[Freeswitch-users] Call FS users from external gateway.

[Bote Man]

This is the job of the dialplan. The example entries that are included with the vanilla FS config provide guidance, but in my particular case I had to test on a different SIP field than the vanilla config was looking for.

 

My inbound calls from CallCentric put the destination number (DID) in sip_to_user so my dialplan test condition reads:

    <condition field="${sip_to_user}" expression="^(12345678900)$">

 

I had to read the FS debug logs to see what it was seeing, and then adjust my dialplan accordingly. 

 

If you have many FS users you might consider looking up the directory entries with XML as is typically done.

 

Here is a good place to start for more information:

https://freeswitch.org/confluence/display/FREESWITCH/Dialplan

 

Hope this helps.

 

Bote

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Nielsen
Sent: Wednesday, 16 September, 2015 08:33
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Call FS users from external gateway.

 

I've got my users in FS in /directory/users.xml in a certain domain and user-context = public.

I've got my FS hooked up to a SIP gateway for connection to the GSM-world.

 

I'm able to route calls to my SIP gateway for outbound calls, but incoming calls to my FS from the GSM-world does not get routed to my users in FS.

 

My users have the ID's "+4412345678", and incoming calls from my SIP gateway does contain +4412345678 at my-sip-gateway-domain.com

 

I guess I need to tell FS somehow that incoming calls from my SIP gateway should match my user IDs in users from my /directory/users.xml.

 

But how to I do this?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150916/9f2e04cd/attachment-0001.html