[Freeswitch-users] Attended transfer to a conference room

[Emrah]

Hi all,

I am experiencing a strange issue with SIP based attended transfers.

If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up.
There is no much activity on the SIP side of things, it seems to be very much related to FS.

Some info:
I call out through a provider configured on the external profile, from a phone registered on the internal profile.
It is not a codec conflict.
Both lines are answered when I actually finalize the transfer.
I tried with multiple phones and softphones.

I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.

Best,
Emrah