Announce: TDM2400P Driver Change

Hello,

Thank you for your support of Digium and Asterisk.

Yesterday, we identified a problem with the driver for the TDM2400P card when used with FXO port modules. The X400M was producing (and expecting) the audio signal at an abnormally high level, which negatively impacted the performance of the echo cancelers, both hardware and software. It also resulted in unusual audio artifacts when the actual signal was already at a high level.

The driver in our Subversion repositories of Zaptel (both 'trunk' and 'branches/1.2') has been modified to properly initialize the FXO ports to alleviate this problem. We encourage all owners of TDM2400P cards with FXO ports to update their driver as soon as possible, especially if they are experiencing audio quality and/or echo issues on the FXO ports. This updated driver will be part of the Zaptel 1.2.2 release as well, if you wish to wait for a tarball release.

Thank you again for supporting Digium and Asterisk!

--
Malcolm Davenport
Digium

Release: AstManProxy - New Version 1.13 Now Available

Hey folks --

Good news! Version 1.13 of AstManProxy is now available for download!

http://www.popvox.com/astmanproxy

Several enhancements have been made, primarily dealing with robustness and also an attempt to add Mac OS X support.

-- AstManProxy 1.13 --

-Changed to read only /etc/asterisk/astmanproxy.conf and not ./astmanproxy.conf
-Added a 'connected' field so we don't try to write to servers which are not yet connected
-Exits when there are no servers able to connect
-Doesn't attempt to re-connect to a server if we get 'Authentication failed'
-Added a connection timeout by using connect_nonb
-Confirmed support for x86-64 processors (added -fPIC; Jennifer Hales)
-Added support for Mac OS X (Tested on 10.3.9); BSD may also work
-Aborts on old config file format (detects incomplete server spec)

I've had much of this code "in the can" for a few weeks now, but have been traveling quite a bit and unable to sit down and package everything up. Please review and let me know your findings!

The next step for Astmanproxy will be to get a branch setup in the Digium Subversion (SVN) repository, which I hope to do sometime in the next few weeks.

Best wishes for a happy holiday season to you and yours!

Cheers,
Dave

Announce: Pending Web-MeetMe Update

[New Features]
1. Added focus to the input textboxes on all pages, so there's one less mouse click on each page. Trivial, but something that would likely trip some users up.

2. Dynamic generation of year/month/day listboxes to prevent invalid date selection. The system still defaults to the current date, but changing the month or year will update the day listbox with the correct number of days.

3. Added 'Extend' and 'End Now' buttons to the monitor page. The 'Extend' button adds 10 minutes to the conference. I have not added an 'Add Seats' button, and am not sure how critical it is. I'd like to avoid interface clutter if possible.

4. Call history report. It is now possible to see who was in a conference and for how long. There might be a small issue with this feature, as I did not see a clean way to add a 'Back' button. At the moment the only way out of this view is to select a menu item from the left side selections.

This functionallity requires a patch to app_meetme to add duration statistics to the meetme_leave event. There are a number of options to accomplish this, but for my environment, this is the best.

It also uses a small PHP script that needs to be running to catch the manager events. The script is a little crude at the moment, and does not deal well if Asterisk not available (crashed/stopped/etc), but works fine under normal circumstances.

I can either make this feature optional and disabled by default, or provide my app_meetme patch (all ready in Mantis). I'd appreciate comments on which people would prefer.


Note: While I am at it, if there happens to be a PHP guru lurking, I'd appreciate any comments on cleaning up the code.

I should have the new package ready by the end of the week.

Click Here for More Information

Dan

Release: AsterFax beta 5 - Sendmail Integration

AsterFax provides an email based Fax gateway for the transmission of faxes using Asterisk.

You can download AsterFax at: http://asterfax.sourceforge.net

The newly released Beta 5 introduces two new features and a number improvements to the configuration of AsterFax.

New Features:

Integration with SendMail/Postfix/Exchange Server etc via procmail
With the release of Beta 5 AsterFax now has two modes of accepting emails for faxing. The exiting mode where AsterFax acts as an SMTP gateway plus a new mode where AsterFax can be configured to monitor a directory where Mime encoded mail files are placed for AsterFax to process. The recommended method is to use procmail to extract the mail messages out of the mail server pipeline.

Audit trail
An audit trail is now written for every fax transmitted. The audit trail is written to logs/audit.log.

Improved Debugging
A number of new debugging options are now available via the AsterFax.xml file to aid it diagnosing problems with AsterFax.

AsterFax has now surpased 400 downloads and is now ranked 305th out of some 100,000 projects on sourceforge, which given that its has only been available for 6 weeks is an indicator of the strong level of interest in the product.

Regards,
S. Brett Sutton

Nerd Vittles: The Music Frontier - Taming Streaming Audio for Music on Hold with Asterisk

Nerd Vittles today presents a detailed HOW-TO for painlessly implementing streaming audio for Music on Hold with Asterisk:

Taming Streaming Audio on Asterisk

Ward Mundy
ward@mundy.org
(404) 795-2227
(800) 942-7620

Release: New Asterisk Management Interface w/ JavaManager Live Console

Hi,

Druid is a new Web-based Asterisk management software. Its quite feature packed and allows you to manage every aspect of Asterisk configuration. It also has a Java Applet based Manager Console so you can visually monitor what your Asterisk box is currently doing.

We will have a live demo up soon but till then enjoy the screenshots.

http://www.voiceroute.net

--
regards
Vikram

Release: Snom Firmware 5.0

Hi,

Snom phones firmware 5.0 is now out.

Try it if you like:
http://www.snom.com/wiki/index.php/Main_Page.

Regards,

Usman Tahir
Snom Technology AG
www.snom.com

Announce: Callware VoiceOne released

We proudly announce that VoiceOne is release in a ALPHA release. We are waiting your suggestions and comment! Try it freely!

http://www.voiceone.it

Main Features:
-Client/Server architecture based on Web services
-Relies on Asterisk Realtime Architecture (ARA)
-SIP extensions management (support for Zap/IAX/mISDN soon added)
-Remote offices via IAX with RSA public key encryption
-Supports VoIP and traditional Telco providers
-Powerful IVR creation system
-Queues system management
-Customizable user's profile
-Powerful configuration of mISDN ports/interfaces (thanks to guys at beroNet for their support)
-Policies system for users/groups management
-SIP, IAX and mISDN general/default options configuration
-Static-like text editor for conf files
-Easy setup wizard
lots more...

Click Here for More Information

Click Here to Download

Thanks,

The VoiceOne development team

News: Asterisk::LDAP Update

Hello All

About six months ago I wrote to announce the release of my Perl module to manage Asterisk configuration in LDAP. I wanted to take a moment and send this follow-up to announce a newly updated version of Asterisk::LDAP, version 0.6.0.

This new release is a significant improvement over the 0.5 series.

Here's a quick summary of the new features:

* Serial Numbers: Contexts are now written with a serial number and are not updated unless the serial number is incremented. This allows for granular and guaranteed consistent updates. Because LDAP updates are atomic the administrator or any external dialplan manager needs only to update the serial number after making any changes to the dialplan to guarantee the new dialplan will be loaded in its entirety.

* API Simplicity: As mentioned above, the API has been dramatically cleaned up. Where before a number of calls were required to set up Asterisk::LDAP before getting any useful data out, only one call is required and three more optional calls can help the developer automate much of the configuration of the module. Also the internal data structures are made available to external programs at more points during the configuration generation so any kind of hooks or post-processing can be done more easily.

* New Output Formats: The developer may now choose to have Asterisk::LDAP simply read the information from LDAP or have it write the contents to a set of files. Future improvements may include support for returning a string containing the entire data output instead of files written to disk.

To find out more about this project or to download the new release please visit the pages at: http://projects.alkaloid.net

Thanks again,

/BAK/
--
Ben Klang ben@alkaloid.net
Alkaloid Networks

Announce: The Buddy System (Beta)

Welcome to The Buddy System.

Do you ever ask the following questions:

* Who is on call?
* Whose turn is it to take the beeper?
* Is the important after-hours call going to be handled?

The Buddy System

* Knows who is on call and all their phone numbers
* Knows who is backup if the primary on call person is busy
* Gets incoming calls the to the right person

Use The Buddy System to make sure that important after-hours calls are handled by the right person.

History:

* 2005/12/14 -- User guide and fixed some production issues introduced by putting 'puts' in controllers
* 2002/12/15 -- Voicemail
* 2005/12/09 -- Support for peering with FWD, IAXTel, and SIPPhone. Versioning information.
* 2005/12/07 -- Completed the "go live" script, team call routing, adding incoming phones to a team, and enabled Asterisk CDR output to PostgreSQL.
* 2005/12/06 -- work on the legal stuff and other "real product" stuff.
* 2005/12/02 -- Got Team List and all other functionality working. Starting setting up a production environment.
* 2005/11/28 -- changed from PyAstre to Fast AGI (talking to the Ruby server) on Asterisk. Implemented calls between users that ring to the scheduled phones.
* 2005/11/18 -- Mastered Ruby of Rails, began Python scripts that live inside Asterisk. Implemented users, phones, voip accounts.
* 2005/11/9 -- Prototyping begins using Ruby on Rails, Asterisk, and PyAstre.

Click Here for More Information

News: Unicall for MacOS X

A new project to bring Steve Underwood's Unicall telephony driver abstraction library to MacOS X has been initiated. Unicall is a replacement for Zaptel, the default telephony driver suite for Asterisk on Linux. The Unicall for MacOS X project aims to support the following hardware:

# Motorola SM56 based Apple Modems, ie. internal and external Apple USB Modem
# Intel i537 softmodem PCI card
# Odin Telesystems interface cards
# cards from other vendors if equipment and sponsorship will be provided

The Astmasters Zaptel for MacOS X project will be shut down in favour of the new Unicall for Mac OS X project. The following is an excerpt from a posting by benjk to the astmasters driver development mailing list on December 15th, 2005:

[Subject: Zaptel is dead - Long live Unicall]
Unicall has got it all

In the place of Zaptel, I would like to establish a project to bring Steve Underwood's Unicall to MacOS X. Unicall is a replacement for Zaptel, which has got all the things which are absent from Zaptel ...
# Unicall has design.
# Unicall has structure.
# Unicall has abstraction.
# Unicall has documentation.
# Unicall has direction and a roadmap.
# Unicall has a cross-platform philosophy.
# Unicall has taken a multi-vendor-telephony-hardware approach.

Lucky Seven. What more could we possibly ask for?!

Click Here For More Information

Nerd Vittles Strikes Again: Newbie's Guide to Asterisk@Home 2.2: Unabridged Soup-to-Nuts Installation Guide

Nerd Vittles today provides its comprehensive HOW-TO Guide for installation and configuration of the latest version of Asterisk@Home, version 2.2:

Newbie's Guide to Asterisk@Home 2.2: Unabridged Soup-to-Nuts Installation Guide

From Site: "Want a rock-solid PBX at a rock-bottom price: free! Yep, it's been a week and here we go again! Asterisk@Home 2.2 has hit the street primarily because of a bug-fix release of Asterisk! Now you get the latest version of Asterisk (version 1.2.1), 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; fax 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 some more bug fixes. And it amazingly still fits on a single CD!
"

Click Here for More Information


Click Here for SourceForge Link

Announce: QueueMetrics 1.0 rc 1 Release Today

Hello,

I am pleased to tell you that we have released a new version of QueueMetrics. This time QueueMetrics "comes of age", as we have almost reached version 1.0. :-)

This version is the result of a feature-freeze of version 0.9.7, and vastly improves memory efficiency, letting you analyze as much as two to four times the number of calls using the same amount of RAM. QueueMetrics is strongly focused to serve the largest commecial call-centers based on Asterisk, and is very often being installed on 100+ agents systems.

Pause handling was improved in order to support even the weirdest cases of agents pausing in/out when not logged on.

We are confident that you'll love this version!

A complete list of improvements can be found here:
http://queuemetrics.loway.it/news.jsp
The latest version of QM can be downloaded from:
http://queuemetrics.loway.it/download.jsp

QueueMerics is free to use and experiment for smaller call centers, home users and general Asterisk hackers. Larger installations can ask us for a free trial key from: http://queuemetrics.loway.it/sendDemoLicence.jsp

--
Loway Research - Home of QueueMetrics

Community: (Montreal Users) Call for technical presentations

Hi everyone,

For the next AMUG meeting (mid-january), I'd like to know if someone would be interested in making a presentation about the following:

*SIP/IAX protocols in depth
*Asterisk source code, files and architecture
*What's a Digium, Sangoma card made of
*whatever technical you'd like to share

I'll take in charge the * source code, and I'd be glad to work with the ones who have a good knowledge of the code / compilation process.

Thanks for your participation,

Adrien
http://amug.modulis.ca/

Project: Implementing Bluetooth Proximity Detection with Asterisk and a TomTom GPS

We've pretty well documented how you can set up Bluetooth Proximity Detection using a bluetooth headset or cellphone with your Asterisk PBX. Once configured, phone calls in your home or office can automatically be transferred to your cellphone whenever you take off carrying your bluetooth device. In our original articles, you'll recall that we recommended a bluetooth headset as the ideal way to track your comings and goings at very little cost. But today, we want to add another bit of magic to the project and also give you something to tell Santa about.

Click Here for the Full Article

Release: Updated Guide to SPA-3000

Note: Kerry Garrison posted details for a updated guide for the Sipura Adapter.


With some of the newer versions of Asterisk and AMP, many people have been having problems getting the Linksys/Sipura SPA-3000 working properly. We have just posted an all-new guide to getting the SPA-3000 up and running.

Click Here For More Information on the Guide
-Kerry

Release: Asteriskguru Queue Statistics version 0.7 released

Hello,

After a long period of inactivity we are proud to bring you a new version of the Queue Statistics.

Main changes in this version are:

- Fixed a nasty bug where calls can't be longer than 999 seconds.
- Added the possibility to see reports for all queues.
- some code cleanups

In the next version we will try to support mysql and pgsql. (currently we only have pgsql).

More information and a free download link is available from:
http://www.asteriskguru.com/tools/
queue_stats.php

Zoa.

Release: AstLinux 0.3.0 Released

Hello everyone,

AstLinux 0.3.0 has been released. This is the first stable release of AstLinux to include Asterisk 1.2. There are (of course) numerous other fixes, more rc.conf variables, a kernel upgrade and some other goodies as well.

AstLinux is now available in the following forms:

- Windows Install Package
- Live CD/Windows CD
- VMWare Virtual Machine Image
- Disk Image

Download now - http://www.astlinux.org

Thanks!

--
Kristian Kielhofner

Release: AsterFax Beta 3

AsterFax provides an SMTP Fax gateway for the transmission of faxes using Asterisk. Asterisk is an Open Source PBX (or PABX) which provides all of the functionality of a high end PBX free of charge. AsterFax builds on the services provided by Asterisk to provide a full fledged SMTP Fax Gateway.

What that means is that with AsterFax and Asterisk you can send faxes from you desktop using your standard email client without have to install any software on your desktop.

AsterFax can translate a normal email message into a fax message. You simply enter the destination phone number in the 'To' address, compose your email message and click send. Its that easy.

For more formal correpsondance AsterFax also supports a growing number of file formats such as PDF, Tiff and in the next release MS-Word. Simply attach the file to your email message and it will be sent as a fax message.

Click Here for More Information

Click Here to Download

News: Asterisk 1.2.1 Released

We are proud to announce that Asterisk 1.2.1 has been released!

This release of Asterisk contains a number of bug fixes over version 1.2.0. See the ChangeLog at:
http://ftp.digium.com/pub/telephony/asterisk/
ChangeLog-1.2.1 for more details.

It is available from the ftp.digium.com FTP servers, as well as the Digium SVN servers (under the '1.2.1' tag).

As of this release, Russell Bryant and Kenny Shumard will become the official maintainers of the Asterisk, Zaptel and libpri 1.2.x release branches (with help from the community bug marshals and others, of course). Russell has been the maintainer of the 1.0.x branch for quite some time and Kenny is a Digium employee who has been helping to manage our internal source trees for Asterisk Business Edition and other projects.

Mark Spencer and Kevin P. Fleming

Contest: Win up to $2000 for Asterisk EnterpriseReferences!

Digium is seeking customer success stories. If you have a nominee, you could win a prize of up to $1000 cash or $2000 in Digium hardware. Digium would like to contact these customers for possible use as references for press quotes and case studies. Depending on the customer and the Asterisk implementation, we will award a prize of up to $2000 in hardware or $1000 in cash. If you have any questions or would like more information concerning this contest, please contact contest@digium.com Submission deadline is Wednesday December 14th!

Please email your entry with your contact information, the customers contact information and a brief description of the deployment to 'contest@digium.com'

Submitted customer references should include contact information. Reference information will not be made publicly available until authorized by the entrant and the ultimate customer. The reference information need not be published by Digium for a prize to be awarded. If you have any questions, please email 'contest@digium.com.'

News: VoIP Providers ignore FCC E911 order

Trixter has posted an interesting follow up to the FCC e911 order.

Excerp:

"Under the FCC mandate, VoIP providers shouldn't be marketing their service to customers in non-E911 areas; initially, the FCC wanted to force providers to stop service to existing customers as well as stop marketing. According to media accounts, companies such as 8x8 are waiting for the FCC to respond to their final E911 filings due on the November 28th compliance deadline so they can get clarity on what service will suffice until they can implement a full E911 solution.

A few research firms believe the FCC will not enforce strict compliance with the order so long as VoIP providers demonstrate forward progress in ultimately meeting the E911 service requirements. And the FCC may intentionally take its time in getting around to handing out fines because of Congressional interest in the matter. Bi-partisan groups of Senators and Representatives have expressed their concerns in a rush to implement E911 due to the lack of staff support and potential for "competitive third-parties" - ILECs - to intentionally hinder the implementation of E911.
"

Click Here for Full Article

Release: Sangoma Technologies New "A104d" Card Provides Major Solution For Bicom Systems

New card with hardware- based echo cancellation and voice enhancement configured automatically "right out of the gate".

TORONTO - December 2, 2005 - Sangoma Technologies Corporation (TSXV: STC) (www.sangoma.com), a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure, has provided a significant technical interoperability need for Bicom Systems' SWITCHware solution.



"For a recent client, we needed to deploy an entire broadband telephone solution," says Stephen Wingfield, CEO of Bicom Systems, a London, UK- based VoIP solutions provider. "Sangoma's newly released A104d Series gateway of cards were a key component to our SWITCHware solution and worked "right out of the gate" with little configuration."

"We had confidence that our A104d had the technical capability to provide Bicom System's with immediate ease of installation, quality of service and reliability," adds Sangoma Technologies President and CEO David Mandelstam. "We are pleased now to be placed on their preferred Hardware Providers List for future projects needing our horsepower."

The A104d includes a miniature voice enhancement sandwich board. The voice enhancement capabilities added to Sangoma's standard AFT-based A104 card include: G.168-2002 echo cancellation with 1024 tap/128ms tail per channel on all channel densities, DMF encoding/decoding and tone recognition, voice quality enhancement and adaptive noise reduction.

"Designed at Sangoma's research and development labs, the A104d PCI card is engineered for today's demanding soft PBX, IVR and VoIP applications, such as Asterisk, Yate, and OPAL, offering a new price/performance standard unparalleled in our industry," says Mandelstam.

Announce: VoIP Recording App

Hi,

For those of you interested, there is an open source project that does VoIP recording, database CDR storage and web retrieval.

http://www.oreka.org

It supports raw RTP, SIP and Cisco Skinny (SCCP) by packet sniffing and runs on both Linux and Windows.

Interesting if you don't want to have asterisk manage the call recording itself (e.g box has high load, want to make it more reliable).

Cheers,
Henri Herscher