Asterisk VoIP News

Thursday, June 30, 2005

AstLinux 0.2.7 Released (please test)

-----------AstLinux 0.2.7------------------------------------------------
-MAJOR init cleanups (conditional start/stops)

-Moved zaptel modules to seperate variable - see /stat/etc/rc.conf

-Improved Sangoma support - should now work for voice and data
(for voice you will still need to edit /etc/zaptel.conf)

-If you are not using Sangoma hardware, make sure to delete

-Improved Zaptel support - should now work for voice and data
(please note you still need to manually run wancfg and/or edit

-removed broken PPPoE startup code - anyone want to work on this?
It is available immediately with "astup testing" or as compressed disk images below:

Enjoy! And let me know if you have any problems (to the list, NOT by direct e-mail).

P.S. - On a personal note, I did most of this work at about 6am in Las Vegas while on "vacation"... Take that to mean whatever you want. Not that it's buggy or I feel the need to make pre-emptive excuses, just an FYI! :)

Kristian Kielhofner

Gizmo: Skype done right?

This has been submitted as a Slashdot story.
I have absolutely no connection with any of the principals. This could have a major impact on the Asterisk community, and VoIP usage in general.

Michael Robertson, of fame, has been battling for open standards in the IP telephony world, in addition to his better-known Lindows (now Linspire, at venture to promote Linux on the desktop. His VoIP operation works great for me, but Michael has been long concerned about the totally closed and proprietary nature of Skype (as well as a lot of the misleading hype surrounding it).

Today his crew released "Gizmo" (at " (a tentative
name until a better one is found) which has the main benefits of Skype, PLUS it is layered upon SIP, DUNDI, and the existing infrastructure, meaning it is fully interconnectable to the world by obvious and nonobvious techniques, Asterisk being on the top of the obvious charts... This is certainly what I've been waiting for, being totally cheesed by the smarminess of Skype and its founders. "Open Standards" is one of the most abused concepts this side of Lake Washington, but this comes pretty damn close!

UTStarcom F1000 WiFi IP Phone Review

weicheng jiang has posted a review of the F1000 Wifi phone:

I bought a UTStarcom F1000 WiFi IP Phone from and tested it with Asterisk. This is a my first impression of the device.

The F1000 supports SIP. It looks and operates like a cell phone, and connects to the Internet through WiFi, so you can use it at any WiFi hotspot. I set up a 802.11b wi-fi network with a Linksys BEFW11S4 Wireless-B broadband router with no security requirement and SSID broadcast enabled. When I turned on the F1000 it automatically picked up the signal and connected to the network.

Usage of the keypad is pretty intuitive, the buttons are the same as on a cell phone, I didn't bother to read the user guide. ;-) I scrolled to the Wi-Fi configuration menu and entered the domain of my asterisk server as the SIP proxy. In the same section I entered my asterisk user name and password. After a reboot the phone was registered to my asterisk server! I made a few calls and the call quality was fine, similar to a regular IP phone.

I then took the phone to the office, it also picked up the wifi signal there and connected to the Internet automatically. Nice! I showed it off to my colleague and they wanted to get one too.

I will follow up with a review of the advanced features such as call transfer and 3-way calling.

Feature List:
-802.11b Wireless
-Frequency band 2.4GHz
-G.711, G.729a/b codecs
-SIP (Session Initiation Protocol)
-3-Way Calling
-Call Forwarding, Call Transfer, Call Waiting
-Talk time up to 5 hours
-Call Hold / Resume
-Call rejecting / redial / mute
-802.1x Authentication
-Comfort noise generation (CNG)
-Voice activity detection (VAD)
-Echo cancellation
-Adaptive jitter buffer
-64 and 128bit Wired Equivalent Privacy (WEP)
-Weight 111g
-Dimensions 11 x 4.5 x 2.2 cm
-100-240VAC 50-60Hz 120mA universal charger
-Transmission output 20mW
-RTP (Real-Time Transfer Protocol) / RTCP
-SDP (Session Description Protocol)
-SAP (Session Announcement Protocol)
-Standby time 50 - 100 hours
-Li-ion DC 3.6V 1500mAh battery
-Charging time 2 - 3 hours

- Lucas

Tuesday, June 28, 2005

HooDaHek 0.2 Released

Just a few changes:
- Added MSN Messenger support to the notification bot
- Added debugging code to the dbhandler script.

Monday, June 27, 2005

Astricon Europe Media Post

Kristian Kielhofner has posted info on the wiki page which contains links to the available media from Astricon:

Hello everyone,

In case you haven't seen it yet, a few of us coming back from Astricon Europe have uploaded our pictures and created a page on the wiki:

Check it out, and I'll see you all in Anaheim.

P.S. - If all 10,000+ of you do show up, I won't be able to meet each and every one of you, but I'll try.

Kristian Kielhofner

Asterisk 1.0.8 Released!!

Russell Bryant has posted details of the release of the latest STABLE version of Asterisk:


Version 1.0.8 has been released of Asterisk, Asterisk-addons, Zaptel, and Libpri. This release contains a significant amount of bug fixes (possibly the most of the 1.0.X releases). Tarballs are available on the asterisk web site as well as the asterisk ftp server.

A complete list of all changes made to the v1-0 branch is available through the archives of the cvs mailing list. See for more information.

ChangeLogs that represent an overview of the larger changes are available in the source, as well as the following web site for convenience -

Thanks for your support,

Russell Bryant

IPSwitchBoard version 0.120 released

Thorben Jensen has posted details of the latest release of the .NET Manager for Asterisk:

Download FREE from:

IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a FREE Windows.NET application which gives you:

-Unattended/attended transfers.

-Park calls and retrieve/forward them again.

-Organize all your SIP, IAX, CAPI and Zap extensions (automatically retrieved from Asterisk).

-Hotel/Call-Shop Billing module

-Monitor all extensions, queues, agents and Parked Calls.

-Dynamically log extensions in and out of queues.

-Integration with CRM software on the web.

-Browse Call Records and make Charts.

-Record calls and transfer wav files to the PC automatically.

-Set Do Not Disturb on Extensions and give a reason.

-Speed Dialing. Speed Dial Numbers can be shared from the server.

Asterisk connecting remote villages in western Uganda

Mark Summer has posted details of a non-profit project being run in Uganda:


I though some of you on this list might be interested in what Inveneo is doing in Uganda. We are a San Francisco based non-profit organization that builds rugged, low-cost, highly reliable and open- source communications systems for under-served communities around the world. We have just completed our first installation in western Uganda, Africa.

The system is up and running since this past Wednesday (June 8th). We have installed 5 units, 4 of which are in villages with with no access to power. The system provides Internet access and phone capabilities to the users. Phone calls among the connected villages are free of charge, with the ability to place and receive calls to / from the Ugandan phone network and voice mail boxes for each station. The systems are linked using 802.11 WiFi links.

For more information please have a look at the following links:

For more detailed information and pictures of the Uganda deployment:

For more information about the solution we have built and implemented, here is a link to our PDF datasheet:

And of course our website:

Thank you!


Friday, June 24, 2005

astGUIclient version released 1.1.4


We've released another update to our Asterisk GUI Client suite: 1.1.4

The client suite runs on Windows, UNIX and Mac, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not an asterisk configuration tool)
This package is geared towards Asterisk installations with SIP,IAX or Zap
phones and Zaptel, IAX or SIP trunks.

For this revision, in addition to adapting the code to the 'Local/' channel
changes made in Asterisk release 1.0.8 and CVS_HEAD, we have added the
ability to use SIP trunks for outbound and inbound lines to the package, as
well as adding an autodial IVR survey example script to VICIDIAL.

We have also created a graph showing possible hardware configurations for
systems running astGUIclient to better understand where astGUIclient fits
in and what it needs to run:

Let me know what you think.



A new way to H323!

Anthony Minessale has released a new Asterisk channel driver for the open source Woomera protocol on PBX Freeware   Woomera was designed to abstract the woes of managing various VoIP protocols from the PBX.

This excerpt was taken from the chan_woomera homepage:


The Woomera protocol, designed by Craig Southeren of OpenH323 fame, makes it possible to put your voice over IP system in one server/process and your PBX in another and connect them with a simple raw-linear-over-udp protocol. chan_woomera is an asterisk channel driver designed to interface the Asterisk PBX with Woomera. Currently this code is working but considered beta. Woomera currently only supports H323 but it should soon support the OPAL VoIP abstraction layer which will allow it to speak many other protocols. The number of protocols supported by the Woomera server is irrelevant to chan_woomera which will support anything Woomera supports because of its thin-client-like design.


Wednesday, June 22, 2005

HooDaHek version 0.1 released

HooDaHek (hoo-dah-hek, as in 'who-the-heck?') is a collection of Asterisk AGI scripts, CGI scripts, and MySQL tables intended to implement your own in-house Caller ID database and notification services. I wrote these scripts originally to enable my household phone system to identify who was calling, be able to modify and specify my own custom CallerID labels, and to have the phone system alert us via AOL Instant Messenger as to who was calling (or had called).

1. Caller calls into your Asterisk system via IAX, SIP, or PSTN
2. Your extensions.conf context sends CLID information to the hoodahek_dbhandle AGI script.
3. hoodahek_dbhandle determines if the caller has called before based on the phone number.
-If YES, it does nothing.
-If NO, it inserts a row into the 'hoodahek' table to identify the caller on subsequent calls. The initial information used is the information supplied by the telco's CLID transmission. If configured, an email is sent notifying someone of the insertion into the system. This information can then be edited by the CGI scripts provided.
4. Control returns to the dialplan.
5. The dialplan passes information to the hoodahek_notify AGI script.
6. hoodahek_notify looks to see if stored information about this caller is in the 'hoodahek' table.
-If YES, hoodahek_notify uses this information to announce the caller via AIM.
-If NO, hoodahek_notify looks to see if the CLID system has passed information about the caller (name, number)
=If YES, it uses that information to announce the caller via AIM.
=If NO, it announces the caller anonymously.
7. hoodahek_notify uses the information it found to write a call queue file in the queue directory for the hoodahekbot.
8. Control passes back into the dialplan to do whatever comes next.
9. hoodahekbot scans the queue directory every second to look for new call files. When it finds one, it opens it, scans in the lines, and deletes it. It then sends an IM to each recipient as configured. ]

For more information go here Hoodahek

Another Look: Asterisk Management Portal (AMP)

Open Source AMP
Coalescent Systems Inc. launched The Asterisk Management Portal project to bring together best-of-breed applications to produce a "canned" (but fully functional) turn-key SMB phone system based on The Asterisk Open Source PBX.

Coalescent Systems has contributed its administration interface to the project.

AMP Features at a Glance:
-Add or change extension and voicemail accounts in seconds
-Supports SIP, IAX, and ZAP clients
-Supports all Asterisk supported trunk technologies
-Reduce long distance costs with LCR
-Route incoming calls based on time-of-day
-Create interactive Digital Receptionist menus
-Design sophisticated call groups
-Manage callers with Queues
-Upload custom on-hold music (MOH)
-Search company directory, based on first or last name
-Detect and receive incoming faxes
-Share administrative duties with AMP Users
-Backup and Restore your system
-Save audio recordings of calls
-View call detail reporting with asterisk-stat
-View extension and trunk status with Flash Operator Pane

Testing Technologies SIP test environment used for IMS testing

Berlin, June 2005 - Testing Technologies announces today that it will take the lead in developing testing solutions for the emerging standardization of IP Multimedia
Subsystem (IMS) in cooperation with Fraunhofer FOKUS. Testing Technologies SIP test
environment can be used to create new test cases for IMS based testing.

IMS defines a generic architecture for offering Voice over IP (VoIP) and multimedia
services. IMS-based services enable person-to-person and person-to-content
communications in a variety of modes - including voice, text, pictures and video, or any combination of these - in a highly personalized and controlled way.

Due to the different standards and technologies being used when deploying IMS
software and services the testing of services like IMS is too complex for traditional
hardware based testing platforms. Manufacturers and carriers alike have realized the
ease in applying and adopting already existing TTCN-3 based conformance testing
solutions like SIP and UMTS to emerging technologies like IMS.

Manufacturers already working with tools from Testing Technologies are ie. Alcatel and
Motorola, even carriers and others getting there networks and backbone ready and
tested for the rapid deployment of IMS.

IMS testing is yet another proof of concept for the flexibility of TTCN-3, a standard for next generation testing, that is the base of Testing Technologies success in testing devices and applications in markets as diverse as automotive, telecoms and datacoms.
" The rapid test development and deployment allow our clients to stay on the cutting
edge of technology without being forced to continuously invest in new proprietary
hardware based testing platforms" says Testing Technologies CEO Theofanis Vassiliou-

The software based SIP testing environment from Testing Technolgies is a validated and customer proven functional test suite containing 534 test cases for SIP. It comes in two packages, TTsuite-SIP Executable to execute the test cases and analyze the test result and TTsuite-SIP Developers in which the SIP ATS sources can be modified for IMS purposes in the standardized test language TTCN-3. After modification new added or changed test cases can be compiled and executed on the fly.

About Testing Technologies IST GmbH

Testing Technologies, a spin-off of Fraunhofer FOKUS research institute, develops and markets universal and high innovative test development tool series and solutions based on the standardized test specification and implementation language TTCN-3.

Astlinux-users: OT: Asterisk and Mambo - help wanted

Kris needs someone to give him a hand with maboifying the AstLinux website:

Hello everyone,

So, this isn't exactly what it seems. I am not looking to integrate Asterisk and Mambo. I am the maintainer/creator of AstLinux, and I have recently decided that I should really have a better web site for it. I would like to use Mambo so that I can do updates easily, from anywhere, without having to waste time learning PHP/HTML/etc. Mambo CMS seems the best and most powerful way to do this. It's not that easy, however, to go from the default Mambo site to a site suitable for an open source project such as AstLinux. I just need some help to get a layout, theme, etc. going. Updates and maintenance I can handle (probably).

So, what I am looking for is someone who is familiar with Mambo (and preferably Asterisk, too) and would be willing to help me jump start Because AstLinux is an open source project, I will be unable to directly compensate anyone (monetarily) for their work at this time. However, any people that help out are more than welcome to plug their own projects, companies, names, etc. on the site (within reason).

Interested? Comments? Questions? Suggestions? Drop me a line.


Kristian Kielhofner

Spanish doc (Asterisk)

Leonardo Federico Bauchwitz has posted details of the completion of the translation of the Asterisk docs to spanish:

We have finished the translation of the FAQ of Digium to spanish.
They are already (in Spanish) available for download:
FAQ Frequently Asked Questions


Hardware compatibility list

Fast Installation Zaptel
All the documentation is available for download in:

Soon the following documents will be finished:
"Volume one" and "Asterisk Gateway Interface (AGI)"


Leonardo Federico Bauchwitz
Coordinator of Asterisk documentation in Spanish
leonardo [at]

Tuesday, June 21, 2005

OrderlyQ released

Orderly Software announces its new call queuing system, OrderlyQ:

OrderlyQ deploys automatically whenever there are too many incoming calls to handle. Instead of waiting on-hold, or hearing the engaged tone, your callers can choose to stay on hold, or call back in, say, seven minutes, when they will be at the front of the queue. It couldn't be simpler, and:
Your callers will be grateful that their time, and money, is not being wasted any more.

You'll save money on your telecoms costs

You can cope with spikes in demand with less agents, if you want to.
The system can even Text callers to tell them when they reach the front of the queue.

OrderlyQ is implemented as a bolt-on for Asterisk (TM) queues using the Asterisk Manager API and Asterisk FastAGI protocols, and is built on OrderlyCalls, the forthcoming Open Source Asterisk CTI application server, also from Orderly Software.

OrderlyQ works with any existing Asterisk queue, and can also protect any other VOIP and PSTN PBX or phones, and provide its own queue service where none exists.

To experience OrderlyQ, please call the demonstration hotline on +44 845 004 5412, and for more information about our patent-pending system, please take a look at

Monday, June 20, 2005 Open for business!

Anthony Minessale II, CTO and implementer of Asterlink, has announced the grand opening of, a new site designed to supply the open source PBX community with access to contributed open source applications. To kick start the release, Anthony has uploaded a few of his own popular open source modules for Asterisk namely res_perl, res_sqlite and app_valetparking as well as a brand new module just released today. "The site is not complete as far as my final vision is concerned but we decided it was important to get it up and running so we can set the code distribution process in motion.", Minessale remarked.

JavaScript module for Asterisk Unveiled!

Anthony Minessale II, CTO and implementer of Asterlink and producer of the upcoming ClueCon Open Source PBX Conference ( has announced the immediate availability of res_js, a JavaScript module for the Asterisk Open Source PBX. The module embeds a live JavaScript interpreter in the Asterisk server and makes it possible to execute .js files as IVR scripts the module comes with a large example script that showcases all of the Asterisk-specific usage. One of the most robust features allows you to white or black list application names, variable names and function calls in a special security mode that makes it possible to extend permission to the end user to manage his or her own IVR scripting and prevent abuse.

Friday, June 17, 2005

Nicolas Gudino: Flash Operator Panel

Nicolas has completed his presentation of the Flash Operator Panel. He discussed the following topics:

What is the FOP? It displays information on your Asterisk PBX activity in real-time
You can also perform actions against Asterisk. You can monitor multiple Asterisk servers at the same time. You can have background images etc.

Monitoring capabilities

Extension status (busy, ringing, available)
See who is talking, to whom and for how long (CLID, Dialed No, Timer)
Meetme room status (number of participants)
Queue Status (users waiting) and statistics
Message waiting indicator and count
Parked Channels
Logged in agents (changing led colour or renaming the button label)
It can monitor almost every channel type; SIP, IAX2, ZAP, MGCP, CAPI, MODEM/I4L, H323, OH323, VPB, etc.

Available actions

Hangup a call by double clicking on the oval Led
Transfer a call via drag & drop
Originate a call
Barge in on a call
Set the callerid before transferring a call
Set the absolute timeout
Mute/Unmute meetme participants

Integrating with Web Based apps

You can embed the FOP into any webpage
FOP can fire screen popups on state ringing channels or directly from your dialplan
FOP can be use to add click to dial
FOP provides a link between Asterisk and Web

FOP Architechture

Client - Proxy server model

Smart perl server and proxy that connects to multiple asterisk servers
Dumb flash client
There is a windows client in the works

Why client/proxy server model?

To not burden the PBX with a lot of connections
Easier to add features and do prototyping without patching asterisk
Flash is slow, so work is done in perl
Proxy can run on another machine

FOP Future

Implement user authentication
Let the FOP send scriptable commands to the Asterisk Manager
Improve queues and agent features for call centers
JOP (a java client maybe)

Thursday, June 16, 2005

Anthm nears 200 points!

Anthony Minsssale, CTO and Implementer of Asterlink, producer of ClueCon and notable Asterisk developer is set to cross the 200 karma point boundary on the Asterisk Bug Tracker this week. Earlier last year Minessale was the first to break the 100 karma point barrier by a sizeable margin but this time fell 3 points shy. When asked, Anthony commented: .I was kind of disappointed just in a fun kind of way. had enough pending issues open on the tracker to get there for a few weeks now but I.m sure I.ll make it soon since I have a bug open that fixes a fatal error in the CDR code =D.

A list of all of Anthony's creations for Asterisk are available at:

Good Work Anthm!

*Golf Clap*

Georges Karam: Asterisk Call Center Strategies

Georges Karam is discussing contact centres using Asterisk:

Why an IP Call Center:

Dramatically lower call center operating costs
Wider Deployment of CTI Applications
Improved agent productivity
Phone email and web on the same platform
Single point of administration
Contact routing to any agent anywhere
Lower cost of expansion

Asterisk call center solutions:

has caused other companies to reevaluate their strategies. Asterisk has become the catalyst for small and mid sized call centers. Telephony in a call centre is not enigmatic anymore.

Advantages of using an Asterisk based call centre solution:

Open Source == free
Lower cost for call center
Lower cost for hardware
Facilitate the implementation of a virtual call center
Large community contributing fixes and features

Aheeva CCS (an Asterisk based contact center solution):

Integrated solution
Fast deployment
Scalable and Distributed
Centalized management
Full audio and video recording

One of their customers saved more than $500,000 in set up costs for a 400 seat call center.


Call centers can benefit extensively from an Asterisk Based Solution through lower cost, increased agent productivity.

Paul Mahler: Asterisk Scalability

Paul Mahler from Signate is now discussing the scaling of Asterisk to thousands of calls:

The main topic will be performance and how you measure it, as well as reliability and scaling up to and including the hardware and configuration.

Asterisk can be grown to any size you like. There is no limit to how large you can make an Asterisk system.

Use faster servers
Use more servers
Use external devices

They used SIPP to do the load testing. Asterisk has now outgrown this test suite, and they are looking at using Asterisk to load test Asterisk in the future. You can find information on the benchmarking suite on their website.

The Dell 750 test machine was a 2.8Ghz P4, RHEL 3, 2GB RAM, SATA, 800Mhz front side bus and a SATA drive.

The Signate Telephony Server 5000 uses 1.4Ghz Itanium, 2GB RAM, NUMA Bus, SATA drive, RHEL3.

They did two benchmarks, taking voicemails and simultaeneous SIP calls.

The source for the benchmark is in XML format.

They used the the milliwatt test to provide audio on bridged SIP calls.

They received the following results:

47 Simultaneous VoiceMail messages
333 Simultaneous SIP Calls
122 Pass through calls
Slightly less than 47% CPU Utilisation

Their Telephony Server 5000 is not an intel based PC, uses 64 bit Asterisk.

Single 2U Chassis
1 or 2 CPUs
NUMA Bus 6.4Gbps
4 PCI/PCI-X slots (3 available)
SATA or SCSI drives
2 to 24 GB Memory
Gigabit ethernet

They were able to get 4997 calls with the series 1 or passthrough ~2500 calls. At this stage, SIPP fell appart. This is the reason they would like to develop a new benchmarking suite based on Asterisk.

They think that with these figures Asterisk will outshine any other system.

The other option is to cluster your Asterisk boxes (I.E. Asterisk server farm).

Using external devices such as gateways and routers so that the load is passed off from Asterisk, allowing it to only do the call switching.

Reliability =< use more servers, fewer boards per server, external gateways, routers.

Use standard Asterisk and Linux.

More Astricon Action: Ed Guy: The fwdOUT Architecture

fwdOUT is a system in which people are able to share their phone lines with each other in order to create a worldwide network.

The system is based on Asterisk, IAX and Apache with AGI's. The system works by transferring credits based on how many people use your phone line.

They also have a system in place to give credits to people who don't have enough calls going through them (I.E. people who live in places that nobody wants to call).

They have around 2000 members.

Wednesday, June 15, 2005

More ClueCon More Fun

ClueCon - is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. ClueCon will offer a great deal of information covering all aspects of Voice Over IP development from core programming to logistical implementation to proper software and hardware selection. ClueCon has invited many of the leaders in the Voice over IP industry as sponsors and attendees and it will prove to be an enriching and valuable resource to any Voice Over IP related business. We also have a board room reserved where crucial decisions will be made that will determine the very near future of Internet Telephony so reserve a spot now.

Asterisk Community Encouraged To Attend

Many members of the Asterisk community have already arranged to come to ClueCon as either a speaker or an attendee. We have verbal confirmation from the author of Asterisk himself, Mark Spencer. Also a few members of the Digium staff are expected to attend. The Asterisk community has a great deal to benifit from attending as many of the core contributors to this conference have made a great deal of changes to the Asterisk codebase including:

Anthony Minessale II (anthm)

Brian K West (bkw)

Josh Colp (file)

Second Half of Kevin's Presentation

-Extreme code review has begun

There are now various people assigned to parts such as commenting, clean code etc

Coding Guidelines:
Can be found in /usr/src/asterisk/doc/CODING-GUIDELINES

If you are changing a module and see that some things are not complying with the guidelines, please submit a patch which ONLY contains the fixes for guidelines first

-Code is generally formatted in same way as the Linux Kernel

-Do not add comments that do not provide direct value (I.E. look at this cool thing I did!)

-Do add comments for things that are not obvious

-Code should have functions that are as short as possible (app_voicemail3 could be written if you like)

-Make liberal use of the operators, statements and block types that C offers

-Function Modules can be named anything you like except they cannot start with 'ast_'

-Functions that are intended to become part of the API should be started with 'ast_' and the prototype MUST be added to the relevant header file in include/asterisk

-Global variables in modules should be named with the 'global_' prefix

-Others should be descriptively named

-Variables exposed outside the module (which should be kept to a minimum) must be named with the 'ast_' prefix

-Remember that Asterisk works on at least 4 operating systems and multiple CPU architectures. Changes should not be platform or CPU specific

-Do not use obscure language, library of system features just for the sake of using them. If it increases performance or maintainability benefit can be demonstrated.

-Document your code so that it can be used with doxygen

-Modules that use a new config file should include a sample config file

-Use IRC, Mailing Lists (Asterisk-Dev, Asterisk-CVS) and the Bug Tracker

-The future may include: Changes to The Asterisk Object Model, Asychronous Event System (for internal modules, in a similar way to the Asterisk Manager Interface works)

-Things in the source will be changing rapidly.

-ALWAYS check #asterisk-bugs (on first before submitting a bug to be sure that the bug report is all correctly filled out.

-People should watch the bugtracker to see when they can help with testing. Also, please post your comments. This will really increase the speed with which fixes are applied

-IAXTel is back up (and has around 10,000 users registered, with 1000-1500 at any given time. It is an average PC with a single processor

Support Zoa's work for increasing the capacity of Asterisk

Kevin Fleming: So, you want to be an Asterisk Developer?

Kevin Fleming discussed the task of becoming an Asterisk Developer:
Discussed Libraries Required


-Easier to debug post-mortem

-Threading models differ by platform

-use make dont-optimize

-check makefiles for debugging defines

-ensure that 'debug' is listed for the console

-always use the -g parameter

-use 'thread apply all bt full' in gdb

-use 'ast_grab_core' (helpful for solving deadlocks)
-always run 'make update'

-use 'cvs diff -u' (or add 'diff -u' to your .cvsrc file)

-read the patch before submitting

-Watch for new/removed files

keeping out of tree patches up to date can be hard work

-Asterisk, Zaptel and LibPRI are dual licenced (GPL and to Digium for commercial distribution)

-Contributions must be disclaimed

-Code can not violate patents etc

-If you make changes you are not required to distribute them

-You are not required to submit changes to Asterisk, although it is recommended

-If your code is committed it will be maintained by Asterisk crew

-The code you have is GPL forever

-The word Asterisk is trademarked

-Always disclaim your code

-You do not lose copyright on the code when you disclaim

-Prototypes are in module.h

-Use Use Count Tracking - needed for dynamically loadable modules

-usecount function is called by the loader

-Use the macros to implement basic usecount management

-Module has to tell Asterisk what they can do

-Application Modules are supposed to work on a channel that they are provided on

-Applications MUST keep use counts

-Channel modules are probably the most complicated part of Asterisk (allow connection to the outside world). Contains usecount again. Provide approx 25 methods that channels may provide

-Codec Modules are the means of converting from one format of audio to another. Quite simple. Many codecs keep internal state from one frame to another.

Format Modules - provide means to read and write audio formats in all of the formats Asterisk supports

Function Modules - look like variables. Currently only in CVS HEAD. They provide dialplan functions. Most of these are built into

CDR Modules - these provide CDR backends to post CDRs into storage systems (DB/txt etc)

Second Half of ManxPowers presentation

We return now after the break.

-Contexts are extemely powerful - they give security, seperation on priveleges. You may for example want one particular phone to not be allowed to make toll calls etc. You would put this in say a "notrust" context and then only include local outbound dialing.

-Macros: very powerful and can be quite simple. For example most of the users want to be able to forward/voicemail etc. Instead of coding this for each extension, you can create a macro and then each phone will only needs a one line entry in the extensions.conf file

-DIALSTATUS: possibly one of the most useful variables in Asterisk.

-/usr/src/asterisk/doc/README.variables contains a list of all of the available variables for use with Asterisk (built in)

-Let carrier do the billing via account codes etc.

-Route calls via Asterisk machines if the dialed number is local to one of your Asterisk machines.

-NoOp can be used to print out variables on the console. For example:

exten => s,1,NoOP(${CALLERID})

-Use a PRI if you can afford it (E1/T1). You get full information from PRI's on the status of dialed calls. At least get a quote on one even if you only have 8 channels. The small additional cost is more than made up for by the extra information provided

-You could probably put two quad PRI cards into a dual CPU box as long as your calls are not too short. This is because call setup is resource intensive.

-Transcoding is also resource intensive.

-Scalability is very much based on what you are using the system for.

-Phone == type=friend. Anything else type=peer and type=user seperately. Many providers expect different details for incoming vs outgoing calls. Some people say never use friends. However a friend is exactly the same as an identical peer and user.

-Use IAX2 native transfers when possible. WARNING: this will make the cdr incorrect on the machine that jumps out of the loop.

-SIP and NAT: Use a static IP for Asterisk.

-Assuming Asterisk is not behind a NAT, and all of the SIP devices are behind NAT. all you should need to do is add nat=yes. Qualify=yes (or a value) so that the connection is refreshed.

-If you don't have qualify=yes, you may find that the phone can only receive calls just after reboot.

-If your * server is on public IP you shouldn't need externip=x or localnet=x

-Say for example you have double NAT, you will need to deal with externip, localnet.

Astricon: Eric Wieling (Manxpower)

Eric is giving a lot of information on how to set up your Asterisk System.

He's speaking currently about ideas to reduce maintenance and diagnose issues.

-don't use multiple types of phones as this will make rollout harder.

-use lspci to view pci stats

-use patlooptest for testing T1 interfaces to see if there is data corruption etc. (use make tests to build these tests

-use setpci to change latencies around between various cards in the system

-ztmonitor is a very usefull tool for monitoring audio levels (helpful in hunting down echo and fax problems - use it when changing volumes with RX/TXgain

-zttool - useful to give you information on channel usage/alarms etc (need newt devel libs on system)

-ztcfg - automatically runs when modprobed - if changing /etc/zaptel.conf you will need to rerun ztcfg. WARNING: Will drop all current ZAP calls.

-Don't share interrupts

-On most motherboards the IRQ is assigned on a per slot basis. Sometimes it will assign the same IRQ to multiple slots.

-cat /proc/interrupts to view which interrupts are used by which card

-If your kernel has been compiled to include APIC then every device will get it's own interrupt. If your board supports this then you _should_ definately use it. Sometimes it's the only way to not share interrupts

-use hdparm to cause the HD to hold the interrupts for less time

-IP/UDP/RTP overhead is 40 bytes/packet

-20ms Packets are common (50 packets per second)

-RTP Header compression can help with this - currently unsupported in Asterisk - not very common, but available

-Increase audio per packet - not currently supported easily in Asterisk currently - ILBC uses 30ms packets - i.e. less packets so less overhead

-if bandwidth is not a problem (LAN) use G711 (ulaw/alaw)

-speex/g726 for IAX2 between servers (speex uses more CPU, g726 uses more bandwidth)

-Use GSM/G729/G726 for SIP

-Only enable codecs you would like to use. I.E. disallow=all, allow=the one you would like to use.

-Don't use allow=all as this will also enable codec which Asterisk can only use in passthrough mode.

-Use context=INVALID in the general section so that calls cannot come in without user/pass. Obviously without having an [INVALID] context. That way if a call comes in and is not authenticated will fail

-make sure you don't have overlapping number space (I.E. don't overlap number)

-use an outside line code (i.e. 9 or 0)
-you need to use an underscore to say that you are using regex
We are now having a break, thought I'd upload it so you can read it as it happens.

Monday, June 13, 2005

Asterisk + Google summer of Code

The site has a note on Asterisk's participation in the Google Summer of Code


The Asterisk Project is proud to participate in the Google Summer of Code. Google is giving a group of lucky students the chance to work on the Asterisk project and other various Open Source projects for the summer and get paid for it too! If you have ever wanted a shot at developing software for the Asterisk Project, this is your chance. We are always looking for new talent to be discovered. Heck, it beats flipping burgers!

Below is a list of requirements and possible project submissions related to the Asterisk Project. Students are welcome to suggest their own project submission but are encouraged to select a project from the possible project submissions below.

Asterisk Community Meeting in Sydney Australia

Shamsul Arefin has posted details of a Asterisk Users meeting in Sydney, Australia:

All those asterisk users in Sydney Australia, we are going to organize frist asterisk users meeting on 7th July around 6 PM in Lidcombe area. I will advise the exect address of meeting, once i have some feeling how many users are coming. In this meeting we will also launch Australia's first "Asterisk Community Telecom" for Australian users, where we all can hook up and stay in touch with each other. So I request all of those from Sydney who like to join us please send me an email on saktek @ asap.


Best Regards
Shamsul Arefin
Iris telecommunications P/L

Saturday, June 11, 2005

AstriCon: The Asterisk Conference... now in Europe!

Digium has posted info of a discount for Astricon:

Time is Running Out!
Astricon Europe... less than ONE week away!
Save $50 by registering in advance today!
Madrid, Spain

Auditorium Hotel (

info @
The AstriCon conference is dedicated to all things Asterisk!

This is the first Asterisk conference in Europe. The event is a combination of tutorials, a trade show, a user conference, and a developer summit. It promises to be an exciting and rewarding event for the Asterisk community. Register today to make sure you'll be part of this event!

AstriCon Europe will take place at the Auditorium Hotel, Madrid - the largest hotel in Europe! Our registration includes hotel registration, taken care of by our partner KonferensBolaget in Sweden.

Topics covered at AstriCon:
Integrating the PBX with the IT infrastructure: Asterisk for the Enterprise

VOIP migration in-a-box: Asterisk for Service Providers

Lower cost, more flexibility: Asterisk for Call Centers

Your VoIP Swiss Army Knife: Asterisk for developers

Managing your Asterisk PBX: from the CLI to the GUI
You may register right away at

For more information, contact Steve Sokol at:
info @

Thursday, June 09, 2005

Digium and Cepstral Announce Text-To-Speech Partnership for Linux Telephony

PITTSBURGH, PA and HUNTSVILLE, AL -- (MARKET WIRE) -- 06/06/2005 -- Digium and Cepstral announced a partnership agreement today to provide Cepstral Swift Text-To-Speech (TTS) languages and voices for Digium's Linux telephony platforms and applications. Under the arrangement, Cepstral becomes a Digium Premier Partner for providing commercial Text-To-Speech.

The offering will include Cepstral's telephony voices, David, Diane and William for U.S. English, as well as options for U.K. English, German, Canadian French, American Spanish, and Italian, for use in multi-channel applications on the Linux operating system. Each will be available from Digium, and incorporated into key products and services.

"Cepstral voices allow our customers to introduce high quality Text-To-Speech into telephone and IP telephony applications that use dynamic content. These voices sound great and provide the flexibility to deliver information over the phone, or on the network," said Mark Spencer, CEO of Digium and primary author of the Asterisk PBX.

"Digium is a proven pioneer who's development of the Asterisk PBX has created a powerful and practical platform for computer-telephony integration," said Kevin Lenzo, CEO and co-founder of Cepstral, LLC. "We are very happy to partner with the company that is spearheading the Linux telephony movement. Our new server-based voices fit the offering perfectly and bring extraordinary vocal quality at an extremely competitive price."

Digium's products and services featuring the Cepstral Swift TTS voices will be available in June of 2005.

About Cepstral, LLC

Cepstral is a speech technology company based in Pittsburgh, PA, USA, which provides speech technologies and services for the spoken delivery of information. Cepstral builds high quality, natural sounding voices for server, desktop, and hand-held applications. Cepstral also provides professional services to customize and tune voices. Cepstral: We Build Voices.

About Digium

Digium is the creator and primary developer of Asterisk, the industry's first Open Source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, and Ethernet architectures. Digium solutions reduce the costs of traditional implementations of PBX, IVR, media gateway, and communications servers through open source, standards-based software and innovative hardware solutions. Digium also offers technical support and development services for Asterisk, and commercial versions such as Asterisk Business Edition, a professional-grade distribution of the acclaimed open-source software.

Wednesday, June 08, 2005

Release: Version 1.5 of SIP-based insta-REACT!T

PLEASANTVILLE, NEW YORK, USA - June 8, 2005 -- Pangean Technologies, specialists in IP collaboration and enterprise communications technologies, today announced the release of Version 1.5 of its SIP-based instant communications products, insta-REACT!T and insta-RELAY!T

Pangean's flagship product, insta-REACT!, is the first of its kind that integrates Presence, Instant Message, Multi-Party Audio Conference, Push to Talk, and recording on a single interface, achieving instant communications enterprise-wide. The product is similar to instant messaging, however, instead of text, it uses voice as the means of communication and allows users to instantly and easily communicate with each other no matter where they are.

With the 1.5 release of Pangean products, companies will experience
increased communications with these new features:

-New and Improved User Interface
-Scalability enhancements
-Easier administration
-Improved Presence
-Integration with Asterisk and Ser IP-PBX's
-SIP-based Interoperability Improvements
-Microsoft Active Directory Integration and LDAP Support
-Voice communications over Satellite
-NASD/Sarbanes-Oxley compliant

The new version of insta-RELAY! also fully integrates with Asterisk for instant conferencing, IP intercom, and IP broadcast. Until now, the product was used only with Pangean's SIP Management Server, but with the integration of Asterisk and Ser open-source IP PBXs, the possibilities of other interoperability scenarios are endless.

"The enhancements to our two products constitute our continual commitment to bringing SIP-based group communication services that are scaleable, user-friendly, and cost-effective to the enterprise market, " said Taa Wongbe, CTO of Pangean Technologies, "and with our integration with open source IP PBXs such as Ser and Asterisk, we continue to take SIP-based interoperability and convergence to new heights."

Pangean Technologies also plans to release a wireless version of its converged client in the fall of 2005 that will run on Pocket-PC devices and integrate Presence, Wifi roaming and location based service. The wireless version will have the same capabilities as the PC version and will target the ever growing mobile workforce.

About Pangean Technologies

Pangean Technologies develops and markets innovative, SIP-based VoIP software applications for internal and interoffice communications. These cost-effective, open-system solutions leverage an organization's existing IP network by delivering a converged communications service to each user's desktop.

To find out more about Pangean Technologies and its VoIP communications solutions, visit or call 1-877-472-6432.

Thank you Taa for the update

LightReading: Digium Echo Cancellation Cards and Cepstral Link

LightReading has posted a press release from Digium about their new Echo Cancellation cards and their new integration with Cepstral Text To Speech.


HUNTSVILLE, Ala. and CHICAGO -- Digium Inc., the creator of open source telephony today announced the availability of two new echo cancellation cards, the TE406P and the TE411P for Asterisk, the industry's first and most widely used open source PBX. Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, and Ethernet architectures.

The TE406P (for use with a 5.0-volt PCI slot) and the TE411P (for use with a 3.3-volt PCI slot) support both E1 and T1 environments and are selectable on a per-card or per-port basis. The TE406P and the TE411P will support ultimate density and performance with echo cancellation for four full T1 (96 channels) or E1 (124 channels) and improves voice quality in situations where software echo cancellation is not sufficient, is not done at the CO, or where CPU utilization must be minimized. The cards are designed to perform in the most difficult of environments while providing capacity/length trade-off by supporting 16ms of echo cancellation over 128 channels, 32ms over 64 channels, or 64ms over 32 channels.

Tuesday, June 07, 2005

ClueCon Update: Mark Spenser, Author of Asterisk, to speak

ClueCon has just confirmed a time slot for Mark Spenser, Author of Asterisk, to speak about his opensource PBX project. Asterisk is probably the most popular Software PBX to date and has many features and configuration options.

ClueCon Update: David Sugar, Core developer of Bayonne, to speak

ClueCon is pleased to announce that David Sugar has decided to speak at ClueCon Aug 4th, 2005. When asked to attend, Mr Sugar replied: "I would be happy to come and speak at this event. Actually it also fits in well with the schedule for introducing a second generation (bayonne2) server."

For More Information: Bayonne2

Asterisk business sightings

Colin Anderson has posted info on a sighting of Asterisk at an Apple store and a call for others to post the same.

So I go into a new Apple store on Sat to buy some stuff for my Mini, and I notice some Snom 360's on the sales counter. Venturing a question, I ask, are they using Asterisk? Guys says yes. Cool! I said: What kind of box are you using. He points to a Mini sitting on the counter! 2 X cool! He's using a SIP-FXO converter.

Plug: great store in Alberta.

Anyone else ran across Asterisk in a setting that they would not have expected?

Iax Phone Pro - Update Released

Steve Sokol has posted a new version of IAX Phone Pro Beta that works outside of USA:

Thanks to everybody who downloaded a copy of the phone over the weekend. Several people discovered a bug in the system related to date formatting. That has (I believe) been fixed. If you would like to try the fix, please download it from:

Be sure to un-install the previous version before installing the new version.



Steve Sokol
Sokol & Associates/AstrCon

Monday, June 06, 2005

SIP changes in CVS head

Olle has posted details of some changes in CVS HEAD (and possibly one in STABLE):
Asterisk can fail an outbound registration

If you enter a register= statement with an incorrect password, wrong hostname or anything else that is wrong, Asterisk will give up registration after 10 attempts. If you do "sip show registry" you will see the registration status as "failed". Doing a "sip reload" will restart the attempts after you fixed the problem.

In earlier versions, Asterisk just kept sending packets endlessly.

Authentication changes Asterisk will now check that the authorisation user name (digest username=) in an incoming authentication is the same as the username part of the From: SIP uri. We've always based peer/user matching on the username part and now we will send an error if the authentication username is different from the From: username.

We have also fixed a problem that is occuring with Sipura devices, where the Sipura sends a proper authentication based on an old nonce.

Coming changes:
Changes that are in the bug tracker and hopefully will be implemented soon:
SIP domain support: As an option, you can configure your Asterisk to be SIP domain aware. If Asterisk gets messages directed to domains that are not configured as local domains, the message will be rejected. Very much like a mail server that doesn't forward or handle mail to other domains than local domains.

This feature also makes it possible to implement SIP transfers in a correct way, since Asterisk is able to judge whether a transfer is to a local or remote extension.

Support for supported/required headers: If a SIP service requires support of a SIP extension by using the Require: header, Asterisk doesn't understand that today. We should fail the transaction if we do not support the extension (which is simple, since Asterisk does not support any extensions at all today). This will also help implementing support for the Replaces: extension in SIP transfers.

Support of SIP call timers: This is something I just started exploring, but we need to implement. There are a lot of reports open in the bug tracker where we do not handle retransmissions properly. In some cases, this is because we retransmit too often according to the SIP standard, so we're getting totally out of sync with other devices.
The SIPURA bug fix will be implemented in the release ("stable") version of Asterisk if Russell approves, but the others are new functions that will only be implemented in the development version.

Meet me in Madrid to discuss chan_sip :-)


[Announcement] Asterisk::LDAP initial release

Ben Klang has posted details of a perl module for creating Asterisk configs from LDAP:

Fellow developers,

My organization uses LDAP extensively to manage our customer information and we were looking for a way to store phone system information with the rest of our data. Unfortunately the solutions we found were not adequate for our needs. To fill that void, I humbly submit Asterisk::LDAP.

Asterisk::LDAP is a Perl module I have written and released under the terms of the GPL which is capable of generating Asterisk version 1.0 compatible configuration files from an LDAP data source. Currently it supports generation of extensions.conf, voicemail.conf and musiconhold.conf. Future plans include sip.conf, iax.conf and meetme.conf. As this is the first release it is sure to include lots bugs so I disclaim any kind of responsiblity for anything including remembering why I wrote any particular piece of code any particular way. That being said it has been tested in my environment and I think it works well. Example scripts are in the tarball, including one which can be dropped in voicemail.conf's 'externpass' argument to update a user's voice mailbox PIN when changed via the telephone interface.

There is much work left to do, not the least of which is documentation. I have made an attempt to document the methods via perldoc ('perldoc Asterisk::LDAP' after installation) and through the example code. I welcome any suggestions or contributions.

I also want to say that I KNOW this is a big ugly hack. The Right Way to do it will be Real Time Asterisk. Unfortunately I needed a solution yesterday and this fit the bill.

If you think you'd like to take a look for yourself, hop on over to

Enjoy! Comments and criticisms to Complaints to /dev/null.


IAXtel update

Russell has posted information about IAXtel coming back. They've managed to get the server to drop from almost 100% CPU to 4% CPU:

Hello Everyone!

Over this weekend, we have updated IAXtel. Before the update, it was running at almost 100% cpu load at an idle state because of the massive amount of database transactions.

We enabled realtime caching and the box immediately crashed. We were able to expose a serious bug related to realtime caching in chan_iax2. Kevin Fleming was able to fix this issue, and also added some experimental code to further enhance performace.

As I write this message, Asterisk is using about 4 percent CPU load on IAXtel. We are hoping that it will become usable again.

"May all of your calls have full-duplex audio!" -- Mark Spencer


New version of Asterisk VConfig

Chris has posted details of the latest version of VConfig:

I repackaged everything into one distribution and cleaned up the installation. Should be a lot simpler to install now.

Asterisk VConfig is a platform for virtual hosting of end users on a single instance of Asterisk using the realtime database structure. Right now the functionality of the web interface is limited to a direct configuration interface. As soon as Realtime SIP/IAX is done I will get to work on adding more features into the web interface.


Sunday, June 05, 2005

Asterisk/ZeroConf at Apple WWDC


Stuart Cheshire (, Apple's Bonjour/ Zeroconf czar has expressed interest in making Asterisk+Zeroconf part of his on stage presentation on Bonjour this coming week at Apple's World Wide Developer Conference aka WWDC.

According to the official schedule at Apple's website the Bonjour presentation is on Tuesday. As this is all pretty short notice, it will be difficult to get an IP phone to Stuart in time for his presentation. Considering that this is a great opportunity to spread the word about Asterisk on the Mac to the wider Mac developer community we don't want to give up just yet.

So, we would like to know from people in the San Francisco Bay area or going there on Monday to attend WWDC and who could lend an IP phone to Stuart for the presentation on Tuesday.

As for the presentation itself, the plan is this ...

Simon Taylor has written a small sample application intended to show Mac developers how to incorporate Zeroconf into their apps and make use of the services Asterisk is advertising through our Zeroconf module. The application is a very simple desktop dialer that uses the remote dial service advertised by Asterisk.

Using this application, it is possible to make Asterisk (running on another Mac) dial a number looked up from the OSX Addressbook and then have that call be connected to any IP phone connected to the Asterisk server. Since the dialer is Zeroconf aware, it can discover the Asterisk server's remote dial service and no configuration is required, other than choosing an Asterisk server from a pop up menu. This would be interesting enough to show given that there is no IP phone with firmware that supports Zeroconf as yet.

Also, Alex Karahalios has added a new service "astcli" to SitePlayer ( that is advertised when Asterisk is running on the system SitePlayer is hooked up to. Connecting to the service will automatically take the user to the Asterisk CLI. Since SitePlayer connects to a serial port on an Xserve, this means an Asterisk administrator can still manage Asterisk even if the network is down. Of course this can already be done using a telnet or ssh service but it makes the life of an administrator a little easier this way.

Anyway, if anybody can help us out with this, please get in touch, thanks.

Astmasters Zeroconf Project

astGUIclient version released 1.1.1

Click Here:
astGUIclient version released 1.1.1 - has moved - Click Here

IAX Phone Pro - Open Beta Test

Click Here:
IAX Phone Pro - Open Beta Test - has moved - Click Here

FCC Puts VoIP E911 Laws Online

Click Here:
FCC Puts VoIP E911 Laws Online - has moved - Click Here

Friday, June 03, 2005

Article: VoiceBlue VoIP GSM Gateway with Asterisk IP PBX How to

Click Here:
VoiceBlue VoIP GSM Gateway with Asterisk IP PBX How to- has moved - Click here

ClueCon - Open Source Telephony Expo and Developer's Conference

ClueCon - is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. ClueCon will offer a great deal of information covering all aspects of Voice Over IP development from core programming to logistical implementation to proper software and hardware selection.

More Info

Thursday, June 02, 2005

Dev: Various chan_skinny enhancements

Simon Lockhart has posted details of some enhancements to chan_skinny:

I posted a patch to the bugtracker a few days ago with a bunch of enhancements to chan_skinny. Would appreciate it if people could test and provide feedback, especially if you currently use chan_skinny.

I would also appreciate it if anyone could volunteer to translate the softkey labels in chan_skinny.c to other languages - so I can add non-english support.


Users: 1.0.8 Release Candidate

Click Here:
1.0.8 Release Candidate- has moved - Click Here

Wednesday, June 01, 2005

AsterTest - The Asterisk Performance Project

Click Here:
AsterTest - The Asterisk Performance Project - has moved - Click here

The 2nd Asterisk conference (AstriCon) in Europe!

Astricon Europe

Welcome to AstriCon Europe, the first European conference dedicated to the Asterisk Open Source PBX.

The Astricon Conference conference is dedicated to all things Asterisk. The event, composed of tutorials, a trade show, a user conference, and a developer summit, promises to be an exciting and rewarding event for the Asterisk community. Register today to make sure you'll be part of this event!

Astricon Europe will take place at the Auditorium Hotel, Madrid - the largest hotel in Europe! Our registration includes hotel registration, taken care of by our partner Konferensbolaget in Sweden.

Additional Information:


Conference Agenda

Hotel Information

Asterisk Management Portal Updated

Click Here:
Asterisk Management Portal Updated - has moved - Click here

Asterisk pbx on wrt54g Wireless Access Point

Click Here:
Asterisk pbx on wrt54g Wireless Access Point - has moved - Click here

Canadian firm scores with open source call center

Click Here:
Canadian firm scores with open source call center - has moved - Click here

Please support the Zeroconf project - Spread the word

Click Here:
Please support the Zeroconf project - Spread the word - has moved - Click here

Asterisk@Home 1.1b1 has been released

Click Here:
Asterisk@Home 1.1b has been released - has moved - Click here