Attended transfer completion during ringing no audio, unless hold/unhold

I have looked for this problem and cannot find it anywhere.

We are running Freepbx 14 with Asterisk 13. Our extensions are pjsip. When we do an attended transfer, if the transfer is completed while the final destination is still ringing, there is no audio.

User 1 calls User 2; User 2 does an attended transfer to User 3. If User 2 completes the transfer while it is still ringing User 3 then User 1 and User 3 have no audio at all. If User 3 puts the call on hold and picks it back up, audio comes back both directions.

An attended transfer where User 2 waits for User 3 to answer, works fine; both User 1 and User 3 have audio both directions. A blind transfer also works fine.

That’s the definition of an attended transfer, otherwise it would be an blind transfer.

In spite of that, we’ll need logs to be able to help troubleshoot this. A SIP Trace would also be a really good idea.

Are the phones on the same network as the PBX or is the PBX hosted off site?

I am with you 100% on that, and I have instructed the users to use blind transfer if they have no intention of talking to the person on the other end; but my boss still wanted me to attempt to get support for what seems like an audio issue.

The phones are part of the same internal network with routers in-between the subnet the phones are on and the subnet the PBX is on (no natting).

I do need to correct my description of the problem a bit. As it turns out, the audio is not completely missing, but simply one-way… which I have a nagging feeling is a network issue…

I am working on getting logs, I’m just new to pjsip so still working out how to get the right logs for those.

That would be correct. Something is happening during the transfer that is not getting the SDP to update and change from one way to two way but when you hold and unhold it updates the SDP properly. You would need to get a SIP trace of the actual SIP packets to look at while trying to replicate this issue.

Here are the Asterisk logs of the call using feature code *2 to transfer.

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