Xorcom TS-1 solid state Asterisk server released

Click Here for Xorcom TS-1 solid state Asterisk server released

OpenSER v1.0.1 released

OpenSER is a project spawned from FhG FOKUS SIP Express Router (SER). The reason for this new venture is the lack of progressing and contributions to the SER project from the other SER team members as well as the reticience to new contributions from project's community members. We want to accelerate the integration of public contributions to the SER project.

OpenSER promotes a new management policy (OPEN) -for new code acceptance and code-through propagation- and development approach -design and architecture. We have decided to bring more dynamics into SIP world by creating this new project that can benefit of TLS and so many other contributions. We welcome your contributions to the success of this project.

Click Here for more Information

Turbolinux Launches IP-PBX Software to Support Broadband IP Phone Services

Turbolinux, Inc., a global leader of Linux-based solutions, today announced the sales launch of IP-PBX software InfiniTalk, which is based on the open source software Asterisk. InfiniTalk improves and upgrades the IP telephone environment for significant cost reductions.

InfiniTalk IP-PBX software is generally considered the best choice for a low-cost, next generation standard IP telephone system. The combination of the Linux and Asterisk open architecture and rich hardware allows customization of the software for specific applications and customers. InfiniTalk software supports the majority of standard telephony equipment. InifiniTalk also supports the newer broadband IP phone services and fiber optic technology provided by NTT East and West Corporation. These capabilities work to create a cost-effective, IP phone system in the business environment.

Click Here for the Full Article

AstriDevCon Europe 2006

From May 8-11 in Pisa, Italy, a group of Asterisk developers will be getting together for four days of hacking, coding, testing, designing and otherwise beating on the Asterisk code base. The event will be hosted at the University of Pisa (thanks to Luigi Rizzo), and will be low-key and open only to serious Asterisk developers and contributors. We are expecting to keep the attendance to 15 people or less, and we already expect to see these people:

Attendees:

Mark Spencer (Digium)
Kevin P. Fleming (Digium)
Luigi Rizzo (University of Pisa)
Christian Stredicke (Snom)
Christian Richter (BeroNet)
Olle Johansson (Edvina.net)
Joachim Vanheuverzwijn (zoa, Securax)

If you wish to participate, please contact me off-list so I can make arrangements with you. We will need to have the final list of attendees in place by March 15th or so, so that hotel accommodations and conference room space can be scheduled. Luigi has suggested that we stay at the Hotel Roma which is within walking distance of the University, the main square and the Tower, so that seems the best choice.

Note that this is _not_ a sponsored event (other than the university providing workspace and network access)... each attendee will be responsible for their own travel, lodging and meals.

The timing of this event means that Asterisk 1.4 will have already entered feature freeze mode and will be in beta-testing, so the goal of the conference is to concentrate on architectural changes and other work targeted for Asterisk 1.6 (to be released around the end of 2006). If it goes well and is productive, we may arrange a similar event in the USA for later this year so that those who cannot attend in Europe can participate.

Posted By: Kevin P. Fleming

Dev: Test my test-branch!

The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback.

For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is written by me. There are large contributions from other developers, code that I maintain in several open subversion branches in order to help them stay up to date with their work.

To assist the testing group and make life easier, I've combined a lot of patches into one superbranch for testing. I've added the README further down.

** PLEASE help the community, please test this branch.

Check it out like this:

SVN Checkout

Then cd into test-trunk and run "make" then "make install"

Report any bugs in the proper open bug in the bug tracker. If you like new functions, add a comment that this works for you. Provide feedback, make our work easier.

Run "svn update" from time to time to get the latest version. Any changes from trunk will be merged into this code. Read the README.test-this-branch file to get more information.

Thank you for your help!


P.S: Obviously, this is test code, not recommended to be closer than 2 miles from your production servers.

----- README.test-this-branch

----------
TESTING BRANCH - WELCOME!!
----------
Asterisk is developed by the Asterisk.org user community. The development team does not only consist of coders, but also testers and people that write documentation and check for security problems.

This is a combined branch of many patches and branches from the bug tracker that needs your testing. Please test and report your results in the bug tracker reports for each patch.

What's in this branch?
----------------------
This branch includes the following branches:

- sipdiversion: Additional support for the Diversion: header
- jitterbuffer: Jitterbuffer for RTP in chan_sip (#3854)
- videosupport: Improved support for video (#5427)
- peermatch: New peer matching algorithm (no bug report yet)
- rtcp: Improved support for RTCP (#2863)
- dialplan-ami-events: Report dialplan reload in manager (#5741)
- sipregister: A new registration architecture (#5834)
- subscribemwi: Support for SIP subscription of MWI notification (#6390)

Coming Here Soon:
- iptos: New IPtos support, separate audio and signalling (#6355)
- metermaids: Subscription support for parking lots (#5779)
- multiparking: Multiple parking lots (#6113)

And the following stand-alone patches
- New CLI commands for global variables (#6506)
- Additional options for the CHANNEL dialplan function

All of these exist in the bug tracker

* PEERMATCH: New object match for incoming calls. Skip the "user" :-)
---------------------------------------------------------------------
In this code, we will match incoming calls like this:

- First user on From: user name
- Then peer on From: user name *** NEW ****
- Then peer on IP and port


This means that in most configurations, you can configure a phone entry as "type=peer" instead of "type=friend". Subscriptions will work much better with just one object to match.

/Olle

Introducing Telephone Reminders 2.5: The Asterisk Telephone Reminder System

Excerpt: Using nothing but a phone call, you can schedule reminders for the near or distant future, specify different numbers for the return calls, and customize a recorded message for each call. In short, it's perfect for appointment reminders, birthday reminders, anniversary reminders, and anything else you want or need to remember.

Click Here for the Full Nerd

Create your own Voice-over-IP PBX using Asterisk

Asterisk is already hard at work in South Africa. Its being used as a PABX, for call-recording, for both small and large call-centres, for voice conferencing, and in CTI (Computer Telephony Integration). Its providing inter-office "free" calling, and very inexpensive international calling. In every case, Asterisk-based solutions are a fraction of the cost of the traditional equivalents.

Voice-over-IP (VOIP) is built right in to Asterisk. Connecting into the traditional phone network is done using interface cards readily available in South Africa, or by connecting into an ITSP (Internet Telephony Service Provider) via VoIP over the Internet.

Asterisk provides all the functions of even the most expensive traditional PABXes simply with software running on an ordinary PC. What's more it comes with all the open-source goodness that Tectonic readers know and love.

At Connection Telecom we like to say that the coming together of VoIP and the open source world is resulting in the most dramatic change in the world of telephony since we last heard "Nommer Asseblief?" ["Number Please" in Afrikaans].

Click Here for the Full Article

Asterisk on OpenWrt

Asterisk is free software that lets you create a fully functional, easily customizable, private branch exchange (PBX). Businesses like Asterisk because they can save money by using it, and because it is open source, they can add functionality to it easily and inexpensively. Asterisk is also becoming popular with home office users -- so much so that it spawned a new project called Asterisk@Home, which released its 1.0 version last year. Now there's even a version of Asterisk that runs on OpenWrt, a Linux distribution designed to run on your wireless router. I found it to be worthwhile, but I wouldn't depend on it for my home office.

I installed Asterisk on OpenWrt White Russian RC4 on a Linksys WRT54GS wireless router. It's my first Asterisk installation. I admit that I scraped the knuckles on both hands getting Asterisk correctly configured, but now that I've done it, I would say it was worth all the frustrations it caused me. Not only do I now have a functional personal PBX, I've also learned a little about the black art of telephony along the way.

nstallation was a snap. All I had to do was point my browser at the WRT54GS's IP address, log in at the OpenWRT Admin Console, and then click the install button next to the Asterisk and Asterisk-sounds packages. The install was finished, but I still had a long way to go.

Click Here for the Full Article

Article: Open source Asterisk PBX getting more Popular

Asterisk, the open source PBX system made by Digium, is gaining ground with companies and governments alike. The cost savings over proprietary PBX systems can be substantial, but Mark Spencer, president of Digium, told TG Daily in a short interview at the Southern California Linux Expo, that "choice" is the main reason companies adopt Asterisk. "Customers have a choice in how they configure they system, in the hardware they buy or the graphical interface they want," says Spencer. This choice also allows companies to add extensions or make changes in seconds compared to days with a traditional PBX.

Click Here for the Full Article

Zaptel 1.2.4 Released

The Asterisk/Zaptel development team is pleased to announce the release of Zaptel 1.2.4.

This release contains a number of bug fixes, along some with new functionality:

* The driver for the Xorcom Astribank has been incorporated into this distribution. Xorcom will provide primary support and driver maintenance for customers using this product.

* The driver for the Digium Wildcard TDM2400P has been upgraded to support revision B of the VPM100M echo cancellation module.

* The special parameters required for the Digium Wildcard TDM400P when used on the Australian PSTN are now automatically set when the opermode is set to 'AUSTRALIA'.

The release is available on the Digium FTP Servers under the name zaptel-1.2.4.tar.gz, and also as a patch from version 1.2.3 (in file zaptel-1.2.4-patch.gz).

In addition, beginning with this release we have included an SHA-1 sum of the files (in files zaptel-1.2.4.tar.gz.sum and zaptel-1.2.4-patch.gz.sum) and GPG signatures (in files zaptel-1.2.4.tar.gz.sign and zaptel-1.2.4-patch.gz.sign) verifying that
this is an official Zaptel release.

You can retrieve the public keys for kpfleming@digium.com and russell@digium.com from the keyserver, pgp.mit.edu.

Next Montreal Asterisk Meeting - 02/21/2006 - Featuring a conference call with Mark Spencer

This is a reminder about our next meeting.

It will be held from 6pm to 8pm, February 21 at Modulis Offices which are at 360 Notre Dame ouest bureau 104, H2Y1T9, Old Montreal.

Thanks to Claude Patry, we will be having a 20 minute conference call with Mark Spencer.

If you'd like to ask Mark a question, please send it to me by email. We are limited to 5 questions, and will do our best to select those to be presented.

Please confirm your attendance at this meeting by replying to this email.

See you Next Week,


--
Adrien Laurent
adrien@modulis.ca
www.modulis.ca
(514) 284-2020 x 202

PIKA Technologies Announces Support for Asterisk PBX

PIKA Technologies today announced that they have integrated PIKA's high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software
layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability between PIKA high-density analog boards (Daytona MM) and Asterisk PBX software.

"The Asterisk development community can now benefit from advanced features for fax and echo cancellation in high density analog applications, made possible by PIKA's DSP processing power on the board," stated Wojciech Tryc, Enterprise VoIP architect at PIKA Technologies. "Because of the native bridging for TDM calls, latency is nearly eliminated in this implementation. The solution is very reliable, as we have witnessed not only in the lab, but in live customer environments."

Asterisk developers can be up and running quickly with PIKA Connect for Asterisk and PIKA hardware. "For those familiar with using the Asterisk platform, no additional training is required. They can take advantage of the PIKA solution with minimal effort or investment," said PIKA Technologies, Wojciech Tryc.

For more information on PIKA Connect for Asterisk go to:
Pika Technologies Asterisk Info

Help Article: Configuring voicemail.conf for Asterisk

Click Here for Configuring voicemail.conf for Asterisk

Firmware version 1.3.1 released for Aastra IP Phones

Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range of IP phones (480i, 480iCT, 9112i and 9133i).

The firmware and release notes (no updated admin and user guides yet) are available for download at:
http://www.aastra.com/support/enterpriseip

Contrary to what the version numbering would suggest, this is a significant update with many new features and bug fixes. See the release notes for full details, but here are some hightlights for Asterisk users:

- Context-sensitive softkeys. Softkeys can now be configured for each of the following call states: idle, incoming, outgoing and connected
- Speed dial using the BLF key
- Per-line outbound proxy
- Use the Icom key to make intercom calls
- Further XML enhancements
- Voice quality (transmit level) issues resolved
- Keypad now continues to work when a second incoming call appears

And much more.

Nerd Vittles: Introducing TeleYapper 2.5: The Free Asterisk Message Broadcasting System

Click Here for Nerd Vittles: Introducing TeleYapper 2.5: The Free Asterisk Message Broadcasting System

Video conferencing over Asterisk unveiled

Adiance have succeeded in becoming one of the world's first companies to add native video support for Asterisk, offering the most advanced video solutions - such as, video conferencing, and video broadcasting.

Released recently, this technology will be integrated into Sidance's existing range of products - including inbound/outbound VoIP Call Center Software Solution - Sidance Enterprise 2006, Voice & Video Broadcasting System, VoIP telecom gateway, Soft phone with multi-user Video Conferencing support and other Asterisk VoIP software solutions.

This technology empowers users to do with video virtually everything they can do with voice on a powerful Asterisk solution platform. Combination of open source Asterisk software and state of the art Sidance technology enables customers to build VoIP video solutions at more than 40% cost savings as compared to available solutions in the market.

Adiance's evolutionary technology will help service providers to offer on-demand video broadcasting/ conferencing services to their Asterisk users at huge savings with the elimination of proprietary video infrastructure hardware. Video conferencing right now is very costly and time consuming technology. With VoIP - Video Over IP, it is cheap and fast. With this technology, Video is sent using normal Internet bandwidth as is the case with audio in Voice over IP. It is also possible to offer Video voice mails, Video on Hold application, Video recording, Text messaging and other services.

Contact:
Siji George
14, Empire Tower
CG Road, Ahmedabad
Gujarat India 380009
Phone: 91 9879200499
http://www.adiance.com

Click Here for the Full Release

Dev Info: Revised Codecs/ Implementation

I noticed that the various files in codecs/codec_*.c contain a large amount of replicated code in the newpvt, framein, frameout callbacks, buffer definitions and so on.

Additionally there are several bugs in there, from null pointer dereferences (e.g. on malloc failures, the code does check, resets the pointer to NULL and then proceeds to use it as if it were good), to (less severe but terribly confusing) comments that have nothing to do with the code that follows (they refer to the file used as a
template).

I have committed in:


http://svn.digium.com/view/asterisk/
team/rizzo/base/

a revised implementation of the codecs interface, where most of the
common functions are moved to translators.c, so the individual
codecs can just use the generic functions in most cases.

Please read the comments in

include/asterisk/translators.h

that describe the architecture (hopefully it is clear enough; if not, ask).

The results are very interesting - codec_*.c reduced from ~5000 to ~3600 lines, and the code is very consistent now.

One area that can still be improved a lot is the generation of 'sample' frames for each codec. Right now, except one or two cases, those frames are just chunks of silence of various lengths, which is not the best input to evaluate a codec's performance (used when building the translation matrix).

I would suggest to move to a slightly different approach where the input is the same for all - a piece of slin data - and we do a first pass using the slin-to-FOOtranslator to generate a frame in format FOO, and then use these frames as input for the actual evaluation. This would remove the need for a 'sample()' callback from all codecs that can do the slin-to-FOO translation, requiring them only for
those (none at the moment) that do not support direct or indirect translation from slin.

Testing and feedback welcome.

cheers
luigi

Asterisk VoIP gets South African Accent

True Voice Communications and Connection Telecom have jointly released a South African package of over 340 free prompts for the open source Asterisk platform. The free prompts, recorded in a neutral South African English female voice, can be used to replace the usual Canadian accented prompts.

"Customers can now replace the standard North American-English pack with one more suited to their customers," says Connection Telecom CEO Rob Lith. "Research and our own experience have taught us that consumers want to deal with the familiar. This is also is a step toward legitimising South Africa as a business destination that should be taken seriously," says Lith.

"Although the previous voice of Asterisk, Allison Smith, will continue to have very dear place in all Asterisk techie hearts, we believe its time to adopt a more local flavour," says Lith.

The package also offers pre-recorded IVR prompts, which can be customised using the same voice talent as in the packages.

Click Here for the Full Article

Asterisk Native Sounds re-release

Hello everyone,

It seems that the letter "s" did not make it into the original release.

Please visit www.astlinux.org and download the latest tarball. Or, if you just want "s" in all of the available formats, just grab this:

http://mirror.astlinux.org/sounds/s.tar.bz2

Sorry!
--Kristian Kielhofner

Help Article: Configuring X-Lite for Asterisk

This tutorial is not a comprehensive review of X-Lite. The purpose of this post is to set up a soft phone for evaluation. Soft phones can save money over the purchase of SIP handsets, but I strongly encourage you to test the soft phones before making a purchase decision. Soft phone quality varies widely depending on network conditions, codecs and protocol. This tutorial will help you evaluate the feasibility of soft phones with Asterisk. Before we start with X-Lite lets set up SIP and dial plan.

1) Change directory to zaptel source directory.
example:
[matt@localhost ~]$ cd /etc/asterisk

2) I'm using nano to edit the file, but pico, vi, emacs, or any text editor will do.
example:
[matt@localhost asterisk]$ nano sip.conf

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 192.168.1.x ; x = Asterisk server IP address
allow = ulaw ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here

[9250]
type=friend
username=9250
secret=password
host=dynamic
context=from-sip
mailbox=9250
nat=no
canreinvite=no

[9251]
type=friend
username=9251
secret=password
host=dynamic
context=from-sip
mailbox=9251
nat=no
canreinvite=no

3) Okay, we have the configuration for two clients on the SIP server. Now we have too make two extensions. These extensions will not include voicemail.
example:
[matt@localhost asterisk]$ nano extensions.conf

[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]

[from-sip]

exten => 9250,1,Dial(SIP/9250,20)
exten => 9250,2,Hangup

exten => 9251,1,Dial(SIP/9251,20)
exten => 9251,2,Hangup

4) Now we need to download X-Lite and install on our Windows PC. Download the soft phone from http://www.xten.net/index.php?menu=download. Run the install. Open X-Lite click on the menu button.



Click on 'System Settings'. Then choose the 'SIP proxy' option. Click on default and continue to proceed with filling out the SIP client configuration. When you are done t should look like this:



5) Now you are ready to login and test some calls.

To Be Continued....Next Week

Note: My name is Matt Birkland, I work as a VoIP Engineer for VoiceIP Solutions an Asterisk Provider in the Seattle area. Every Week I will be submitting a one page Asterisk/VoIP tip of the week on the blog. Next week we will build on this subject by reviewing codes and at some point we'll move on to common network issues or IAX tunnels... haven't decided yet!

LinuxWorld Magazine - Asterisk: Getting Connected Part 2

Excerpt:

"
In December's column, we installed the Asterisk PBX and configured two IP phones as extensions. This month, we'll connect our PBX to the telephone network for incoming and outgoing calls and set up the Digital Receptionist to route our incoming calls.

Connecting Asterisk to the Telephone Network Asterisk supports a myriad of ways to connect to the public telephone network. Asterisk provides standard technologies such as IAX (Inter-Asterisk-eXchange), SIP (Session Initiation Protocol), and PRI (with appropriate hardware). We'll be connecting with IAX because it's simple and widely supported by VoIP providers."

Click Here for the Full Article

Interview: Sangoma CEO David Mandelstam

Here is a link to an interview with Sangoma's CEO David Mandelstam with Ronald Lewis.

Nerd Vittles: Installing Asterisk@Home on Your Windows PC for Free

Click Here for Nerd Vittles: Installing Asterisk@Home on Your Windows PC for Free

Announce: New issue tracker for handling licensing issues for Asterisk, Zaptel and related projects

In an effort to ensure that every licensing issue brought to our attention is handled fully and openly, we have created a new issue tracker for this purpose. It is located at:

http://licensing.digium.com

The tracker is open to the public, and we encourage all interested parties to post their concerns and participate in the discussions involved in resolving them. Digium's will actively respond and pursue resolution of each and every issue posted, and other concerned parties may also participate.

In the future, if you have a question or concern related to the licensing of any of these projects (including packaging, trademark and other related issues), please open an issue in the tracker rather than posting to one of the mailing lists.

Thank you for supporting Asterisk and Zaptel!

Dev: Automerge changes... finally worked out

I finally figured out the source of the problems with automerging of developer branches from the trunk (this was not an issue with developer branches based on branches/1.2).

The core issue that the 'svnmerge-integrated' property was being used for two conflicting purposes: to track the branches/1.2 fixes that had been forward-ported into the trunk and also to track the trunk changes that had been merged into the developer branch. Obviously this cannot work :-)

To solve the problem, the branches/1.2 forward-porting properties on trunk are no longer using the standard svnmerge property names; they are now called 'branch-1.2-merged' and 'branch-1.2-blocked', and there are _no_'svnmerge-integrated' or 'svnmerge-blocked' properties on the trunk. This means that forward-porting patches from branches/1.2 requires specifying the property names to svnmerge; see the
branching/merging page on asterisk.org for an example.

All new developers branches made from trunk should work with automerging without any problem, but existing branches have an issue: the developer branch contains a property 'svnmerge-integrated', but the trunk does not, and running 'svnmerge merge' will try to update that property, resulting in a conflict. To resolve this issue, for each branch that you maintain, you will need to manually use svnmerge to bring it up-to-date to at least revision 9163 (where the property was removed from the
trunk); from that point forward, automerge will be able to manage the merges for you.

In the meantime, you will probably start to receive conflict notices because your branches cannot be merged... but once we are past this issue, automerging should work well and should no longer generate 'false' conflicts like it was doing before.

Posted by: Kevin Fleming

Release: ClearOne Introduces VoIP(SIP) Tabletop Conference Phones

ClearOne Communications Inc., the developer of the industry-first and award-winning MAXAttach wired and MAXAttach wireless conference phones, today introduced the latest addition to its tabletop conferencing product line, the new MAXAttach IP and MAX IP VoIP tabletop conference phones.

These products represent ClearOne's entry into the rapidly growing VoIP telephony market space, and are based on the industry-standard SIP signaling protocol.

Up until now, users have had a limited choice for adding high-quality audio conference phones to VoIP telephony systems. With the introduction of MAXAttach IP and MAX IP, customers with SIP-based VoIP systems can now enjoy the outstanding audio clarity and superior room coverage that ClearOne's analog MAX wired and wireless products have provided for several years, at a price point well below the competition.

Click Here for the Full Release

Asterisk native sounds now available!

Hello everyone,

As I promised at eTel last week, I have finished up work on my "Asterisk Native Sounds" project. Here's a little diddy from astlinux.org:

-----------------------------------

Asterisk Native Sounds are a collection of audio prompts for Asterisk. They will improve quality, reduce CPU usage, reduce latency, and (in some cases) eliminate the need for G729 licenses! The Asterisk Native Sounds are a collection of alternative sounds prompts for Asterisk. Here's how it works. I had Allison Smith (the voice of Asterisk) re-record all of the sound prompts present in Asterisk 1.2. She provided them to me in the best audio format possible. I then converted them into several native Asterisk sound formats. Why would I do all of this?

The default audio prompts provided with Asterisk are in the GSM audio format. GSM audio is nice because it doesn't use much disk space. However, because GSM is a loss-based compression format, there is no way to recover the audio quality lost when they were converted to GSM. Also, because few commercial products (including phones) include support for GSM, you can all but guarantee that Asterisk has to transcode the
prompts when a device needs them (to access voicemail, for example).

With the Asterisk Native Sounds collection you will be using audio prompts with the same voice (Allison) as the standard prompts, saying the same thing as the the standard prompts. The only difference is that they are provided in several different audio formats (most with better quality) so that Asterisk doesn't have to transcode them to the format that is being used by the current channel.

Installation is very simple. Simply download the prompts to a directory on your Asterisk server. Any will do. Once you have downloaded theformats you desire, simple follow these steps:

cd /var/lib/asterisk/
mv sounds sounds.orig
tar -xvjf /path/to/sounds.tar.bz2
[repeat last step for other formats]

The audio prompts are available from the "Extras" category in the Downloads section of astlinux.org.

While you're thinking about how much processor time you are saving and how much happier your users will be with better sounds, why don't you send me some money? Paypal donations are accepted at paypal@krisk.org.
Thanks!

Asterisk will immediately being using your new, high quality, audio prompts. Enjoy!

---------------------------------------------------------------
Kristian Kielhofner

Cape Town to respond to disasters using Asterisk

An emergency management centre in Cape Town will soon be using open source VoIP telephony to deal with and respond to disasters in the region. Cape Town-based Voice over Internet Protocol (VoIP) gurus, Connection-Telecom, recently implemented an Asterisk solution at the Provincial Emergency Management Centre based at Tygerberg Hospital.

"Our side of it is the control centre for the disaster management exercise, where the decision-makers and their support people are deciding what to do about a situation," says Steve Davies, co-founder and "chief stuff officer" at Connection-Telecom.

"The big driver for them was the necessity to keep a recording of everything that has happened," says Davies. "On traditional systems, call recording is an expensive add-on. In Asterisk, we're able to do it on the platform itself as we go along."

"They are also expecting to be using a lot of conference calling, which is also built into Asterisk."

While headlines normally focus on the cost-savings potential of VOIP, especially between branches over long distances, this call management solution reveals Asterisk's other core features.

Click Here for the Full Article

Signate Announces Availability of Telephony Stack Software

Signate, a leading global provider of VoIP telephone solutions based on industry standard hardware and open source software, today announced the general availability of the Signate Telephony Stack, a complete software infrastructure for 32-bit IP telephony applications on a single CD. Signate's installer loads all the software and creates a ready to configure VoIP telephony server in about fifteen minutes.

"Signate's Telephony Stack is the same software foundation that Signate uses for its telephony installations around the world," said Paul Mahler, Signate's CTO and author of VoIP Telephony with Asterisk. "Signate downloads it from community sources semi-annually, or as major enhancements are released, and tests it to ensure that the entire stack operates as expected. We make bug fixes when called for and return them to the appropriate project," he said.

Click Here for the Full Release

Dev Info: Svnmerge Updated

svnmerge in our repotools repository (http://svn.digium.com/svn/repotools/svnmerge)
has just been updated to the latest upstream code, which drastically improves avail/merge speed due to checking fewer revisions. This new version is also required for 'automerge' setup, because it allows the names of the properties to be specified on the command line.

There still seems to be an issue with automerge, though... it applies the updates to the branch, but does not update the property showing what it did, so the next time it runs it believes there are still revisions to be merged.

Help: Monitor Recording Legal Issues

Note: I'm not a lawyer so this isn't official legal advice but I have significant legal experience and offer some information as a guide to doing research on legal issues regarding recording phone conversations. Having said that, here are important concepts to research. The information here is probably better than what you would get from the average lawyer, but not as good as advice you would get from a lawyer who has experience in this area of practice.

Click Here for a List of Terms and Concepts

Nerd Vittles: Newbie's Guide to Asterisk@Home 2.5

Excerpt:

Want a rock-solid PBX at a rock-bottom price: free! Well, it's been two days since our tutorial on AAH 2.4 but here we go again! Asterisk@Home 2.5 has hit the street because of another serious bug-fix release of Asterisk. Now you get version 1.2.4 of Asterisk, and you also get the latest and greatest version of Linux, CentOS 4.2; the latest Festival Speech Engine (1.96); the latest version of the Asterisk Management Portal (1.10.010); the Flash Operator Panel (version 0.24); Open A2Billing; Digium card auto-configuration; NVfaxdetect support; loads of AGI scripts including weather forecasts and wakeup calls; xPL support; the SugarCRM Contact Management System with the Cisco XML Services interface and Click-to-Dial support; plus lots more. And, yes, it still fits on a single CD!

Click Here for more Nerd

Asterisk at SCALE 4x

Hello,

Asterisk will have strong presence at SCALE 4x, the 2006 Southern California Linux Expo next week. On the exhibit hall floor both Digium and SwitchVox will have booths demonstrating asterisk and related products.

The event will be held on Feb 11th and 12th at the Los Angeles Airport Radisson. In addition to Asterisk focused sponsors, we will have 3 talks on the topic of Asterisk and open-source VoIP.

Speakers:
* Mark Spencer (Digium) - IP Communication: Open for Business
* David Mandelstam (Sangoma) - It's a whole new world -- open source at the PBX, ready for prime time
* Tim Fritchel - Case Study Switching from Motorola to Asterisk

Other speakers include: Hans Reiser, Chris Dibona, Andi Gutmans and more.. For further details see the conference website at:
http://www.socallinuxexpo.org

Those interested in attending the show can use the promo code "AST06" to get 40% off full access passes.
(Click Here to Order)

Announce: New version of Snom soft phone

Hey we have made a new version of our soft phone which fixes an important bug in the SRTP SSRC part... It is compatible with our latest version 5.3 of the hard phones.

http://www.snom.com/download/snom360-5.3.exe

Enjoy,
Christian

News: An Old-School VoIP Deployment

Dennis Hock, a collector of antiquated telephony Relevant Products/Services from 3COM gear, wanted to make the treasures in his home ring. He figured that would require a manual switchboard, however, and his wife wasn't keen on helping out by playing operator.

"You start collecting phones, those are cool," says Hock, whose day job is network engineer at DTE Energy in Detroit. "Then you display them. The next logical step is that you'd like to be able to see them work."

As it turns out, fellow members of the Telephone Collectors International (TCI) were having similar thoughts, and the result might actually help demystify one of the big issues surrounding VoIP Relevant Products/Services from 3COM: how to make new packet-based systems work with old -- in this case, really old -- circuit-switched gear.

Click Here for the Full Article

News: Asterisk, Gizmo Project and RadioHandi

I'm not a VoIP geek by any stretch. Some people just get insanely excited about the idea of running their own PBX at home or what not, but I'm really not that type of person. However, I have to say that this video demo of Asterisk over at Kevin Rose's SYSTM really piqued my interest. Using an inexpensive telephone adapter - $100 for the Sipura 1000 - they were able to get a SIP-based VoIP server with full-on IVR functionality up and running in no time. It's pretty damn cool, I have to say - having calls routed from your phone to your desktop whereever you are, or using one of the new WiFi SIP handsets (the Hitachi 5000 for example) to make calls by connecting back to your home server through any WiFi Access Point. I'm definitely impressed enough to want to geek around with it.

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News: Skype-to-Asterisk(SIP): Progress - Part 2

Click Here for Skype-to-Asterisk(SIP): Progress - Part 2

Nerd Vittles: Newbie's Guide to Asterisk@Home 2.4

Note: He's done it again, another great read.

"Want a rock-solid PBX at a rock-bottom price: free! Well, it's almost Ground Hog's Day so here we go again! Asterisk@Home 2.4 has hit the street because of a serious bug-fix release of Asterisk. Now you get version 1.2.3 of Asterisk, and you also get the latest and greatest version of Linux, CentOS 4.2; the latest Festival Speech Engine (1.96); the latest version of the Asterisk Management Portal (1.10.010); the Flash Operator Panel (version 0.24); Open A2Billing; Digium card auto-configuration; NVfaxdetect support; loads of AGI scripts including weather forecasts and wakeup calls; xPL support; the SugarCRM Contact Management System with the Cisco XML Services interface and Click-to-Dial support; plus lots more. And, yes, it still fits on a single CD!"

Click Here for more Nerd Vittles