Comments for The World of VoIP, Asterisk and Linux

Comment on Calling Multiple Extensions and Let Them All Answer Using Standard Dialplan by emma

emma — Thu, 16 Feb 2012 11:35:03 +0000

Thanks, this is what i needed, I am looking for a production solution so a lot of enhancements will be done but this gives a good starting point, primarily am trying to do a VoIP to GSM calls with multiple numbers and this is perfect. Has anyone done anything like this before?

regards

Comment on Calling Multiple Extensions and Let Them All Answer Using Standard Dialplan by Zeeshan A Zakaria

Zeeshan A Zakaria — Fri, 10 Feb 2012 15:12:29 +0000

Hi Emma,

Thanks for reading my blog post.

The method described in this post is very basic, not applicable for a large scale or any serious production system. I did something similar to what you are looking for using an AGI script written in PHP which called a list of dynamically added phone numbers, and as the callees picked up their phones, they were added into a conference room. That included web interface and MySQL, so it was pretty comlicated.

However using the same technique as above, what you can do is define all the numbers you want to call in variables, e.g. if you have same 3 numbers to call, do this

[callnumbers]
exten => 100,1,Set(number1=”7778884457″)
exten => 100,n,Set(number2=”7784445544″)
exten => 100,n,Set(number3=”7775544887″)
exten => 100,n,Set(count=1)
exten => 100,n,GotoIf($[ ${count} < 4]?6:9)
exten => 100,n,System(asterisk -rx “originate SIP/number${count} extension s@mycode”)
exten => 100,n,Set(count=$[${count} + 1])
exten => 100,n,Goto(5)
exten => 100,n,NoOp(end of loop)

As I mentioned earlier, for dynamically entered numbers the dialplan would be much more complicated and would require AGI scripting. I don’t remember where are my old scripts at this moment, and because they were specific to that client’s need, so they won’t work in a general scenario.

Comment on Calling Multiple Extensions and Let Them All Answer Using Standard Dialplan by emma

emma — Fri, 10 Feb 2012 11:56:35 +0000

hello Will
Did you get this working, i have also been trying to achieve the same but using multiple numbers i.e. dial multiple numbers at the same time and whoever answers is added to a conference automatically. Do you have a sample of the configs?

regards

Comment on Getting started with AEL by Zeeshan A Zakaria

Zeeshan A Zakaria — Mon, 30 Jan 2012 17:59:44 +0000

You are welcome Rajeev.

Unfortunately there is no detailed documentation on AEL. The only one page into to AEL on voip-info is all where I learnt it from, and later gained experience by practice. Actually its not that hard. All I can suggest is start writing small dialplans and soon you’ll be good at it. If you have an existing dialplan, start converting it into AEL. All you have to do is delete extensions.conf and Asterisk will automatically look for extensions.ael (unless your version of Asterisk requires some other configuration, I never moved beyond Asterisk 1.4).

Comment on Getting started with AEL by RT

RT — Mon, 30 Jan 2012 17:51:05 +0000

Zeeshan, Thanks for such a benign intro. I too was apprehensive about dialplan and syntax. AEL is a good thing, but I do not see any conceptual documentation of AEL (the way it works – or how does it co exist with extensions.conf) The vioi-info is a good place but I wanted to know if any officially supported document was there? The Asterisk site does have it. The way you have mentioned looks better to me but would prefer more elaborated details, if possible.

Can both co-exist?

Comment on How to setup a VoIP Server by Roo

Roo — Fri, 20 Jan 2012 22:16:12 +0000

As the last comment I too would like to use a VoIP server while I’m abroad.
I’m assuming if I setup a server as suggested above, I could setup extensions for mobile phones (which have sip software) and access my server via internet access abroad.

This would then go from my phone – to my server at home – and then to (for instance) my wife who is in the same country as me?

Very helpfull info thank you.

Comment on Install OpenSIPS on CentOS 5.2 with MySQL by Sam

Sam — Fri, 21 Oct 2011 04:56:18 +0000

Zeeshan,
Do you have any experience working with Kamailio?

Comment on From Scratch to Finish: Setting up and Configure Asterisk for Production by Sam

Sam — Fri, 21 Oct 2011 04:52:28 +0000

Yes, but it was only for Realtime Queue.
I asked for hardware configuration only because being not so good at asterisk dimensioning I have/had no idea what it would take to have around 200 Agents, 200 simultaneous calls(Dahdi 2 SIP), every call getting recorded, around 200 queues in an asterisk system.

Comment on From Scratch to Finish: Setting up and Configure Asterisk for Production by Zeeshan A Zakaria

Zeeshan A Zakaria — Fri, 21 Oct 2011 03:50:05 +0000

So many different configurations, I don’t even remember.

Have you also used the real-time architecture?

Comment on From Scratch to Finish: Setting up and Configure Asterisk for Production by Sam

Sam — Wed, 19 Oct 2011 12:23:28 +0000

BTW, let me tell you, you have been doing great job on your site as well as on Asterisk-users group.