[Freeswitch-users] sip_to_user and destination number

Tanguy phenix at vfemail.net
Sun May 3 21:39:06 MSD 2015


Hi Sergey

I also tried like you ( note the initial post ) using directly a 
condition field with with ${sip_to_user} variable instead of destination 
and it worked even with the bad sip profile but it's better to avoid 
specific dialplan when using a GUI, and i was not sure of the To: field 
with my provider who send correct INVITE.

Regards

On 03/05/2015 18:54, Sergey Safarov wrote:
> Hi Tanguy
> I has tested case where:
> 1) gateway linked to public context;
> 2) gateway has parameter "<param name="context" value="did_processing"/>";
> 3) FS configuration has following dialplan.
>
> <context name="did_processing">
> <extension name="info" continue="true">
> <condition>
> <action application="info" />
> </condition>
> </extension>
>
> <extension name="did_processing">
> <condition field="${sip_to_user}" expression="^11_my_did_number_22$" 
> break="on-true">
> <action application="transfer" data="sales XML defaul" />
> </condition>
> <condition field="${sip_to_user}" expression="^33_my_did_number_44$" 
> break="on-true">
> <action application="transfer" data="support XML defaul" />
> </condition>
> <condition>
> <action application="hangup" data="INCOMING_CALL_BARRED"/>
> </condition>
> </extension>
> </context>
>
> It it working correctly.
>
>




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