[asterisk-users] force sip URI call through PBX

[D'Arcy J.M. Cain]

On Mon, 30 Nov 2015 16:42:02 +0100
Julien Sansonnens <julien at jsansonnens.ch> wrote:
> When I do a SIP URI call from my softphone, the call is made directly
> to the destination host (p2p), bypassing the PBX. So I lose the
> possibility of recording, making statistics, etc ...
> 
> Is there a way to force URI calls through the PBX? I have found no
> configuration at the client or at the server level. Do you know any
> softphone that will allows me to do this ?

If two phones are calling each other directly then there is no server
setting that will reach across the Internet and grab the call.  You
need to insure that the call is proxied through your PBX.  That's just
a setup in your softphone.  You will need to ask about that on a
mailing list for that software.

One thing that I do so that I can call SIP phones from my regular phone
through an ATA is set up extensions for them in Asterisk like this.

exten => 6135553638,1,Dial(SIP/my.friend at example.com)

Looks like a regular call to my users but bypasses the PSTN.  This
might work for you as well.

Whatever solution you use, you may want to look at directmedia
settings.  If you can talk directly to another SIP client Asterisk may
step out of the picture anyway not allowing you to record the call.
Turning that off forces all the calls to be proxied through you even if
they could talk directly.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net