Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response and Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). As SIP protocol is much simpler and doesn’t rule out any of the features required in the PBX world, nowadays many people prefer it. Basic SIP building blocks include, for example, SIP proxy and registrar. Asterisk can’t really be described as either of these. It is closer to a media gateway with SIP proxy/registrar- type features that make it possible to build a feature-rich PBX system (or network).
Asterisk was originally written by Mark Spencer of Digium dba Linux Support Services Inc. Code has been contributed from Open Source coders around the world.
All snom phone models can be used with Asterisk.
Basic Asterisk configuration
The relevant files for SIP phones in Asterisk are sip.conf, extensions.conf and voicemail.conf.The configuration depend on the desired dial plan and usernames e.g. preference to use phone extensions as a usernames.
sip.conf describes some general SIP parameters and all the SIP devices in the Asterisk PBX system. Please check the Asterisk sample files that come with the software or see our sample file section.
Extensions.conf describes the dialplan for the Asterisk PBX system. It can be used in many ways. Please check the Asterisk sample files that come with the software or see our sample file section.
voicemail.conf describes how all the mailboxes should behave. Please check the Asterisk sample files that come with the software or see our sample file section.
snom configuration for Asterisk interoperability
In order to use snom phones with Asterisk, you will need to configure some SIP parameters.
- Call Completion : OFF
- Challenge Response on Phone: OFF
- Use user=phone: OFF
- Filter packets from Registrar: OFF
- Go to the Identity/ Line page
- Fill in the name, account and registrar. Use Asterisk’s IP address in the registrar field. Do not configure the mailbox here.
snom and Asterisk both support several codecs but unlike snom, a separate license is required for Asterisk when using g.729 codec (Contact Digium inc.)
Message Waiting Indication (MWI)
MWI also works with Asterisk. If someone has left you a voicemail, you will receive indication of this (MWI). In snom 3xx, this will be displayed in two ways: a yellow LED will blink and there will be an MWI on the display. This is cleared only when you check your voicemails (and delete them).
In order to set this scenario up, configure the following:
Multicast Streaming is possibile in Asterisk starting from version 1.8. You must use the MulticastRTP channel like these lines of extension.conf:
When you dial the 201 extension Asterisk starts sending a multicast stream to 18.104.22.168 port 5555, so you need to configure multicast_listen/mc_address on your phones. At the moment of writing (2013/03/14) there is an Asterisk open issue that breaks this feature if you're using a file as a audio source.
Add to your extensions.conf
Please note that Asterisk doesn't act as a pure proxy SIP, if you need to use the intercom function key you must preserve the Alert-Info Header with a similar dialplan:
This isn't currently supported by Asterisk.
Extension Monitoring (BLF) & Call Pick-Up
Snom phones will only support this feature on unpatched Asterisk versions. See the requirements and possible implementation in our "Call Pick-Up" article.
Here you can find a little howto explaining hot to take advantage of Asterisk "device states" feature.
To reach your snom phone, you can, for example, have the following three lines for extension 910 in extension.conf:
- Although it is also possible to use a username other than the extension number, using the extension number makes management and documentation a little easier.
- Details for 910 are in sip.conf:
- 910 is the mailbox number.
- 1234 is the password.
- Using the mail address makes it possible to receive voicemails as a mail attachment (wav).
- (ip-phone-forum.de -german): Configuration of "Extension Monitoring & Call pickup" for * 1.2 / Konfigurationsanleitung der Funktion "Nebenstellen- Überwachung und Rufübernahme" für * 1.2
- (voip-info.org): Tweaks to make the SNOMS happier with Asterisk
- More information about supported features can be found at http://www.asterisk.org.
- Hardware for Asterisk can be purchased from Digium Inc.
- Asterisk Echo Cancellation