Configurations Guide

Hi all, I’m from the UK, new to Asterisk / FreePBX and I’m lost in the configuration!

I have 3 analogue lines at my work premises, one of which has two lines. Currently, for testing purposes I only intend to use the single line number.

I’ve downloaded and installed AsteriskNOW 3.0.1. I have an OpenVOX A400P with 4 FXO cards on it and it is showing green lights on the back panel. I followed the guide over on the AsteriskNOW Wiki page so I did the hello world test and everything seemed fine. I set up two extensions, (at the time I was using Zoiper on two Android phones) and tested them.

Now I am using Snom 370 phones which I have pointed to the server, registered and have working internal calls.

Now the problems began. Reading through the Wiki, there was no apparent next move after the hello world test. I have managed to get inboud calls working, (albeit a little delayed in starting to ring and it rings on for a while after the caller hangs up. I cannot get my head around the outbound calls config. I’ve setup everything I can think of, plus I compiled DAHDI and the tools following another web page.

My main question really: Is there any definitive guide on getting all this working? I look at a lot of the google results I’ve found and they seem to be for Asterisk not for the FreePBX / Asterisk system I have here. I don’t mind using seperate guides for each part, as I appreciate people use different cards etc.

Any help / guidance / direction would be very welcome!

Many Thanks


To setup Openvox card and Freepbx I combined this two guides (probably you already know for them):

But I cant get work inboudn and outbound calls trough POTS line. If you have idea what did I miss please let me know.

Here is my post:

Hi Greenfrog,

I’ve managed to get some incoming calls working, however I get a delay between the phone ringing to the caller and the the deskphone ringing, sometimes up to 6 or 8 rings by which time we are getting close to the voicemail cut off provided by the supplier. (I’ll cancel this once I’m satisfied with the settings. I’m working with the CLI alongside me to figure out the outgoing calls bit.

I’ve acutally already read your thread as I’ve been fishing though page after page of help requests, and I currently have many tabs open on my browser so maybe I can write a guide at some point once I have it all working.

Are you in the UK? I may be able to send you some entries for the config files to get you running, can’t promise it’ll work though as I am very new to this myself. I also need to weed out the errors in the log files to see what else needs altering to remove or fix the error messages!

Thanks for commenting! I was beginning to lose hope of a response!


We have guides on the wiki for configuring via the DAHDI Module

Hi reconwireless,

I’ve not been able to watch the videos at the moment due to a half built desktop and no speakers / audio drivers. The joys of running my own IT department whilst also being out on site most days working!

The delay before the SIP phone rings seems to be random, and sometimes I don’t get CallerID sometimes I do. I’ve managed to get the phone to stop ringing as soon as the caller hangs up though.

Hopefully I’ll be able to watch the video at some point this week, lots going on!

Cheers for the reply!

Thank you for answers Fastbluelion. I am from Europe your config files might work for me. Could you sent it? As I am also new in this field I do not know how to help you.

Thank you for link Reconwireless, I already tried to setup with this guide, but I think something else is missing.

I noted that you have used Hello World Test wiki page and get 2 soft phones working. Please share that URL as I intend to do exactly the same.

Hi miteshpc,

That was on the Asterisk Wiki. The link to the Getting started section is as follows:

The specific section regarding the Hello World phone call is:

Hope this helps!


Just an update, in the section about configuring the softphone, point 6 says:

“6. Enter your SIP peer’s password in the Password field”

This confused me and had me googling SIP peer password. It actually refers to:


in the example config file shown above the configuration instructions for the SIP softphone.

Make sure you backup the relevant files and restore them afterwards as you can set the configuration from the web interface.

Hope this makes sense?


Thank you for your reply.

I have followed the wiki guidelines and also identified the way forward. Please see my first experience to get it working on the URL below if you are interested:

Apologies for the time taken to get back to this thread, I’ve had a lot of work on so snatching the odd moment to work on my install has been my only option.

I do however now have incoming and outgoing calls working on a single POTS line. Turns out it was down to dialing rules. I hadn’t set them up correctly. Reading and rereading the Wiki and other sources eventually highlighted the issue. The calls still have a small echo on the office side which is annoying however I’d say this was a teething problem rather than a major issue.

Now all I need to get working is the other two lines and remove the echo. In addition I would like to get a directory working so we all have access to the same phone numbers for outbound and caller id purposes. I’ve not found anything solid on this yet though.

How are you getting along miteshpc? And what about you greenfrog?


hello guys,
I am so new to AsteriskNOW .I’ve managed to install AsteriskNOW 3.0.1 and configured too softphones to use it everything is working fine internally but my boss is more concern about inbound and outbounds calls. To be honest I don’t have an idea how to get this going. I have been doing a lot of reading and research but I couldn’t come up with any definitive guide. I will appreciate it if anyone can help me with a documentation or even a book to get this task kicking. my afraid the death line is closing on me.I will appreciate any help. Thanks


You didn’t tell us what kind of connection you have to the phone network so hard to point you in right direction.

What did you learn about trunks and routes in all of your reading? What specific part it confusing you?

Have you spent time in our wiki? It contains all of the documentation.

Your peer is incorrect. You can’t have two host entries in one peer.

The directive “peer ip address” means nothing to Asterisk, just use

Lastly the registration string is all the matters for registration and yours is correct.

No idea. I did not know Skype even supported direct SIP. There is a chan_skype available from Digium I thought you had to use.

I take the simplest path, there are many providers that directly support FreePBX, why go through all this effort to use Skype? The Sipstation product on this web site has a module for FreePBX you can order your service and it will setup the trunk.

Chan_skype is dead

From google modified to be freepbx appropriate


You MUST have a business account
Remove g.729 if you dont have a license for it.

Actually two experts told you now that it won’t work.

I don’t think you will have any success.

Please not this article:

I am sure that it could be hacked, but that is not a business solution.

I would tell your supervisor to do their research prior to purchasing and not make you responsible for their incorrect decision.

Don’t remove the whole line, replace g729 with the CODEC you want to use.

g.729 is a licensed compressed CODEC