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@abv.bg> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Wed, 2 Jul 2008 19:16:03 +0300 (EEST) To: <freeswitch-users@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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