[Freeswitch-users] sip to spa3102 fallback

[Eric Masson]

Le 03/10/2010 20:55, Eric Masson a écrit :

Hello,

[Responding to myself]

> I have two sip providers configuration files in
> $(fs_conf}/conf/diaplan/default :
> - 00_keyyo.net.xml
> - 00_spa3102.xml
> 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 found a solution that seems to work but a little problem stays.
I deactivated 00_keyyo.net.xml & 00_spa3102.xml then created a new file 
in the same directory :
00_outbound.xml
<include>
   <extension name="outbound_calls" continue="false">
     <condition field="destination_number" expression = "^0(.*)$">
       <action application="set" data="effective_caller_id_name=**MyID**"/>
       <action application="set" 
data="effective_caller_id_number=**MyPhone**"/>
       <action application="bridge" 
data="sofia/gateway/keyyo.net/0$1|sofia/internal/4$1 at spafenmsf:5061"/>
     </condition>
   </extension>
</include>

If keyyo.net is registered, FS uses keyyo.net gateway, if not FS uses 
the spa.

The problem remaining is that FS takes a long time to detect that 
connection to keyyo.net is lost (I've tested by disconnecting the box 
from the internet).

keyyo.net gateway is defined the following way :
keyyo.net.xml
<include>
   <gateway name="keyyo.net">
     <param name="username" value=**MyUsername**/>
     <param name="password" value=**MyPassword**/>
     <param name="register-proxy" value="keyyo.net"/>
     <param name="register" value="true"/>
     <param name="ping" value="60"/>
   </gateway>
</include>

Any idea, please ?

Kind Regards

Eric Masson