[Freeswitch-users] sip to spa3102 fallback

[Michael Collins]

How about using pipe-separated dial strings in your bridge app?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover

-MC

On Sun, Oct 3, 2010 at 11:55 AM, Eric Masson <emss.mail at gmail.com> wrote:

> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20101004/c176e75d/attachment.html