[Freeswitch-users] How to Configure SIP DID to IVR

[Ken Rice]

You don't need a extension created for the cisco... Just set it up to
forward the DID to the freeswitch boxes IP on its dial peer.. Then on
freeswitch you set up a profile w/ auth calls turned off then have a
separate context for that profile that does IP auth for the cisco something
like this
     <extension name="cisco">
       <condition field="network_addr" expression="^192\.168\.2\.1$"/>
       <condition field="destination_number" expression="^<DID number>">
         <action application="answer"/>
         <action application="sleep" data="2000"/>
         <action application="ivr" data="demo_ivr"/>
       </condition>
     </extension>

Setting up gateways is ONLY required if you are going to have to register
and use sip username/password auth

K



> From: Hristo Benev <foxb at abv.bg>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 2 Jul 2008 19:16:03 +0300 (EEST)
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] How to Configure SIP DID to IVR
> 
> Hi,
> 
> I'm new to FS and trying to configure DID only configuration.
> 
> Here is the setup:
> PSTN <->Cisco AS(realIP/maybe multiple ones in production) <-> FS(realIP)
> 
> Cisco box is configured to send SIP to IP (real IP nor 192.168.x.x type) and I
> do not have much control over it. No authentication is needed.
> 
> I'm using FS 1.0.0
> 
> What I need to configure to send incoming PSTN calls to demo IVR
> What I've changed?
> Created cisco.xml file in /conf/directory/default
> ----------------
> <include>
>   <gateway name="Cisco1">
>     <param name="extension" value="<DID number>"/>
>     <param name="realm" value="<CiscoIP>"/>
>     <param name="proxy" value="<CiscoIP>"/>
>   </gateway>
> </include>
> ------------------
> Added to /conf/dialplan/default.xml
> -----------------------------
> <!-- test -->
>         <extension name="cisco">
>       <condition field="destination_number" expression="^<DID number>">
>         <action application="answer"/>
>         <action application="sleep" data="2000"/>
>         <action application="ivr" data="demo_ivr"/>
>       </condition>
>     </extension>
> ------------------------------
> When I call DID it just rings.
> If I connect to FS with SoftPhone on extension and I dial DID.
> 
> I was able to get this configuration working with Asterisk(but had some sound
> quality issues and wanted to try something else) so there is no HW problem.
> 
> Where is my misconfiguration(hopefully just this)?
> 
> Thanks
> 
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