The Asterisk Development Team has announced security releases for Asterisk
branches 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which
contained a bug which caused duplicate manager entries (issue #18987).
The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues:
* Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
* Remote crash vulnerability in TCP/TLS server (AST-2011-004)
The issues and resolutions are described in the AST-2011-003 and AST-2011-004
security advisories.
For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-003 and AST-2011-004, which were released at the
same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
Security advisory AST-2011-003 and AST-2011-004 are available at:
http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
http://downloads.asterisk.org/pub/security/AST-2011-004.pdf
Thank you for your continued support of Asterisk!
Sunday, March 27, 2011
Wednesday, March 23, 2011
Asterisk Multi Tenant PBX
SimplePBX is a dial plan and configuration manager for the Asterisk Open Source VoIP and Telephony server. SimplePBX is written in Object Pascal using the excellent Open Source compiler Freepascal. All configuration data is stored in a MySql database.
Unlike other dial plan managers SimplePBX is strictly a console application. This has its advantages and disadvantages but was done this way because I am an old school Turbo Pascal programmer and I just wanted to create an application the best way I knew how.
The main advantage in creating a console application is speed. Applications can be created and debugged much faster than web based applications. Console applications also benefit from the built in SSH when being accessed from a remote system. Another benefit is the size of the application and the amount of files that need to be installed on a host system.
The reason that SimplePBX was created was as a project for me to learn Linux and Asterisk. So far this has turned out be a great decision because I have extended my skills as well as my professional interests. I have learned a great deal about Voip and Telephony servers. There is a lot to learn still and I hope to apply future knowledge back into this project.
I have chosen to support a multi tenant environment because there is a need for this. This is not just for TSP’s but also small business and homes offices. Many small business share office space, a receptionist, Internet, and now a PBX system. With single tenant system companies are forced to setup a PBX for each company. This is time consuming and expensive. With a Multi tenant system a PBX can serve multiple companies by sharing the costs and benefits evenly.
SimplePBX is available at www.totek.ca
Unlike other dial plan managers SimplePBX is strictly a console application. This has its advantages and disadvantages but was done this way because I am an old school Turbo Pascal programmer and I just wanted to create an application the best way I knew how.
The main advantage in creating a console application is speed. Applications can be created and debugged much faster than web based applications. Console applications also benefit from the built in SSH when being accessed from a remote system. Another benefit is the size of the application and the amount of files that need to be installed on a host system.
The reason that SimplePBX was created was as a project for me to learn Linux and Asterisk. So far this has turned out be a great decision because I have extended my skills as well as my professional interests. I have learned a great deal about Voip and Telephony servers. There is a lot to learn still and I hope to apply future knowledge back into this project.
I have chosen to support a multi tenant environment because there is a need for this. This is not just for TSP’s but also small business and homes offices. Many small business share office space, a receptionist, Internet, and now a PBX system. With single tenant system companies are forced to setup a PBX for each company. This is time consuming and expensive. With a Multi tenant system a PBX can serve multiple companies by sharing the costs and benefits evenly.
SimplePBX is available at www.totek.ca
Preparing for Asterisk on Ubuntu 10.10
Through the next few posts, I plan on showing you how to install Asterisk 1.6.x on Ubuntu 10.10, link it to an Avaya S8700/8300 through H.323 and configure it to successfully handle all voicemail. This project is a great solution for anyone still using an Octel voicemail box, as support is quickly coming to an end (June 2011). This solution solves a few problems others have been having – including MWI (message waiting indicators) on Avaya telephones AND figuring out what extension was dialed.
I plan on showing an exact step-by-step on my setup, so duplicating it should be very straight forward. Let’s start with the installation.To begin, you’re going to need a copy of Ubuntu 10.10 Before we go any further, I’m sure there are plenty of users that would love to debate the usage of a certain flavor of *nix over another. I’m not interested. I’ve worked with Red Hat, CentOS and Mandrake/Mandriva, but *most* of my experience has been with Ubuntu. I first installed Dapper Drake on my home machine years ago, and forced myself to become familiar with a Windows alternative. I am NOT a Linux power user, but feel fairly comfortable with Ubuntu – so I choose to use it for my Asterisk solution.
Installing Ubuntu should be cake, and reproducing a walk-through for this would be repetitive and unnecessary.
After installing Ubuntu, we’re going to need a few packages/libraries prior to installing Asterisk. If your install requires more than setting up Asterisk as just a voicemail server, you may need additional packages. In some future posts, I’ll go over this along with step by step instructions on getting things to work (for example – I have my Asterisk box working with Google Voice. This required 2 more packages and recompiling Asterisk.)
Let’s start by getting a command prompt, and elevating our permissions.
sudo su
Next, let’s install some packages (some might already be installed)
apt-get install ssh
apt-get install gcc
apt-get install ncurses-dev
apt-get install g++
apt-get install flex
apt-get install libcurl3-dev
apt-get install libxml2-dev
apt-get install libiksemel-dev
apt-get install build-essential
Rockin and rollin now! The system is getting close for our Asterisk install.
If you plan to allow voicemail users to receive voicemail via email, we’ll also want to install Postfix. To do that type:
apt-get install postfix
During the install of postfix, you will be asked a few questions about how you want Postfix to behave. On my system, I set Postfix up as a relay server, forwarding mail to our Exchange boxes to be processed.
Optionally, you may want to install Postgres or MySQL to store CDR (call data records) or leverage Asterisk’s Realtime component. I will cover this in a later post, as my setup uses MySQL to store all voicemail box information – making management a breeze and eliminating the need to reload the voicemail configuration files after each modification.
Monday, March 21, 2011
How To Install Asterisk PBX System
Installing Asterisk PBX can be a daunting task and were it not for the numerous individuals and companies and their amazing contributions to the Asterisk open-source project, the Asterisk PBX software would not be where it is today. There are many, many sources of information regarding the proper installation of an Asterisk PBX system. Because Asterisk can be and is used for so many purposes, finding an installation guide for your specific scenario is difficult. It is therefore best to get your feet wet with a simple getting started document that can get you familiarized with the installation process and some of the basic functionality that Asterisk can provide. The best document for this purpose is the Getting Started With Asterisk document written by Andy Powell. This document can be found at: http://www.automated.it/guidetoasterisk.htm
Asterisk PBX Total Cost of Ownership
Based on Asterisk, the most widely used open source telephony software, Switchvox provides all the functionality and stability of a sophisticated IP PBX platform at a fraction of the cost.
In-Stat predicts that by 2013, nearly 80 percent of businesses are expected to have adopted voice over IP (VoIP) phone systems, a rapid rise from half that number in 2009. Full-featured unified communications systems combine VoIP, messaging and presence with business data to improve employee efficiency. However, traditional UC vendors can charge SMBs big fees—in some cases, hundreds of thousand of dollars over a five-year period—for their phone systems. Most of this cost is the result of fees for system “add-ons.” This includes fees for adding separate voicemail servers, fees for mobile accessibility, fees for integration with other business software or CRM systems, and charging for critical features such as conferencing, interactive voice response (IVR) and call queues.
Leslie Conway, vice president of global marketing at Digium, commented: “When SMBs are evaluating UC solutions, it’s worthwhile to do a feature by feature comparison and evaluate the associated costs for the number of users needing those features. The potential savings on that total investment could translate to $80,000 on the average phone system over time. Our customers are in the unique position of making a buying decision that could eliminate $50,000 or more off their initial investment, while gaining enterprise-class features from a true unified communications solution.”
NP Information Systems, a Seattle-based Digium certified partner and reseller of Switchvox, often works with its SMB customers to compare features across business phone systems and identify potential cost savings. A recent customer, Clear Creek Telephone & Television, needed to replace its phone system with VoIP functionality but also required substantial customization. Formed as a co-op in the 1900s, the company has a century of experience providing telephone, cable and Internet service to businesses and residents of Oregon City and Redland, Oregon. “They’d first considered a ShoreTel system, but getting all the features they needed would have exceeded their budget,” said Vince Preece of NP Information Systems.
In-Stat predicts that by 2013, nearly 80 percent of businesses are expected to have adopted voice over IP (VoIP) phone systems, a rapid rise from half that number in 2009. Full-featured unified communications systems combine VoIP, messaging and presence with business data to improve employee efficiency. However, traditional UC vendors can charge SMBs big fees—in some cases, hundreds of thousand of dollars over a five-year period—for their phone systems. Most of this cost is the result of fees for system “add-ons.” This includes fees for adding separate voicemail servers, fees for mobile accessibility, fees for integration with other business software or CRM systems, and charging for critical features such as conferencing, interactive voice response (IVR) and call queues.
Leslie Conway, vice president of global marketing at Digium, commented: “When SMBs are evaluating UC solutions, it’s worthwhile to do a feature by feature comparison and evaluate the associated costs for the number of users needing those features. The potential savings on that total investment could translate to $80,000 on the average phone system over time. Our customers are in the unique position of making a buying decision that could eliminate $50,000 or more off their initial investment, while gaining enterprise-class features from a true unified communications solution.”
NP Information Systems, a Seattle-based Digium certified partner and reseller of Switchvox, often works with its SMB customers to compare features across business phone systems and identify potential cost savings. A recent customer, Clear Creek Telephone & Television, needed to replace its phone system with VoIP functionality but also required substantial customization. Formed as a co-op in the 1900s, the company has a century of experience providing telephone, cable and Internet service to businesses and residents of Oregon City and Redland, Oregon. “They’d first considered a ShoreTel system, but getting all the features they needed would have exceeded their budget,” said Vince Preece of NP Information Systems.
Sunday, March 20, 2011
Asterisk PBX Manager
A commercial Asterisk management software is available from Thirdlane. Asterisk PBX Manager allows for easier configuration of the Asterisk PBX.
Asterisk PBX Manager includes a simple GUI interface allowing experienced and not-so experienced users to make changes to their Asterisk configuration without having to modify text based configuration files.
Asterisk PBX Manager is a webmin based module and can be easily installed into a webmin environment. The following links will guide you to the information required to begin using Asterisk PBX Manager
* View some Asterisk PBX Manager screenshots.
* If you don't already have Webmin, it is available here.
* Asterisk PBX Manager is available from Thirdlane.
* View a live demo of Asterisk PBX Manager from Thirdlane. User is: demo Password is: insecure
Here is a list of tasks that can be done with Asterisk PBX Manager webmin
* PBX Options - general PBX configuration and system defaults management
* Generator - bulk generation of extensions, devices, and mailboxes
* Trunk Manager - management of PBX connections to service providers or other PBXs
* Route Manager - management of outbound dialing
* Inbound Call Manager - management of inbound calls and DIDs
* Extensions Manager - dialplan management and assignment of scripts to extensions
* Phone Manager - management of devices (phones)
* Mailbox Manager - configuration of user mailboxes
* Voice Menu Manager - configuration of Auto Attendant and multi level voice menus
* Queue Manager - management of ACD and call queues
* Music-on-Hold Manager - management of on hold music
* Script Manager - creation of scripts for call handling (used by Extensions Manager)
* Conference Manager - configuration of conference rooms
* Configuration Editor - direct access to Asterisk configuration files
* Command Shell- web interface to Asterisk command line interface
* File Manager - intelligent upload and download for various configuration and support files
* Call Detail Records - display of Asterisk CDR
Asterisk PBX Manager includes a simple GUI interface allowing experienced and not-so experienced users to make changes to their Asterisk configuration without having to modify text based configuration files.
Asterisk PBX Manager is a webmin based module and can be easily installed into a webmin environment. The following links will guide you to the information required to begin using Asterisk PBX Manager
* View some Asterisk PBX Manager screenshots.
* If you don't already have Webmin, it is available here.
* Asterisk PBX Manager is available from Thirdlane.
* View a live demo of Asterisk PBX Manager from Thirdlane. User is: demo Password is: insecure
Here is a list of tasks that can be done with Asterisk PBX Manager webmin
* PBX Options - general PBX configuration and system defaults management
* Generator - bulk generation of extensions, devices, and mailboxes
* Trunk Manager - management of PBX connections to service providers or other PBXs
* Route Manager - management of outbound dialing
* Inbound Call Manager - management of inbound calls and DIDs
* Extensions Manager - dialplan management and assignment of scripts to extensions
* Phone Manager - management of devices (phones)
* Mailbox Manager - configuration of user mailboxes
* Voice Menu Manager - configuration of Auto Attendant and multi level voice menus
* Queue Manager - management of ACD and call queues
* Music-on-Hold Manager - management of on hold music
* Script Manager - creation of scripts for call handling (used by Extensions Manager)
* Conference Manager - configuration of conference rooms
* Configuration Editor - direct access to Asterisk configuration files
* Command Shell- web interface to Asterisk command line interface
* File Manager - intelligent upload and download for various configuration and support files
* Call Detail Records - display of Asterisk CDR
Thursday, March 3, 2011
Asterisk ConfBridge
ConfBridge is an application for Asterisk starting with the 1.6.2.* series. ConfBridge is very similar in features to MeetMe, but unlike MeetMe, ConfBridge does not perform audio mixing using DAHDI. Instead, audio mixing is performed within the internals of Asterisk.
To get an up2date description of ConfBridge for your used Asterisk version execute "core show application ConfBridge" on the Asterisk CLI.
The option string may contain zero or more of the following characters:
* 'a' — Set admin mode
* 'A' — Set marked mode
* 'c' — Announce user(s) count on joining a conference.
* 'm' — Set initially muted.
* 'M' — Enable music on hold when the conference has a single caller. Optionally, specify a musiconhold class to use. If one is not provided, it will use the channel's currently set music class, or 'default'
* '1' — Do not play message when first person enters
* 's' — Present menu (user or admin) when '#' is received (send to menu)
* 'w' — Wait until the marked user enters the conference
* 'q' — Quiet mode (don't play enter/leave sounds).
The join sound can be set using the 'CONFBRIDGE_JOIN_SOUND' variable and
the leave sound can be set using the 'CONFBRIDGE_LEAVE_SOUND' variable.
These can be unique to the caller.
NOTE: This application will not automatically answer the channel.
To get an up2date description of ConfBridge for your used Asterisk version execute "core show application ConfBridge" on the Asterisk CLI.
The option string may contain zero or more of the following characters:
* 'a' — Set admin mode
* 'A' — Set marked mode
* 'c' — Announce user(s) count on joining a conference.
* 'm' — Set initially muted.
* 'M' — Enable music on hold when the conference has a single caller. Optionally, specify a musiconhold class to use. If one is not provided, it will use the channel's currently set music class, or 'default'
* '1' — Do not play message when first person enters
* 's' — Present menu (user or admin) when '#' is received (send to menu)
* 'w' — Wait until the marked user enters the conference
* 'q' — Quiet mode (don't play enter/leave sounds).
The join sound can be set using the 'CONFBRIDGE_JOIN_SOUND' variable and
the leave sound can be set using the 'CONFBRIDGE_LEAVE_SOUND' variable.
These can be unique to the caller.
NOTE: This application will not automatically answer the channel.
Subscribe to:
Comments (Atom)