[Freeswitch-users] Incoming call configuration

[Midiclorian]

Hi Brian,
   
  thanks in advance for your quick reply.
   
  Actually I am using the following default_context.xml
   
  <!-- Valid fields in conditions: -->
<!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, -->
<!-- rdnis, destination_number, uuid, source, context, chan_name" -->
  <!-- *NOTE* The special context name 'any' will match any context -->
<context>
  <extension name="from-unidata" continue="true">
    <condition field="destination_number" expression="390640043101">
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="continue_on_fail=true"/>
      <action application="bridge" data="sofia/sip/513 at 192.168.1.213"/>
    </condition>
  </extension>
  
</context>
   
  where 390640043101 is the SIP account number given me from my SIP provider.
  With the current configuration when the INVITE is received by FS, it answers with 404 Not found
  When I receive a call, I would like to route it towards 513 at 192.168.1.213 via SIP, so which sofia profile should I use?
   
  Thanks a lot and regards,
  Francesco.


Brian West <brian.west at mac.com> ha scritto:
  Example 1:  

  <extension name="from-sip-provider">                                                                                                                                       
      <condition field="destination_number" expression="55512512">                                                                                                    
        <action application="bridge" data="sofia/profile/extension at ip"/>                                                                                              
      </condition>                                                                                                                                                
    </extension>
  

  

  /b
  

  

  
    On Oct 3, 2007, at 10:54 AM, Midiclorian wrote:

    Hi all,
   
  I have managed to have FS registered to my SIP Provider.
   
  Now how can I set the routing rules so that when I receive an incoming call form my external SIP provider, the call is routed to an extension?
   
  Thanks a lot and regards,
  Francesco.
  

  
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