[Freeswitch-users] Sending SIP calls via outbound proxy

[Erick Johnson]

Thanks Dave,

Actually I realized my problem (stupid mistake of course).  For anyone else
trying to use the fs_path variable the value needs to be a fully 
qualified SIP
URI, e.g. "bob at bar.com;fs_path=sip:host.domain.net", notice it being 
prefaced
with the "sip:", my problem was that  I was only entering
the host name.  Then somewhere down in mod_sofia it must have decided that
it didn't like that and just closed the channel.

Hope this helps somebody who gets stuck like I did.

Cheers,

Erick

> Hi Erick,
>
> Not sure if you've tried this (or if it'll help), but you can force
> routing in the dialplan like so:
> <action application="set" data="sip_h_Route=<sip:@11.22.33.44;lr>" />
> <action application="bridge" data="sofia/gateway/gw/$1"/>
>
> Cheers --
>
> Dave