Getting started with FreePBX – Part 2 Setting up an extension
So we’ve got our trunk setup and now we need an extension so we can make a test call via the trunk.
For testing I’m going to be using x-lite. A free sip softphone available for Windows and Linux from here.
Adding an extension in FreePBX
First we need to click on Extensions on the left hand main menu
Now we can click on “Submit” to add a “Generic Sip Device”
This page is too big to fit on a single screenshot but there are only a couple of things we need to change to get a basic extension
First enter a “User Extension” – I called mine 1000
Next enter a “Display Name” – I used the same as above but you could enter a real persons name if reuiqred
Now scroll down and enter a secret. This is going to be the password for the extension and you really want to have one, especially if you’re connected to the internet
And that’s all we need so scroll further down and click on “Submit”
Now, as always, we need to click on the orange “Apply Configuration Changes” for the extension to take effect
Configuring X-lite
Now we’re going to configure our softphone. I’m using x-lite that you can download using the link at the top of the page. I’m using the Windows version, the Linux configuration screens look a little different
Right click on the display part of the screen and you should see an option called “SIP Account Settings…”
Next click on “Add…” so we can enter our extension account settings
Now we enter our extension details as below. You will need to -
- replace 1000 with the extension number you created
- replace Password with the secret you chose above
- replace Domain with the IP address or DNS name of you Asterisk server
- replace proxy with the IP address or DNS name of you Asterisk server
And click on Ok
And that’s it! Now, if you dial *65 Asterisk should read back your extension number to you
We probably can’t the outside world just yet as we need an Outbound Route. That will be in the next article 
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Related posts:
- Getting started with FreePBX – Part 1 Setting up a trunk
- Getting started with FreePBX – Part 4 Setting up a DID number
- Getting started with FreePBX – Part 5 Setting up an IVR
Avaialble systems include FreePBX, PBX-in-a-Flash, Elastix, A2Billing and FusionPBX.
More details and prices can be found at sysadminman.net
Joseph:
I got a registration error: 404 not found with my x lite softphone..
I am sure the address i use in the x lite account is valid.
17 June 2009, 3:54 ammatt:
Hi Joseph, it does sound like you’re using either the wrong IP address or server name to connect to. Or maybe if it’s your Asterisk server the connections are being blocked on there by iptables?
17 June 2009, 5:35 amluke:
im getting a 408 timeout in x-lite. this is my first attempt at asterisk as well as voip telephony so im sure its something quite obvious but it escapes me.
23 July 2009, 7:32 pmmatt:
Hi Luke,
That just means X-lite can’t connect to Asterisk. Could iptables be running on the Asterisk server (try – “iptables –list”), is Asterisk definitely running (try – “asterisk -rv”), is the Asterisk server behind a NAT router?
23 July 2009, 7:40 pmMikeP:
Works like a champ with 3CX softphone.
21 December 2009, 3:26 ammatt:
Thanks for the info. I didn’t know that. Matt.
21 December 2009, 8:42 amPachang:
I had same problem as Luke.
7 February 2010, 3:59 amIt wasn’t to do with iptables/firewalls. The port was open but asterisk wasn’t listening on that port for some reason.
I had to add a variable: bindport = 5060 (or whichever port you are setting the extensions with) in one of the config files.
Hope that helps someone.
Basit:
Hi,
Hope you will be fine. I am running asterisk 1.4.18.1. when i connect to asterisk using Xlite and then try to listen the recorded message it says
[Mar 2 16:57:01] NOTICE[5198]: res_odbc.c:530 odbc_obj_connect: Connecting voipodbc
[Mar 2 16:57:01] NOTICE[5251]: app_voicemail.c:4838 open_mailbox: Resequencing Mailbox: /var/spool/asterisk/voicemail/brights/220/Old
[Mar 2 16:57:17] NOTICE[5198]: res_odbc.c:544 odbc_obj_connect: res_odbc: Connected to voipodbc [Mysql-asterisk]
[Mar 2 16:57:49] WARNING[5251]: file.c:644 ast_readaudio_callback: Failed to write frame
== Spawn extension (office, *, 1) exited non-zero on ‘SIP/203-09da2080′
and the message did not play but it says you have message from 203 and then take a long pause and then says press this for new message, It is not playing the message. Any idea. Is this firewall issue?
2 March 2010, 6:13 amThanks