Just Days Remaining Until ClueCon!

ClueCon is almost here! If you were planning on waiting until the last minute to register, that time has come. We can still accept registrations on the internet but we can no longer guarantee the hotel package. Please call for availability at 1-877-7-4-A-CLUE (1-877-742-2583). Also take note that you can call this number for directions or other information at any time between now and the end of the conference because, being sponsored by a Toll-Free provider, we are able to redirect the number to any location!

If you plan to register in person, remember to have your ID and payment method ready to help speed up the registration process. Students, remember your Student ID's!

Registration begins at 8:00 AM Wednesday morning and will last until about 10:30 AM

Digium to Sponsor a Pizza party at Cluecon

Digium, the creator and primary developer of Asterisk, the industrys first Open Source PBX, will be hosting a pizza party from 4pm to 6pm on the first day of Cluecon. We look forward to everyone coming out to enjoy this opportunity to meet fellow developers and users in a more casual environment.


I would like to personally thank Mark Spencer and Digium for their support.


Thanks,
Brian West
Asterlink.com

Announcement of upcoming Asterisk book published by O'Reilly Media

I found this great information at Asteriskdocs.org. Finally a book about Asterisk will be release by O'Reilly. I have read many books in the past by this press. They are good at putting out quality reading material and I look forward to this one as well. Here is what they said on there site:

So you are probably wondering what we've been doing over the last seven months - well . . . we've been writing a book! Thanks to the generosity of O'Reilly Media, the dreams we had for our first two volumes have been realized in the upcoming book, Asterisk: The Future of Telephony

The manuscript is finished, the book is in production, and you can pre-order the book now to ensure your copy doesn't get backordered.

The authors, Jim Van Meggelen, Jared Smith and Leif Madsen will all be on hand at AstriCon 2005 in Anaheim, California from October 12th through the 14th to officially launch Asterisk: The Future of Telephony. We hope you can attend as we'd love to meet you all!

You can find the page on there site here: Full Article

Asterisk version 1.2 :: What's new?

In response to a large number of questions on the mailing list I've decided to publish a presentation I have been running in the Asterisk bootcamp - our one-week training class.

This presentation covers many, but does not claim to cover ALL, new features of Asterisk version 1.2. I hope it will wet your appetite to help us test the new code.

The presentation is available here:
* http://www.astricon.net/asterisk1-2/

For information on how you can help with Asterisk 1.2, see here:
* http://www.voip-forum.com/?p=176&more=1

If I've forgotten a new feature that you think is important, I'm grateful for all input.

/Olle

JAJAH-User - new version released

Now everyone lets go test the new version and find all the bugs.


We are happy to inform you that a new release of our revolutionary JAJAH webphone is now available for free download.

We have considered our users feedback and suggestions and so the new
version features:

* Improved registration procedure
* Improved firewall traversal
* Display of length of SMS (text message) to mobile phones
* Revamped Skype bridge - no more bottlenecks!
* Local Skype mode for outbound Skype calls with personal Skype ID
* An inbound service: Skype users can now call JAJAH users

For the complete list of changes and download go to: http://www.jajah.com/en/download.asp

If you have any questions, check out our FAQs (http://www.jajah.com/en/index.asp) or simply drop us an email !

Asterisk 1.2 Release Plans

This was posted by the lively Ken Fleming. Detail the process they go through when putting together another release for Asterisk.



As previously mentioned on the lists by Olle Johannson, we are actively trying to get Asterisk in shape for a 1.2 release within the next 60 days. To accomplish this, we need a few things to happen:

1) A feature freeze - This will occur at the end of this month, with no new feature submissions accepted after July 31st. Any _pending_ feature patches in Mantis that have passed architecture review and functionality
testing before August 1st can be accepted into 1.2, if they make it through the remainder of the review processes and are able to be merged before August 15th.

2) Progress on open bugs - There are a number of bugs open in Mantis that are waiting for the poster to provide additional information, test results, call traces, etc. We would much prefer to not release 1.2 with suspected problems already identified, but we cannot solve them without adequate input from you. If you have an open bug and are not in a position to continue providing assistance in solving it, please post a message to the mailing lists asking for volunteers to help replicate the problem so it can get resolved.

3) Testing - We need a _lot_ of help testing. If you have not previously tested CVS HEAD, please download it, read the UPGRADE.txt file and install it on one or more systems to play around with. Please do _not_ put it into a production environment unless you are willing to accept the consequences of that action. If you do find a bug or other issue, when you open a bug in Mantis, please try to provide _all_ the configuration information, call traces, etc. that the bug guidelines request, so that we don't waste 3-4 days just going back and forth requesting more information from you. If possible, join the #asterisk or #asterisk-dev IRC channel to find out exactly what debugging information will be required and how to produce it, if you don't already have that knowledge.

4) Release Candidates - I will produce the first release candidate on August 20th, with followup versions produced every week until we deem the release ready for public consumption. I expect it will require at least three -RC releases for us to get things in shape, so that means that 1.2 itself may be ready by September 15th.

We are very thankful for the community's help and support, and we want Asterisk 1.2 to be as important a release as 1.0 itself was. The number of new features, performance improvements, bug fixes and interoperability enhancements in CVS HEAD is astonishing, and a very large percentage of them came directly from community contributions. We hope that all of the 'non-developers' in the community will be able to help us 'shake out' the bugs and problems remaining in the code, so we can be assured of the most stable 1.2 release possible :-)

Huge Give Away @ ClueCon!!!!! Free Swag

Now we are getting to the good stuff, free swag time. Read below on how to get in on the action:


Cluecon's (http://www.cluecon.com) premier sponsor, Sangoma (http://www.sangoma.com), will be giving away 3 A101 single-port T1 cards during Cluecon. The A101 is Sangoma's next generation hardware designed for optimum support of data and voice over T1 and E1. Register for Cluecon now for your chance to win.


Be sure to register by this wednesday, it's the last day I can squeeze in room registrations so please register and pay by that date if possible.


Thanks,
Brian West
Asterlink.com

ClueCon in Two Weeks!

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount!

ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based on Asterisk and to top it off, we have invited many of the greatest names in VoIP to attend and share their experiences with
you.

* Mark Spencer - President of Digium
* Kristian Kielhofner - AstLinux
* Craig Southeren - co-author of OpenH323
* David Sugar - Lead developer of Bayonne
To name a few (there are many more).

We have done extensive research and development and had success in enterprise deployment, configuration and clustering of Asterisk. We don't just daydream about it we actually have it in production and send back our changes to the Asterisk CVS.

In addition, Asterlink has donated may resources to the community such as:

* The Infamous 996 Audio Conference
* The Digium CVS mirror
* PBXFreeware.org and all it's free asterisk add-on's
* The dev.pbxclue.com forum

Asterlink developers have produced a great deal of technology that we hope to offer as a service in the near future:

* Speech Recognition IVR Input
* IMAP Based Voicemail
* The app_confcall software that powers the 996 conference.

We have produced 2 embedded language modules that are freely available:

* res_perl - Embedding Perl into Asterisk
* res_js - Embedding JavaScript into Asterisk

All of this for the modest cost of $350.00. You could learn enough the first day to justify the price and then you get 2 more days on top of that!

Asterisk 1.2 is getting closer - please help

Dear Asterisk Community,

Asterisk 1.0 was released at Astricon 2004, in September last year. It's been almost a year and we haven't been able to go ahead and release a new version. Now is the time to try to move forward again.

As we've outlined before, the process is this:
---------------------------------------------------------------
* Code freeze: At this point, we'll stop accepting new additions (new functions) to the source code. Bug fixes are more than welcome, but additions will be postponed until after release and added to the 1.3dev source code base (the new HEAD).

* Release candidate: A release candidate will be produced as a tar.gz file on the FTP site.

* Release of 1.2: The new release version of Asterisk, that replaces Asterisk 1.0

* Release of 1.2.1: The working version :-) of the new version of Asterisk

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

So why 1.2.1? Well, the common feeling among developers is that "No one really tests anything until we release, so we will receive bug reports from the hour we release 1.2.0". Let's try to prove that they are wrong!

What can you do to help this process?
-----------------------------------------------------
* Set up a test system, and test CVS head in something that resembles your production environment. Scripts, phone, dialplan - make sure you use as many of the features as you can and use in production to make sure they work as expected in version 1.2

* Go wild and test at least two of the new features in 1.2 just for fun and make sure they work as documented. Or document how they work if it's not documented. Test the new realtime architecture, voicemail ODBC storage, AEL - the new scripting language, the new dialplan templates and constructs, the #exec config directive, attended transfers, native music on hold... The list is long.

* If you have reported bugs or filed patches in the bugtracker (bugs.digium.com), make sure you reply quickly when a bug marshal or developer ask you questions or require more information. At this point, we're working very hard to clear out outstanding bugs and stabilize the additions that is waiting for inclusion in the CVS. We will close reports that we can't move forward if we do not get any responses. We can re-open later, but need to move forward. If we have a report of a proven bug that needs fixing, those will not be closed. Only unclear reports with no responses will be closed.

* Visit the bug tracker at bugs.digium.com and help us test patches. Postitive and negative reports are both equally needed. There's no way a small team of core developers and bug marshals can test everything in there now. We need to decide which patches that are ready for inclusion, that are tested and documented.

* If you find that we're missing documentation, please add to the readme files, write new ones. The Asterisk documentation team is ready to help you if you need assistance in this effort.

* Disappoint the developers by making sure that the CVS head gets a thorough testing phase now, before release!

* Update the Wiki on the 1.2 version. Make sure that you make it very clear that new features only work in 1.2 and releases after that so you won't confuse readers that use older versions.

* Test Asterisk CVS head on other platforms than Linux: FreeBSD, OpenBSD, MacOS/X, Commodore VIC 20 - will it work?

When is 1.2 scheduled to be released?
-------------------------------------
At usual with Open Source, we release when the software is ready for release. We do not release when it suits the marketing department, when we need a positive stock report or when customers require it.

That said, we now are trying to focus on getting a release out of the door around September 1st. No promises, it all depends on your help and assistance to move forward. Please ask your boss for some time and resources to help the project with testing or dedicate resources within your company to help us. It's Open Source, meaning that everyone works together to make sure we get the software that works for our home, our company or our organization.

Finding information
-------------------
If you have questions about the developer version, the base for the 1.2 release, use the #asterisk-dev channel on the freenode.net IRC. If you have questions about bug reports and patches, find a bug marshal in the #asterisk-bugs channel. To find out how to download or connect to the IRC channel, please visit http://www.asterisk.org

Thank you for your assistance!

/Olle

VoIP on a bike - Infoworld Article

The fine guys at SineApps report on this article also. Of all the great services Asterisk provides, the one I hope is that it helps speed up the process of building infrastructure in 2nd and 3rd world countries. The only we will all be able to eat at the same table is if we can all have the same basics services. This is an inspiring article.

Here is a snippet of the article:

"The mission of Inveneo, a nonprofit group of inveterate high-tech adventurers, is to bring developing communities that never reached a 20th century level of infrastructure into the 21st century. Its bicycle-powered system brings not just VoIP but also e-mail and Web browsing to remote areas, using a combination of Linux and the Asterisk (Overview, Articles, Company) open source PBX."

Full Article Here

Editor Note: I have opened this post for comments. I would like to read people opinions of the article and if enough responses are posted I might make another post to open a discussion about this topic.

Xorcom Rapid 1.1 Released

Xorcom Rapid 1.1 is here.

* Asterisk 1.0.9
* Flash Operator Panel
* improved Zaptel hardware detection: should hopefully detect E1, T1 ZapHFC and qozap. No more channel numbers guessing in zapata/zaptel.conf
* and much of the "extra" software available for it

You can get the full details at:
http://www.xorcom.com/rapid

so I'll just highlight the points that I believe are more relevant to the people here rather than a standard press-release. Warning: long post.


Xorcom Rapid is based on the current Debian Stable. This is not just Asterisk built from source on a certain system: we use native distribution packages. You can install just the parts you like. E.g: spandsp and the h323 channel (with their extra dependencies) are optional components. It is also possible to upgrade packages or the whole system.

It is a binary distribution. Some people really don't like that idea. They think that if you didn't built it from upstream source it's not worth it. Well, if you have such an attitude then why are you running a Linux/*BSD distribution? use LinuxFromScratch to build a system "like a real programmer", and come back to report how long it took you and if you eventually did get a better and more manageable result.

Binary distribution is by no means locked down. You need to apply some fixes to the source? a decent packaging system provides simple ways of extracting the original source, patching it, and building the result. Build your own debs. True, you may need to set up a separate build system, but then again, the whole build tool-chain is not needed for a PBX to run.

We tried to separate the configuration to smaller files. This should make it safer to use newer configuration that fixes and enhances the default, and yet maintain your local changes. We do want to make it easier for you to upgrade your system, so you won't be stuck with an old, broken Asterisk that "happened to work and you don't touch it".

That said, we do realise that the voodoo factor is still considerably large. We can't and won't try forcing upgrades on anybody's precious PBX system.


This version is based on 1.0 . However it seems that the CVS head is really not that far from becoming 1.2 . The next version of Rapid will be based on it. Debian is also supposed to start working with Asterisk "1.1" packages in the Experimental branch. In the near future we will probably continue backporting required packages from Unstable when necessary and maintain compatibility with Stable.


Vim is included, along with syntax highlighting for asterisk configurations. vim is not the default vi (nvi is much smaller, you know) but if you edit many files, you'd probably want to install it.

I am looking for improvements: e.g: when editing Apache's httpd.conf files or CSS files, the syntax highlighting is very good at spotting syntax errors. I have already added something simple in that direction (a line that begins with '#' and is not an 'include' will be coloured as an error), but I'd like to see more.

Also included in this release is a web-based configuration interface called DeStar . I'm interested to expose it to a larger crowd, so have a go with it.

I've included some scriptary to play convert MP3s (off line) to "phone quality" WAVs, and to play the WAVs with sox for the music-on-hold . I would appreciate input on what you'd expect there. e.j: add some randomisation to the "wav-player" scriptlet?


The detection of Zaptel PRI and BRI cards should detect channel numbers correctly. But the span parameters and such are generally my simple attempt to give sane defaults. If it doesn't work in your case, please let me know. As a general note, if a simple shell script can detect channel numbers so easily, why can't chan_zap do all the work by itself?


And another small thing to simplify the initial testing: iaxcomm.exe is included on the CD. For windows people it should run off the CD. One less thing to download.


We have also set up a mailing list for Rapid, so feel free to subscribe there and post questions to a smaller, more focused crowd:

http://xorcom.com/mailman/listinfo/users_xorcom.com

Ollie needs YOUR help: Test CVS HEAD Voicemail ODBC Storage

As we are getting closer to release of CVS head as version 1.2, we're in need of your help.

One of the cool new features in CVS head is the ability to store the actual voicemail messages in a database. This is not using ARA, the Asterisk Realtime Architecture, but directly interfaces with ODBC from app_voicemail. It stores both meta data and audio in the database, which will give you real scalability if you have the voicemail accounts in realtime as well - you can propably have many servers handling the same voicemail accounts without bothering with NFS file locking and such problems.

We need extensive testing of this new feature to make sure it's ready for release!

There's also a patch in the bug tracker that extends this functionality a bit that needs testing and confirmation. Please check http://bugs.digium.com/view.php?id=4403 for a patch that adds information to the stored message.

Thank you for helping us testing CVS head and finding bugs before we release this new exciting version of Asterisk!

/O

astGUIclient project looking for consultants

Here's your chance to make an impact in Asterisk:

Hello,

The astGUIclient/VICIDIAL project is looking for consultants to help users install and troubleshoot astGUIclient/VICIDIAL on top of Asterisk
installations.

We have seen our mailing-list volume go up to the point that we are referring more people to consultants and the consultants we use are getting busy. We want to have a few more consultants on our list to give to people who need help or want to do installations from scratch. You can charge what you like but you must have thorough knowledge of Asterisk, MySQL, Linux and have installed and used astGUIclient/VICIDIAL on at least one server in a production environment.

Please email me off-list if you are interested.

astGUIclient is an open-source, GPL suite of applications extending the functions of Asterisk at the user level. Included in the package is VICIDIAL a contact-center application that scales across multiple Asterisk servers.
http://astguiclient.sourceforge.net/
Thanks,

MATT
mattf@vicimarketing.com

Announcement: YAACID (Caller ID for Asterisk)

I grabbed this hot off the email list:

I'm announcing YAACID: Yet Another Asterisk Caller ID v0.9 it's the first public release. we've been using it in our office for a couple of months now.


YAACID is a native Windows (.NET) program that sits in the notification area and logs into the manager interface. it waits for a call to come in on a monitored channel and then pops up the callerid info in a very intuitive interface (like the callerid boxes for normal phones)

It has a number of other features also:


1. Has the ability to play any custom wav sound when a call comes in

2. Has the ability to spawn a web browser and when a call comes
in, and it can pass any of the following variables to a given web
page:

$CALLERID,$CALLERIDNAME, and $UNIQUEID

YAACID can be configured to have the following url:
http://www.shatterit.com/callinfo.php?cid=$CALLERID&name=$CALLERIDNAME&id=$UNIQUEID

if a call comes in from: 716-852-5872, it will spawn a
browser and go to:

http://www.shatterit.com/callinfo.php?cid=17168525872&name=Mark+Musone&id=12345.3

3. YAACID can hold up to 50 calls in its Call ID History.

Feel free to download it and take a look:

http://www.shatterit.com/opensource/yaacid

AstLinux creator to speak at Cluecon

Kristian Kielhofner, the lead developer of the AstLinux project, will be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a complete Asterisk distribution built to run from Compact Flash and uses less than 32mb.

Thanks,
Brian

Asterisk bounty: Email TTS - $150.00

I pulled this off the User list. Let see if one of our readers and enjoy wealth and fame for solving this problem. If you answer this from seeing the post here, email me so I can post an update.

Bounty:

(forgive the brief interruption to -users with a mostly -dev issue, just
wanted to publicize this on behalf of the larger community)

If there are any ambitious coders out there (not too many shekels yet,
but I expect some folks may pony-up) please see:

www.voip-info.org/?page=Asterisk+Bounty+Email+TTS

We are at $150 & counting.

Maybe lobby your exec's for $50 to contribute to this, you know how
those exec' types dig stuff like this, imagine the CFO using this from
his Treo650.

:)

Jason Sjobeck
ICQ 5579183

ClueCon Announces Academic Admission

ClueCon is giving students an oppurtunity to receive a special academic admission to the upcomming VoIP conference as a way to help recruit new interest in voice-over-IP technology.

There will be many speakers at the show and it is our hope the students will take an interest in VOIP technology and we feel it is important to give them a chance to learn something about the growing industry.


To apply for the program call 1-877-7-4-A-CLUE (1-877-742-2583) or email to students@cluecon.com

Wireless VoIP Congress - Examining the Latest Technologies and Business Strategies for Wireless VoIP and the Threats and Opportunities they Represent

I just got this email sent to me. I think we all need to watch these very close. I am sure within the next 6 to 9 monthes some rulings will be made that will greatly affect what will happen to VoIP in general in the future.

Date: 8-9 November 2005

Venue: Crowne Plaza Europa, Brussels

Website: www.telecoms.com/wvoip

For further information, please contact emma.abrahamson@informa.com or call Emma on +44 (0) 20 7017 4202

This inaugural event will be addressing:

-Implications of Wireless VoIP for communications providers business models
-Opportunities created in the enterprise and residential markets
-Realistic capabilities of dual mode handsets and their time to market
-Ensuring network security and QoS for Wireless VoIP
-Implications of UMA, SIP and IMS for Wireless VoIP services

Leading industry speakers from:
-Skype
-O2
-T-Mobile
-Swisscom
-Telabria
-Telewest
-Coffee Telecom
-The Cloud
-WIND
-Motorola
-Cisco Systems
-Tulip Mobile

IPSwitchBoard shows Call Charges

Version 0.122 - 12 July 2005
Call charges are now shown on the Calls page

IPSwitchBoard will check for a live connection every minute and reconnect if the connection is lost for some reason (asterisk restart etc.)

Bug fixes
FREE Download: http://ipswitchboard.thorben.dk

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.

Astricon 2005 :: Call for speakers and Asteriskprojects

I know all you rabid Asterisk fans have been waiting for the next Astricon installment. Here is your information:

Astricon 2005 will take place in Anaheim, California October 12-14th 2005. Astricon is the Asterisk conference, arranged by IPsando LLC in cooperation with Digium.

We are now looking for speakers. The conference will be bigger than last year, so we are looking for more speakers in the conference part of Astricon.

In addition to the tutorials, we're adding a three-track conference day with a focus on call/contact centers, carriers and enterprise use of Asterisk. The full program for Astricon 2005 looks like this:

* Tuesday, Oct 11th: Developer meeting
Meet Asterisk introduction for newbies
* Wednesday, Oct 12th: Tutorials
* Thursday, Oct 13th: Three track conference
* Friday, Oct 14th: One track general conference

Please go to http://www.astricon.net/2005/speakers/ to read more on how to submit your speaker's proposal. If you have any questions,contact me off-list at oej@ipsando.com ! We need your proposal latest july 30th.

*** Asterisk Solutions Showcase
For the first year, we're running an Asterisk Solutions Showcase on Astricon! This will be part of the Astricon Exhibition. Asterisk-related Open Source projects will be able to showcase their solutions in this area. Also, if you have a good, funny, scary or just impressive solution based on Asterisk that you want to share, please apply for a booth in the Asterisk Solutions Showcase. Brian Capouch will be there showing Asterisk on Linksys Routers amongst other projects.

Please send proposals describing your solution to info@ipsando.com no later than july 30th. If you want to show commercial solutions that you sell, you need to buy a booth in the exhibition. Ask for more information by e-mailing info@astricon.net.

Looking forward to meeting you in Anaheim! Early Bird registrations will open later this week.

Best regards,
/Olle and Steve

InfoWeek Article on VoIP

Micheal from the Asterisk Users List posted this nice little article from InfoWeek about VoIP. Thanks Mike keep up the good work. Here is the post:

"InfoWeek VoIP Article

The bottom line is that they compare retail VoIP providers like Comcast Cable, Time-Warner Cable, AT&T, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VoIP services don't yet match the PSTN for reliability & call quality.

It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VoIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones.

My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access.

Michael Graves


--
Michael Graves
Sr. Product Specialist
Pixel Power Inc."

chan_capi-cm-0.5.4 release

Hi all,

on sourceforge.net I added the fixup release 0.5.4 of
chan_capi-cm driver.

The changes from 0.5.3 to 0.5.4 are:

- fixed 'group' setting according to Asterisk defaults.
- use SetCallerPres(prohib_not_screened) instead of CallingPres(32) for CLIR.
- full CallingPres support added.
- use mutex when debug/verbose messages are printed.
- set dnid on incoming call.
- catch errors in wrong dialstring.
- set correct DIALSTATUS and HANGUPCAUSE.
- set PROGRESS and PROCEEDING when the network signals them.
- increased voice send buffer a little bit.
- fixed seg-fault when unallocated number was dialed.

Have fun
Armin

--
Cytronics & Melware
Weinbergstrasse 39
55296 Loerzweiler / Germany
Tel: +49 6138 98110-0
Fax: +49 6138 98110-9

JAJAH says Hello to you! (Emailed me too)

Today I received an email with information about this new SIP and IAX compliant. It looks to have the same look and feel as Skype and Gizmo. Has Anyone used this yet? If you have a review please email and post on the comments below. The email stats this is a "Beta-Test" so please download and put this VoIP app through its paces.



" Eventually you already know of our new webphone named JAJAH
(http://www.jajah.com). We are continuously working on the improvement of our application and we do seriously consider the feedback by our users.

At this stage we would like to invite you to join our exclusive group of beta users. As you are interested in VoIP related topics we would be glad to receive specific and constructive feedback of yours.

If you are interested in experiencing the advantages of our free VoIP communication tool we are ready to award you 60 minutes of free international phone calls (jajah-out).

The main advantages of JAJAH are its compatibility to SIP, IAX, Skype, Gizmo and other services, the low bandwidth requirements and superior audio quality. By the use of JAJAH and one of our SIP-phonesets you can enjoy free telephony to 100 mil. people, 24 hours a day, even while your computer is asleep. High quality live video calls, free voice mailbox, live chat, instant messaging , real-time translation, call forwarding, conference calls, skins, ringtones and other great features (http://www.jajah.com/en/features_infinite.asp for the complete functionality of JAJAH).



The latest version of JAJAH is available for free download at:
http://www.jajah.com/en/download.asp

Currently we are also finalising our all new JAJAH mobile application. This revolutionary solution allows JAJAH calls with any customary cell phone with data connectivity (e.g. WAP, GPRS, HTML) at the low JAJAH rates.

In two weeks time we will introduce JAJAH mobile to an exclusive group of beta users. If you are interested in joining this group, just send me an email titled "JAJAH goes mobile" and apply for participation. This is your invitation to go JAJAH mobile well before its public launch.

Kind regards,

Sonja Malzer


JAJAH Technologies GmbH
Wohllebengasse 19/23
A-1040 Vienna, Austria

Peter Nixon to Speak at Cluecon

Peter Nixon will be making the trip to Chicago to speak at Cluecon,
he'll be speaking on the topic of "Real world deployment of Open
Source". Peter has done tremendous amounts of work on the
FreeRadius project. In addition if you're wanting to get sponsorship
in this is the week to do so, we are sending everything off to the
printer to get printed by friday.

Thanks,
Brian West

ClueCon Exhibitor booths now available

ClueCon has announced the immediate availability of 25 exhibitor booths to give VoIP related companies a chance to show off their products and services. The booths will be sold on a first-come-first-serve basis for $1,000.00 and will include an all-access pass for one individual. Call 1-877-7-4-A-CLUE (1-877-742-2583) to reserve a booth today.

New Projects: Adding SS7 FGD trunking capabilities to the Asterisk PBX

I came across this project in progress. Below is an excerpt:

"This project is to add SS7 FGD trunking capabilities to the Asterisk PBX as enabled by the OpenSS7 SS7 and IP stacks. This project focuses on the ISUP capabilities of the OpenSS7 stack. It utilizes M3UA for distribution, redunancy and reliability and supports TALI integration to Tekelec IP7 and other TALI compliant products."

Asterisk integration with sphinx2

Here is a link to a nice little article on a voice recognition project that is still in the Alpha testing stage. It deals with integrating Asterisk with voice recognition.

Vikrant Mathur lead developer for the open sourceOSP Toolkit to speak at Cluecon

Vikrant Mathur is the lead developer for the open source OSP Toolkit available on SIPfoundry. Mr. Mathur began his career in telecommunications as a software engineer at Hughes Software Systems where he focused on softswitch development. After completing his Masters degree in Electrical Engineering at North Carolina State University he joined TransNexus as a senior software engineer developing solutions for secure peer to peer routing, access control and accounting of VoIP traffic on the Internet.

If you haven't registered yet please do so ASAP so we can make sure
to reserve you a room!

Thanks,
Brian West
http://www.cluecon.com

News Issue Resolved w/ SineApps

Hello All,

I just wanted to drop a line to let all my readers know that I am in contact with the fine people at SineApps.com about this issue. One of the editors has mentioned a few articles that were taken from his site and we are taking steps to properly source them. Also I am not trying to take any of the hard work the SineApps guys.

Sidenote: The only goal of this site is to provide useful information about "Asterisk" and "VoIP". I think Asterisk is one of the greatest piece of open-source software next to Linux.

-Dal

asterisk-oh323: New version 0.6.6

Hello all,

A new bug-fix release of asterisk-oh323 for the *stable version*
of Asterisk is available. This version has the option to compile
with latest OpenH323/Pwlib libraries but we recommend to stay
with the Janus version.

The updated version that is compatible with the *CVS HEAD version* of
Asterisk will delay for a while.

Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323

Regards,
Michael

NewsForge: No software patents in Europe

Source: SineApps.com

A great step forward for the European software development community.

Excerpt from NewsForge story:

Press Release - After years of struggle, the European Parliament finally rejected the software patent directive with 648 of 680 votes: A strong signal against patents on software logic, a sign of lost faith in the European Union and a clear request for the European Patent Office (EPO) to change its policy: the EPO must stop issuing software patents today.

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"This outcome does not affect patents on high-tech inventions in any way," explains Stefano Maffulli, Italian representative of FSFE: "High-tech innovation has always been patentable, and even if the directive had been passed with all proposed amendmends, it would have remained patentable. It is important to point this out because the proponents of software logic patents have tried to confuse people about high-tech inventions being subject of this directive."

FSFE's president, Georg Greve adds: "The parliament understood this when it amended the directive in the first reading to keep high-tech innovation inside and software outside the patent system."

Astlinux-announce: AstLinux Install CD

Source: SineApps.com

Kristian Kielhofner has posted details of the AstLinux installer CD:

Hello Everyone,

I finally wrote the "installer" for the AstLinux CD. It is available in the following ISO image:

http://mirror.krisk.org/AstLinux-0.2.7-2.iso.gz

As usual, burn it, boot it. At the ISOlinux prompt, you should be able to type "install" to start the install process... Right now (much like AstLinux) it only works on IDE/SATA hard drives and USB drives...

It has a few quirks, but it has worked %100 for me on my test systems. Let me know how it goes!

Flash Operator Panel: 0.23-snapshot

Nicolas Gudino has posted details of the latest snapshot of the Flash Operator Panel:

Hi all,

The new version has some bug fixes and internationalization support.

Please test and contribute with translating it to your native language.
You will find very brief instructions on UPGRADE.

I would also like to provide nice help files...

Thanks!

SIPfoundry Board Member Bob Andreasen to speak at ClueCon

ClueCon has just confirmed the attendance of Bob Andreasen from SIPfoundry who will be giving a presentation entitled "SIPfoundry - Open Source SIP Transforming Communications" The presentation will last an hour and will take place August 4th, 2005 at 2pm. We are very excited to have such a diverse array of open source VoIP projects to showcase.

ClueCon Welcomes Another Speaker! Craig Southeren, CO-Author of OpenH323 Scheduled To Attend

Through the generosity of our Premier sponsor, Sangoma Technologies we are proud to welcome Craig Southeren all the way from Australia. Mr. Southeren.s work has pioneered the development of open source telephony applications with his ground-breaking Open H323 protocol stack that stood alone as the only open source VOIP software for quite some time. Today, Craig continues to raise the bar with the next generation OPAL VOIP abstraction layer and WOOMERA a brilliant approach to overcome software incompatibles. Hope to see you all at Clue Con.

OT Mark Spencer lunch in Paris Fri July 8th

Mark Spencer will be in Paris this weekend:

There is going to be another great Paris lunch with Mark this Friday.

The restaurant will probably be in the southern part of Paris in the 14th arrdt. like last time.

Please contact me off list if you are able to attend.

Idefisk iax2 softphone - new version

We just released a new version of the idefisk iax2 softphone, version 1.21 beta, available for download at http://www.asteriskguru.com/tools/idefisk_beta.php

Some bugs were fixed, some new bugs might have been introduced :) - The problem with delays is finally gone!!! (one of the bugs was a memory leak, everybody using an older version is encouraged to upgrade.)

Privacy Warning:
Version 1.21 of the softphone will send 'usage statistics' to the asteriskguru webserver, this can be disabled in the configuration menu (uncheck send usage statistics). The only info sent is the version of idefisk used.

Many thanks to digium, stevek and others for the iaxlib and iaxclient libraries.

Zoa.

New Astmanproxy 1.1 now available

Hey there folks --

I have been continuing development on the multi-threaded, c-based Asterisk
Manager Proxy program, AstManProxy.

I've incorporated several ideas I received at the recent Astricon Europe,
including:

- Supports proxying of multiple Asterisk servers at once
- Abstracted, modular I/O handlers (implemented as shared objects)
- Existing handlers: XML, Standard, CSV, HTTP

One really cool feature that I'd like feedback and testing on is HTTP
support. With this, you can POST or GET HTTP to the proxy and receive XML
back, thus allowing a very simple (REST-like) web interface to the
Asterisk Manager.

Please download astmanproxy 1.1 and try it out:
http://www.popvox.com/astmanproxy

We are also putting together an 'astmanproxy' mailing list. If you would
like to join, please e-mail me and I will include your name in the initial
mailing list.

Thanks, and for you fellow yanks out there, have a great holiday!

Regards,
Dave

--
David C. Troy
President/CEO
popvox, LLC
dave@popvox.com
Phone: +1-410-647-5812

Digium Second Generation Firmware for T1/E1/J1 Cards

Digium has posted details of an improved firmware for their PRI cards.

Excerpt:

Digium cards are now smarter! Digium is proud to announce second-generation firmware for Digium multispan TE-series hardware PCI cards. This firmware provides several new features. These features allow Digium cards to do more work in hardware and leaving the server's CPU open to do other tasks, all of which contribute to less CPU overhead, and therefore improved performance, and more channels running through a single PC or server.

-SNIP-

This is a 67% increase over previous benchmarks

-SNIP-
Improvements in the firmware include:

-TDM channel alignment now done in hardware (instead of software), for greater voice integrity and reliability.
-Master clock source distribution for synchronized timing across multiple cards, assuring synchronization of clocks and increasing reliability and quality of data transmission.
-Zero-latency TDM direct hardware-level cross-connect
-Supports new echo-cancellation module which further increases performance.
-DTMF detection can be done in hardware with echo cancellation module.
-Field upgradable firmware for future updates.

new chan_sccp release

Sergio Chersovani has posted details of the latest chan_sccp release:

http://chan-sccp.berlios.de/ (you can find a mailing list here and a bug tracker)

20050701 chan_sccp.20050701.tar.gz

I did rework the original chan_sccp. Asterisk cisco phones channel driver.

-fixed all the issues, added localized softkeys and display messages (the phone load XML localized strings from the tftp server).

-complete rewrite of the call flow.

-cleaned the console. Now you can use sccp debug to set the debug level from 0 to 10 (verbose)

-added new directives (incominglimit, tos and rtptos, digittimeout, firstdigittimeout and more. I need to modify the config file)

-added support for callwaiting

-no need to edit the makefile. It does discover the asterisk version. You just need to extract and make install.

-new locking system

more and more :-)
I will work on it to add native transfer/blind transfer and call forward capabilities

Please test it.

Sergio Chersovani

Operators Panel for Asterisk

Thorben Jensen has posted details of the latest release of IPSwitchBoard:

IPSwitchBoard Version 0.121 - 02 July 2005
Extensions can be added to speed dial number. This can be used to dial speed dial numbers from any phone connected to your asterisk system. This requires that you configure your dial plan to take advantage of this feature. See sample Dial Plan in the IPS Manual.
Download: http://ipswitchboard.thorben.dk

Asterisk Business Edition

A New Professional-grade Version of the Asterisk PBX Software Suite!
Available now!

Digium, the leader in open source telephony, is now shipping the new Asterisk Business Edition, a professional-grade version of its acclaimed open source PBX for the Linux operating system. This version provides tested reliability of critical functions and features, tailored for small- and medium-sized business applications. An all-new Asterisk technical manual and quick-start documentation supplements the package, making Asterisk even easier to install, configure, and use. Asterisk Business Edition is backed by Digium's professional support team with a full one year limited warranty. This provides enterprise environments with a PBX and telephony platform suitable for critical business applications.

Digium's comprehensive test program ensures Asterisk Business Edition's reliability, performance, and interoperability with key hardware, software, and protocols. Digium hardware cards are tested for full compatibility with Asterisk Business Edition, as are several select models of servers, VoIP, and TDM devices. All major software features in Asterisk Business Edition are thoroughly tested for functionality and reliability. Test bed systems are also subjected to extreme stress conditions using Empirix™ test equipment to simulate hundreds of thousands of calls in various real-world combinations and configurations.

As a result, customers can rely on their combination of proven Asterisk software and Digium hardware to work together to provide a feature-rich PBX or VoIP system.

Digium's Asterisk Business Edition may be ordered from an authorized Digium distributor, or directly through Digium by clicking here
... or by calling +1-256-428-6262, or toll free +1-877-LINUX-ME (+1-877-546-8963 - press "one" for sales).

A Day in the Life: How to setup a Cisco 7960G with SIP

Tracy has put up a blog entry on setting up a Cisco 7960G with SIP.

Excerpt:

Man, I had a beast of a time setting up a Cisco 7960G to use a SIP image instead of the default SCCP image that came installed on it.

I purchased the 7960G from VoIPSupply.com about a week ago, and just got time to mess around with it yesterday. Surprisingly the system comes with zero documentation, either hardcopy or a pdf on a CD. Zip, nil, notta, nothing… not even a CD period.

I skimmed through some "how-to's" on VoIP-Info.org and got the jist of what needed to be completed before the Cisco 7960G would boot up and connect to the TFTP server and download the new SIP image. It all seemed easy enough from what I had read, after all it isn't rocket science that we were dealing with.