Is it possible to forward a call from one server to another using asterisk?

Here is my scenario I have multiple servers with same dialplan and calls are distributed with Kamailio among these servers but when it comes to conferencing it simply does not work out.

Suppose Caller A wants to join a conference C and dials the number assigned to C
Call is routed to Server S1 and since he is the first user so conference is started at S1

Now another caller B calls and this time call lands to S2 but the conference is started at S1, so I need to forward the call to S1 in such a way so that the caller can join the conference at S1,
and leaves S2 gracefully so that S2 is not involved in conferencing calls between A and B

And same for the Vice versa when the first call lands at S2.

I am not able to forward the calls from S1 to S2 for now, so please help me to understand if we can do this or I am doing it wrong.


Use Dial or Transfer depending on whether or not you want the signalling path to be retained.

I don’t know if PJSIP supports Transfer (I had to guess you were using SIP, as well, as Transfer isn’t supported for most technologies). The error handling in Transfer may not be particularly robust.