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.
Asterisk 1.4.40, 1.6.2.17 and 1.8.3 released.
The Asterisk Development Team has announced the release of Asterisk 1.4.40, 1.6.2.17 and 1.8.3.Asterisk 1.4.40
The Asterisk Development Team has announced the release of Asterisk 1.4.40. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.40 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman)
* Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson)
* Resolve issue in res_odbc where it may crash when a query fails.
(Closes issue #18243. Reported, patched by ks3)
* Fix CPU spike when pressing DTMF after agent login.
(Closes issue #18130. Reported by rgj. Patched by jpeeler)
* Fix cross-compiling issue.
(Closes issue #18301. Reported, patched by abelbeck)
* This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman)
* Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes.
* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.40
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.17
The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman)
* Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman)
* When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman)
* Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson)
* Fix regression causing forwarding voicemails to not work with file storage.
(Closes issue #18358. Reported by cabal95. Patched by jpeeler)
* This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman)
* Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes.
* Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47)
* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.17
Thank you for your continued support of Asterisk!
Asterisk 1.8.3
The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you!
The following is a sample of the issues resolved in this release:
* Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman)
* Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
(Closes issue #18464. Reported, patched by IgorG)
* Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing.
(Closes issue #18350. Reported by gbour. Patched by Marquis)
* When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman)
* Resolve memory leak in iCalendar and Exchange calendaring modules.
(Closes issue #18521. Reported, patched by pitel. Tested by cervajs)
* This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman)
* Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47)
* Resolve a memory leak when the Asterisk Manager Interface is disabled.
(Reported internally by kmorgan. Patched by russellb)
* Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486.
(Reported internally. Patched by mnicholson)
* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
* Resolve deadlock involving REFER.
(Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)
Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3
Thank you for your continued support of Asterisk!
Wednesday, March 2, 2011
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory.
For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announc"> .msg_list {margin: 0px;padding: 0px;width: 100%; } .msg_head {padding: 5px 10px;cursor: pointer;position: relative;background-color:#f4f4ff;border: 1px solid #ccc;margin:0px; } .msg_body {padding: 5px 10px 15px;background-color:#F4F4F8; display: none;border: 1px solid #ccc;border-top: 0px; } .msg_body_2 {padding: 5px 10px 15px;background-color:#F4F4F8; }
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Asterisk and Kamailio (openser) realtime integration
August 5, 2010 Average Vote: 9.9
Daniel-Constantin Mierla posted a writeup on combining Asterisk and Kamailio.
Interview with Mark Spencer
November 26, 2004 Average Vote: 9.9
We have managed to get an interview with Mark Spencer AKA Markster. Mark Spencer is the creator of Asterisk and by far the most active developer.
ZORG new C++ and Java ZRTP implementation public release
January 13, 2011 Average Vote: 9.9
Andrea Cristofanini from PrivateWave has sent us across a press release about a new Open Source encryption implementation.
Asterisk IPv6 update
February 1, 2010 Average Vote: 9.8
Olle has posted an update on IPV6 in Asterisk and a link to a blog post of his.
Proposal for T.38 transparent gateway design in Asterisk
April 29, 2010 Average Vote: 9.8
Kevin Fleming has posted a proposed design for a transparent T.38 gateway for Asterisk:
A2Billing 1.9 (Cuprum) released
March 2, 2011
Areski has posted details of the latest release of A2Billing.
Using exec to set externaddr in sip.conf
March 1, 2011
Leif Madsen has put together a script to use cURL via PHP to set externaddr in sip.conf.
Asterisk 1.4.40, 1.6.2.17 and 1.8.3 released.
March 1, 2011
The Asterisk Development Team has announced the release of Asterisk 1.4.40, 1.6.2.17 and 1.8.3.
Introducing the new ConfBridge
February 24, 2011
David Vossel has posted details of the new ConfBridge in Asterisk.
Adhearsion 1.0.1 Released
February 24, 2011
Ben Klang has posted details of the latest version of Adhearsion - the Open Source Ruby language framework for creating telephony applications.
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
February 23, 2011
The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8.
AST-2011-002: Multiple array overflow and crashvulnerabilities in UDPTL code
February 22, 2011
The Asterisk Project has release a security advisory.
AstLinux 0.7.6 Released
February 21, 2011
The AstLinux Team have announced the release of the latest version of their embedded operating system with Asterisk.
First HD Conference In Asterisk
February 21, 2011
Leif Madsen has posted details of a successful public HD conference using Asterisk.
Chan SS7 2.0.0 released
February 18, 2011
Anders Baekgaard from Netfors has posted details of the latest release of their SS7 Channel for Asterisk.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory.
For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announc"> .msg_list {margin: 0px;padding: 0px;width: 100%; } .msg_head {padding: 5px 10px;cursor: pointer;position: relative;background-color:#f4f4ff;border: 1px solid #ccc;margin:0px; } .msg_body {padding: 5px 10px 15px;background-color:#F4F4F8; display: none;border: 1px solid #ccc;border-top: 0px; } .msg_body_2 {padding: 5px 10px 15px;background-color:#F4F4F8; }
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Tue, 22 Feb 2011 17:12:23 -0400 The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory.
For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announcement.
For a full list of changes in the current release, please see the ChangeLog:
ChangeLog-1.4.39.2
ChangeLog-1.6.1.22
ChangeLog-1.6.2.16.2
ChangeLog-1.8.2.4
Security advisory AST-2011-002 is available at:
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
Thank you for your continued support of Asterisk!
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Comments (Click to post)
Name: Subject: Website: Message: Similar Articles (Based on Title)DUNDi is available for use with v1-0 - October 23, 2004
BKW has released a .tar.gz file so that you can use DUNDi with Asterisk 1.0
Asterisk-Users: Asterisk 1.0.2 rpms now available for FC1 - October 27, 2004
Andrew McRory has posted details of the rpms for Asterisk 1.0.2
Asterisk-Users: RPMS for Fedora Core 2 now available - November 9, 2004
Andrew McRory has posted information on the RPMs of Asterisk for Fedora Core 2.
asterisk-oh323: New versions available - December 22, 2004
Michael Manousos has announced new versions of chan_oh323 from inaccessnetworks.
New 4-Port BRI card Sirrix.PCI4S0 with Asterisk support available - January 9, 2005
Oskar Senft has posted details of the 4-Port BRI card Sirrix.PCI4S0 with Asterisk support to the Asterisk-biz mailing list.
*-Dev: New jitterbuffer and Packet Loss Concealment preview/prototype patch available in tracker. - January 21, 2005
Steve Kann has posted details of the latest patch added to the bugtracker.
Linux Bridge + QoS Shaper HOWTO available - January 28, 2005
Ron Senykoff has posted details of a HOWTO he has written.
DIAX version 0.9.10a available for download - February 9, 2005
Dan has posted details of the latest version of his IAX softphone - DIAX - to the Asterisk-Users mailing list.
DIAX 0.9.10f available for download - March 12, 2005
Dan has posted details of the release of the latest version of the DIAX softphone.
UNISTIM channel driver available - March 12, 2005
Andres has posted details of a release from Cedric Hans of a UNISTIM channel driver for Asterisk.
Iaxclient-devel: Kiax 0.8.3 available - March 18, 2005
Emil Stoyanov has posted details of the latest release of Kiax.
AstLinux 0.2.5.5 now available for testing (includes ISDN/BRI) - April 15, 2005
Kristian Kielhofner has posted details of the pre-release of what will become 0.2.6 of AstLinux.
Original Content (C) 2004-2010Matt Riddell
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Asterisk: The Definitive Guide book available for pre order and public review
October 21, 2010 Average Vote: 10
The 3rd edition of the Asterisk book is on its way and available for public review and pre-order.
Submit your own stories
September 21, 2010 Average Vote: 10
We have made a pretty big update to the Daily Asterisk News. You can now submit your own stories! There is now a link at the top of the Daily Asterisk News. Just click the link, fill out the form and we will check over your story and submit it.
First HD Conference In Asterisk
February 21, 2011 Average Vote: 10
Leif Madsen has posted details of a successful public HD conference using Asterisk.
Introducing the new ConfBridge
February 24, 2011 Average Vote: 10
David Vossel has posted details of the new ConfBridge in Asterisk.
Tropo Now Speaks Asterisk Gateway Interface (AGI)
October 4, 2010 Average Vote: 10
Jason Goeke has posted a link to an article about Tropo support for Asterisk via AGI.
The Everything Asterisk Video Collection
August 5, 2010 Average Vote: 10
Steven Sokol has posted a blog entry on Asterisk Video Resources.
AstriDevCon: October 29th, Washington DC
August 23, 2010 Average Vote: 10
John Todd has posted a note about the AstriDevCon conference which occurs within the Astricon conference.
Using locked PAP2 and PAP2-NA with Asterisk
August 23, 2005 Average Vote: 10
VoIPHacker has posted details on how to unlock a PAP2 and PAP2-NA.
Asterisk 1.10 Update
February 17, 2011 Average Vote: 10
Russell Bryant has posted some information about Asterisk 1.10.
Asterisk-Fax
May 19, 2005 Average Vote: 10
Just thought I'd provide a pointer to the Asterisk-Fax site (it came up on Asterisk-Users).
Asterisk and Kamailio (openser) realtime integration
August 5, 2010 Average Vote: 9.9
Daniel-Constantin Mierla posted a writeup on combining Asterisk and Kamailio.
Interview with Mark Spencer
November 26, 2004 Average Vote: 9.9
We have managed to get an interview with Mark Spencer AKA Markster. Mark Spencer is the creator of Asterisk and by far the most active developer.
ZORG new C++ and Java ZRTP implementation public release
January 13, 2011 Average Vote: 9.9
Andrea Cristofanini from PrivateWave has sent us across a press release about a new Open Source encryption implementation.
Asterisk IPv6 update
February 1, 2010 Average Vote: 9.8
Olle has posted an update on IPV6 in Asterisk and a link to a blog post of his.
Proposal for T.38 transparent gateway design in Asterisk
April 29, 2010 Average Vote: 9.8
Kevin Fleming has posted a proposed design for a transparent T.38 gateway for Asterisk:
A2Billing 1.9 (Cuprum) released
March 2, 2011
Areski has posted details of the latest release of A2Billing.
Using exec to set externaddr in sip.conf
March 1, 2011
Leif Madsen has put together a script to use cURL via PHP to set externaddr in sip.conf.
Asterisk 1.4.40, 1.6.2.17 and 1.8.3 released.
March 1, 2011
The Asterisk Development Team has announced the release of Asterisk 1.4.40, 1.6.2.17 and 1.8.3.
Introducing the new ConfBridge
February 24, 2011
David Vossel has posted details of the new ConfBridge in Asterisk.
Adhearsion 1.0.1 Released
February 24, 2011
Ben Klang has posted details of the latest version of Adhearsion - the Open Source Ruby language framework for creating telephony applications.
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
February 23, 2011
The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8.
AST-2011-002: Multiple array overflow and crashvulnerabilities in UDPTL code
February 22, 2011
The Asterisk Project has release a security advisory.
AstLinux 0.7.6 Released
February 21, 2011
The AstLinux Team have announced the release of the latest version of their embedded operating system with Asterisk.
First HD Conference In Asterisk
February 21, 2011
Leif Madsen has posted details of a successful public HD conference using Asterisk.
Chan SS7 2.0.0 released
February 18, 2011
Anders Baekgaard from Netfors has posted details of the latest release of their SS7 Channel for Asterisk.
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