[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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