Open source Asterisk PBX


Opensource Asterisk is complete telephony PBX that run on Linux (Recommended), BSD and MACOS Proxy all telephony features that you expect from a PBX beside a lot of other features. Asterisk maybe can use on all major operating system Asterisk™, the Open Source PBX, is taking the world by storm. Asterisk makes a complete business- or carrier-class PBX out of an ordinary Linux computer, integrating with the telephony network as well as the new world of voice over IP and Internet..

Asterisk does VoIP in any protocols and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware Asterisk needs no additional hardware for VoIP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk’s sponsors, Digium. Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. Asterisk supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next generation voice/data integrated networks and existing infrastructure. Asterisk provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.

Be Sociable, Share!

7 Responses to Open source Asterisk PBX

  1. Daniel Reardon August 22, 2007 at 10:34 pm #

    I hope you dont mind me emailing you with a question regarding the configuration of my Asterisk installation.

    I have been all over the internet trying to find an answer but so far haven’t been able to find anything.
    I have Asterisk installed and I can successfully call each extension.

    We have an external Sip gateway which has 4 fxo ports.

    I can not find any information regarding how to get Asterisk to connect to the external sip gateway via an ethernet cable

    Do I need to install a driver?
    I hope you can help

    Many thanks
    Danny Reardon

  2. Figo July 8, 2008 at 1:31 pm #

    I need help,
    I’m dialing extension and call drops. I see this on console
    == Auto fallthrough, channel ‘SIP/XXXXXXX’ status is ‘UNKNOWN’

    What does this mean. Please help

  3. Sesy September 14, 2008 at 3:48 pm #

    I’m facing following error while compiling Asterisk please help:

    checking for ZT_EVENT_REMOVED in zaptel/zaptel.h… yes
    checking for ZT_TCOP_ALLOCATE in zaptel/zaptel.h… no
    configure: error: *** termcap support not found

  4. David September 14, 2008 at 6:20 pm #

    I’ve want to compile Asterisk-Addon package but it gives me following error, please if anyone help to findout the reason?

    *** Install ncurses to use the menu interface! ***

    I’m installing Asterisk Addons 1.4.9.

  5. Bishant April 9, 2009 at 11:28 am #

    Anybody can tell guide how I can save Asterisk CLI console messages in my Linux box on some path?

  6. sameer June 3, 2009 at 5:35 pm #

    I’m using asterisk ver 1.4.25, the problem is when i try to start Asterisk service I see the following error:

    [root@ivrdev asterisk-1.4.25]# service asterisk start
    Starting asterisk: /usr/sbin/safe_asterisk: line 75: ulimit: open files: cannot modify limit: Operation not permitted
    [ OK ]

    However asterisk gets started. Anybody can tell me why i see this error?

  7. juan gomez December 4, 2009 at 2:04 pm #

    I’m using Asterisk 1.6.x and during analyzing Asterisk logs i often see the following message. Please indicate what is wrong.

    [Nov 30 05:12:44] WARNING[29617] file.c: Failed to write frame

    Also please guide how can we rotate the asterisk logs.


Leave a Reply