[SR-Users] Call transfer between pbx servers behind kamailio proxy

Sebastian Damm damm at sipgate.de
Mon Sep 4 08:25:17 CEST 2017


Hi,

On Fri, Sep 1, 2017 at 2:35 PM, Iskren Hadzhinedev
<iskren.hadzhinedev at ikiji.com> wrote:
> Hi everyone,
> I'm having a hard time transferring calls when the users are on different
> FreeSWITCH servers behind kamailio.

We had to solve that problem a while ago, too. Only with Asterisk
instead of Freeswitch. In newer versions, Asterisk has support for
remote transfer built in, but their own page says you should avoid it
if possible.
https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers

I don't know whether that's a possible solution for you, but we found
a solution to make sure whenever there are more than one call for the
same user, they both run through the same Asterisk. So we remember (in
Kamailio whether there are running calls for a user and where they
came from or got dispatched to, and whenever a second call comes in
for that user, it gets sent to the same Asterisk. This way we have
only local transfers, and those do work.

Maybe this is something you can implement in your setup, too. However,
if you really need to send the call to user3 via FreeSWITCH2, then
maybe it could be a solution to first send the ourbound call from
user2 first to FreeSWITCH1 and then dial something on FreeSWITCH2.
This way you will also have the two channels to bridge on FreeSWITCH1.

HTH
Sebastian



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