Category Archives: Asterisk

Asterisk

A Quick Look at Asterisk Highlights for this Month

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Much has happened with Asterisk this month both in terms of versions released and the upcoming Astricon 2007.

Asterisk Versions Released:

  • Asterisk 1.2.24 and 1.4.10 were released early this month (August 7). Asterisk 1.2.24 was announced to be the final version of Asterisk 1.2 and contained the usual bug fixes. The version will still be supported even though only security releases will be issued from time to time. Support will continue until ALL Asterisk users have migrated to newer versions. Asterisk version 1.4.10, on the other hand, contained both bug and security fixes.
  • Asterisk 1.4.11 was released August 21st. The 1.4.11 release contained numerous bug fixes and a security fix for problems with the chan_sip, which previously caused all SIP dialog history to be stored automatically.
  • Zaptel 1.2.20.1 and 1.4.5.1 were released last week (August 24). Both versions contained bug fixes, specifically in the Makefile of both Zaptel 1.2.20 and 1.4.5.

AstriCon 2007 Developments:

The whole Asterisk community is abuzz with anticipation over the upcoming conference to be held September 24-28 in Arizona. Earlier this week it was announced that registration for the conference was open. New things that can be expected from AstriCon 2007 includes a job fair, an Open Source Telephony Executive Summit, and a commercial track. Interested developers can register for AstriCon 2007 here.

Registrant can choose from the following options:

  • All Access Pass – Allows you to join all all conference and pre-conference activities; Registration Fee: $550
  • Tutorials and Conference – Allows you to join all conference activities ; Registration Fee: $450
  • Conference Only – Allows you to join conference activities held on Thursday and Friday only; Registration Fee: $250
  • Tutorials Only – Allows you to join activities held on Wednesday Only; Registration Fee: $250

Online registrants who register on or before September 24 can enjoy a 20% discount by using “Digium-Astricon-2007″ as discount code. For more details regarding AstriCon 2007 please visit the AstriCon 2007 website.

Digium Now Accepting Submission for the Digium Innovation Awards

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Digium announced a call for entries for the Digium Innovation Awards. Digium’s aim in giving the awards is to “recognize developers, customers and partners for outstanding achievements that are improving business processes, overcoming technology challenges and enhancing the company’s bottom line.” In short, it’s a way to say thank you to Asterisk users and give the deserving ones a time to shine. The Digium Innovation Awards Submission Forms can be downloaded via Digium’s website. Deadline of submission of entries is October 1, 2007. The entries will then be judged and the awards given out on October 31 during the Asterisk World Boston, a special event open to all Asterisk users that will run for two days (October 30-31).

Award categories are as follows:

  • Pioneer Award – Will be given to the most innovative Asterisk implementation.
  • Big Biz Asterisk Award – Will be given to the largest Asterisk enterprise-class installation.
  • ROI Award – Will be given to the entry that can show the best measurable ROI from implementing Asterisk.
  • Inside Out Award – Will be given to the entry that shows the best/most innovative use of Asterisk in a business outside of telecommunications.

Though the award lists no monetary prize it is guaranteed to boost the entrant’s reputation. Aside from the actual presentation of the award during the conference winners will also get a congratulatory press release from Digium, Inc. and will get link love from Digium’s website.

 

Raketu Offers Free International Calls

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Raketu, the known P2P VOIP provider has announced their promo that will allow users to make free international calls to over 40 countries.

Raketu is not your ordinary VOIP provider. On its website, it’s stated that Raketu is where social networking meets communication meets media services in one application. It does not only offer VOIP services – Raketu also has video on demand, IM, SMS, multi-media player, podcasting, slide show, info feeds and other features. And to add to that, they’re offering these features both on their desktop client and as web services through their Rakweb site – a way to use RakeTu services without downloading anything. RakWeb also has a mobile-friendly version that can be found in www.raketu.mobi.

The promotion, which started last August 20, will run for 3 months after a user has made a payment of either $9.95 or $24.95. The three-month counter will start after the user has made the payments. This will allow the user to make up to 1200 minutes per month free calling to locations in 40 countries. Any usage that exceeds this free 1200 minutes will be charged with the regular rate. The promo is not cumulative though, so one user cannot have more than the 1200 alloted free call minutes, even if he makes another payment. This will only extend his original 3-month timer

This promo will only be available to RakOut users: for RakWeb users, regular VOIP rates will apply. Also, all current paying customers will be automatically signed up for this promo from the day the offer started last monday, August 20.

The 40 countries included in the promo are: Argentina, Australia, Austria, Belgium, Brazil, Canada, Chile, China, Cyprus, Czech Republic, Denmark, France, Germany, Greece, Hong Kong, Hungary, Ireland, Israel, Italy, Japan, Korea South, Mexico, Netherlands, New Zealand, Norway, Poland, Portugal, Puerto Rico, Russia, Singapore, Spain, Sweden, Switzerland, Taiwan, Thailand, United Kingdom, United States, US Virgin Islands, Vatican City State, Venezuela

To dial numbers, dial as if you are calling from the US. To make calls to international numbers, dial 011+(country code)+(city code)+(phone number). For North American Number Format, simply enter the 10 digit number: (area code) (number).

Asteriskâ„¢ : The Future Of Telephony book: 2nd Edition Released

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Safari.java.net announced the availability of the second edition of the widely popular and respected book “Asterisk: The Future of Telephony”. This bestselling book has proven to be an excellent resource and guide for building and deploying phone systems with the Asterisk technology. The second edition of the book has been revised to incorporate the features from the 1.4 release of Asterisk software.

If you are planning to work on an Asterisk project or you just simply want to learn the essentials in building systems using the popular open source telephony software, this book is a must-have. Traditional telephone systems are expensive and inflexible because they are usually made with a particular set of vendors in mind. This is one of the problems that is solved by Asterisk’s powerful and highly-configurable offering. But with configurability and power comes the price of complexity and steep learning curve. This is what this book is hoping to address: The book contains many informative material on deploying a complete asterisk-based telephone system. From installing, configuring and integrating your asterisk deployment with your existing telephone system. The book will also help you setup up speech applications like speech synthesis and voice recognition.

The New edition includes the following:

  • A new chapter on managing/administering your Asterisk system
  • A new chapter on using Asterisk with databases
  • Coverage of features in Asterisk 1.4
  • A new appendix on dialplan functions
  • A simplified installation chapter
  • New simplified SIP configuration, including examples for several popular SIP clients (soft phones and IP telephones)
  • Revised chapters and appendicies reviewed and updated for the latest in features, applications, trends and best-practices

The book is targeted to people who are new to Asterisk, but the book assumes that the reader is familiar with basic Linux administration, networking and other IT disciplines. The book is written though with the assumption that the reader is new to telecommunications, both traditional, switched telephony and the new voice over Internet protocol (VOIP).

The book contains 15 chapters of excellent material on Asterisk, including chapters on dialplan, the Asterisk gateway interface, the GUI framework, managing Asterisk sytems, and many other things.

If you have always wanted to try the Asterisk software for your telephone system needs but don’t know how to start the ball rolling, this book is an excellent start.

The Skype Outage: A Reality Check on VOIP Reliability

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When Skype experienced a massive service outage two weeks ago, it sent a kind of panic to its users and the industry watchers. During the outage, the number of Skype users who are online, which usually fluctuates in the number of 5 million to 8 million users, stayed on a flatline at 1 million. For the more than 4 Million users affected by the outage, the service outage caused inconvenience, frustration, and very probably, disruptions to businesses who rely on the popular peer to peer voice over internet application. But beyond these effects to the users, the outage highlights a very important point about the reality of VOIP: as people and businesses become more and more dependent on VOIP technology and services, VOIP reliability will become more and more important.

Two weeks after the outage, the cause is clearer now, having been discussed and analyzed by the blogosphere. The culprit of course lies in Skype’s server software, specifically, on the part of the Skype system that handles user logins. The bug has been sitting in the code and was just waiting to be triggered. And last August 16, Thursday, the trigger happened. Having just received a security update to the operating system, millions of Skype users who are also using Windows almost simultaneously rebooted as a part of the autmated update. This caused a peak in number of users trying to login to the Skype peer to peer network. The added effect of these millions of skype users simultaneously trying to login and the few available node resources (which are important in a peer to peer network) caused a massive outage to Skype’s servers. The outage lasted for almost three days before things were fixed and the root cause of the problem eventually identified.

While it’s clear that the peer to peer nature of the Skype nature was an integral part of the problem that caused the outage (and the Windows udpate causing a reboot on many users the trigger), this is actually of the reality of any distributed software system with a massive user base. And VOIP software fits very well to this profile. So this brings up an important point: if VOIP is indeed the future of voice telecomunnication, the technology must be improved to the point where outages like this would be remote if not impossible. VOIP, before it can become truly mainstream and replace traditional telephone systems must pass the reliability test with flying colors.

Use of Asterisk’s IP Telephony Platform Boosted by Genesys’ Support

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Asterisk’s userbase has gotten even bigger now that Genesys Telecommunications Laboratories, an Alcatel-Lucent company (PARIS: ALU) and (NYSE: ALU), has announced its decision to support the Asterisk® open source IP PBX platform. According to the president and SEO of Genesys, Wes Hayden, they decided to formally support Asterisk to be able to meet customer demands. Furthermore he stated that, “The increased importance of SIP and standards-based technology has paved the way for using open source for contact centres. We have reached a point where Asterisk and SIP are mature, reliable and proven technologies.”

The response of Genesys customers was very positive with companies like Groupama Rhône Alpes Auvergne in France, E.Novation Contact Centers in the Netherlands, and Global Speech Networks in Australia all announcing their own deployment of their use of Genesys on Asterisk. It is expected that more companies will soon follow suit as the use of of open source is highly beneficial to them cost-wise. In addition to these using an open platform gives them much flexibility, which is very important to be able to develop the highly customised environments that is required of contact centre service providers.

The move by Genesys is clearly advantageous not only to Digium and Genesys’s clients but to the VoIP technology as a whole. Since Genesys is the leading provider of software for major contact centres then their use of Asterisk is sure to help hasten the acceptance VoIP technology in the contact centre industry. Currently Genesys is the only company worldwide focusing its entire resources on software meant to manage customer services through the phone, web and email. The customer-base Genesys’ 4,000 clients interact with each day is approximately about 100 million (individual) people. The impact of even just a percentage of the 4,000 Genesys clients shifting to IP technology or at least incorporating it with the existing technology they use will be an unbelievable boost to IP. At the moment though only a few, as mentioned earlier, have actively embraced the new platform. Although contact centres are enthusiastic about adopting this new standard being pushed by Genesys it will of course take some time to transition from their old systems. Transitioning to IP technology is expected by most contact centres to start in about 2 years.

Asterisk 1.6 Updates: Part 2

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Queue changes
Added functionalities include:  a keepstats option to keep queue statistics during a reload,’Strategy’ field to represent the queue strategy in use, an option to run macro when a queue member is connected to a caller, option to control the minimum amount of time between queue announcements for use when the caller’s queue position changes frequently, and other additional information to the queue log.

MeetMe Changes
Added functionalities include: ability to relate callers that come and go from long standing conferences by storing IDs, and application called MeetMeChannelAdmin that does operations on a channel by name, and a ‘C’ option to Meetme which causes a caller to continue in the dialplan when kicked out.

Music On Hold Changes
Added functionalities include: a new option called “digit” that allows callers listening to music on hold to switch music

AEL Changes
Added functionalities include: ability for users to define their own local variables in macros.

Call Features Changes
Added functionalities include: parkedcalltransfers option, ability to have the transferee sent back to the person that did the transfer if the transfer is not successful, support for configuring named groups of custom call features resulting in ability for features to be written a single time, and then mapped into groups of features for different key mappings or easier access control.

Language Support Changes
Added functionalities include: Brazilian Portuguese  and support for the Hungarian language for saying numbers, dates, and times.

Tribox 2.4 Beta Released

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The latest version of Tribox 2, which is the most popular Asterisk-based telephony platform in use now, was released by Fonality last week. Tribox 2.4 is includes the latest releases of CentOS 5, Asterisk 1.4 and FreePBX 2.3. Tribox 2.4 is still just a beta version with the general release set to be announced shortly after the latest release of Aterisk 1.4 comes out and they are able to incorporate needed changes. The newest release boasts an overall better performance especially in terms of choice/flexibility, compatibility, and usability. Applications included in Tribox 2.4 Beta include Apache, Asterisk (version 1.4), FreePBX, Flash Operator Panel, MySQL, phpMyAdmin and SugarCRM.

According to Andrew Gills, the founder of Tribox,they are just beginning to develop and deliver according to their user’s demand. As he said, “Listening to the trixbox community is our first priority. They wanted to see Asterisk 1.4 incorporated into trixbox, so we are delivering just that. This beta release is just the first step and by having trixbox 2.4 based on CentOS 5, trixbox is compatible with the newest hardware on the market.” So while it seems that the latest version has already increased Tribox’s compatibility we can be pretty sure that the compatibility issue is just a step towards a larger goal.

For more information on Tribox please visit the Tribox website.

Asterisk 1.6 Update

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Asterisk 1.6 is starting to shape up with some features of Asterisk 1.2 and 1.4 already successfully merged and new dialplan functions already in place. Listed below is a summary of latest update in the functionality CHANGES file of Asterisk.

AMI – The manager (TCP/TLS/HTTP)
Added functionalities include: TLS support for the manager interface and HTTP server, URI redirect option for the built-in HTTP server, GetConfigJSON (action that returns the contents of an Asterisk configuration file in JSON format), a “Bridge” action which allows you to bridge any two channels that are currently active on the system, and a “ListAllVoicemailUsers” action that allows you to get a list of all the voicemail users setup.

Dialplan functions
Added functionalities include: a DEVSTATE() dialplan function that allows retrieval of any device state in the dialplan, a new option to Dial() for telling IP phones not to count the call as “missed” when dial times out and cancels, LOCK(), TRYLOCK(), and UNLOCK().

CLI Changes
Added functionalities include a ‘core show channels count’ CLI command and the ability to set the core debug and verbose values on a per-file basis.

SIP changes
Added functionalities include: an improved NAT and STUN support, new way of matching incoming requests, “busy-level” for setting a level of calls where asterisk reports a device as busy, new realtime family called “sipregs” (used to store SIP registration data), more support for T.140 realtime text in SIP/RTP, new variables for call transfers, and a new header that is displayed for cancelled calls answered by another phone.

IAX2 changes
Added functionalities include: trunkmaxsize configuration option to chan_iax2, srvlookup option to iax.conf, and support for OSP.

DUNDi changes
Added functionalities include: the ability to specify arguments to the Dial application when using the DUNDi switch in the dialplan, ability to set weights for responses dynamically, dialplan functions (DUNDIQUERY and DUNDIRESULT) that will allow you to initiate a DUNDi query from the dialplan and find out how many results there are as well as access each one.

ENUM changes
Added functionalities are two new dialplan functions (ENUMQUERY and ENUMRESULT) that will allow you to initiate an ENUM lookup from the dialplan access the results without doing multiple DNS queries.

Voicemail Changes
Added functionalities include:  the ability to customize which sound files are used for some of the prompts within the Voicemail application, the ability for the “voicemail show users” CLI command, “tw” language support, support for storage of greetings, and the ability to customize forward, reverse, stop, and pause keys for message playback.

More of the latest updates on my next post. To view the detailed updates including modified and removed functionalities you can check out their changes file.

Skype Inks Deal with SpinVox

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SpinVox, the pioneer and leader in Voice-to-Screen messaging, just announced that it has made a deal with Skype to offer its voicemail-to-text service for the VOIP software. Initially, the service will be available in English, Spanish, German and French.

The SpinVox voice-to-message is a simple and practical solution that converts your voicemail to text and delivers them to your mobile phone, email inbox or blackberry device. This makes the voicemail experience easier and more manageable since it makes the message more accessible.

In its deal with Skype, SpinVox will allow Skype users to have all their voicemail messages converted and sent to their cellphones as an SMS when they are not on their computer. The service also supports the Skype Caller ID, which it will include on the SMS message of the converted voicemail. This gives skype users even more immediate and convenient access to their messages even when they are on the move. This also enhances the apparent reliability of the skype platform, since people will be more confident that their messages will reach their contacts immediately even when the latter are offline or away from their computer.

Because it uses the industry standard SMS to deliver the converted message, SpinVox will reach any mobile phone, obviating the need for skype users to download new software onto their phone to benefit from this voice-to-text messaging service.

This is a powerful example of how SpinVos is enabling leading Web 2.0 services to extend their reach from PC to mobile,” said Christina Domecq, CEO and co-founder, SpinVox. “It’s also proof of our ability to integrate our technology with an existing servie platform, whether fixed-line, cable or wireless and now internet calling, with market leaders Skype.

The globan nature of both Skype and SpinVox services makes for a natural and exciting partnership that will deliver a simpler, more effective communication experience for both the caller and the Skype user receiving a message,” continued Domecq.

Though the service will launch at the end of the year, this deal is a step towards a world of convergence and connectivity. Soon, skype users will become even more connected and always updated with their voice mail messages, thanks to SpinVox’s voice-to-text services. Having seen the renewed focus of the industry to bring new things to the age-old voice mail technology, we can only expect more innovation in this space.