[Freeswitch-users] Sending SIP calls via outbound proxy

[Michael Collins]

On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson
<erick at junctionnetworks.com> wrote:
> 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.

Erick, thanks for the clarification! I'll get it put on the wiki right away.
-MC

>
> 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
>
>
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