[asterisk-users] Beginning to use Asterisk and tests with extensions

[Dana Harding]

Hello Daniel,

You will find the information at http://www.voip-info.org/ and 
http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the 
"Online Book" link) very useful.

The asterisk package by itself should be adequate for SIP/IAX calls.
I don't think you need libpri unless you are planning on connecting asterisk 
to a digital connection such as ISDN or a PRI.
You will need Zaptel (for Asterisk versions 1.2,1.4) or DAHDI (Asterisk 
versions >=1.6) if you choose to install an internal card (OpenVOX, Digium, 
Sangoma, etc.)      I do not know if or how well this will work with a VM.

I suggest testing your SIP softphone with the Echo() and/or Playback() 
dialplan applications before attempting to call another 
softphone/hardphone/etc.   This will allow you to confirm that the one 
endpoint functions properly before adding more complexity by calling another 
endpoint.

some things that allow you to call a conventional telephone:
    an ATA with an FXS port
    an internal card (such as OpenVOX, Digium, Sangoma) with an FXS port
    call a conventional phone number through the PSTN (below)

To connect to the PSTN you can use any of:
    an ATA with an FXO port (plug an analog phone line into it)
    internal card with an FXO port (also to plug an analog phone line in)
    account with an ITSP (there is occasionally discussion on the list about 
advantages/issues/opinions/and flames with various ITSPs - google 
"site:lists.digium.com ITSP")


An ATA (Analog Telephone Adapter) is basically an analog to digital 
converter.  CISCO/Linksys does manufacture some ATAs, but this is not the 
only option for an ATA.   There are two types of analog ports - FXO, and 
FXS.
You plug a telephone into an FXS port.  You plug a conventional phone line 
into an FXO port.
Bad things can happen if you plug a phone line into an FXS port - do not do 
it.

In your example of an ATA with two FXS ports,  you will use two conventional 
telephones - one plugged into each port.
The ATA will use a different user account for each of the analog ports. 
This requires configuring each user account on both the ATA and in 
Asterisk's sip.conf.         The ATA functions simply as a passthrough 
device:
"phone calls for user1 = ring port1"
"phone calls for user2 = ring port2"
"phone calls from port1 = use user1 account"
"phone calls from port2 = user user2 account"

The ATA does not decide which port is connected to which extension number. 
This actually happens in Asterisk's dialplan.
A very basic extensions.conf to illustrate different extensions to call each 
port, and one to call both:
[default]
exten => 101,1,Dial(SIP/user1)
exten => 102,1,Dial(SIP/user2)
exten => 103,1,Dial(SIP/user1&SIP/user2)

Hope that gets you going in the right direction.

http://www.voipsupply.com/ is a good source to see what equipment is 
generally available to end users.