Having just set up an asterisk box to connect to Faktortel I thought I'd post the settings I used as the ones on the
Faktortel support website do not work quite right:
For a default asterisk install on Ubuntu do the following:
$ sudo apt-get install asterisk
You then need to edit the following files three files:
sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
; sets the codecs you can use. First disallow all defaults, then set our own ind ividually.
disallow = all
allow=gsm
; Setup a user with the phone number 1000 on our network.
[1000]
tos=reliability
type=friend
username=1000
host=dynamic
context=default
reinvite=no
canreinvite=no
secret=abc123
nat=yes
qualify=yes
iax.conf
; Setup the general options, including our default port for use with IAX
; the amount of bandwidth we have on our connection to the server (best left as low)
; jitterbuffer, this isn't enabled at the moment in the latest CVS but we don't want it
; to suddenly come on when it is enabled.
[general]
bandwidth=low
jitterbuffer=no
tos=lowdelay
; This line registers your server with the FaktorTel main server, what this means is that
; it tells us where you are on the internet so you can receive incoming calls from our network
; don't forget to replace
"YOUR-USERNAME" and
"YOUR-PASSWORD" with the ones supplied to you.
register => YOUR-USERNAME:
YOUR-PASSWORD@iax.faktortel.com.au ; The following section sets up an IAX user called
"faktortel", this is the user which will place
; the call from your server to our network. First we tell it not to use any other codec by setting
; our
"Disallow" and
"allow" settings. Then we set the type to
"friend", then set our password and
; username for the faktortel network (don't forget to change these to the ones supplied to you).
; using the
"host" command we tell the user which server to connect to, then using
"auth" we tell
; it the type of password to look for, and finally using
"qualify" we tell it to keep in touch
; with the server ever 3 seconds. (This is fast enough).
[faktortel]
disallow=all
allow=gsm
type=friend
username=YOUR-USERNAME
secret=YOUR-PASSWORD
host=iax.faktortel.com.au
auth=md5
context=incoming
qualify=yes
extensions.conf
; First we setup a context called "Default" this is our "Encapsulation" context. The only point of this
; context is to point to all our other groups in order. We only have one other outgoing context at the
; moment so we are just including one "faktortel-outgoing" however this will allow you to make a much
; more sturdy network later if you use it as a full PBX. This is also the context we use to place calls out of.
[default]
include => faktortel-outgoing
; FaktorTel-Outgoing is another context, this tells anyone who is trying to make a call out from the PBX
; which starts with a "0" to put that call out to the PSTN via FaktorTel. The format here is best
; described at "www.voip-info.org" under "configuration files" "extensions.conf".
[faktortel-outgoing]
exten => _0.,1,DIAL(IAX2/YOUR-USERNAME@faktortel/${EXTEN:1})
exten => _0.,2,Congestion
; Adding an extension for the number 1234 which will give you the Time.
exten => 1234,1,SayUnixTime
exten => 1234,2,Hangup
; This is our "incoming" context, when a call comes in from the faktortel network it is sent here
; because we set "context=incoming" in our IAX.CONF file under [faktortel]. This then tells it
; that ANY call that comes in should ring the SIP based phone at extension "1000" which we setup
; earlier.
[incoming]
; Add extension for you DID number
exten => YOUR_DID_NUMBER,1,Answer
exten => YOUR_DID_NUMBER,n,SayUnixTime
exten => YOUR_DID_NUMBER,n,Hangup
Hope this helps you get going with Asterisk. Cheers, Mark
It works great, but when I tried it in my new install the sound was all choppy, i.e. I would get every half of each second of the audio.