softphone is directly bridging the active channels.

The problem is that my call recording stops exactly when this bridge occurs. The last console lines look like this:

Channel SIP/vivo-00000027 left ‘simple_bridge’ basic-bridge <422956f8-97ad-48f8-81d2-8737f9fbe20b>
Channel SIP/6180-0000002c left ‘simple_bridge’ basic-bridge <2700e82a-7a7d-406c-bb8c-92f97443ec09>
Channel SIP/vivo-00000027 swapped with SIP/6180-0000002c into ‘simple_bridge’ basic-bridge <2700e82a-7a7d-406c-bb8c-92f97443ec09>

Does anyone know how to keep this type of assisted transfer under Asterisk’s control

example:

[david551]

Although a phone could end up doing the bridge (more likely for three way), your logs show Asterisk handling a REFER/Replaces itself.

This sort of case has come up before, on the forum, so I’d suggest doing a search; I can’t remember the details of what was said. However, I would point out that the details will depend on which leg the monitor was set, and on which channel the REFER/Replaces was sent.

Also note that Asterisk cannot distinguish between an enquiry and the start of an attended transfer until the transfer is completed.