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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • 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.
Server Stats
  • Execution time: 0.42s
  • Memory usage: 2.28MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 1.33

Asterisk at Home

Image

Up and running in one hour


Asterisk@Home has been replaced by trixbox

Please visit trixbox page for current information


This page has been viewed 385318 times since being created on Fri 30 of Sep, 2005


News







Quick Links (Project Pages at SourceForge)

  • trixbox - Official Asterisk@Home/trixbox Home Page
  • The Asterisk@Home Handbook Wiki - A MUST read! Please read the handbook before posting any questions to the forums
  • Asterisk@Home Forums (Please be sure to do a search for your problem at this forum first. Chances are, someone else has already had the exact same problem and the solution already exists in the forum. Once you decide to post, it would be better to post it at the sourceforge forum instead of the wiki forum so we can keep the knowledgebase in one place).

NOTE:

Asterisk is a trademark of Digium Inc., and is used by permission. The Asterisk@Home project is not sponsored, endorsed, or supported by Digium, and its authors and maintainers are not affiliated with Digium. Digium does not provide free Technical Support for Asterisk@Home

Intro

The Asterisk@Home project enables the home (or small office) user to quickly set up a full featured Asterisk PBX with a web based interface in about an hour on a dedicated PC. Even if you are new to Linux, Asterisk@home handles that by handling the complete Linux install for you. In order to get up and running all you need to do is download the Asterisk@Home .iso and burn it to a CD. Boot that CD and you will get a very complete Asterisk and Linux install.

Asterisk@Home provides a nicely integrated install of some of the best software from the Asterisk community, such as the Asterisk Management Portal, which provides an intuitive Web GUI for configuring asterisk, and the Flash Operators Panel, which lets you see and control your Asterisk PBX in realtime, and FAX support through span-dsp.

Asterisk@Home also provides an xPL (home automation) interface for easy interaction with other devices in the home.

The quickest and easiest way to get Asterisk up and running at home is to download the asteriskathome .ISO file and burn it to a CD, while reading the Asterisk@Home Handbook Wiki!


Installation Requirements


Hardware

  • any old pc with:
    • Minimum CPU of 500mhz PIII or equivalent.
    • 256Mb of RAM minimum.
    • At least a 4gb hard disk drive that you don't mind overwriting with the Asterisk@Home install
    • A cdrom and a network card.
    • Internet Access for downloading updates.

You can have a functional Asterisk installation with no additional telephony hardware, using a VoIP provider and some soft phones.
Optionally if you want to be able to use Analog / POTS lines or analog telephones, you will need either PCI cards or network based Analog Telephone Adapters. See Asterisk Hardware for more information.

Software


  • Asterisk@Home CD (it has everything you need or will download it)


The software that is currently installed as of Asterisk@Home version 2.8 (04/13/06) is:


  • Asterisk (1.2.7.1) - http://www.asterisk.org/ An open source software implementation of a telephone private branch exchange (PBX). A PBX connects one or more telephones on one side to one or more telephone lines on the other side. A good example of this is a small company with 100 internal telephones sharing 20 outgoing/incoming telephone lines. A PBX can be more cost effective then having 100 direct telephone lines.
  • AMP (1.10.010) - http://www.coalescentsystems.ca - Asterisk Management Panel is a web based GUI that allows you to easily manage Asterisk without having to edit sometimes complicated text configuration files. This package can really make a difference in learning and configuring asterisk easily.
  • Flash Operator Panel (023.001) - http://www.asternic.org/ - Flash Operator Panel is a switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your PBX activity in real time. You can see what all of your extensions, trunks, and conferences are doing. You can also hang up, transfer, initiate a call or create a conference call.
  • MPG123 Music On Hold (0.59r) - Asterisk@Home now uses native music on hold so the MP3 music on hold interface in AMP will not work. The old mpg123 is still running. If you change the config files to use MP3s you can upload with AMP.
  • SugarCRM (4.0.1a)with Cisco XML Services interface + Click to Dial - http://www.sugarcrm.com/crm/ - SugarCRM is designed to be a complete customer/contact manager. Using SugarCRM we can manage all types of communications (faxes, text messages, phone calls, emails, and even tasks and scheduling) within one single system. Otherwise all these systems are separate and isolated from each other. One way it is integrated with A@H is once you enter all your contacts all you need to do to dial them is use the "click to dial" feature without having to dial the numbers manually. Your phone rings and when you pick up, A@H calls the contact you've requested.
  • Festival Speech Engine version (1.96) - http://festvox.org/festival/ - Festival is a speech synthesis system. It allows you to enter text that the Asterisk@Home server "reads out loud" to anyone calling the server. Using this, you can be sure the same voice is used across the whole asterisk server.
  • Asterisk Span DSP (0.0.2pre25) (Fax Support) - Optional Software based FAX. Automatically detects and receives incoming fax (on zaptel hardware). It sends the fax as e-mail with a MIME .PDF attachment.
  • Open A2Billing () http://www.areski.net/a2billing/ - A2Billing with Asterisk is trying to meet the needs of large to medium-sized companies and start-ups engaged in the Calling Card business. A2Billing allows you to craft a calling card management system for your Asterisk Server. With A2Billing & Asterisk, prepaid/postpaid calling card services are easy to implement via a user-friendly but powerful web interface.

  • Linux CentOS (4.3) - http://www.centos.org/ - CentOS is 100% compatible rebuild of the Red Hat Enterprise Linux (RHEL), in full compliance with Red Hat's redistribution requirements. CentOS 2, 3, and 4 are built from publicly available open source SRPMS provided by Red Hat. CentOS conforms fully to the upstream vendor's redistribution policies and aims to be 100% binary compatible. CentOS mainly changes packages to remove upstream vendor branding and artwork. CentOS is for people who need an enterprise level operating system with stability to match without the associated cost and support.
  • Apache Web Server (2.0.52-22.ent.centos4) - http://www.apache.org/ - The Apache HTTP Server Project is a collaborative software development effort aimed at creating a robust, commercial-grade, feature rich, and freely-available source code implementation of an HTTP (Web) server. The project is jointly managed by a group of volunteers located around the world, using the Internet and the Web to communicate, plan, and develop the server and its related documentation.
  • PHP (4.3.9) - http://www.php.net/ PHP is an open-source, reflective programming language used mainly for developing server-side applications and dynamic web content, and more recently, other software.
  • PHPMyAdmin (2.7.0-pl2) - http://www.phpmyadmin.net/ phpMyAdmin is a tool written in PHP intended to handle the administration of MySQL over the Internet. Currently it can create and drop databases, create/drop/alter tables, delete/edit/add fields, execute any SQL statement, and manage keys on fields.
  • MySQL Database (4.1.12-3.RHEL4.1) - http://www.mysql.com/ MySQL is a multithreaded, multi-user, SQL (Structured Query Language) Database Management System (DBMS) with an estimated six million installations. MySQL AB makes MySQL available as free software under the GNU General Public License (GPL), but they also sell it under traditional commercial licensing arrangements for cases where the intended use is incompatible with use of the GPL. It is used in A@H Call Detail Reports and optional configuration information.
  • VSFTPD (2.0.1-5.EL4.3) - http://vsftpd.beasts.org/ Very Secure FTPD is a GPL licensed FTP server for UNIX systems, including Linux. It is very secure, stable and extremely fast.
  • sendmail (8.13.1-2) - http://www.sendmail.org/ - Sendmail is an open source mail transfer agent. A mail transfer agent or MTA (also called a mail server, or a mail exchange server in the context of the Domain Name System) is a computer program or software agent that transfers electronic mail messages from one computer to another.
  • Nwebmail ( 0.1.80 ) - http://nwebmail.sourceforge.net/ Nwebmail is a webmail client written in ANSI C. It allows users to check and send email from any web-browser. It accesses the mail spools directly for fast and efficient mail processing. It supports MIME attachments and can import/export address books.
  • OpenSSH (_3.9p1) - http://www.openssh.com/ - OpenSSH (Open Secure Shell) is a set of computer programs providing encrypted communication sessions over a computer network using the SSH protocol. It was created as an open alternative to the proprietary Secure Shell software.

  • xPL () - We have a built in xPL connector that sends out information on Voicemail and CallerID.
  • Integrated WebMeetMe GUI (A@H 2.7) - WebMeetMe is a front end to the MeetMe add-on. It gives users full control and the ability to monitor telephone conferences over on a web browser.
  • Digium card auto-config (A@H 2.7) -
  • Weather agi scripts (A@H 2.7) - Weather agi scripts Fetch the weather from weather.noaa.gov. At weather.noaa.gov is the weather stored in a text file that this script downloads and converts to a sound that is sent to the phone call. Default is Andrew's home city New York;-) This covers only US locations.
  • Wakeup calls (1.11) - Wakeup calls This is a wake up call feature. By dialing a phone number you can set the wakeup time when you would like to get a wakeup call.
  • Cisco SIP phone support () - We have a web interface and TFTP server that can configure Cisco SIP phones like the 7960
  • uLaw Sound Files
  • Java based SSH client
  • Samba Auto-Setup Script
  • VMware support -


Resources







Created by agillis, Last modification by james.zhu on Sat 04 of Aug, 2007 [09:34 UTC]

Comments

Comments Filter

what are requirements to connect public ip

by geetha on Tuesday 03 of April, 2007 [10:26:55 UTC]
hi friends,,

I installed and configured trixbox-VoIP(CentOS) with softphone in intranet successfully. We have another office in remote location. so that i route with our public ip. In the softphone setting also configured. We forwarded the port number as 10,000 for TCP and 5060 for UDP. After picked the call, we couldnt get the voice.. what is the problem.. can we add extra things..

Probelm with incoming calls to my DID-Please help me

by Chandra on Friday 01 of September, 2006 [11:15:50 UTC]
Hi friends,

Thank you to all for your response and cooperation to me. I have a doubt.

We have registered with Teliax and got DID number. We are making calls to USA successfully using your service. But, We are unable to receive incoming calls to our DID. Here I am sending my config files and error message on Asterisk console.

Contents in IAX.CONF file:

disallow=all
allow = ulaw

general
register => teliaxusername:teliaxpassword@voip-co1.teliax.com
                  
teliax
context=telincoming
type=friend
host=voip-co1.teliax.com
auth=md5
secret=teliaxpassword
disallow=all
allow=ulaw
allow=alaw
allow=gsm

Contents in Sip.conf file:

105
type=friend
username=105
secret=ravi
callerid="RaviKanth"
host=dynamic
context=leader
canreinvite=no
nat=yes
dtmfmode=rfc2833
allow=all
mailbox=605@vmail

107
type=friend
username=107
secret=suresh
callerid="Suresh"
host=dynamic
context=administration
canreinvite=no
nat=yes
dtmfmode=rfc2833
allow=all
mailbox=607@vmail

Contents in Extensions.conf file:

telincoming
exten => 303xxxxxxx, 1, Answer()
exten => 303xxxxxxx, n, Wait,2
exten => 303xxxxxxx, n, Goto(incoming,s,1)
include => internal
include => incoming

incoming
exten => s,1,Wait(3)
exten => s,n,Answer
exten => s,n,SetMusicOnHold(default)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(/tmp/virg2)
exten => s,n,Goto(s,1)
exten => s,n,Hangup()
include => internal

internal
exten => 105,1,SetMusicOnHold(default)
exten => 105,2,Dial(SIP/105,7,t,m,T)
exten => 1605,1,VoiceMailMain(605@vmail)
exten => 105,3,VoiceMail(605@vmail)
exten => 105,4,Hangup

exten => 107,1,SetMusicOnHold(default)
exten => 107,2,Dial(SIP/107,7,t,m,T)
exten => 1607,1,VoiceMailMain(607@vmail)
exten => 107,3,VoiceMail(607@vmail)
exten => 107,4,Hangup

uscall
exten => _1XXXXXXXXXX,1,DIAL(IAX2/teliaxusername@teliax/${EXTEN},30,tr)

manager
include => uscall
include => internal

The error message on Asterisk console:

  • CLI> — Executing Dial("SIP/105-007951e0", "IAX2/teliaxusername@teliax/1303xxxxxxx|30|tr") in new stack
   — Called teliaxusername@teliax/1303xxxxxxx
   — Call accepted by 207.174.202.2 (format ulaw)
   — Format for call is ulaw
   — IAX2/teliax-1 is ringing
   — IAX2/teliax-1 is making progress passing it to SIP/105-007951e0
   — IAX2/teliax-1 is ringing
   — IAX2/teliax-1 is busy
   — Hungup 'IAX2/teliax-1'
 == Everyone is busy/congested at this time (1:1/0/0)
 == Auto fallthrough, channel 'SIP/105-007951e0' status is 'BUSY'


What is the problem? Can you please tell me the solution. Looking forward to your response. Thank you.

Regards,
Chandra.

2 Strange probs with Asterisk@Home.....

by Paul Brown on Wednesday 02 of August, 2006 [09:39:19 UTC]
I am running Asterisk@home 2.7 and I get a few strange things happen...

1) If I call internally the voicemail kicks in, if I call from an outside trunk (Via sipgate for example) it never kicks in.
2) DTMF, when I call a place that requires DTMF it doesn't seem to work. I hear it but the other side doesn't recognise it.

Any clues? Its on a Cisco 7960 if that helps

Thanks

how to connect mysql front to asterisk@home

by rootx on Sunday 23 of July, 2006 [03:01:03 UTC]
I try to connect mysql front to mysqld run on asterisk@home. It deny and I don not know how to do my working. Help me !

H323 is not there after installed....

by vanna on Tuesday 28 of February, 2006 [02:29:52 UTC]
tafilaj : I have same problem with you. I installed this H323 and at the end it said installed, but when I tried to look for the file H323 under /usr/lib/asterisk/modules .. there is no such of this file in there. as far as I know, this file should be there if we install correctly. if anyone can get the H323 to work, please post the guide here.... thanks;

+447xxx macro help.

by Matt Martin on Tuesday 03 of January, 2006 [14:24:49 UTC]
I have a slight problem and have posted to several forums but without any joy.

Basically I have a queue that forwards to a mobile number in the UK. If the mobile is off or out of service - the network quite rightly says so and sends but an unavalible message.

However, Asterisk sees this and then sends the person on hold a message saying all circuits are busy - please try the call again later.

What I am trying to do is if it calls a +447 (07) number and gets that message to simply try another number in that queue but NOT give the error message to the caller.

Any help would be appreciated.

Re: Dial Patterns

by bjornta on Sunday 01 of May, 2005 [15:30:02 UTC]
The two Outbound Routing rules are competing against each other.
Right now it picks the *44+XXXXX rule, which adds *44 to 5-digit dialled numbers. Delete this rule, and it will work.

The + adds prefixes, and | removes prefixes

Btw: Everyone is using 363 as prefix for FWD, if you want to standardize.

Dial Patterns

by tafilaj on Sunday 01 of May, 2005 [15:20:22 UTC]
Hi there

quick question if anyone is willing to give me a hand on a problem.

how can set AMP so when i dial 0800800800 it goes to FWD and connect me.

now i use a custum extention


free-calls
exten => _0800.,1,SetCallerId,XXXXX
exten => _0800.,2,Dial(IAX2/?????:??????@iax2.fwdnet.net/*44${EXTEN:1},60,r)
exten => _0800.,3,Congestion

I did the trunk with 6310|XXXXX.
and Outbound routing
  • 44+XXXXX.


But it will not work.
it calls the number as *446310XXXXXXX.

it will not take the 6310 out of the dial string.

any ideas?

oh323

by tafilaj on Tuesday 19 of April, 2005 [22:46:03 UTC]
Great job by the way, a calling card aplication and it's perfect

( people dont want much do they , (:smile:))

im installin H323 support from the script that was provided at download site.

IT does not install.
I get h323 installed
but it is not on the system
when I did some snooping in to the instalation i noticed


make1: *** /usr/src/openh323/lib/libh323_linux_x86_r.so.1.13.5 Error 1
make1: Leaving directory `/usr/src/openh323/src'
make: *** opt Error 2

and than
make1: Entering directory `/usr/src/asterisk-oh323-0.6.5/asterisk-driver'
ERROR: No PWLIB library found!
make1: *** chan_oh323.so Error 1
make1: Leaving directory `/usr/src/asterisk-oh323-0.6.5/asterisk-driver'
make: *** subdirs_build Error 1

after that it tells me it's instaled
im a beginer at this, so anyone that can tell me how can i sort this out???


thank you for your time in advance..

How to turn off FOP Server?

by bulgin on Monday 18 of April, 2005 [21:13:58 UTC]
Hello and thanks for making Asterisk@home available to the open source community. It's great! A simple question — if I would like to turn off the FOP server on startup, how would I go about doing that?

Thanks

B.

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