[Freeswitch-users] sip to spa3102 fallback

[Eric Masson]

Hello,

I have two sip providers configuration files in 
$(fs_conf}/conf/diaplan/default :
- 00_keyyo.net.xml
<include>
   <extension name="keyyo" continue="false">
     <condition field="destination_number" expression = "^(.*)$">
       <action application="set" data="effective_caller_id_name=**MyID**"/>
       <action application="set" 
data="effective_caller_id_number=**Myphonenumber"/>
       <action application="bridge" data="sofia/gateway/keyyo.net/$1"/>
     </condition>
   </extension>
</include>

- 00_spa3102.xml
<include>
   <extension name="To PSTN">
     <condition field="destination_number" expression="^(.*)$">
       <action application="bridge" 
data="sofia/internal/${destination_number}@spafenmsf:5061" />
     </condition>
   </extension>
</include>

I'd like to implement the following scenario for outbound calls :
- dial using "keyyo" if available
- fallback to "To PSTN" if keyyo isn't available (not registered, 
already used, for example)

I've googled for freeswitch failover setups but didn't find anything 
that could have enlightened me (I'm kind of new in fs).

Where could I find docs to achieve the exposed goal please ?

Kind Regards

Eric Masson