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Asterisk

Created by: jht2,Last modification on Tue 25 of Nov, 2008 [23:35 UTC] by liyuqian

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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 1.2.30, 1.4.22 and 1.6.0.1.


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


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


Comments

Comments Filter
222

333Dedicated Conference Bridge

by flunnon, Monday 10 of November, 2008 [16:11:04 UTC]
I want to create a dedicated Asterisk conference bridge to host more than 100 simultaneous calls in multiple rooms at the same time. It will be used solely for managing conference calls and rooms. There is little information on this area and looking for any help. Is it better to use the AppConference plugin or MeetMe?
Is there any advice on load balancing for managing larger capacity?
Thanks in advance

Frank
222

333REDIAL / RECALL on Asterisk

by zipge, Monday 09 of June, 2008 [11:37:36 UTC]
Hi all!
I've a problem with my Asterisk. In my configuration, to make an external call, I've to put 0 before the number that I want to call. It's work! But if I recieved an answered/unaswered call and I want to recall it, Asterisk don't recognize the number without initially 0 to route it outside.
Can someone help me?

I'm sorry for my english, I hope u understand it!
Thank you all

Zipge

222

333chrt for realtime asterisk process

by lingolep, Wednesday 14 of May, 2008 [10:42:27 UTC]
With GNU/Linux I've found that running

/usr/bin/chrt --rr -p 99 `</var/run/asterisk.pid`

after starting the asterisk process seems to help with its performance. It attempts to set the process to a realtime state.

Here are a couple ways to confirm that it did its job:

1. You can confirm by running /usr/bin/chrt -p `</var/run/asterisk.pid`

2. Run a process monitor like top and you should see the asterisk entry marked as RT, instead of a normal priority of 20

I know that asterisk has a whole bunch of threads for its normal operation. I don't know whether overall there is a positive benefit or not, but perhaps you can try it and report back.
222

333Re: no voice sip phone to my asterisk for moments

by osiris2985, 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
222

333help,mgcp softphone+asterisk

by pylonion2008, 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
222

333help,mgcp softphone+asterisk

by pylonion2008, 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
222

333help,mgcp softphone+asterisk

by pylonion2008, 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
222

333Asterisk to Panasonic PBX

by Aubichon, 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
222

333voip or sip?

by habibhayek, 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
222

333Asterisk/Voicemail

by Luisha, 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