[Freeswitch-users] how to sip uri call

Ken Rice krice at freeswitch.org
Wed Feb 17 09:26:12 MSK 2016


Its all about ordering of the dialplan... first thing that matches wins...



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle
Boogie
Sent: Wednesday, February 17, 2016 12:18 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] how to sip uri call

Hello All,

I'd like my freeswitch server to make sip uri calls to various URI's.

I see this in dialplan/default.xml:
   <!-- If sip_req_host is not a local domain then this has to be an
external sip uri -->
    <!--
    <extension name="external_sip_uri" continue="true">
      <condition field="source" expression="mod_sofia"/>
      <condition field="${outside_call}" expression="^$"/>
      <condition field="${domain_exists(${sip_req_host})}"
expression="true">
        <anti-action application="bridge"
data="sofia/${use_profile}/${sip_to_uri}"/>
      </condition>
    </extension>
    -->

If I enable this, I can call any sip URI but then I can't call an extension
local to the FS instance nor out on the PSTN. I see in

In the same file, I do see this:
<action application="bridge"
  data="sofia/${use_profile}/$1 at conference.freeswitch.org"/>

But I think there must be a method to enable URI calling for any domain--not
just explicitly listed URI.


If there's a confluence page that already covers this, feel free to send the
article to me so I can try it out.


Thanks for any assistance!

--
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si

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