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

[Iskren Hadzhinedev]

Hi Sebastian,
Sorry for the very late reply. I took your advice on originating all 
calls if they exist to the same server and keeping the transfers local 
to the same node.
That way the CDRs stay clean and easy to read/parse at a later stage if 
need be.
I used the htable module to store information for the active calls and 
it works pretty nice.
Thanks for your help!

Kind regards,
/Iskren Hadzhinedev/

On 4.9.2017 г. 9:25, Sebastian Damm wrote:
> 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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