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[vladrodionov at 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

[dujinfang at gmail.com]

mod_easyroute?

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

[vladrodionov at gmail.com]

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:

[msc at freeswitch.org]

On Wed, Aug 5, 2009 at 8:57 PM, Vladimir Rodionov <vladrodionov@gmail.com (vladrodionov@gmail.com)> wrote:

[pete at privateconnect...]

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

[vladrodionov at gmail.com]

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 (pete@privateconnect.com)> wrote:

[vladrodionov at gmail.com]

Thanks, I will give it it a try and let you know.

On Wed, Aug 5, 2009 at 8:40 PM, Michael Collins <msc@freeswitch.org (msc@freeswitch.org)> wrote: