Number analysis and manipulation

Hello,

I have two questions:

  1. Is AsteriskNOW running smoothly on VMWare? I’m planning to run it on a Win XP box and this is the only option I have.
  2. Can I achieve the following (and how):

Incoming call is coming from a registered SIP client with extension 123 to number 088xx. A-number analysis is performed on the call and when the “123” pattern is matched the B-number is prefixed with 123#088xx and sent towards a GSM gate (which can analyze only B-numbers so it needs them to be prefixed to select the proper outgoing SIM card and strip the prefix before dialing) defined as a SIP trunk in Asterisk configuration. Now it comes to my mind that to enable SIP-to-SIP calls through Asterisk I’ll have to analyze both A- and B-numbers so that for example 123 dials 456 the call is not routed towards the GSM gate.

My VoIP skills got a bit rusty in the last few years but I guess you got the idea. My experience with Asterisk is 0 so far but if someone can provide a simple example this should be sufficient.

Thank you in advance,

Stefan

Much better if you run in a separate server, but if is imposible look this: asteriskw32, beacuse using virtual machine degrade the sound quality

I know the Win project but ain’t it a bit outdated? The current version is built on Asterisk 1.2.26.2 and Asterisk current release is 1.6.1…

Yes is very very outdated, the last week Digium release asterisk 1.6.2.0. But the use of virtual machine really decrease the perform on DTMF tones and sounds because use the same network card in the host and many packets are lost.

But you can try and view if your results are the desires.
Sorry my English