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May 31, 2005

Flash Operator Panel: Version 0.21 released

Version 0.21 is out. It has lot of changes and new features. You can see them at http://www.asternic.org/CHANGES.html

You can also rate the project on freshmeat (please be kind):

http://freshmeat.net/rate/53045/

Updated info from asterisk-users:

After a while, version 0.21 of the Flash Operator Panel is out.

Flash Operator Panel displays information about your Asterisk PBX activity in real time via a standard web browser with Flash plugin. It can monitor several asterisk servers at once. It can integrate with CRM software, by poping up a web page (and passing the CLID and CLIDNAME) when a specified button is ringing. It also can be used to enable click-to-dial for web based applications.

It can monitor almost any asterisk channel type available: ZAP, SIP, IAX2, H323, OH323, MGCP, CAPI, MODEM/I4L, VPB, mISDN, etc.

Monitoring Features:
Monitoring of Agents (logged in/off)
Monitoring of Queues
Monitoring of Parked calls/slots
Monitoring of conferences
Shows ip address of sip/iax2 peers
Shows sip/iax2 status/reachability
Shows callerid/called number
Shows timers/countdown for absolute timeout calls or parked slots
Shows statistics on agents and queues
Available Actions:
Hangup a channel
transfer via drag&drop
originate via drag&drop
Set absolute timeout when transferring
Set callerid when transferring
Reload Asterisk
Mute/Unmute meetme participants
Barge-in on a call (optionally barge-in muted to avoid being noticed)
Commands optionally restricted by security code
Commands optionally restricted to a specific button/channel
The new version has lots of new features, such as:
REGEXP buttons (they replace wildcard buttons)
Set timeout for transferred calls
Change state/label/text for buttons based on astdb values or dialplan userevents
Fire popups from the dialplan passing any channel variable to the web application

Best regards,

--
Nicolas

Pictures of the Digium booth at ISPCon 2005

Hello Everyone,

Even though a lot of it was a bit last minute, several of us from the community made it to Baltimore to help Digium with their booth at ISPCon. It was a great time.

Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian Kielhofner (me), and John Todd (not pictured) were there (as well as others), and some pictures were taken (the up close ones of me were very unfortunate) :) .

I was there demoing AstLinux on the Soekris Net4801, and Brian Capouch was showing off the WRT54GS running Asterisk (as well as some of his INSANE home automation work).







Anyways, on to the link:

http://www.krisk.org/gallery/ispcon05

Enjoy!

--
Kristian Kielhofner

Open Source SIP/IAX/Zeroconf soft phone for MacOS X

I am responsible for the SFLphone project (http://www.sflphone.org/).
FLphone is an open source software phone aimed to provide a professional look & feel and features. It currently runs on Linux but there is a lot of job being done. It supports SIP, and IAX is planned for one of the next two releases. Also Zeroconf is to be supported as soon as we finish migrating our audio layer to the Portaudio library, this shouldnt take more than a few days. Our code has been rewritten a lot since the last 0.3 release, so we can support many protocols/GUIs/audio drivers easier.

We are currently working at a MacOS X version of the phone. Most of the job has done, we just have a few os-specific issues to fix : for example dealing with the mac's 32-bit float audio format and most codec's 16-bit int. I suppose there is an API to convert between these formats in MacOS X, but we have no clue.

Also, it's quite easy to plug a new GUI to our code, that would permit plugging a native Cocoa GUI instead of the current QT one. But once again, our programmer does not have experience with this kind of technologies.

This is a call, I know there may be people here interested in such a project. So if anyone would like to help us by contributing code, graphics, ideas, coffee, remarks or anything, feel welcome !

Our mailing lists are reachable at :
Developers: http://forge.novell.com/mailman/listinfo/sflphone-dev
Users: http://forge.novell.com/mailman/listinfo/sflphone-user
And finally our generic contact address, contact "at" savoirfairelinux dot com

Regards and happy asterisk-ing,
-Jerome

May 27, 2005

Cisco Call Manager & Asterisk for Voicemail

Shaun Ewing has posted detail of a pdf he's written for integrating Cisco Call Manager & Asterisk:

Okay, I've whipped up a little guide.

It assumes you have a working oh323 configuration on Asterisk. I'd appreciate any feedback.

It's not all that well laid out (I created it in around 10 minutes) and some if it (dialplan, etc) is specific to my configuration (four digit extensions, phones starting with 7, system features in 88xx and 89xx), but it should be enough to go by:

http://asterisk.edropbox.net/ccmasteriskvm.pdf

-Shaun

May 26, 2005

Digium Announces: Asterisk Business Edition

A Professional-grade version of the Asterisk PBX!

Digium, the leader in open source telephony, announces Asterisk Business Edition, a professional-grade version of its acclaimed open source PBX for the Linux operating system. This version provides tested reliability of critical functions and features, tailored for small- and medium-sized business applications.

An all-new Asterisk technical manual and quick-start documentation supplements the package, making Asterisk even easier to install, configure, and use. Asterisk Business Edition is backed by Digium's professional support team with a full one year limited warranty. This provides enterprise environments with a PBX and telephony platform suitable for critical business applications.

 

Digium's comprehensive test program ensures Asterisk Business Edition's reliability, performance, and interoperability with key hardware, software, and protocols. Digium hardware cards are tested for full compatibility with Asterisk Business Edition, as are several select models of servers, VoIP, and TDM devices. All major software features in Asterisk Business Edition are thoroughly tested for functionality and reliability. Test bed systems are also subjected to extreme stress conditions using Empirix test equipment to simulate hundreds of thousands of calls in various real-world combinations and configurations.

Click Here for More Info

 

May 25, 2005

Asterisk based Call Accounting software

Hello Asterisk community,

After numerous request from various companies where we have implemented * as a phone system and also from many other * users all over the world, yesterday we released the 1st version of Asterisk module for Call Accounting Mate (www.callaccounting.ws) . As some of you know we also use Asterisk internally as our phone system and as developers for Call Accounting Mate, we felt it was necessary to implement a decent Call Accounting software for *. Call Accounting Mate runs on Windows and is completely web based. It ships with the necessary source files and Asterisk modules to interface Asterisk via tcpip to Call Accounting Mate.

We have set up a Asterisk - Call Accounting Mate forum so we can gather input from the Asterisk community. You can access the forum at:
http://www.callaccounting.ws/forum/index.php?board=5.0

Regards,

San Singhania
Tel : +1 718 5762066
 
 

May 24, 2005

MacOS X build for Asterisk 1.0.7 available

A MacOS X build for Asterisk 1.0.7 is now available for download at

http://www.astmasters.net/Asterisk-1.0.7-OSX-build-BETA.3.tar

Instructions are here.

Feedback collected/discussed on the Macintosh Asterisk Mailing List

Previous pre-1.0 releases of Asterisk have been rather unreliable on MacOS X. The 1.0.7 release looks very promising, however. If it proves to be reliable, it will form the basis for a new MacOS X installation package.
 

May 23, 2005

E911 on the Asterisk Open Source PBX

Enhanced 911 service or E911 service is a North American telephone network feature that automatically associates the physical address of a caller with the calling party's telephone number. This is generally done by a form of reverse telephone directory that is supplied by the telephone company. This provides emergency responders with the location of the emergency without the person calling for help having to provide it.1

 

In the late 90's some states started to require that E911 functionality be provided for when installing PBX telephone systems. Regulations and requirements very by state. In Washington State, PBX phone systems are required to be E911 compatible in certain circumstances when installed in schools, or when installed in multi-tenant office buildings.2 This enables an emergency services operator to know not only the address of a caller, but also the building (if there are multiple buildings), floor and suite or room number from which a call originated. This information enables faster response to emergencies and helps to eliminate confusion about a caller's exact location.


The Asterisk Open Source PBX provides a cost effective way of achieving E911 compatibility when used in conjunction with a telephone carrier that offers E911 database management services to its customers.This database management service consists of a reverse directory maintained by the customer, but residing with the carrier, that assigns location information to private (DID) telephone numbers that the customer uses.

In order for E911 to function on an Asterisk PBX, each extension phone must be assigned a 10-digit DID number. This DID number is then registered with the telephone carrier along with the location of the phone. An ISDN PRI link connects the Asterisk PBX to the carrier. This digital link utilizes a dedicated signaling channel (D-Channel) to handle call setup and signaling. It is the D-Channel that enables E911 functionality through the carrier. When a 911 call is placed on the PBX, Asterisk needs to be scripted to set the outbound CallerID (ANI) for the call to the registered DID for the calling extension. When the call reaches the Public Safety Answering Point (PSAP)this ANI number is matched with the DID number and location information which was pre-registered with the carrier. The exact location of the caller is then displayed to the emergency services operator.

Asterisk offers a cost effective way to implement E911 services, increasing safety and security within an organization. Even organizations that are not legally required to implement E911 can greatly benefit from its implementation.

About the Asterisk Open Source PBX


Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice-over-IP (VoIP) in multiple protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. For more about Asterisk see the official Asterisk website at: http://www.asterisk.org/.

VoiceIP Solutions is a telephone systems integrator specializing in Asterisk Open Source PBX phone system integration. To see how VoiceIP Solutions may be able to help your organization please see our website at http://www.voiceipsolutions.com/.

1http://en.wikipedia.org/wiki/E911

2http://www.nena9-1-1.org/9-1-1TechStandards/state.htm
 

May 20, 2005

Asterisk - Open Source PBX

"They are developing a sophisticated PBX on a PC with the (capability) of a $100,000 PBX...It will be a world class PBX that runs on Linux. You can have a PBX for the cost of a PC." VoIP guru Jeff Pulver

What is Asterisk?


Asterisk is open-source software IPBX. It operates on Linux operating system (known to work well with - Debian, Red-Hat, Fedora, Gentoo, SuSE, Mandrake and other
distributions). Asterisk represents a revolutionary approach to the world of telephony; it brings the word of open-source to a field which was 100% proprietary. The two major advantages in this approach are:

1) Significantly lower costs (software is free).

2) Rapid development: today thousands of people all over the globe work with asterisk, many of them contribute to the code. Asterisk literally grows from day to day.

As a result, using Asterisk makes it possible to build high end telephony systems in a fraction of what they cost in the traditional way.
Asterisk is fully capable of working with IP telephony as well as POTS (plain old telephony service) and analog telephones.

In my opinion, this new approach to the world of telephony will change the enormous IPBX market dramatically in the near future. It will enable smaller companies to come in that market and offer solutions competitive to those of huge corporations with an Asterisk based platforms, and will present a problem to those who carry huge R&D expenses of proprietary code in that field.


Asterisk Features
Asterisk is feature rich and grows rapidly. Apart from basic capabilities such as call routing (including DID - direct inbound dialing), call forward, music on hold ect., it can also perform as a conference bridge, send voice mail to email, perform as IVR (Interactive Voice Response) and much more. For more information about Asterisk features see http://www.voip-info.org/wiki-Asterisk+PBX+functions
Asterisk Usage
Asterisk is used today world wide by many different types of users, from private and small business implementations to large call centers and service providers. Since it is open source, it can be implemented as a PBX or IPPBX (see below) or be used for a special purpose, such as voice mail or conference bridging.

How did it all begin?
Asterisk was written by Mark Spencer who founded Digium which is the main sponsor of Asterisk. He started writing it to save the high costs of telephony for his own business. After he saw the great potential in it, he made it the main business of his company. As for today, Mark and his "bug marshals" still head the ship.

Asterisk as a traditional PBX
Asterisk can perform as a PBX for traditional analog telephony. This means you can upgrade an old telephony without the high cost of changing all of your telephone sets to IP phones. You can enjoy all the features without making unnecessary investments in equipment. It also allows you to gradually start using IP service providers and IP telephone sets with the old equipment.

In order to use Asterisk as a PBX for traditional telephony, it is necessary to use certain hardware with telephony interfaces, such as channel banks, PCI cards or small gateways.

Asterisk as IPPBX
Asterisk is fully capable of functioning as an IPPBX. All that is required for such usage is an Asterisk unit (PC), LAN (local area network) and IP telephone sets or IP gateways for connecting analog phones.

Asterisk can work with a few IP telephony protocol such as SIP, MGCP, H323, SCCP (Cisco proprietary protocol), however, at this point of time it is known to have some problems with certain protocols, and it is recommended to work with SIP. Asterisk also works with IAX2 (Inter Asterisk eXchange) protocol, an open source protocol that was written for Asterisk and handles NAT and firewalls better than SIP and other protocols.

Since it works with analog and digital telephony protocols and with several IP protocols, Asterisk can also be used as a gateway between different protocols.
Astricon
"What other Linux applications have their very own conference?" Mark Spencer,

Astricon 2004.
Astricon is a conference dedicated to asterisk. The first Astricon took place in Atlanta, on September 2004. About 450 people from 5 continents came to hear about the last developments in Asterisk and where it is going in the future. The fathers of Astricon only expected about 150. For a summery of Astricon 2004 see http://www.xorcom.com/astricon.html

In 2005 there will be two Astricon conferences, one in Europe in May, and one in Atlanta, GA, USA in October. For more information see http://www.astricon.net/

The Future of Asterisk
Asterisk grows at an extraordinary rate. VoIP guru Jeff Pulver states: "They are developing a sophisticated PBX on a PC with the (capability) of a $100,000 PBX...It will be a world class PBX that runs on Linux. You can have a PBX for the cost of a PC." For complete article see http://www.techweb.com/wire/networking/55801111.
Jon 'Maddog' Hall, president of Linux International, states: "I predict that over next three years, VoIP using an open-source solution, such as Asterisk; will generate more business than the entire Linux marketplace today".

By: Eran
 

FREE Asterisk music for downloading (Music on Hold)

Need new Music on Hold for your PBX?

Signate is happy to make a variety of classical music selections available, sampled at rates that are appropriate for telephony. There is no charge.

The selections feature Elena Kuschnerova, pianist, and Lev Guelbard, violinist, playing public domain pieces that will give callers a classic impression of you or your company . Click on the link to see a list of the available music and download what you want from our ftp site.
http://www.signate.com/moh.php

Thanks to Greg Camp, who graciously provided us with the original files. We plan to add other types of music over time.

Legal Stuff Follows:
SIGNATE MAKES NO WARRANTIES, EXPRESS OR IMPLIED, REGARDING THE FREE MUSIC ON HOLD FILES, INCLUDING, WITHOUT LIMITATION, ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. SIGNATE SHALL NOT BE LIABLE TO YOU OR ANY OTHER PERSON OR ENTITY FOR ANY GENERAL, PUNITIVE, SPECIAL, DIRECT, INDIRECT, CONSEQUENTIAL OR INCIDENTAL DAMAGES, OR LOST PROFITS OR ANY OTHER DAMAGES, COSTS OR LOSSES ARISING OUT OF YOUR USE OF THE FREE MUSIC ON HOLD FILES, EVEN IF SIGNATE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES, COSTS OR LOSSES.

Paul Mahler
www.signate.com
 

May 19, 2005

ztdummy accuracy improvements on kernel 2.6

I've been using MeetMe via IAX with no problems on a FC1 box with the 2.4 kernel and zaprtc for timing.

Recently I've set up a FC3 box with the 2.6 kernel, and have been using ztdummy for timing. Using the same IAX sources to a MeetMe conference, I found that there was an increasing delay between a participant speaking and the others hearing him. Over 10-20 minutes this crept up to several seconds!

The only difference between the two boxes were the kernel and the timing, so I looked at ztdummy. I think the use of add_timer() and the jiffy counter doesn't give enough accuracy for MeetMe use (nor probably for IAX trunking, although I'm not using trunking).

This got me thinking and looking again at zaprtc.

In 2.4, in order to use zaprtc it was necessary to recompile the kernel to make the rtc a module or not included (CONFIG_RTC=m or CONFIG_RTC=n).

In the 2.6 version of drivers/char/rtc.c I found a new feature to hook a function into the rtc interrupt, using the functions rtc_register(), rtc_unregister() and rtc_control(). There is an example of their use in sound/code/rtctimer.c. This would enable an rtc-based zap timer to be implemented without any kernel changes.

So I rewrote the 2.6-specific parts of ztdummy.c to use these rtc funcs instead of add_timer(), adopting the same technique as in zaprtc, which is to set the rtc irq to 1024Hz (it must be a power of 2), and then skip 3 out of every 128 irqs (evenly spaced) to give 1000 zaptel interrupts each second.

Repeating my tests, I found that the increasing delay was no longer present. This confirmed to me that the jiffy-based ztdummy is not accurate enough, and than an rtc-based one would be a big improvement.

I intend to submit this to mantis, but could approach it in two ways:
1. Make it a completely new module called ztrtc, and only for 2.6.
2. Make it an update to ztdummy.c, replacing the add_timer code with the rtc_request() code.

Any recommendations which of the two I should do?

Cheers
Tony Mountifield

He has since responded with:

Well, no-one replied to this part of my original posting, so I have submitted it to Mantis as a patch to ztdummy, since the performance of the current ztdummy under 2.6 is definitely poor.

Please see http://bugs.digium.com/view.php?id=4301

Cheers,
Tony

 

AstFax - Asterisk Fax

Here is a snippet from there site:

AstFax provides an outgoing email to fax gateway for the Asterisk PBX package. Incoming fax to email can also be configured so your Asterisk server can act as both an outgoing and incoming fax server.

Ast Fax

May 18, 2005

Equation adds voice logging to existing PABX

Voice logging - the recording of in and outbound telephone calls - has traditionally only been used in selected instances because of its high cost, but thanks to the Asterisk IP-ready PABX solution this security feature is now readily affordable for most businesses, says Allan Gee, managing director of Equation Business Solutions.

"And importantly, it's technology that can be added to your existing PABX. Many organisations have been reticent to take advantage of the recent liberalisation of telecoms in the belief that they will have to ditch their existing equipment, but this is not the case. New-generation tools, like the Asterisk open source solution, can be bolted on to an existing system to bring about an immediate increase in features and benefits."

 

Asterisk is a Linux-based product that has the ability to interact with all open standard IP PABX protocols, alleviating the problems often associated with proprietary brands.

"As an example, the increasing popularity of call centres has seen a dramatic shift in the need for voice logging and recording facilities," says Gee.

"Traditionally the costs of implementing such a feature were onerous or, in some cases, required replacement of the entire PABX. Not so with Asterisk. And, typically, Asterisk costs about 30% less to implement than conventional recording solutions!

"Because Asterisk is a software solution it's totally customisable for the specific application. Both in and outbound calls can be monitored, alternatively only certain lines. And reports can be drawn based on a host of pre-established criteria - date, time, number of calls, by operator, by project, etc. Recording capacity is based on the hard drive selected with backup to CD storage. Flexibility is the key word!"

In addition to the options offered by traditional PABX systems, Asterisk offers a host of value-added features such as IVR, voicemail by handset or via e-mail, video-phone support, least-cost routing and much more.

"Communications in SA is undergoing a revolution. And it makes sense to be a part of that process because there are already significant savings to be made, even for relatively small enterprises. So speak to your technology partner about how you can maximise the business benefits already on offer," says Gee.

Source: By Allan Gee

IP telephony firm launched - OpenVoice

OpenVoice, which designs, delivers and implements telecoms solutions by way of open source platforms, was launched officially in Johannesburg last night.

The company's focus is open source application Asterisk, a converged telecommunications platform that allows different types of Internet Protocol (IP) telephony hardware, middleware and software to interface with each other consistently.

 

"The market is certainly ready to take advantage of a host of telecommunications solutions, from a basic PBX setup to the more advanced VoIP (voice over IP) solutions now available," says OpenVoice director Clayton Hayward.

"VoIP is poised to join the World Wide Web and e-mail as the third ubiquitous and critical Internet application to date and the next decade will prove this."

Hayward predicts that by the start of the next decade, analogue telephony will have become rare in the developed world.  He says the use of an open platform gives companies a flexible, low-cost alternative that can easily be upgraded.

Asterisk supports both VoIP and legacy public switch telephone network connectivity.  Director Justin Colyn says OpenVoice is finalising a black economic empowerment deal, and "we are very excited about the partner we have on board".  Hayward says the company recently won a significant deal to implement Asterisk at a call centre in Cape Town. He adds that the project took only five weeks.

Source: Iain Scott

A pipe length tester for TDM400 cards

Hi,
 
I finally put together a simple program for testing what happens to the pipe length out from software, through a TDM400 card, and back to software again. I haven't done much with it yet, or really studied in depth if there are pitfalls in what I have done. However, you can grab the current source from ftp://ftp.soft-switch.org/pub/sliptest.c, and compile it with

gcc sliptest.c -lspandsp

(which, obviously, means you need spandsp installed). When you run it, you need to give it a parameter which is the channel number of a circuit on a TDM400 card. Don't have anything plugged into the card. The software listens to the echo of what it sends. When it runs, it lists 50 loop lengths per second. The loop lengths are measured in audio samples.

The initial results I get are odd. I suspected I might get jumps of 1, 8 or 160 samples. These seem to be the significant sizes of things in the zaptel world. However, I get a variety of oddly sized changes, that I haven't yet explained.

The loop length is evaluated by looking for the peak in the cross-correlation of the Tx and Rx signals. This isn't foolproof, so you may get a few samples which hiccup a little. However consistent changes in loop length are real.

Treat this code as an interesting (or possibly very dull) experiment, rather than a proof of anything.

Regards,
Steve

May 17, 2005

Big telco players embrace Asterisk

Today I had an interesting conversation with a large telecoms equipment provider from the U.S. For obvious reasons I will refrain from mentioning the name. It turns out that this company, and I'm sure many others, are going to be using Asterisk as a feature-set provider for their softswitches.

  
Asterisk has a huge developer base that these companies cannot compete with. As a result the number of features being developed and integrated into Asterisk is phenomenal and these comapnies can't keep up. Similarly, creating new services in the Asterisk environment is easy because of its open source nature - comapre this with the conventional proprietary systems we are all used to. The main reason however is the design which enables developers to create new services using the Asterisk Gateway Interface (AGI) or the C API.

I am sure there are many companies already doing this, and if not, will be doing it soon if they have any sense. However, I don't expect these companies to advertise the use of Asterisk in their systems, which is possible via the purchase of a commercial Asterisk license from Digium.

well done Asterisk

Editor Note: I found this blurb while reading about Asterisk. A nice comment might I add.

Source: Jay Bee Pee

Managed Linux services on show at LinuxWorld

Synaq, a provider of managed Linux services, will be showcasing its solutions at this year's Linux
LinuxWorld.  Synaq's Yossi Hasson says the company hopes to demonstrate the "clear-cut advantages" behind open source software to visitors to the show. He says that these advantages, together with the benefits of managed services, makes Linux a compelling offering for all businesses from small and medium sized companies through to large enterprises.

 

Synaq's Yossi Hasson says the company hopes to demonstrate the "clear-cut advantages" behind open source software to visitors to the show. He says that these advantages, together with the benefits of managed services, makes Linux a compelling offering for all businesses from small and medium sized companies through to large enterprises.

"There is no doubt that the popularity of Linux and open source technology is on the rise. Our key objective is not only to interact with resellers and prospective partners, but also to showcase the potential behind a range of solutions including Asterisk, an open source PABX solution with VoIP capabilities, PinPoint Securemail, a comprehensive anti-virus and overall e-mail protection service, and a host of other Managed Linux Services," says Hasson. "We also aim to reach those who may be considering an investment in open source technology, but require more details in order to make an informed decision."

The key message that Synaq wants to communicate to the market is that open source software in conjunction with managed Linux services is a credible alternative that is easily managed, cost effective and can enhance the networking and communication, and other critical business operations for a business.

*Synaq will be located at Stand A8 at LinuxWorld in the Sandton Convention Centre.

Source: LinuxWorld.com

 

May 16, 2005

VoIP encryption options

Colin Anderson has posted details of how his network is set up and how he encrypts network traffic (vtun):

"I'm looking for solutions that work when one end of the call is connected to the pstn, and the entire media stream needs to be encrypted."

In my scenario, I have Snom's in a remote LAN and they get dialtone to the PSTN thru my Asterisk server here via the VPN. I also use soemthing that you might want to consider something like this:

 

SIP phone ---SIP--->Asterisk server NIC # 1
|
|
Asterisk server NIC #
2<---IAX---VTUND---INTERNET---VTUND---IAX--->Asterisk server
|
|
PSTN

The Asterisk server NIC # 1 is on a non routable subnet so you don't have to worry about snooping for the SIP part, and the IAX data is encrypted by the time it hits the Internet. I have this running in several locations as well, with the remote Asterisk server running the Locustworld meshbox distribution:

www.locustworld.com

We use a single Meshbox with a second nic added to the Meshbox WiFi bridge using brctl. The single Meshbox acts as firewall, dhcp server, WiFi access point, and Asterisk server all in one. I use Compaq Deskpro En's P-II 400's with 64 meg of RAM and an SMC EliteConnect 2512W PCI card and everything runs nicely. The Meshbox assigns DHCP IP's to the Snoms and an instance of Asterisk is run on the meshbox to provide registration for the Snom. When the Snom dials out, iax.conf on the Meshbox is set to dial into the dialplan on our primary Asterisk server connected to the PSTN. Traffic is encrypted using VTUND. Works good, my salespeople are pleased with it because they can do fancy stuff like call forward, juggle multiple lines, MeetMe, IVR menus, and blind call transfer to the PSTN. Coming from a single POTS line with basic calling features to these remote locations, it's like a different world for them.

Although, the encryption part I'm not too worried about, that's just a bonus. It's not as if we have state secrets or anything.

If you want to use a bolt on in your own distro from server to server, without using the Meshbox distro, you can just run vtund by itself:

http://vtun.sourceforge.net/

 

May 13, 2005

Sipura SPA-3000 Reviewed

Overview
The Sipura SPA-3000 is labeled as a 1 Port FXS + 1 FXO Analog Telephone Adapter. Just what that means to non-telephony guru's is anybody's guess. What it does mean is that the SPA-3000 has both a PSTN (Public Switched Telephone Network) port for plugging in a standard analog telephone line as well as a POTS (Plain Old Telephone Service) jack for plugging in a standard analog telephone.

Along with the two telephony jacks, the SPA-3000 is fitted with an RJ45 plug for connecting to your Ethernet network. While the label may not tell you what is so interesting about this unit, what it does is actually very unique. The SPA-3000 can not only act as an ATA (analog telephone adapter) to connect a regular analog phone to a VoIP service provider or VoIP PBX, it can also act as a gateway between the PSTN line and the VoIP Provider.

 




Still confused? I don't blame you. Let's take a couple of examples of how you might utilize the SPA-3000.



VoIP Service
You are using a service such as Vonage or Broadvoice to reduce your long distance costs. The SPA-3000 can be configured to sit on a land line which you can dial into, and using authentication, allow you to dial back out through your VoIP provider. Calls through the VoIP provider can also be made directly using an attached telephone. Should the connection to the VoIP provider not be available, the SPA-3000 will revert to sending calls out the land line.

VoIP PBX
For us serious Asterisk PBX geeks out there, the SPA-3000 provides a cost-effective means of bring a PSTN trunk into the PBX while still functioning as an ATA. Not only can you use the SPA-3000 as inbound and/or outbound trunk, you can also easily configure the SPA-3000 as a PSTN failover should the primary trunk into Asterisk fail. Considering what you can buy the SPA-3000 for right now, this is one of the best deals going.

Features

The SPA-3000 Supports Key Telephony Features:
* VolP to PSTN (USA) Service Call Origination and Termination
* PSTN (USA) to Val? Service Call Origination and Termination
* Single or Dual Stage Dialing
* Service Authentication via PIN, Digest, Caller ID (Bellcore Type I)
* Per Call Authentication and Associated Routing
* Least Cost Routing Support
* Telephone Impedance Agnostic - 8 Configurable Settings
* Call Waiting, Cancel Call Waiting
* Caller ID Detection (Bellcore Type I)
* Caller 10 with Name I Number (Multinational Variants) Display
* Caller ID Blocking
* Call Waiting call& ID with Name I Number
* Call Forwarding to PSTN or VolP Service: No Answer I Busy I All
* Do Not Disturb
* Call Transfer
* Three-Way Conference Calling with Local Mixing
* Message Waiting Indication - Visual and Tone Based
* Call Return and Call Back on Busy
* Call Blocking with Toll Restriction
* Distinctive Ringing
* Off-Hook Warning Tone
* Selective 1 Anonymous Call Rejection
* Hot Line and Warm Line Calling
* Speed Dialing of 8 Numbers I Addresses
* Music on Hold


The SPA-3000 supports one POTS FXS port for connection to existing analog phones and fax machines. The SPA-3000 also supports one PSTN FXO port for connection to a Telco or PBX circuit. The SPA-3000 includes an Ethernet interface for connection to a home or office LAN. The SPA-3000 FXS and FXO lines can be Independently configured via software controlled by the service provider and/or end user.

Setup
Configuring the SPA-3000 isn't actually difficult per-say, the problem is a lack of good, solid documentation. The well-written Quick Start Guide gets you through setting up the IP address or discovering the DHCP-assigned IP address so that you have access to the web interface that has hundreds of configuration settings. There probably isn't a phone system or VoIP provider in existence that you can't get the SPA-3000 to work with. It's figuring out what the settings are actually for is the hard part. There is no doubt that were the SPA-3000 is one of the most feature-rich devices of its kind, the documentation side really hurt our overall marks.

Some of the VoIP providers we checked with will have no problem helping you get your device up and running with their service so we will refrain from going through the motions of setting it up with a provider.

Using the SPA-3000 with Asterisk, on the other hand, is a highly requested topic and the current documentation available on the net ranges from instructions that simply don't work to 90 minute plus setup routines that serve only to get a call from the PSTN to go into the Asterisk PBX.

With that in mind, we set out to find out if we could actually get the SPA-3000 to really function with Asterisk the way it "should". It did take a good week of tinkering and some of the instructions on the net actually got fairly close. In the end we came up with a setup that should take anyone that is comfortable using the AMP interface about 5 minutes to get up and running.,

To begin with, go to Sipura's website and make sure you have the latest firmware installed on your device, there is a fair to good chance this will not work properly with earlier firmware versions. Next, we will setup the trunk within Asterisk. For the sake of simplicity we are going to demonstrate this using the AMP interface which is available separately or part of the Asterisk@Home install package. Advanced users should be able to figure out how to modify the config files based on our information.

Usage
Documentation aside, the Sipura SPA-3000 works like a charm. The audio quality on the ATA is as good or better than anything we have tried so far. The vast amount of settings in the web admin ensure that the device will probably be able to be made to work with systems that haven't even been created yet. As for the gateway function, it may be a little tricky to get setup the first time or get tweaked to fit your particular requirements, but once it is in place, it just works.

While preparing this review, problems with our VOIP service provider reached a critical state. A single outage lasted two days. This obviously was not good for business. We needed something a lot more reliable than what we had. The Sipura SPA-3000 was already in the test lap being configured for the Asterisk server. No sooner had we verified the settings with some outside testers than the SPA-3000 took up permanent residence as a PSTN <-> Asterisk gateway providing routing of the incoming PSTN line to a VoIP provider that hands off to the Asterisk server as well as outbound routing of 911/411 calls. On top of basic call routing, the SPA-3000 also functions as a failover outbound trunk in case the VoIP Provider is not available and while providing all of these functions, it still acts as a normal ATA.

Results
The Sipura SPA-3000 is really an incredible little box. Only slightly larger than two decks of cards and amazing amount of functionality into a tiny case. There are settings in there that should enable advanced users to get the SPA-3000 to work with virtually any SIP-Compatible VoIP service. It might not be as attractive as Sipura's other devices but it does feature mounting holes on the bottom so it will fit nicely on the wall next to your other phone equipment.

Whether you are an experimenter, hard core PBX user, or a VoIP service user, the SPA-3000 is a great value by adding only a few dollars to the cost of an ATA you get an entire gateway package. As we mentioned, it works so well it is now OUR Gateway device!

Features: 5/5
Setup: 2/5
Usage: 5/5
Results: 5/5
Value: 5/5

Overall Index: 4.40/5

Sipura
http://www.sipura.com

Asterisk PBX
http://www.asterisk.org

BroadVoice
http://www.broadvoice.com

Source: Geek Gazette


VoIP Is Killing Traditional Telephony

Source: Info-Tech
 
Half of small to medium-sized companies will use voice over IP by 2008, Info-Tech Research says.  The rapid adoption of Voice over IP (VoIP) is killing off traditional telephony, with 50% of small- to mid-sized enterprises expected to rely on VoIP by 2008, according to a new study by Info-Tech Research.

 

VoIP is growing even more quickly than expected, according to Info-Tech research analyst George Goodall. The study found that 23% of small- to mid-sized enterprises are already using VoIP technology and the firm expects the number to grow to 50% by 2008. For all the promise of converged networks, however, the speed of the technology changeover has put a strain on IT managers, who are, Goodall notes, scrambling to implement the technology."

Goodall expects the majority of small to medium-sized enterprises to have switched at least part of their networks to VoIP within the next five years. That could spell disaster for traditional telephony equipment vendors, and whether they can keep up with the market is anyone's guess.

"Companies like Nortel and Avaya are aggressively introducing new VoIP products to the SME market. It may be too late," Goodall said in a statement. "They're racing against a group of young companies with products that specifically address the infrastructure limitations of SMEs. These products aren't just scaled down version of large-enterprise systems. Potentially, they're category killers."

 

May 12, 2005

Hit "Asterisk" for More Options

Source: By Business 2.0 (c)2005

When Mark Spencer was starting a Linux company six years ago, he had $4,000 and some cheap, leftover hardware from a company where he had interned during college. His first conundrum: How were customers going to call him? A private branch exchange -- the specialized hardware that routes calls around an office -- was going to set him back $6,000.

 

So Spencer decided to program his own Linux-based PBX. "Telecom was not really our core business," he says. But he released the software as open-source, and as contributions of code started coming in, Asterisk was born. Today, his company, now called Digium, focuses entirely on developing Asterisk and selling related hardware and software. He won't disclose the revenue of his closely held company, but he says it is profitable.

The success of Asterisk shows the growing power of open-source. Digium could have tried to roll out its own proprietary PBX -- and likely would have been crushed by the likes of Avaya, Cisco (CSCO), and Nortel (NT). But by sharing his code, Spencer has created an ecosystem full of niches waiting to be filled. That should keep his phones ringing for quite a while.

In addition to routing calls, Asterisk also serves as a voice-mail system, and you can plug in all manner of phones, from analog lines to voice-over-Internet-protocol handsets, to hardware running Asterisk. Best of all, you can run it on a repurposed PC rather than buy expensive, custom PBX hardware. A fully supported version from Digium, available later this month, will cost $750 per machine, though users will still be able to download the source code for free.


As with Linux, though, I suspect most users will want to buy a finished product rather than compile the code themselves. Joe Kraus, co-founder and CEO of JotSpot, a Web-based software startup in Palo Alto, toyed with the idea of installing Asterisk, but his operations chief told him that any time spent configuring that software would be time not spent on developing his company's product. Chastened, Kraus shelled out for a conventional PBX. Having bought a PBX, Kraus isn't about to rip it out. (That's one of the many problems with old-school PBXs: You can't reuse the hardware for something else.) But he talks about his decision wistfully. As it is, his employees rarely use their phones, preferring Skype -- a VoIP service Kraus could have run in-house with an Asterisk PBX.

"It's a common misunderstanding that just because it's an open-source solution, you have to do it yourself," Spencer says. That's a big opportunity for the likes of Fonality and SwitchVox, which have already built ready-to-use Internet PBX products around Asterisk. Other resellers make a specialty of configuring Asterisk to suit a company's particular needs; branch offices, for example, may want voice-mail that's integrated with headquarters, while a call center operator would like sophisticated call-transferring capabilities to get the customer on the line with the right service representative.

The ultimate customer for Asterisk might be the phone company. Some VOIP startups, like VoiceGlo and VoicePulse, use Asterisk to run their services. The incumbent providers have been more hesitant: For now, Spencer says, only overseas telecoms have deployed Asterisk for customers. Some regional phone companies in the States are testing it, as is AT&T, which showed off Asterisk running on a system connected to its Internet backbone at a recent industry conference.

"We thought we should take a look at Asterisk and see what all the Web hubbub was about," says Mark Vince, a technical consultant for AT&T Labs who's been testing the software. One of the best aspects of Asterisk, he says, is the wide range of developers it has attracted. When he wanted to see if Asterisk could support an obscure VOIP standard, he simply had to search the Web and download a software driver from a programmer who'd already solved the problem. Just one more example of the productivity multiplier that is open-source software.

 

May 11, 2005

AreskiCC V2.2 - Asterisk CallingCard Application

Here the version 2.2 a new version of your dear CallingCard Software !!!
http://www.areski.net/areskicc-doc-v2/

Many new features have been added and several enhancements made!

Newest features :
-A new re-build rate-engine
-LCR & LCD management (OOOOHHH YESSSSS)
-Billing Increment
-Progressive Rate
-Scheduled Rates (days of the weeks)
-Expiration rates
-Buy rates configuration
-importation ratecard from csv file
-Simultaneous access for same card
-SIP/IAX Friends Management
-Generate conf file for SIP/IAX Friends
-Reload Asterisk from UI
-AGI flexibility - many options such as use DNID, Directcall, saybalance, ...
-SIP/IAX Friends on AGI (press 9)
-USE DID
-New Graphic design :) Is it cute no ?
-Internal help/info
more...
 
Unfortunately, the number of improvements make me impossible to support the old database schema and to let it compatible with the previous callingcard version...
Sorry for those that are running the application in production but the future release wont have this problem anymore, I swear ;)  I will provide, in the future, script to update the Database too, sorry about that guys.

If someone is willing to help on an user guideline, it would be greatly appreciated ! (contact me please) I hope you will enjoy this new version (an other is already on the way)!
 
-Areski
 

May 10, 2005

AstriCon Europe 2005: June 15 - 17 in Madrid Spain

AstriCon Europe 2005 will be held at the Auditorium Hotel Madrid in Madrid, Spain in less than six weeks. This is the first ever all-Asterisk conference to be held in Europe and we expect it to be a blast. The first AstriCon, held this last fall in Atlanta, drew in nearly 500 Asterisk users.

Reasons To Attend AstriCon Europe:

* Hear Mark Spencer, creator of Asterisk give a keynote address.
* Meet hundreds of fellow Asterisk users, developers, and enthusiasts.
* Tapas, Tapas, Tapas!
* Learn valuable Asterisk tips and tricks.
* Find an Asterisk consultant or employee.
* Discover what others are doing with Asterisk.
* See many products and services which add value to Asterisk.
* Connect with potential resources, partners, and customers.
* Drink mucho sangria with the Asterisk community!
* Hear about upcoming enhancements to Asterisk.
* Provide input to the development process.
* Take the dCAP certification exam.
* Learn to flamenco dance!

AstriCon will include:

* A Tutorial Day: June 15
* A General Conference Day: June 16
* A Developers Summit: June 17
* An Asterisk Trade Show and Exhibition: June 15 - 17

For more information or to register, please see our web site at: http://www.astricon.net/europe

If you would like to speak at AstriCon, please send a proposal to: speakers@astricon.net

If your company would like to exhibit an Asterisk-related product or service, please contact us at: info@astricon.net

Thanks,
Steve Sokol & Olle Johansson
IPSando - The AstriCon Company
 

May 06, 2005

Grandstream GXP-2000 reviewed

Rob from gladstonewireless has written a review of the new Grandstream phone:



I've been hounding Grandstream for weeks to get an evaluation GXP-2000 phone. From all the rumours, pictures and variously incorrect information on them, there hasn't been much in the way of real, actual, information on them. Well I've done my best to fix it. I've added a couple of pages to the gladstonewireless.net wiki with my review of the phone, and a list of bugs that I've found so far.

Click Here

Also, in other news, there is some user-contributed firmware on the PA1688 page - it's in 'attached files' on the wiki, but I'll be tidying it up tonight and linking it from the page. Blame bkw for it, he's been dissecting the code for the last couple of days and is starting to get the hang of it. No news on the Sourceforge project page yet, but, it is a week-long public holiday in China at the moment, so don't expect too much more news this week.

-Rob

May 04, 2005

Asterisk dialplanner

Source: Ahmad Faiz

I'd like to inform you that Lanvik ICU has put together a Java-based application to assist with dialplan creation. You can check it out here.


The dialplanner is a web-based tool that provides a point-and-click interface to create an Asterisk dialplan. You can create contexts and extensions, then select the appropriate command from a list. Then you'll be prompted to enter the arguments for that command. The dialplanner will show a nested tree-based view of your dialplan.

Once you're done, you can click on a button to export the dialplan. Simply type in your email address, and your new creation will be emailed to you in a flash.

We're hoping that this will help new users in creating their dialplan, and hopefully get more people interested in using Asterisk.

Thanks,

Ahmad Faiz
Lanvik ICU Sdn Bhd
Level 2, Block B, Plaza Damansara
45 Medan Setia 1, Bukit Damansara
50490 Kuala Lumpur

May 02, 2005

The Asterisk Realtime Architecture

Source: By Olle Johansson

The Asterisk Realtime Architecture

----------------------------------
The Asterisk Realtime Architecture is a new set of drivers and functions implemented in Asterisk 1.1dev (and the following v1.2 stable). The benefits of this architecture are many, both from a code management standpoint and from an installation perspective.

Additional information on the configuration of Realtime with Asterisk can be found in README.extconfig.

The ARA is designed to be independent of storage. Currently, most drivers are based on SQL, but the architecture should be able to handle other storage methods in the future, like LDAP.

 

The main benefit comes in the database support. In Asterisk v1.0 some functions supported MySQL database, some PostgreSQL and other ODBC. With the ARA, we have a unified database interface internally in Asterisk, so if one function supports database integration, all databases that has a realtime driver will be supported in that function.


Currently there are three realtime database drivers:

* ODBC: Support for UnixODBC, integrated into Asterisk
The UnixODBC subsystem supports many different databases,
please check www.unixodbc.org for more information.
* MySQL: Found in the asterisk-addons cvs archive on cvs.digium.com
* Res_perl: Found in the asterisk-addons cvs archive on cvs.digium.com

* Two modes: Static and Realtime
--------------------------------
The ARA realtime mode is used to dynamically load and update objects. This mode is used in the SIP and IAX2 channels, as well as in the voicemail system. For SIP and IAX2 this is similar to the v1.0 MYSQL_FRIENDS functionality. With the ARA, we now support many more databases for dynamic configuration of phones.

The ARA static mode is used to load configuration files. For the Asterisk modules that read configurations, there's no difference between a static file in the file system, like extensions.conf, and a configuration loaded from a database.

* Realtime SIP friends
----------------------
The SIP realtime objects are users and peers that are loaded in memory when needed, then deleted. This means that Asterisk currently can't handle voicemail notification and NAT keepalives for these peers. Other than that, most of the functionality works the same way for realtime friends as for the ones in static configuration.

There is some work to create a solution for Realtime SIP devices that loads from database and stays in memory for the duration of a call or a registration, but that work is not integrated into Asterisk yet.

* New function in the dial plan: The Realtime Switch
----------------------------------------------------
The realtime switch is more than a port of functionality in v1.0 to the new architecture, this is a new feature of Asterisk based on the ARA. The realtime switch let's your Asterisk server do database lookups of extensions in realtime from your dial plan. You can have many Asterisk servers sharing a dynamically updated dial plan in real time with this
solution.

* So what can you do?
---------------------
The realtime Architecture lets you store all of your configuration in databases and reload it whenever you want. You can force a reload over the AMI, Asterisk Manager Interface or by calling Asterisk from a shell script with

asterisk -rx "reload"

You may also dynamically add SIP and IAX devices and extensions and making them available without a reload, by using the realtime objects and the realtime switch.

* Configuration in extconfig.conf
---------------------------------
You configure the ARA in extconfig.conf (yes, it's a strange name, but is was defined in the early days of the realtime architecture and kind of stuck).

The part of Asterisk that connects to the ARA use a well defined family name to find the proper database driver. The syntax is easy:

{family} => {realtime driver},[,{table}]

The options following the realtime driver identified depends on the driver.

Defined well-known family names are:

* sippeers, sipusers SIP peers and users
* iaxfriends IAX2 peers
* voicemail Voicemail accounts

There is documentation of the SQL database in the file README.extconfig in your Asterisk source code tree, the /doc directory.

For voicemail storage with the support of ODBC, there is a README.odbcstorage documentation file.

* Please test this architecture in order to make it stable
-----------------------------------------------------------
The Asterisk CVS head, v1.1 dev, is there for you to test. In order to move it forward to a stable release (v1.2) we need more tests, more bug reports and more fixes.

You will find download instructions for Asterisk CVS head on the www.asterisk.org web site. As usual, do not install a development version on a production server.

 

Asterisk-java 0.1 released

Source: By Stefan Reuter

Stefan Reuter has posted details of the latest release of Asterisk-Java. These are classes to interface with the Manager or AGI from Java:

Asterisk -java 0.1 a Java control for the Asterisk PBX has been released.

The Asterisk-java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API.

The FastAGI implementation supports all commands currently available from Asterisk.

The Manager API implementation supports receiving events from the Asterisk server (e.g. call progess, registered peers, channel state) and sending actions to Asterisk (e.g. originate call, agent login/logoff, start/stop voice recording).


Asterisk-java is available under Apache 2.0 license at:
http://asterisk-java.sourceforge.net

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