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Asterisk PBX and phone lines

Posted on 2014-09-11
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Last Modified: 2014-10-14
Hello All,

I have inherited a number of Asterisk servers running Asterisk SIP Phones in several offices at the company I am currently working with. I am currently running:

Asterisk 1.4.19.1 ON CentOS 4.4

As far as I know, there is no GUI front end and I can only configure anything from the CLI.

One of the offices is moving location and as part of this, they are also changing the phone numbers.

1- Do I need to make any changes inside Asterisk?
2- How and where?

I hasten to add, that I'm not a telephony engineer, so a lot of stuff is a bit foreign to me. I am a Windows engineer really :)

Please could anyone help?
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Question by:Tommy_Cooper
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by:Phonebuff
ID: 40317564
Tommy,

   
Asterisk 1.4.19.1 ON CentOS 4.4

   Is very old --   But the direct answer really depends on how the lines / services are brought into your Asterisk server and rather you have Firewalls and IP addresses changing between the old & new facilities.  

  ============
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by:Tommy_Cooper
ID: 40318833
Phonebuff,

:) Yeah, I know! I think the guys that were here before me should have been sued for criminal irresponsibility! There were no virus scanners running anywhere either. And the firewall rules are still laughable. But slowly we're getting there :) But that's beside the point!

So I believe that the lines come in to our offices to a piece of telco kit. I think that the office in question then has a Cisco ATA 188 that converts it to a CAT5 / RJ45 connection that is then connected to a PCIe card in the server.

Does that help at all?

I've been through all the config files in /etc/asterisk and I can't see anything about the DDI phone numbers that we have, so I'm hoping that if the equipment setup by the telco is all similar, I should be able to just plug and go :)

OR am I fooling myself?

Cheers
Tommy
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Expert Comment

by:Phonebuff
ID: 40319116
Tommy,

So I believe that the lines come in to our offices to a piece of telco kit. I think that the office in question then has a Cisco ATA 188 that converts it to a CAT5 / RJ45 connection that is then connected to a PCIe card in the server.

This is not your dial tone source. the 188 is an FXS device for extensions and Fax machines and such.  Usually in Asterisk they will show up as as a SIP extension --  

Your dial tone will show up as SIP or IAX2 some form of Copper lines (T1, PRI,  FXO)  zapTel or Dadhi into the box.   Then your phones will move out from there.  

From a command line do a "asterisk -rx 'sip show peers' >/tmp/sipPeers.txt

Then look through the resulting file for extensions and trunks..  

=================================================================

FXS - Foreign Exchange Station - (Supports a station and provides -48Vdc)
FXO - Foreign Exchange Station -  (Dial tone from a CO or another PBX expects to see -48 Vdc)
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by:Tommy_Cooper
ID: 40319371
:) Yup - You're losing me!

But... running the command only shows me a list of extensions and ip addresses.  There is nothing in there at all about the connection to the outside world.

How is the connection to the outside world configured in Asterisk? Or is it dumb to the outside world and it will just work if I can connect the wires in the same way as in my existing office?
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Phonebuff earned 500 total points
ID: 40319401
okay,

   Two questions what part of the world are you in and do you want to talk offline --

    =====
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by:Tommy_Cooper
ID: 40319425
Currently in London (UK). Yes - can talk offline. Does EE have PM?
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