About To Take The Asterisk Journey

Fatman

Member

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Messages: 84

Hi everyone,

I would like to start looking into using asterisk. Could someone please post some links to good howtos or recommend some books.

Thanks,

Fatman
 

Oko

Daemon

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Messages: 1,620

Fatman said:
Hi everyone,

I would like to start looking into using asterisk. Could someone please post some links to good howtos or recommend some books.

Thanks,

Fatman

You can download Asterix the future of telephony from this link.
Click on the covers of the book
http://www.asteriskdocs.org/

There is also a fine FreeBSD distro devoted to implementation of Asterix on FreeBSD

http://www.askozia.com/pbx/
 
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Fatman

Member

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Messages: 84

Thanks for the advice. My boss at work passed me the following book: AsteriskNOW

I'm somewhat confused about VoIP and was hoping someone could help me understand something. According to this book, if I have a TDM400 with 3xFXO modules and 1xFXS module, I would do the following:

-Any cellular number (07XXXXXXXXX) should be directed to
FXO module 1.
-All other local numbers should be directed to FXO module 2.
-All international calls should be routed to the international
SIP trunk.
-All faxes should be directed to FXO module 3.

Am I forced to have three seperate lines, each to it's own Telco? Is there no way to route everything out to SIP trunks? I thought the Idea of VoIP was to allow the termination elsewhere which means you don't need a T1.

With the setup in this book, i would require a couple of T1s since every call takes a line. So if I have a business with 500 desks, I would need the many T1 to handle all the calls.

I have a feeling that this might be a stupid question so please do not flame me. :p
 

lbl

Member

Reaction score: 8
Messages: 40

Well if you have a SIP interconnect you can send and recive miltiple calls on then you just need that.

Tho it is important that you use g.711a or g.711u as a voice codec since its the only voice codecs that supports FAX.

If you want a small local setup all you need to do is fix sip.conf and extensions.conf.

This is a quick example with 3 SIP phones in house and a fax with a SIP adapter on.

sip.conf:
Code:
[general]

dtmfmode=auto ;DTMF mode auto so people can use the tones during a call.

disallow=all ;Disallow all codecs.
allow=alaw ;Allow only alaw (G.711a).

; Register to a sip provider to recive calls.
register => USERNAME:PASSWORD@SIPPROVIDER_IP/PHONENUMBER

[sipprovider]
type=friend
fromuser=PHONENUMBER
username=USERNAME
secret=PASSWORD
host=SIPPROVIDER_ip
canreinvite=no
qualify=300
insecure=port,invite
context=incomming

; Phone 1
[2001]
type=friend
username=2001
secret=2001
host=dynamic
canreinvite=no
qualify=5000
context=internal

; Phone 2
[2002]
type=friend
username=2002
secret=2002
host=dynamic
canreinvite=no
qualify=5000
context=internal

; Phone 3
[2003]
type=friend
username=2003
secret=2003
host=dynamic
canreinvite=no
qualify=5000
context=internal

; FAX
[2004]
type=friend
username=2004
secret=2004
host=dynamic
canreinvite=no
qualify=5000
context=internal

This will give you 4 user accounts on the server so people can connect to the server but they cant call eatch other or dail out.

Hence the extensions.conf its the dial plan.

extensions.conf
Code:
[default]
include => internal

[internal]
; A phone number for incomming calls. That calls all users except the fax.
exten => 2000,1,Dial(SIP/2001&SIP/2002&SIP/2003)
exten => 2000,2,Hangup()

; Phone 1
exten => 2001,1,Dial(SIP/2001)
exten => 2001,2,Hangup()

; Phone 2
exten => 2002,1,Dial(SIP/2002)
exten => 2002,2,Hangup()

; Phone 3
exten => 2003,1,Dial(SIP/2003)
exten => 2003,2,Hangup()

; FAX
exten => 2004,1,Dial(SIP/2004)
exten => 2004,2,Hangup()

; For all other calls send em thrug the interconnect.
exten => _X.,1,Dial(SIP/${EXTEN}@sipprovider,600,r)
exten => _X.,2,Hangup()

; To recive calls we need the incomming group.
[incomming]
exten => PHONENUMBER,1,Answer()
exten => PHONENUMBER,2,Dial(LOCAL/2000,600,r)
exten => PHONENUMBER,3,Hangup()

In this dial plan all the users god a phone number 2001, 2002, 2003 and the fax got 2004.

And the phone number 2000 simply calls 2001, 2002 and 2003 until some one picks up. The PHONENUMBER from the sip provider will then call the local number 2000 when a calls comes in.

The fax in this example is useless unless the local users wanna send a fax to it.

I can recommend the linksys SIP adapters they work great with asterisk, but i guess most adapters work. Zoiper is a great choice for softphones on windows clients, on linux i prefer Ekiga.net witch should run on freebsd as well. Once you have set this up its quite straight forward.

I use it as a local PBX at home so all computers have either Ekiga or Zoiper installed. Both me and my girlfriend has a Nokia N95 witch has a SIP client build in and works great. And then ofcause the wireless home phone just uses a linksys SIP adapter. I have been running this setup on a old box for around 7 months now and so far its working great.

Its not the same setup as posted here i have implemented queus and stuff like that to make it more realistic compared to a business setup.

/lbl
 
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