add dialing by sip (thanks trixter)

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@6398 d0543943-73ff-0310-b7d9-9358b9ac24b2

diff --git a/conf/dialplan/default.xml b/conf/dialplan/default.xml
index 8211c82700055a846cc7d4d40451f5e81066f7a6..76cc2dc24fab166c2aa1e00e7a2a8975dc9f9c9f 100644 (file)
--- a/conf/dialplan/default.xml
+++ b/conf/dialplan/default.xml
@@ -34,8 +34,15 @@
       <anti-action application="bridge" data="sofia/$${domain}/${dialed_ext}"/>
       <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/>
     </condition>
-  </extension>                                                                                                                                                 
- 
+  </extension>
+
+  <!-- dial via SIP uri -->
+  <extension name="SIP_URI">
+    <condition field="destination_number" expression="^sip:(.*)$">
+      <action application="bridge" data="sofia/${use_profile}/$1"/>
+    </condition>
+  </extension>
+
   <!--
   start a dynamic conference with the settings of the 
   "default" conference profile in conference.conf.xml