Pete, Thank you for script. I can not find find channel variables rdnis, sip_to_user and all others which start with "sb" on wiki page http://wiki.freeswitch.org/wiki/Channel_Variables Are they undocumented? -Vladimir Rodionov On Wed, Aug 5, 2009 at 8:45 PM, Pete Mueller <pete@privateconnect.com>wrote:

Disclaimer: I'm not familiar with all the mods of FS, There may be one that does this already. There are probably many ways to do this, I am just offering one that works well for me.

Item #1 - Findout the callee #. "destination_number" can be set to several different things based on the gateway configuration (forced override with an extension) and may or may not start with a "+" so the example below may not work. To make matters worse, different gateways set fields differently when they hand off the call. The most reliable I've found is "rdnis" or "sip_to_user" , however if you know you are going to stay with one gateway, you can relay on the oddities of the way they are configured. I had to write something relatively generic, so I moved all processing to a script (see #3 below)

Item #2 - Find the caller ID. This is located in "caller_id_number", but remember in your processing that caller ID may be "anonymous", "restricted", "unknown" or some other word when dealing with blocked/private numbers. You cannot looks for just numbers.

Item #3 - Routing. As I mentioned I have 100s of numbers across many gateways, so I needed a way to route the calls to the right places AND know which gateway the call came in on, so I can bridge the call out the same gateway. I handled this by creating a small DB table (using postgreSQL) and connecting using LUA and luasql. The table has three fields: number, gateway, and extension to route to. In my public.xml I list all the places a call can be routed to and the last entry is a unconditional transfer to the "switchboard" script. The switchboard script matches "rdnis" and "sip_to_user" to find the callee and then performs a lookup for the extension to route to.

If you would like a copy of my switchboard script I can provide it to you in a PM. -pete

-------- Original Message -------- Subject: Re: [Freeswitch-users] Multiple DIDs per SIP trunk (how to configure?) From: Vladimir Rodionov <vladrodionov@gmail.com> Date: Wed, August 05, 2009 6:57 pm To: freeswitch-users@lists.freeswitch.org

No, it is more like static routing. I need my *script program* be invoked when somebody dial in. That is it. One script for all inbound DIDs. Suppose I have thousand of them. I think I know how to accomplish this but I am not sure yet.

in my dialplan I need to define:

<!-- Launch a JavaScript application if dialed in--> <extension name="ProviderABC"> <condition field="source" expression="mod_sofia"/> *<condition field="destination_number" expression="^1NXXNXXXXXX$">*

<action application="javascript" data="/usr/local/freeswitch/scripts/myapp.js"/> </condition> </extension>

In provider configuration:

<gateway name="voicepulse"> <!--/// account username *required* ///--> <param name="username" value="your-username"/> <!--/// auth realm: *optional* same as gateway name, if blank ///-->

<param name="realm" value="nyc.voicepulse.com"/> <!--/// account password *required* ///--> <param name="password" value="your-password"/>

<!--/// extension for inbound calls: *optional* same as username, if blank ///--> * * *<param name="extension" value="1NXXNXXXXXX"/> * <!--/// proxy host: *optional* same as realm, if blank ///-->

<param name="proxy" value="nyc.voicepulse.com"/> <!--/// expire in seconds: *optional* 3600, if blank ///--> <param name="expire-seconds" value="600"/>

<param name="register" value="true"/> </gateway>

Something like this, yes? I can use regular expressions in destination_number?

Q: There is object Session in JavaScript, Lua. Is Session.destination == destination_number from incoming call? It is not clear for me from what I have read so far.

TIA,

-Vladimir Rodionov

On Wed, Aug 5, 2009 at 6:26 PM, Seven Du <dujinfang@gmail.com> wrote:

...

mod_easyroute?

2009/8/6 Vladimir Rodionov <vladrodionov@gmail.com>

...

Hi, everybody

This is a newbie question: Suppose I have XX (variable dynamic number) DIDs assigned to one sip trunk (from VOIP provider ABC ). All calls coming from VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it possible in FS? If yes, how everything should be configuered? Dialplan, sip gateway? One more question: suppose it is doeable as I hope then how can I get in my script CalleeID (not a CallerID)? Basicaly,

I want to acomplish the following:

1. Avoid re-configuring FS every time I got new bunch of DIDs assigned/released from/to my Voip provider. 2. Have a way of extracting CalleeID in my script.

TIA,

Vladimir Rodionov

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