In internal.xml for sip profiles. 

On Monday, November 23, 2015, Michael Nielsen <mic.niel84@gmail.com> wrote:

Sorry, I meant:

<param name="odbc-dsn" value="dsn:user:pass"/>

Should that take care of everything?

On Mon, Nov 23, 2015 at 1:17 PM, Michael Nielsen <mic.niel84@gmail.com> wrote:

Is it the

<param name="track-calls" value="true"/>

I need to add or an entire postgres connection string?

On Mon, Nov 23, 2015 at 12:47 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:

you must put sofia in database, too

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18

Il 23/Nov/2015 12:38, "Michael Nielsen" <mic.niel84@gmail.com> ha scritto:

But shouldn't FS handle this automatically when they use same database as core?

On Monday, November 23, 2015, Ashwin Rath <ashwinrath@gmail.com> wrote:

Normally the way this should be done is that the server which handles the incoming call should connect to the destination UA through the server which the destination is registered on . So the call flow should be "Caller-->FS One----> FS Two--->Calee".

On Mon, Nov 23, 2015 at 4:48 PM, Michael Nielsen <mic.niel84@gmail.com> wrote:

I've got 2 Freeswitch 1.6.2 setup on different servers: fs1.my-domain.com and fs2.my-domain.com.

I've got a DNS record with 2 A-records for the servers.

When two users ends up on he same server they can call each other. If they land on different servers they cannot. Then they endt up in voicemail.

I'm running the servers with a shared database in postgresql.

Should I do anything else besides db in the core for FS?

They are registrering with the same "main" domain which is fs.my-domain.com.

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