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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.57s
  • Memory usage: 2.64MB
  • Database queries: 39
  • GZIP: Disabled
  • Server load: 2.91

Asterisk

Page Contents

Image

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

Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows 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.

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 1.2.23 and 1.4.9.

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



Reference

Starting Out


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.
  • Asterisk Visual Dialplan - innovative development platform for Asterisk dialplan development. Simply drag, drop and connect dialplan blocks to make Asterisk dialplan.
  • 1videoConference - 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.
  • JAGIServer: Open Source Java Application Server using the FastAGI protocol.
  • 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
  • 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
Created by jht2, Last modification by james.zhu on Wed 01 of Aug, 2007 [04:16 UTC]

Comments

Comments Filter

by Pavel Siderov on Monday 16 of July, 2007 [08:38:37 UTC]
TO: kanderson

Hi,

Try setting different listening port to every sip phone behind nat (default is 5060 - set different port to every phone).

Regards,
Pavel Siderov

Caller ID

by Craig on Thursday 12 of July, 2007 [15:56:34 UTC]
We are having what seems to be a common problem. Caller id from our POTS lines only comes through about 80% of the time. I think the solution is to delay * from answering the line by a second or two. No body seems to know how to do this.

Any help would be appreciated.

Vonage Home account and Trixbox 2.2

by Darrin on Tuesday 10 of July, 2007 [19:43:33 UTC]
Is there a way to use a regular Vonage in Asterisk/Trixbox 2.2?

Vonage Home account and Trixbox 2.2

by Darrin on Tuesday 10 of July, 2007 [19:43:05 UTC]
Is there a way to use a regular Vonage in Asterisk/Trixbox 2.2?

Strange Sip behavior, please help!

by K Anderson on Monday 09 of July, 2007 [18:23:03 UTC]
The meat of the issue is SIP phones at remote clients stop ringing, can not call eachother, and sometimes ring/answer for the wrong extension. They can always call out and registration never seems to be lost. It almost always occurs after very heavy call volumes but sometimes before any calls are placed. Rebooting the phones always corrects the problem, as does restarting asterisk or standard re-registration (after timeout).

Details
- Phones: Polycom (430 or 601)
- rtpkeepalive=10 (seems to make no difference if 0, 10, or 100)
- rtptimeout=120
- Codec: g729
- Registration: every 3600s on username and secret (works fine)
- insecure=no
- No firewalls
- Different internet connections
- Different Routers (but all SOHO)
- Server: Asterisk 1.2.17 with public IP (no nat) in colo (data center)

Some things I have tried:
- Took five phones home and tried recreating issue through Linksys, D-Link, Belkin, and Trendnet. Could not get the D-link to recreate issue so replaced a few clients routers with a d-link, did not correct problem.
- Tried different settings for rtpkeepalive, did not fix problem
- Called Digium (will no longer support open source version)
- Tried limiting to two phones at remote location, did not fix problem

At first it seemed like a network problem (NAT, slow router) but several clients now have this issue and all are connected to this gateway. Clients not connected to this gateway have never reported this issue.

Food for thought
- I suspect that reducing the registration time could correct the issue but this gateway will carry so many phones the load may become excessive
- during compiling zaptel I got the following notice but have found no reference to it anywhere: /usr/src/zaptel-1.2.16/xpp/xpp_zap.c:411:2: warning: #warning "HZ != 1000. PCM would be good only with Astribank sync"

Any help would be greatly appreciated, I am running out of ideas short of higher end networking equipment (not an option). We are willing to hire a freelancer if you wish to offer your services.


Re: time counter?
Would a cron job work?

Refernces

by Doug Eamer on Tuesday 19 of June, 2007 [14:55:52 UTC]
I am currently installing ABE in a local lawfirm. There is nobody in the area currently using Asterisk and they are looking for a reference to speak to so they have the peace of mind the technology works and is reliable. If anyone is willing to share a reference or actually be one for me it would be appreciated.
Thanks
Doug

sip 2.0 603 declined (no dialog)

by Mike on Sunday 17 of June, 2007 [03:53:58 UTC]
I can't make an outbound call on my new trunk. Inbound works great but I get the message above when I look at the debug

sip 2.0 603 declined (no dialog)

by Mike on Sunday 17 of June, 2007 [03:53:43 UTC]
I can't make an outbound call on my new trunk. Inbound works great but I get the message above when I look at the debug

sip 2.0 603 declined (no dialog)

by Mike on Sunday 17 of June, 2007 [03:52:35 UTC]
I can't make an outbound call on my new trunk. Inbound works great but I get the message above when I look at the debug

time counter?

by Carlos on Thursday 31 of May, 2007 [20:20:40 UTC]
hi

i need a time counter in asterisk, for example:

i run a script with the System command and i need to do again after 20 minutes

how can i set the timer or counter to execute something after 20 minutes.

if some have an idea please tell me a nd i´ll prove it.

Regards Carlos

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