Asterisk source code distributions (release, svn trunk) compiles and runs on OS/X.
If you are new to Asterisk and VoIP and you don’t have the time or the nerve to concern yourself with building software from source code and editing configuration files, then a Mac running OSX is the best choice for you to get started. Installing Asterisk and setting up a basic home IP-PBX system on your Mac is as easy as anything else you do on your Mac, just as you would expect it to be.
Installing Asterisk and Configuration Tools
Asterisk server software
First you will need to install the Asterisk server software on your Mac. To do so, simply download and double-click the Asterisk Installation Package for MacOSX below. The installation is straightforward and only takes about 10 seconds on a G4/867MHz and about one minute on the original iMac G3/233MHz running MacOSX 10.3.5.
Asterisk server control and configuration tools
Next, you should download the tools to control and configure the Asterisk server from the MacOSX GUI.
- Asterisk AppleScripts to start and stop your Asterisk server by double clicking on the applets
- Asterisk Assistants for MacOSX to configure your Asterisk server using the MacOSX GUI interface
Starting and Stopping Asterisk
To start and stop your Asterisk server, simply double click on the appropriate script applet from the Asterisk AppleScripts collection.
- double click on the Start Asterisk script applet to start your Asterisk server
- double click on the Stop Asterisk script applet to stop your Asterisk server
Configuring Asterisk
Setting up phones and associated extensions
First, you need to configure your Asterisk server to recognise your telephone sets and assign extension numbers to each of them. You can do this from the MacOSX GUI, using the New Extension Assistant. Let’s assume you want to connect three VOIP Phones to your Asterisk server and assign them the extension numbers 2001, 2002 and 2003 as shown in the diagram below:
To configure this, open the New Extension Assistant and enter the details for each phone and user in the appropriate dialogs as shown in the screenshots below:
First, enter the extension number and password you want to assign to the phone …
Note that the assistant will use the extension number also as a login name for the phone and that choosing a password is up to you. The user’s name is used only for setting the caller ID display. It is not used for authentication.
Next, set the ring timer and select the VoIP protocol and codecs …
Note that the settings for protocol and codecs depend on your VoIP phone. Check the phone’s manual to find out which protocols and codecs are supported by the phone. Do not select a protocol or codec that the phone doesn’t support.
Repeat this for each phone. You can use the “Start Over” button to create multiple entries without quitting the assistant. Once you have configured Asterisk to recognise the phones and assign extensions to them, you will need to enter the settings into the phones.
At the very least you will need to enter your Asterisk server’s IP address, the login (which in our case is identical to the extension number) and the password, but depending on the phone, there may be more settings. Let’s assume you want to configure a Grandstream BudgeTone SIP phone, then you would need to enter the following settings for Fred Astairisk on extension 2001:
- SIP Server: 192.168.0.2
- Outbound Proxy: (leave this blank)
- SIP User ID: 2001
- Authenticate ID: (leave this blank)
- Authenticate Password: (enter the same password you entered in the assistant)
- Name: Fred Astairisk
- Preferred Vocoder: (choose PCMU and PCMA and disable all others)
- Register Expiration: (set this to a low value, ie 1, 2 or 3 minutes)
- NAT Traversal: No
Once you have configured all the phones, you can test your setup by making test calls between the phones. Simply dial the extension number of another phone to make a test call. On some phones you need to press # (the hash key or pound sign) in order to tell the phone that you have finished dialling, not unlike the “Send” button on a mobile phone.
You can also dial the built-in demo of your Asterisk server:
- Dial 500 for a recorded greeting and to get connected to Digium testing PBX to PBX calls over the Internet
- Dial 600 for an echo test where everything you say is repeated back to you by Asterisk
Setting up a phone connecting over the Internet
this section has not been completed yet. Please come back within the next day or two.
Setting up a phone connecting over the Internet from behind a NAT router
this section has not been completed yet. Please come back within the next day or two.
Connecting your Asterisk server to the FWD VoIP service
this section has not been completed yet. Please come back within the next day or two.
Connecting your Asterisk server to VoicePulse Connect, a VoIP/PSTN gateway service
Coming soon.
Connecting your Asterisk server to an analog telephone line from your local phone company
Coming soon.
Other Useful Utilities
Dialling a phone number from your MacOSX Addressbook via Asterisk
- Download Jon’s Phone Tool
Back to Asterisk MacOSX Support | Asterisk AppleScripts | Asterisk Assistants for MacOSX
April 7, 2023
Phone System 