[Freeswitch-users] Send calls through registered gateway

[David Villasmil]

Hello,

Thanks for the reply.

Not what I'm looking for though.
I register an asterisk with with, say with username 1001. I receive a call
from user 2 and want to terminate what 1002 sends with the gateway (not a
gateway  per se, but a registered user).

in the dialplan I have:

          <action application="bridge"
data="sofia/internal/1001%domain^NUMBER_TO_DIAL"/>


But all asterisk gets is a call to extension "s" in the INVITE, even though
the "To" header is OK.... how do I set it correctly??

INVITE sip:s at 5.6.7.8 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgmBy3gy8BSKZB
Max-Forwards: 69
From: "David Villasmil" <sip:1002 at 1.2.3.4>;tag=1ep74eBrB2Fcj
To: <sip:912074419 at 1.2.3.4>
Call-ID: 3393725d-e8d7-122f-6395-0013725ca38a
CSeq: 25549284 INVITE
Contact: <sip:mod_sofia at 1.2.3.4:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 482
X-FS-Support: update_display
Remote-Party-ID: "David Villasmil" <sip:1002 at 1.2.3.4
>;party=calling;screen=yes;privacy=off


On Thu, Mar 15, 2012 at 1:23 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Wed, Mar 14, 2012 at 5:58 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
> > I seem to be unable to send calls through a registered gateway... is
> there
> > any sample dialplan available?
>
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways
>
> Best,
> Gabe
>
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