I'm going to give setting up an Asterisk server a bash.

Anybody manage to do it ??
Anybody got config settings ??

try this..

[general]
register => 076XXXXXX:PASSWD@freespeech.ie/076XXXXXX

[freespeech]
type=peer
insecure=very ; otherwise authentication errors
nat=yes
username=076XXXXXXX
fromuser=076XXXXXXX
authuser=076XXXXXXX
fromdomain=freespeech.ie
secret=PASSWD
host=freespeech.ie
context=(your inbound context)
dtmfmode=inband
canreinvite=no

I tried that - works perfectly

Brilliant.
What would you do if you wanted more than one line?

you would do soemthing like this

[freespeech1]
type=peer
insecure=very ; otherwise authentication errors
nat=yes
username=076XXXXXXX
fromuser=076XXXXXXX
authuser=076XXXXXXX
fromdomain=freespeech.ie
secret=PASSWD
host=freespeech.ie
context=(your inbound context)
dtmfmode=inband
canreinvite=no

[freespeech2]
type=peer
insecure=very ; otherwise authentication errors
nat=yes
username=076YYYYYYY
fromuser=076YYYYYYY
authuser=076YYYYYYY
fromdomain=freespeech.ie
secret=PASSWD
host=freespeech.ie
context=(your inbound context)
dtmfmode=inband
canreinvite=no



Edited 1 time(s). Last edit at 05/19/2006 10:21AM by moderator.

A note on this

Unlike traditional phone lines your freespeech account can handle multiple calls, so you would only add an aditional line if you needed to be reached on more than one number OR you wanted seperate accounting of call charges.

:-) ok I get it, get it, get it... ta.

I downloaded a copy of Asterisk@home a couple of months ago. From what I can see I just burn it to CD and bung it in any old workstation/server that I have lying around and let it run from CD. Then I'll be able to manage it from anywhere on the web

It looks like a good introduction to VoIP...

... I was going to do it this weekend, but now I'm away working away at the weekend again... but I'll try and do it soon.

Any other hints and or tips for the set-up...?

regards
David

yeah...use this for G729 support - [www.readytechnology.co.uk]

be aware of patent issues

you can download binary and pop into /usr/lib/asterisk/modules/ then configure your channels for G729 in sip.conf

i solved this



Edited 3 time(s). Last edit at 01/11/2007 12:51PM by belgarath.

1. Check if your Asterisk box registered by logging into freespeech account.
2. Peform a trace on the Asterisk command line and see what is happening

>SIP DEBUG

( oops... probably not a good idea to post your passsword to the forum )

Note: you have a spelling mistake here
>>register => 076xxxxxxx:xxxxxxx@freespeach.ie/076xxxxxxx

! as peachy as the service is

Hi,
I am using Trixbox version 2.
I added the above settings via the 'Trunks' Setup menu.
'Asterisk Info' page shows that the sip peer is registered.
Should i be able to dial out using x-lite if i topup my account or is there anything else that i need to setup.

***************
Sip Peers

Host Username Refresh State
freespeech.ie:5060 076XXXXXXX 105 Registered
Verbosity is at least 1
***************

Thanks guys



Edited 1 time(s). Last edit at 02/20/2007 04:46PM by pratheesh.

Hi Everyone

just wondering if some one could give me a hand with setting up Asterisk with Freespeech. i have got it dialing out through freespeech but i am unable to receive incoming calls from it, every time i ring from my mobile i get freespeech's voice mail. i have added a dialing rule and also an incoming dialing rule from freespeech.ie but still nothing.

i am only using the free account for now till i am satisfied that Asterisk will work correctly. I followed the instructions from this forum and managed to get it to dial out but as i said nothing in

Thanks in advance for the help on this

Al.

Ps on the freespeech.ie website it does say that i am online

It's likely you do not have your inbound context/dialplan correctly defined.
in sip.conf you have define the context where the calls comes in

context=(your inbound context)

You need to make sure you have a dialplan set up correctly in this context for receiving the call.


hint: Enter the asterisk command line (asterisk -rvvvvv
) and you should be able to debug the problem when making a call. If you want to see the call coming in then run >sip debug to trace the messages.