[Freeswitch-users] Incoming call configuration

[Brian West]

Give me the console output on the failed call inbound.

/b

On Oct 4, 2007, at 2:14 AM, Midiclorian wrote:

> 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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>> Yahoo! Mail
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