Digium's VON Announcements Part 1: Asterisk v. 1.4

The new version of Asterisk is packed with so many features that we can only highlight the most important of them.

by Alex Goldman ISP-Planet Managing Editor [September 12, 2006]

It's the week before VON, everybody at Huntsville, Ala.-based Digium, the private company that is the creator and primary developer of Asterisk, "The Open Source PBX", is quite busy. But Kevin Fleming, officially Digium's senior software developer and unofficially the company's number two, is taking some time to update us on the latest version of software.

Version 1.4, he says, will be demonstrated at VON Fall 2006, but will be delivered in early October. Currently, final bug fixes are underway. "It's our first major release since version 1.2, in November, 2005," he says.

He explains that Asterisk's version numbers follow Linux version numbering (thus, each beta release has an odd end number, and each final release has an even end number).

The final release will have at least 29 changes, but Fleming says seven are key:

1) Generic jitter buffer. In the past, Asterisk has had this but only for its own Inter-Asterisk eXchange protocol (known as IAX, now at version 2). Now, the jitter buffer will improve connections for all interfaces, including other VoIP protocols and TDM interfaces.

Fleming says this improvement was contributed by a community member who are a major call center in Europe and who do Asterisk development work for business customers. "There was a huge demand and they had the people with the talent and the desire to do it."

Asked whether the company gets public credit for their work, Fleming says that Asterisk code comes with a credit file. "In this case, they contributed a significant amount of code. It's complex. It has to interact with all the channel drivers that we have."

2) Dial plan programming language. The original code was written by Mark Spencer, the company's president and the creator of Asterisk, and included as an experimental feature in Asterisk v. 1.2. A community member, Fleming says, took the code and made sure that it supports everything Asterisk needs. "They are now an employee of Digium," Fleming adds.

The programming language should be easy to use for anyone who has programming experience in C or Java, Fleming says.

Dial plans are not just call forwarding. Such features are already part of Asterisk. The dial plan for a small business would map each phone number called to a specific phone and also provide business features from automatic call distribution to least cost routing.

3) Pass through ITU Standard T.38 fax calls. The Asterisk server can now pass fax calls through to a fax machine. The project is not yet ready to terminate fax calls. "In the past, if you were using Asterisk, you would have needed a traditional phone line for your fax machine," Fleming explains.

So is the community working on terminating fax calls and sending faxes to computers? "The community has decided that termination has value and so have we."

The problem is licensing. Asterisk software is open source. A license holder contributed the pass through code but, obviously, is unwilling to provide the modem library. Furthermore, for speeds higher than 14.4 Kbps, the Asterisk project would have to deal with other code owners, who have patents.

4) Call to IM. Asterisk calls can be connected to Jabber (open source instant messaging) and any other IM software that supports the Jingle protocol.

Asterisk servers can also appear on the Google Talk network. Asked whether that network is proprietary, Fleming explains, "the client is only available for Windows, but it was built on fairly open specifications that made interoperability possible."

5) Built-in voicemail system. "Until recently," Fleming explains, "you could either store voicemail as files on the Asterisk server or on an external database. Now voicemail can be retrieved via IMAP on any IMAP-compliant storage system. The benefit here is this: unified messaging."

Fleming explains that voicemail is stored in the e-mail database as audio attachments. File sizes depend on the attachment type, varying from larger, uncompressed .wav files to small, compressed .gsm files.

6) Whisper paging. This is a feature that's becoming standard in the PBX business market. It allows an assistant or colleague to talk to someone else in the same office when they're on a call, perhaps conveying time-sensitive or important information, without the person on the other end hearing what's said, which is important if the information is "your wife's on the other line and. . ." or any other personal event.

Fleming says that Asterisk already had a supervisor function, developed for call centers, that allows a supervisor to listen in on a call. "So I decided to implement this and I did it. It was relatively straightforward, though not trivial."

7) Improved sound prompts. Digium works with a specific voice talent for sound prompts, and hired her to re-record all voice prompts in a single session (lasting three days). Asterisk now includes higher quality sound files. In addition, companies can pay her to record specific messages (at rates starting at $12 per prompt, up to 20 words, with volume discounts available). She trained in Spanish (though she doesn't speak it) and recorded prompts in Spanish as well. In addition, Digium hired a separate French talent to record French prompts.

Finally, the functionality in Asterisk that handles the speaking of dates, times, and numbers has been made more flexible, allowing the community to generate rules covering the grammar of other languages.

— End