I need to set up a simple IVR system for a friend's company, through the menu by pressing the caller Lets navigate.
Press '1' for today's schedule, '2' for the program tomorrow and so on.
This is only an information system, i.e. any navigation route will end with a real person, but only audio messages will be played.
Now, I have never set up such a set before and have diged a little on Google. Looks like I will be able to get it using Asterisk
- What do I need hardware-wise?
- Is a Simple Linux Server and a VoIP Account Enough with a Provider in Germany?
- Will a VPS handle the work?
- How about several concurrent incoming calls?
- Are they controlled by Asterisk?
This is entirely possible.
What you need to know:
-
There are some problems with the
H323in the asterisk. If your provider asks forSIP, instead ofSIP. -
You have a whole
IVRin yourextensions.conf, but for complex tasks, thisAGIIs better to use. These arepearlorpythonor any language scripts that apply yourIVRargument > - < P> Multiple concurrent
- Call is not a problem, my installation of asterisk on a simple PC < Em> hundreds of combined calls
-
Things that affect actually display sound conversion and < Strong> Tone detection .
To improve the performance, you must:
- Paste on
a codec (μLawwhich I use) Force allSIPconnections to use that codec, and you'll do this as soon as your code isSOCS-T UL, file bytes of all Astro Operation Disks On reading and networking they are just like sending with original wrapping. -
There is no math for your provider to identify the tone on your behalf and send them outside the band ,
RFC 2833 By using. To detect tone a lot of CPU consumption is operating, it does it for its own sake. I personally run Asterisk on2,66 MHz Celeron IV2048 MB RAM,Under Fedora 10 X86_64. 150 connections work just once, there is no delay.Approximately 9.6 KByte / sec per connection should not be a problem for modern VPS.
- Paste on
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