All posts under tagged ‘Asterisk 1.4’

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Practical Asterisk 1.4 being (written) published online

Source: snapvoip.blogspot.com

If you can read German and interested in Asterisk, you may have already read the book by Stefan Wintermeyer, Asterisk 1.2 + 1.4. Now this book is being translated by Stephen Bosch under the name Practical Asterisk 1.4. This early version is a work in progress and online to find bugs and get your feedback.
Stephen plans to get the translation completed by end of 2007. So get over there and proof read or give your feed back, so we can have another good Asterisk 1.4 book soon. The book is being published under GNU Free Documentation License.

How do you Emulate Shared Lines or SLAs in Asterisk

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com
"The term "shared lines" in Asterisk means implementing the functionality of having a line appearance on a phone that maps to a trunk. So, multiple phones can have a "line1" appearance that is mapped to a SIPi trunk, for example. Then, when any phone is using this line, the line shows as in use on all of the other phones."

Did that confuse you? It did in the first glance but after reading Russell’s post about it made a whole lot of sense and the configuration example and the walk through did a lot of help!
So head over and read the article and the Rewritten SLA is a part of Asterisk 1.4 source tree.
For more information, look at the following files in the Asterisk 1.4 source tree:

* configs/sla.conf.sample
* doc/sla.pdf

Feel free to report any problems to http://bugs.digium.com/.

Published on June 6th, 2007 under , ,

FreePBX will support Asterisk 1.4 (Updated)

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com
Update:
One of the FreePBX people, James had this to add! Thanks James. It is bit hard to think in another language and write in another.!;

"Topic sentence is a little confusing. freePBX isn’t ‘dropping’ support for asterisk 1.2. It’s adding 1.4 support in freePBX version 2.3"

If you did not already hear that it was decided at the recent Asterisk Developers Conference, to drop Asterisk 1.2 and move whole heartedly to Asterisk 1.4
“One decision that we took was to stop maintaining 1.2 as a current release from August 1st 2007. At that date, we will move 1.2 of Asterisk, Asterisk-addons, libpri and zaptel to security maintenance status. 1.4 will at that point be the recommended release.”
So where do we go? and the answer comes from Phillipel at the FreePBX in behalf of the team.But before lead you to the answer, let me remind you that the FreePBX team have been supporting you all these years. Providing you with guidance and of course FreePBX. Now the FreePBX team needs your help and support or we need our support. Here is the gist of Phillipel’s post;
"So what about freePBX® and Asterisk 1.4?

As a community leader we will respect Digium’s desire to get more users onto Asterisk 1.4 and will do our part to support their efforts. We’ve been planning to enable Asterisk 1.4 on our freePBX® 2.3, which is the current development branch. freePBX® 2.3 is currently very stable on Asterisk 1.2.

What this means

We will branch the development trunk into its proper 2.3 release candidate shortly, and start the formal beta program. If you run the freePBX® 2.3 beta on Asterisk 1.2, you should have a very stable system. If you choose to start “kicking the tires” with Asterisk 1.4, you should consider it alpha quality. We will address bugs as they surface and quickly find out just how much work there is to do.

Things are going to be heating up for the team with this plan; we’ll try to get you more details soon. In the mean time – moral support to help accelerate the effort is always appreciated, especially the kind that you can show with that nifty donate button off on the left."
So head over to FreePBX, read more and click that button.
Also do not forget to leave a note for Rob Thomas. He has had a hard time lately. All my sympathies are with him. I am so sorry.

Published on June 6th, 2007 under , , ,

Web-Meetme, Scheduled conferencing for Asterisk

Source: snapvoip.blogspot.com

I came across this package a while back and have been playing a bit with it. But I really got it working after Dan Austin released the version 3.0 just a two days ago. Then again I had to switch to SVN version in order get it completeley working.
Managed conferencing solution using Asterisk, Web-MeetMe and a database-driven scheduler. From the projects readme.
Web-MeetMe for Asterisk
Web-MeetMe is a collection of PHP pages that leverage
the database for scheduling and the Asterisk Manager interface
to monitor and control active conferences.

The scheduler currently supports these features:
+ Conference number conflict avoidance
+ Recurring conferences
* Daily, weekly or bi-weekly
+ Enforces a user password if a admin password
is set
+ Seperate views for past, current and future
conferences
+ Editing future conferences
* Includes a series of conferences
+ Deleting past conferences
+ CDR-like view of past conferences
* Includes a link to conference recordings
+ One click features to-
* Extend an active conference (time)
* Kick out all participants
* Out-call to include participants
+ Optional early join features
* fuzzystart allows a caller to enter a
room early (in seconds, set in cbmysql.conf)
* earlyalert notifies a call that their
conference has not yet started, if they
enter a valid conference number and the
conference is scheduled to start soon.
(in seconds, set in cbmysql.conf)

App_cbmysql
When combined with a database, MySQL or Postgress,
CBMySQL can authenticate a caller, verify that the conference
they are attempting to join is scheduled and enforce the
maximum participent count.

CBMySQL can distinguish between a conference
administrator and a standard user based on pins. Different
options may be set in the configuration file to join the
caller to the conference based on their status. A standard
user might be joined in a listen only mode as an example.

CBEnd
CBEnd is a small PHP script that monitors for active
conferences. When it identifies a current conference, it checks
the scheduler database and enforces the conference duration.
The default behaviour is to announce that the conference is to
end five minutes before the scheduled end time. This allows the
conference owner or administrator to update the conference using
Web-MeetMe to extend the duration.

CBEnd uses the Asterisk Manager interface to issue commands
to the conferencing application. The current commands include
identifying active conferences and the ‘Kick all’ command.

This script is also responsible for maintaining the
CDR tables. The script is optional, but not using it limits
the overall system functionality.

** Installation
* Dependencies
- Asterisk 1.4 or later
- MySQL 3.23 or later.
!! Other databases may work, but testing
!! and development have been against MySQL
- PHP 4.3 or later
!! Reports depend on mod_gd and GD-2.0.28
!! or later
!! PEAR::DB (php-pear)

* Install Web-MeetMe
Download the latest Web-MeetMe tgz file from
http://www.sf.net/projects/web-meetme and extract it to your webservers
document root.
* Setup the database
Create a table called booking in your database. The
table is described in ./cbmysql/db-tables-v5.txt

* Compile and install CBMySQL
App_cbmysql is now included in the web-meetme package,
located in ./cbmysql. To install just run make; make install

Copy the sample cbmysql.conf to /etc/asterisk and create
a dialplan similar to the one in cb-extensions.conf.sample
Modify the settings to suit your system. The location of the
mysql.sock file is likely not correct, check /etc/my.conf for
the correct location.

Modify the ../web-meetme/lib/defines.php to match you
database settings.

Copy ../web-meetme/phpagi/phpagi.example.conf to
/etc/asterisk and modify it to match the settings in your
manager.conf

If you intend to use the authentication functions,
it is strongly recommended that you use SSL and force all
conenctions to the Web-MeetMe pages to use HTTPS.

* LDAP or Active Directory integration
Edit ./lib/defines and set the AUTH_TYPE to adLDAP.
Set the ADMIN_GROUP to appropriate list, the default is
"Domain Admins", but you might not want the network folks
controlling your telephony systems. Optionally you can
chanege the browser session timeout.

Edit ./lib/adLDAP and replace all instances of
"yourdomain" and "yourserver" with the values for your
network.

* Usage
If all of the steps have been followed, you should be
able to open your browser, connect to the Web-MeetMe page and
schedule a conference. A new conference start time defaults
to the time it is scheduled, so modify it if needed. Any
required fields that are left blank will be auto-generated.
Note the conference room number and password.

Dial the number you have assigned for conferencing, and
try the conference you have just added. If the time and password
are correct, you will be joined to the conference.

You can now use the Web-Meetme monitor function to manage
your conference.

Links;
Web-Meetme download
Asterisk IPPBX

Published on January 4th, 2007 under , , , , ,

Asterisk for OpenWRT, My linksys wireless router is a Asterisk 1.4 server

Source: snapvoip.blogspot.com

As you know Asterisk have been loaded on everything from servers to workstations, routers and virtual machines. But you may know that it is possible to run the Asterisk server from a wireless router too. My favorite wireless router is WRT54G and I have been using OpenWRT so much that I had to get multiple WRT54G routers so that I had a working wireless router for the rest of the household that use notebooks.
So while browsing around OpenWRT forums, I noticed that User Sskol has written how to get Asterisk 1.4 on white Russian RC6 (White Russian is a one version of OpenWRT) and I got to work following his instructions, only to find user Zandbelt has done all the work. He/She also have package for the Asterisk GUI.
So now I am playing with his version. If you are so inclined please follow the links and make your wireless router speak Asterisk. But be warned, these may damage your router and render it non functional.

Links;
OpenWRT Developer Forum discussion
Zandbelt’s work
OpenWRT site

Published on January 1st, 2007 under , , ,

One last beta before the final Asterisk 1.4 release

Source: snapvoip.blogspot.com

Asterisk has released information about a new release of Zaptel 1.2.12 and Asterisk 1.2.14 together with Zaptel 1.4.0-beta3 and Asterisk 1.4.0-beta4.
According to the Asterisk Development Team, This will very likely be the last beta release of Asterisk 1.4 before the final release, which is targeted for next Friday.

Also please check Asterisknow for Asterisknow versions of the above. Links at the end.

Here is the complete news release;

Zaptel 1.2.12 Released

The Asterisk Development Team is pleased to announce the release of Zaptel 1.2.12.

This release contains a number of updates:

* compatibility with Linux kernel 2.6.19
* bug fixes to the Xorcom Astribank driver (XPP)
* various other bug fixes

Thanks for supporting Asterisk and Zaptel!

Asterisk 1.2.14 Released

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.14.

This release contains a number of updates:

* a bug fix for the ExternalIVR application and addition of ’silence’ sound files to support it
* various SIP interoperability improvements
* memory and dialog leaks in the SIP channel driver
* a fix to music-on-hold random mode that was not really random
* an improvement to app_voicemail to ensure that the message duration is properly included in email notifications when voicemail messages are forwarded
* corrected a segfault issue during reload of the PostgreSQL CDR driver
* a change to no longer include a header file that does not exist on Linux kernel 2.6.18 (and caused a problem on Fedora Core 6)
* many other bug fixes

Thanks for supporting Asterisk and Zaptel!

Zaptel 1.4.0-beta3 Released

The Asterisk Development Team is pleased to announce the release of Zaptel 1.4.0-beta3.

This release contains a number of updates:

* compatibility with Linux kernel 2.6.19
* bug fixes to the Xorcom Astribank driver (XPP)
* support for Digium’s TE110P Rev C, VPMOCT064 and new revisions of the S110M and S400M FXS modules
* various other bug fixes

Thanks for supporting Asterisk and Zaptel!

Asterisk 1.4.0-beta4 Released

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.0-beta4.

This release contains a number of updates:

* a bug fix for the ExternalIVR application and addition of ’silence’ sound files to support it
* various SIP interoperability improvements
* memory and dialog leaks in the SIP channel driver
* a fix to music-on-hold random mode that was not really random
* an improvement to app_voicemail to ensure that the message duration is properly included in email notifications when voicemail messages are forwarded
* corrected a segfault issue during reload of the PostgreSQL CDR driver
* a change to no longer include a header file that does not exist on Linux kernel 2.6.18 (and caused a problem on Fedora Core 6)
* logging of dynamic queue member addition and removal in queue_log
* a minor redesign of many CLI commands to be more similar to previous Asterisk releases
* significant improvements to IMAP storage support for voicemail
* a change to the SIP channel to avoid offering formats (codecs) that cannot be transcoded due to lack of available transcoders (along with dynamic activation/deactivation of transcoders)
* support for G.722 16KHz (wideband) audio passthrough, recording and playback
* support for standard prompts in G.722 format
* many other bug fixes

Some of the changes in this release are behavior modifications from the last release; please review the UPGRADE.txt file.

This will very likely be the last beta release of Asterisk 1.4 before the final release, which is targeted for next Friday.

Thanks for supporting Asterisk and Zaptel!

Links;
Asterisk IP PBX And Zaptel

Asterisknow

Published on December 19th, 2006 under , , , , , , ,

Install Asterisk 1.4 and Asterisk GUI afterwards.

Source: snapvoip.blogspot.com

AstRecipes had two recipes for Asterisk users or budding Asterisk users. (And one more if you need to uninstall older Asterisk). The document carries one through steps needed to install new Asterisk 1.4 on your box. According to the document the installation / compilation is a bit different from the Asterisk 1.2.
The icing on the cake is that this allows you to install Asterisk IPPBX on a box already populated wit TrixBox!
It is also advised that you remove Asterisk (older version) before attempting to install/ compile the new version. The tutorial uses the following versions of the application;
Asterisk Version 1.4.0-beta3
Zaptel Version 1.4.0-beta2
Libpri Version 1.4.0-beta1

Once that is done you can install the new ajax gui for the embedded web server on the Asterisk server / machine. Here too yo are guided through necessary routes get the application installed and to look at your Asterisk 1.4, through your favorite web browser.

Links;
Install Asterisk 1.4 (beta)
Install Asterisk GUI
Remove Asterisk

Published on November 15th, 2006 under , , , , ,

Install Asterisk 1.4 and Asterisk GUI afterwards.

Source: snapvoip.blogspot.com

AstRecipes had two recipes for Asterisk users or budding Asterisk users. (And one more if you need to uninstall older Asterisk). The document carries one through steps needed to install new Asterisk 1.4 on your box. According to the document the installation / compilation is a bit different from the Asterisk 1.2.
The icing on the cake is that this allows you to install Asterisk IPPBX on a box already populated wit TrixBox!
It is also advised that you remove Asterisk (older version) before attempting to install/ compile the new version. The tutorial uses the following versions of the application;
Asterisk Version 1.4.0-beta3
Zaptel Version 1.4.0-beta2
Libpri Version 1.4.0-beta1

Once that is done you can install the new ajax gui for the embedded web server on the Asterisk server / machine. Here too yo are guided through necessary routes get the application installed and to look at your Asterisk 1.4, through your favorite web browser.

Links;
Install Asterisk 1.4 (beta)
Install Asterisk GUI
Remove Asterisk

Published on November 15th, 2006 under , , , , ,

Install Asterisk 1.4 and Asterisk GUI afterwards.

Source: snapvoip.blogspot.com

AstRecipes had two recipes for Asterisk users or budding Asterisk users. (And one more if you need to uninstall older Asterisk). The document carries one through steps needed to install new Asterisk 1.4 on your box. According to the document the installation / compilation is a bit different from the Asterisk 1.2.
The icing on the cake is that this allows you to install Asterisk IPPBX on a box already populated wit TrixBox!
It is also advised that you remove Asterisk (older version) before attempting to install/ compile the new version. The tutorial uses the following versions of the application;
Asterisk Version 1.4.0-beta3
Zaptel Version 1.4.0-beta2
Libpri Version 1.4.0-beta1

Once that is done you can install the new ajax gui for the embedded web server on the Asterisk server / machine. Here too yo are guided through necessary routes get the application installed and to look at your Asterisk 1.4, through your favorite web browser.

Links;
Install Asterisk 1.4 (beta)
Install Asterisk GUI
Remove Asterisk

Published on November 15th, 2006 under , , , , , ,

Install Asterisk 1.4 and Asterisk GUI afterwards.

Source: snapvoip.blogspot.com

AstRecipes had two recipes for Asterisk users or budding Asterisk users. (And one more if you need to uninstall older Asterisk). The document carries one through steps needed to install new Asterisk 1.4 on your box. According to the document the installation / compilation is a bit different from the Asterisk 1.2.
The icing on the cake is that this allows you to install Asterisk IPPBX on a box already populated wit TrixBox!
It is also advised that you remove Asterisk (older version) before attempting to install/ compile the new version. The tutorial uses the following versions of the application;
Asterisk Version 1.4.0-beta3
Zaptel Version 1.4.0-beta2
Libpri Version 1.4.0-beta1

Once that is done you can install the new ajax gui for the embedded web server on the Asterisk server / machine. Here too yo are guided through necessary routes get the application installed and to look at your Asterisk 1.4, through your favorite web browser.

Links;
Install Asterisk 1.4 (beta)
Install Asterisk GUI
Remove Asterisk

Published on November 15th, 2006 under , , , , , ,

Digium launches Asterisk 1.4 version

Source: voipcentral.org

digium

Digium Inc., the creator of Asterisk technology has announced the launch of its Asterisk 1.4 version one year after the release of 1.2 version.

It is the major open source VoIP IP-PBX system that consists of 20 new functionality including IPFAX compatibility, unified messaging capabilities and Gtalk protocol compatibilities.

Features of Asterisk 1.4

1. Generic jitter buffer improves connections for all interfaces, including other VoIP protocols and TDM interfaces. Therefore, the users can get better call quality during network congestion.

2. Pass through ITU Standard T.38 fax calls enables Asterisk server to pass fax calls through to a fax machine. This feature was not available in the early version.

3. The Call to IM system connects Asterisk calls to Jabber or other IM software that supports Jingle protocol.

4. Built-in voicemail system retrieves voicemail through IMAP on any IMAP-compliant storage system.

5. Asterisk Extension Language Version 2 makes programming and dial plan configuration easier.

Asserting version 1.4 is the best version of Asterisk, companys president Mark Spencer comments,

With the support of the Asterisk community, we have been able to develop an advanced platform that will make it even easier for users to migrate to VoIP, especially those in the enterprise community.

Read

Published on September 14th, 2006 under ,

Asterisk Beta 1.4 by the end of the week and an interview!

Source: snapvoip.blogspot.com

Sineapps has posted a and interview with Kevin Fleming, a senior software engineer at Digium.
But what stood out in the article was;

Question 2: So, a couple of things point to the fact that the beta of Asterisk 1.4 might be coming out this week, can you confirm this?

Yes. We will be producing the first beta of Asterisk 1.4 by the end of this week, and then actively working to resolve all known issues as quickly as possible.

I think the interview itself a good one as always. Also those who is wondering what ASTERISK 1.4 may bring, here is an answer from the post;

Question 6: What are some of the changes coming in the version 1.4 release (compared to 1.2)

Well, the list is quite long, but here are about twenty that we’ve already identified (and we haven’t done an exhaustive review yet):

Generic jitter buffer
Variable Length DTMF (proper RFC-2833 support)
Asterisk Extension Language Version 2
Shared Line Appearance support
ODBC() dialplan function
T.38 FAX passthrough support
IAX2 scalability improvements
Re-architected build system with better maintainability and portability
Jabber/Jingle/GoogleTalk support
New high-quality sounds in English, French and Spanish
IMAP storage support for voicemail
RADIUS support for CDR storage
SNMP monitoring
HTTP Asterisk Manager Interface (with AJAX components)
SIP transfer interoperability improvements
Cisco SCCP (Skinny) channel driver improvements
Whisper paging
Better language support for speaking dates, times and numbers
IAX2 media-only transfers
Memory usage and thread locking reduction
Addition of a simple ‘users.conf’ configuration file for SOHO/SMB users
RTP native bridging

Published on September 7th, 2006 under , , ,

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