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  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
  • Christopher Faust, Wed 07 of May, 2008 [15:28 UTC]: When I try to startx I ge input not supported. Though before installing asterisk I had no video issue to start the GUI
  • Christopher Faust, Wed 07 of May, 2008 [15:26 UTC]: Hi Nick, I got centos 5.1 and asterisk up But now I cannot start startx I have set the depth from 24 to 16 for the video i810 driver for the i845 on my netvista machine but I cannot start GNOME. Please advise
  • Nick Barnes, Wed 07 of May, 2008 [10:01 UTC]: Howard - You'll need to provide a lot more information if you really want help.
  • Nick Barnes, Wed 07 of May, 2008 [10:00 UTC]: Christopher - Search the Wiki and you'll find a page I wrote detailing exactly what you have to do for Asterisk 1.4 + CentOS 5.1.
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Asterisk

Page Contents

Image
Original Website - http://www.asterisk.org/

Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows (emulated) and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four 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, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. Check the Features section for a more complete list.

Asterisk needs no additional hardware for Voice-over-IP, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism (for certain applications such as conferencing). A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers)

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 sponsor, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks. In addition, single to quad port analog FXO and FXS cards are available and are popular for small installations. Other vendors' cards can be used for BRI (ISDN2) or quad- and octo- port BRI based upon CAPI compatible cards or HFC chipset cards.

For interconnection with the cellular network (GSM or CDMA), Asterisk can use the Celliax channel driver or chan_mobile that is in the trunk now and there is also a unofficial backported version.

Lastly, standalone devices are available to do a wide range of tasks including providing fxo and fxs ports that simply plug into the LAN and register to Asterisk as an available device.

The current release versions of Asterisk are 1.2.27, 1.4.19.1 and 1.6.0-beta8.

This Wiki covers both the stable and the development branch of Asterisk. When adding new commands, applications and options, please also add a note on *when* this was added so that users may compare with their version date.

News


Old asterisk news


Reference

Starting Out

Books



Introduction


Hardware


Administration and system layout



Configuration


Management


Troubleshooting


General Reference






Country-Specific Information





Commercial support


SIP Service Providers


User Groups



Weekly SIP Asterisk Users Conference

  • x2z.eu THis conference is open to users at all levels of asterisk expertise


Howtos and Tutorials



Third party software

Asterisk works in conjunction with other programs in some installations.
  • Evolution Call Center An asterisk-compatible free, comprehensive call center software solution.
  • Activa for Asterisk Activa Asterisk TSP enables TAPI and TAPI-compatible applications to use the Asterisk IP-PBX: MS Outlook, ACT!, TapiCall, MS Dialer...
  • DM Link release chan_astsky, which is hot using by small and middle class company.
  • Skip2PBX Now with recently added SIP support can interface with Asterisk for calling with Skype
  • ztloop - The way to test zaptel based applications without any telecom card.
  • Asterisk Visual Dialplan - innovative development platform for Asterisk dialplan development. Simply drag, drop and connect dialplan blocks to make Asterisk dialplan.
  • VMukti - Open source web2.0 video conferencing software for Asterisk.
  • Attractel Predilux - a fully optimized solution for outbound calling, with real predictive dialing; multiprotocol softphone, developed to run on Windows, Linux, Mac OSX
  • SineDialer - Professional Predictive Dialing and Message Broadcasting - tested from 1 to 3000 lines
  • Tello Zero Cost Routing - Make free phone calls between registered Asterisk PBXs. Register today for this free service
  • ADM - Asterisk Desktop Manager Integrate your desktop with Asterisk and hardware IP phone. Bluetooth presence detection redirects calls to your mobile when you walk out of the office. One click dialling (paste numbers from clipboard)
  • AsterFax - Email to fax gateway for Asterisk
  • Jabber/XMPP Integration
    • Asterisk-IM The Open Source project of Jive messenger has recently released Asterisk-IM, a plugin that integrates Asterisk features with The Open Source XMPP instant messenger, including integrated presence and call notification.
    • AstJab Communication bridge for connecting an asterisk server to a jabber server, in order to obtain presence information for the Asterisk dialplan.
  • FastSMS connects Asterisk for worldwide delivery of SMS text messaging.
  • Festival: Open Source Speech Synthesis software used by the Festival application
  • OrderlyQ: Extension to Asterisk Queues that lets callers hang up, then call back later without losing their place.
  • Asterisk Java API: Open Source Java Application Server using the FastAGI protocol.
  • JAGIServer: Open Source Java Application Server using the FastAGI protocol (not active anymore).
  • Contaque V5: New version AVIS e Solutions Launched VoIP based Predictive Dialer system Contaque
  • Asterisk Dial Plan Compiler: A simpler form of programming dialplans, if you use lots of Goto's and GotoIf's.
  • Asterisk Dialplanner: A Java-based point-and-click web tool to help you create your dialplan.
  • Sphinx: Speech recognition
  • Voiceglue: VXML interpreter for Asterisk.
  • mpg123: MP3 Player for Linux and *BSD, used by the MusicOnHold application.
  • The VoiceMail application uses /usr/sbin/sendmail to mail voicemail messages to users. This can be any sendmail-compatible MTA. In practice you can use Sendmail, Postfix, Exim or any other MTA. It is recommended to use the default one that comes with your distribution.
  • If you want a flexible and reliant database connection, use the ODBC connections that is built upon the UnixODBC libraries
  • The Receptionist Console Desktop application for receptionist to
  • QueueMetrics turns your Asterisk BPX into a full-fledged ACD system with extensive reporting.
  • Asterisk Codec Negotiation Patch Patch that improved performance and scalability of the Asterisk by preventing costly audio transcoding when it's not necessary.
  • AstQueueIcon: Alternative free queue solution utilizing the call parking feature instead of the queue application. As the queue app stays out of the media path, features like call forwarding will not be denied for the agents.
  • VXML 2.1 for Asterisk IEC - SIP/2.0+VXML 2.1 Inexpensive VXML IVR Platform, MRCP ASR/TTS Client/Server, Linux/Windows, Call Analysis, Outbound, more ... Free Trial Download.
  • SafiWorkshop - Visual Call Flow Editor and Server for Asterisk: Enterprise level Asterisk call flow editor and server. Trial Download
Created by jht2, Last modification by lolo on Fri 09 of May, 2008 [08:36 UTC]

Comments

Comments Filter

Re: no voice sip phone to my asterisk for moments

by Gerardo on Thursday 03 of April, 2008 [05:20:53 UTC]
Hi there... so...you establish a call and hear the other person, but you are not heard??... you might want to check the codec list of the extension, i had the same problem...set the allow=all (sip.conf) between the extensions and so in the phone (softphone)(s) you are using....this will let asterisk decide wich codec is the best for the conversation...it most likely use a/u law or gsm.

try this and let us know who it comes out

help,mgcp softphone+asterisk

by jordan on Monday 31 of March, 2008 [07:12:22 UTC]
hi,I am jordan,who can tell me ,how to configure asterisk connect to softphone with mgcp.thanks!
jordan

help,mgcp softphone+asterisk

by jordan on Monday 31 of March, 2008 [02:51:20 UTC]
hi,I am jordan,who can tell me ,how to configure asterisk connect to softphone with mgcp.thanks!
jordan

help,mgcp softphone+asterisk

by jordan on Monday 31 of March, 2008 [02:24:39 UTC]
hi,I am jordan,who can tell me ,how to configure asterisk connect to softphone with mgcp.thanks!
jordan

Asterisk to Panasonic PBX

by Rod Aubichon on Friday 01 of February, 2008 [23:27:09 UTC]
Situation: WIFI SIP phones need to work behind a Panasonic TDA600 PBX. The SIP phones will mostly be used for calling internally to each other as well as to extensions on the Panasonic PBX. Rarely will they call to PSTN.

Flow:
PSTN PRI to Panasonic PBX PRI card in, another Panasonic PRI card into Sangoma A101 PRI card into Asterisk server


Minimal experience with Linux command line so a nice GUI interface like Trixbox or Asterisk Now would be preferred.

As you can tell by my description, I am using asterisk for very little PBX functionality. Besides the SIP phones calling each other, most of the call routing will be done in the Panasonic PBX.

The only reason I am doing this is because the Panasonic system can't support SIP registrations.

Give me some ideas!

rod at firetec dot ca

voip or sip?

by habib hayek on Thursday 10 of January, 2008 [13:47:35 UTC]
hi my name is habib and looking to this informations plz:
first i am new user of sip and voip i am in canada and i want to call our office out of canada in dubai for example if i have an internet connection or a dsl connection in my office there i ahve a ata device here in canada and dubai and i have sip software on both computer what do i need to call dubai over voip or sip and i want to connect my land line in dubai to my device or modem if my computer so i can make calls to dubai via voip or sip but i will pay local call not international because i am using my dubai land line is that posssible or i am dreaming i hope i can find something thx for u all plz send me email to hayik@hotmail.com

Asterisk/Voicemail

by Luisa on Tuesday 13 of November, 2007 [20:39:59 UTC]
Hello!

I am doing this thing.. i have to take one attachment .WAV from an email, and take it an put it in the folder that correspond to it extention... (/var/spool/asterisk/voicemail/default/105/INBOX for example..) i could get the .wav with this thing.. ripMIME... an whith sox i convert it to .gsm, but my problem is that i have to create the .txt that the asterisk voicemail needs to interpret the message, i mean there is all the information about the mess... i mean, from who, what time, etc etc,,,,

example: (this is a real thing.. ... i have 6 mes...each one have a .txt file....)

root@central:/var/spool/asterisk/voicemail/default/107/INBOX# ls
msg0000.gsm msg0001.gsm msg0002.gsm msg0003.gsm msg0004.gsm msg0005.gsm
msg0000.txt msg0001.txt msg0002.txt msg0003.txt msg0004.txt msg0005.txt
msg0000.wav msg0001.wav msg0002.wav msg0003.wav msg0004.wav msg0005.wav
msg0000.WAV msg0001.WAV msg0002.WAV msg0003.WAV msg0004.WAV msg0005.WAV

this is msg0004.txt
root@central:/var/spool/asterisk/voicemail/default/107/INBOX# cat msg0004.txt
;
; Message Information file
;
message
origmailbox=107
context=macro-stdexten
macrocontext=DefaultOutgoingRule
exten=s-BUSY
priority=6
callerchan=SIP/107-b6b00468
callerid="Aneury Disipulo" <107>
origdate=Tue Oct 30 06:21:10 PM AST 2007
origtime=1193782870
category=
duration=47


I need to find a way to generate this file for files.wav that i want to put into this folder....

CAN SOMEONE HELP???

thatks...

LMPJ

Newbie: Asterisk Bridge

by Rocky on Sunday 04 of November, 2007 [05:13:39 UTC]
I've just setup an asterisk server (my first), and after a few days of struggle trying to work out all the functions, going through tutorials, and trying to take shortcuts using the gui, I think I've finally got my SIP's registered. That said, I apologise if this has been covered somewhere else, but I've done searches but to be honest I'm not sure exactly what to search for.

What I'm trying to do is provide a connection from my Netcomm V100 to my VoIP provider KMoo, who appears to be using some odd VoIP equipment. The problem with directly connecting my ata with KMoo is I can make calls, but not recieve, and apparently my ata is incompatible with their stuff. So my plan here is to use an asterisk server to "bridge" the two if possible.

So far so good, the Netcomm ata registers with asterisk, and asterisk registers with KMoo (I think). Now my problem lies in setting up the extensions.conf file simply so when I dial a number it routs the call through KMoo, and when I recieve a call its routed to the phone (using the Netcomm ata).

What happens currently is I dial on the phone and it comes back with a hangup (not unexpected, as problem is in the dial function), and when I call the phone it says its not available (out of range or switched off).

My setup is as follows:
extensions.conf

kmoo-in
exten => s,1,Dial(SIP/home,25,r)

from-user
exten => s,1,Dial(SIP/kmoo,25,r)
exten => s,2,Hangup


sip.conf
kmoo
context = kmoo-out

"phone number"
context = kmoo-in

home
context = from-user



Any ideas on how I can pull this off?
I figure if I'll try pulling this off before venturing too much further as this is all I need for the moment.

no voice sip phone to my asterisk for moments

by garnier on Monday 15 of October, 2007 [10:25:09 UTC]
short of time when a phone arrived to my sda/sip phone, i receive the voice, and i answer, but some of times i don't hear the other person.

I said, other person heard me ???

Why ? what can y do ?

Thanks for your help


New Asterisk Motherboard (chipset) compatibility forum

by Justin on Thursday 04 of October, 2007 [21:17:08 UTC]
A new forum for testing zaptel cards and motherboard chipsets.

http://www.asteriskmotherboards.com

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

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