PJSIP version 0.7.0 is released with ICE and Symbian support.

VoIP IP Telephony @ http://snapvoip.blogspot.com

Some time back, in January, Ravenii pointed me to PJSIP, a free and Open Source SIP stack for Presence, IM and multimedia communication.
Since then the application /library has come a long way. Now the Version 0.7.0 is out. The PJSIP now has ICE and Symbian port and the usual. There are two versions and the following table explains the differences.
What ever the way you plan to go about testing, using the PJSIP, the best place to start is at Getting started document page. It is well written for the purpose and not boringly long.
There is also a mailing list, TRAC development site and..... the blog, where I picked up the info.

What version 0.7.0 brings!

Features	Stable (0.5.x, 0.6.x)	Development (0.7.x, 0.9.x)
PJSIP and PJMEDIA	Yes	Yes
Simple STUN Client	Yes	Yes
Speex version and date	r12136 2006/11	r12832 2007/04
PortAudio version and date	v19 ~2005	r1198 2007/05
PJNATH		Yes
STUN and TURN Library		Yes
ICE Implementation		Yes
Windows, Linux, uCLinux, and MacOS Port	Yes	Yes
WinCE and Windows Mobile Port	Yes	Yes
Symbian Port		Yes
RTEMS Port	Yes	./configure-legacy currently broken

Tags: pjsip, SIP, open source SIP stack, TRAC

TrixBox Appliance! Sold to the first ever customer!

VoIP IP Telephony @ http://snapvoip.blogspot.com
The TrixBox Appliance that is a hardware appliance designed only by Trixbox for Trixbox has been sold to it's first customer at VON.
The product is slated to ship by June 24th 2007, and available in Seven configured forms;
From the base model to the single span T1 in the standard form. There is also a Enterprise version again expanding from Enterprise base to Dual T1 model.

According to the Trixbox Appliance site; (Where you will also be able to find the models and pricing)
TrixBox Appliance features,

• Rackmount Options – The trixbox Appliance can be mounted five different ways: 19” telephony relay rackmount, cabinet slide-mount, wall mount and free-standing floor or tabletop.

• Telephony Expansion Capability – trixbox Appliance can scale up to 4 T1s or 48 analog lines. The trixbox appliance supports Asterisk®-compatible cards from virtually any telephony card manufacturer.

• Enhanced Reliability – With the dual power supply, the trixbox Appliance Enterprise Edition offers high reliability with the ability to run on multiple power sources; a feature you won’t find on any other telephony appliance.

• Data Security – Mirrored hard drives ensure that your data is always safe from a hardware failure.

• Easy Networking – Featuring network connectivity on the motherboard and an onboard 4-port switch, the trixbox Appliance can be networked in a variety of configurations to meet your specific business needs.

• Increased Air Flow – What’s that loud noise you hear? It’s not your trixbox Appliance! With its intriguing one-of-a-kind custom front panel and limited fan noise, the trixbox Appliance was designed to be seen, not heard.

• LCD Display – With a back-lit 4 line LCD included prominently on the front of your trixbox Appliance, you can check the status of the system, including call and queue metrics/details, all in a single glance.

• Quality Hardware – The trixbox Appliance Enterprise Edition comes with pre-configured Sangoma line cards with industry-leading Octasic echo-cancellation hardware inside to give you high quality phone calls.

Customer who purchased the first Trixbox Appliance.

Tags: trixbox, Asterisk, trixbox appliance, Octasic echo-cancellation, Sangoma, VON

How do you Emulate Shared Lines or SLAs in Asterisk

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/.

Tags: Asterisk 1.4, Asterisk SLA, shared lines

FreePBX will support Asterisk 1.4 (Updated)

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.

Tags: FreePBX, Asterisk, asterisk 1.2, asterisk 1.4

To converge or not to converge, with my VoIP?

VoIP IP Telephony @ http://snapvoip.blogspot.com
With VoIP ringing in almost every ear, (By the way today is ears day! June 6th, 6-6)!! Many business and their IT staff is presented with the questions of convergence.
Shall we run voice and data on the same network? or shall we not.
Shall we outsource the VoIP all together or shall we handle our own IPPBX, connected to our traditional PBX during the transition?
Well all those are valid thoughts and questions. But today I am going to lead you to an article that attempt to answer some of the questions. It was published by ZDNet UK. Here is the summery but follow the link for the complete article.

"To converge or not to converge — that is the question. But the answer isn't as clear-cut as we'd like. There are many advantages to convergence, and it appears to be the way of the future, especially in budget-conscious environments.

However, separating voice and data transmissions can provide better security for both. In addition, it may be a more cost-effective alternative — at least in the short run — if the existing network would require a complete redesign to support VoIP.

A single network can make management and troubleshooting easier if IT personnel have the proper training to deal with VoIP technology. But it also makes the network more complex, and there may be a learning curve before the company can realize the full benefits.

As with so many issues in IT, then, the answer to the question is: "It depends." It depends on the network infrastructure you're starting with, your security requirements, budgetary considerations, personnel considerations, and many other factors. It's essential to perform an assessment — not just of the network but of your company's particular needs in regard to both data and voice, and a projection of future needs — before you make the decision."

Tags: hosted VoIP, VoIP business, voip, convergence, pbx, ippbx