how to sip uri call

17 Feb 2016

      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@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@sip2sip.info
xmpp: jungle-boogie@jit.si

jungle Boogie

Ken Rice

tags

participants (2)