Recently in Asterisk Category

When I met with PIKA Technologies  at ITEXPO they told me support for FreePBX was coming. Well today, PIKA Technologies announced that PIKA WARP the Appliance is now compatible with the Asterisk-based FreePBX GUI (Graphical User Interface) application. I reviewed the PIKA Appliance recently and was pretty impressed with it. Having FreePBX support is a huge milestone for the PIKA Appliance. FreePBX is a popular user-friendly web application  that makes it easy to setup and configure Asterisk.

According to PIKA, "While customers often develop their own GUIs, many have told PIKA that if WARP were compatible with industry-standard GUIs like FreePBX, they would be much more inclined to adopt the portfolio. With today's announcement, PIKA has once again demonstrated its responsiveness to the needs of its user base."

"We are very pleased to have supported the effort to adopt the FreePBX application to run in the PIKA Warp environment," said Terry Atwood, vice president of sales, marketing and customer care at PIKA Technologies. "Used in many Asterisk implementations around the world, including Trixbox, FreePBX has proven its value, time and again. When the FreePBX team expressed their willingness to work with us to port to the Warp Appliance, we jumped on the opportunity."

"FreePBX has become the de facto standard for enterprise grade PBX functionality delivered to the SMB business and includes a very rich set of functionality and customization potential," said Philippe Lindheimer, open source community director of Bandwidth.com and leader of the FreePBX project. "But no GUI is complete without a wide range of hardware options to complete the package. We are delighted that PIKA can now include FreePBX in the PIKA WARP and bring our two eco-systems together."

Today's announcement from PIKA follows news of a new partnership between FreePBX and Bandwidth.com, a complete business communications provider offering advanced VoIP, Internet services and managed network services to small and medium businesses. Bandwidth.com will devote significant resources to expand the scope of FreePBX while protecting its charter to remain open source and free.

"The partnership with Bandwidth.com is great news as it gives FreePBX the support it needs to grow while ensuring it remains a free GUI for the entire open source eco-system," said David Clarke, business development manager at PIKA and director of the PIKA Warp Community. "I know the choice of Bandwidth.com was a decision that Philippe made only after months of consideration and sound input from the key developers and contributors to the FreePBX project."

Out of the box, FreePBX provides a long list of features including many typically found only in an enterprise-grade PBX, some examples are:

• Unlimited number of voicemail boxes
• "Follow me" functionality
• Ring groups and call queues
• Unlimited number of conference bridges
• Paging and intercom functionality
• and much more

The PIKA WARP Appliance product portfolio is ideal for deploying small- to medium-sized IP-PBX systems, IVR self-service systems, predictive dialling systems, fax servers and many other features typical of a traditional, purpose-built business telephone system that are often lacking in a computerized system. Compatible with a variety of open-source development platforms, including Asterisk and Linux, the Appliance offers a cost-effective alternative to traditional off-the-shelf computers and plug-in-card network connectivity in a smaller footprint.

Bandwidth.com has just made an investment in FreePBX, the popular front-end interface to Asterisk-based distros. I discussed this news with Philippe Lindheimer just a couple hours ago. One of the questions I asked was if Bandwidth.com would get "preferred treatment" within the FreePBX interface, since Bandwidth.com offers SIP trunking. Obviously, if FreePBX gives Bandwidth.com a prominent position in the GUI or they make it "easier" to configure FreePBX (i.e. plug-n-play) that could be a huge boon to Bandwidth.com Philippe said that that isn't part of the investment announcement being made today, however, that is something they are looking at.

As for the purpose of the investment, Philippe said it was mostly due to Bandwidth.com's desire to grow the market and help build the FreePBX community. The idea is that the more IP-PBXs out there, the more SIP trunks, and hence more revenue for Bandwidth.com. I have some further thoughts on this, but I'm pretty busy today and wanted to share the news.

Philippe Lindheimer said, "Part of assuring the success of FreePBX is to make sure that we continue to have strong leadership, community participation and a thriving eco-system of users and partners. I would like to announce a new partnership that will help the project tremendously. I have joined forces with Bandwidth.com as their Open Source Community Director, where we will be devoting significant resources and effort to expand the scope of FreePBX while protecting its charter to make sure it remains open and strong."

One significant piece of news is that Bandwidth.com helped protect the FreePBX's project several months ago when the FreePBX trademark (which FreePBX.org nor Phillipe never owned) was "being shopped around to parties that did not have this project's best interest in mind" according to Phillipe. Thus, Bandwidth.com preemptively purchased the trademark with Phillipe's blessing in order to assure FreePBX was not jeopardized.

You can read Phillipe's blog post about this here which has more details.

AudioCodes has now entered the IP phone arena with a phone that supports wideband codecs (HD) for superior sound quality. Seems a bit odd for a company that makes VoIP PCI and cPCI communication boards and VoIP media gateway modules (PMC form factor) and Analog Media Gateways (2/4/8/24 ports) to be entering the crowded VoIP arena, but enterprise IP phone market is expected to quadruple, from $2.1 billion in 2007 to $8.4 billion in 2001, with some 63 million endpoints being shipped by that time, according the Synergy Research Group. So there is a pretty big pie to go after. Traditional network hardware vendor Adtran has also recently entered the IP phone arena with their Adtran IP700 series (see Adtran IP706 review).

The AudioCodes 300HD Series includes three models: The 310 HD entry level phone with a basic display and interface; the 320HD premium endpoint with a larger screen, and the 350HD executive phone with a color LCD screen. All three models are based on AudioCodes newly announced VoIPerfect software, and include the most commonly used wideband codecs. Of course they are SIP-based so they should work on standards-based IP-PBXs such as Asterisk and these phones are also PoE (Power over Ethernet) compliant.

For more details, check out TMCnet reporter Erik Linask's article. One interesting take Erik has is when he writes, "First and foremost, it owns the DSPs that enable the higher quality, which means it can provide its HD VOIP-enabled handsets at a price point comparable to other high-end non-HD devices".

Now I just have to get my hands on on to review. Stay tuned...

Analog Devices unveiled the new Blackfin BF51x series, the newest members of their convergent-processor family. Blackfin processors are very popular when building Asterisk-based appliances, including the Digium Asterisk Appliance AA50 and Astfin. The Blackfin convergent-processor architecture offers reduced cost, power consumption, and software complexity. Although the processor is popular in creating Asterisk appliances, it can be used for a variety of low-cost, low-power consumption required applications.

The new Blackfin processors are the BF512, BF514, BF516 and BF518. According to Analog Devices, "All are single-core convergent processors that surpass outdated, heterogenous MCU+DSP approaches in reducing part-count, system cost, board space, and power consumption. Like traditional DSPs, the BF51x processors feature high clock rates and low power dissipation per unit of processing (MMACs/mW), and like traditional MCUs, these convergent processors are OS and compiler-friendly."

All four of the new 16-/32-bit BF51x processors are available at clock speeds up to 400 MHz (800 MMACS) and include 116 kBytes of RAM plus an optional 4 Mbits of serial (SPI) flash memory. Each also integrates Lockboxâ„¢ security for code and content protection.

The Blackfin processors on-chip integration assures easy connection to a variety of audio, video, imaging and communications peripherals and memory types. Integrated features include support for sixteen stereo I2S digital-audio channels, twelve peripheral DMA channels, and an advanced memory controller for glueless connection to multiple banks of external SDRAM, SRAM, Flash, or ROM. Each processor includes two dual-channel synchronous serial communication ports (SPORTs), a high-speed parallel peripheral interface (PPI), an I2C compatible two-wire interface (TWI), dual PC-compatible UARTs, and 2 SPI-compatible serial peripheral interface ports.

"System solutions ultimately determine how much power any particular application will consume," said Jerry McGuire, vice president, General Purpose DSP, Analog Devices, Inc. "It's quite intuitive that a single convergent processor with the right mix of integrated peripherals is always going to lead to lower BOM costs and power consumption than an inelegant combination of disparate processors and parts can possibly achieve. Many companies today talk about the lowest power or the highest performance. But what is important for today's applications is the highest levels of performance at low power."

All of the new Blackfin processors, delivering 8.5 MMACs/mW (100 MHz), include dynamic power management (DPM) functionality that lets developers match the processor's power consumption to processing requirements during program execution. ADI pioneered the application of DPM more than seven years ago with the release of the first Blackfin processors.

The BF512 is the new low-cost entry point in the Blackfin processor family. The device balances performance, peripheral integration, and price, and is well suited for the most cost-sensitive applications including portable test equipment, embedded modems, biometrics, and consumer audio. All members of the BF51x family also include a new 3-phase PWM generation unit for inductive motor control applications and a quadrature interface for rotary encoders.

The BF514, BF516, and BF518 all extend the convergent processor family further into the portable application space with on-chip removable-storage interfaces. All three devices include Secure Digital Input Output (SDIO) for connectivity to standard flash memory and Wi-Fi cards; a power-optimized CE-ATA storage interface for small form-factor handheld and consumer electronics applications; and an embedded multimedia card (eMMC) interface for integrating mass-storage flash memory in a wide range of consumer electronics, wireless, navigation, and industrial applications.

For developers of network-connected industrial and instrumentation applications, the BF516 adds an Ethernet 10/100 MAC with Media Independent Interface (MII) and Reduced Media Independent Interface (RMII). Highly integrated for industrial, portable and VoIP applications, the BF518 Ethernet MAC supports the IEEE-1588 clock synchronization protocol for networked measurement and control systems.

An increasingly wide variety of applications are viewing the contemporary convergent-processor approach as the soundest choice for cost- and power-sensitive designs. For example, some voice-over-IP (VoIP) telephony system developers have designed in separate DSP and microcontroller chips to implement the required media and control functionality. With BF51x Blackfin processors, however, a single architecture enables full VoIP telephony functionality in a unified software development environment with faster system debugging and deployment, lower overall system cost, and the lowest possible system power demand.

"GIPS VoiceEngine media processing capabilities meet the highest requirements of VoIP equipment manufacturers and paired with Analog Devices' Blackfin processors we can assure customers a consistently high quality VoIP experience. The performance, power and functionality profile of Blackfin is a superb fit for VoIP technology," said Larry Golob, Senior Director Business Development, Global IP Solutions.

With the Global IP Solutions (GIPS) VoiceEngine package of VoIP software components available for Blackfin processors, and a VoIP reference platform available on uClinux, the feature-rich Blackfin family has driven down the price required to easily design and deploy a fully scalable range of VoIP telephony designs across multiple market spaces.

Pricing and Availability
The BF51x family includes the BF512 at $4.95, the BF514 at $7.75, the BF516 at $8.75 and the BF518 at $11.85. Processors are sampling immediately. All prices are based on 25,000-unit quantities.

How do save money in this worldwide tight economy so that you can have a Merry Christmas with lots of gift giving? (Not that gift giving is the main point of Christmas) Well, one way is by using VoIP to telecommute. Research done by Aastra found that commuters driving into the UK's largest cities could potentially save enough money by Christmas to buy more than half a kilometer of wrapping paper if they worked from home just one day a week. Based on commuters with 50-mile round trips, the average transit time soars in London to 111 hours - almost fourteen working days a year. Researchers found that London was the most expensive and time-consuming city to commute into, followed by Leeds and Bristol.

                                                    Aastra 57i CT phone

Telecommuting just once per week could save on average, could save £19.36 per day of telecommuting or £174 ($271 U.S. dollars) in the nine weeks running up 'til Christmas - enough to buy 17 turkey crowns, with change to spare for cranberry sauce - if they were equipped to work from home one day a week. In London, this figure soars to £41.90 a day - more than twice the national average due to parking costs, traffic delays, and petrol.

I wonder if this study was done since the recent petrol/oil price jobs? In any event, the research commissioned by Aastra revealed that commuters making 50-mile round trips by car, on average, could save £19.36 per day by working from home. In London, this figure soars to £41.90 a day - more than twice the national average.

I have an Aastra 57i (see review) at home that I use to telecommute myself. One nice thing about the Aastra VoIP phones is that they licensed Packet8's NAT technology for their firmware, which solves those pesky VoIP-over-NAT issues.

Working from home one day a week could also save penny-pinching parents with young children more than £460 in day care (£286) and travel costs (£174) in the build-up to Christmas - enough to buy all of this year's top 5 most wanted presents, as predicted by the Toy Retailers Association, with change to spare for more than 130 bags of chocolate coins for their Christmas stockings.

According to Aastra, as Christmas looms and inflation hits a decade high, more people are looking to home working as a means to enjoy a better work/life balance and save money.

Michael Calvert, UK General Manager of Aastra, who commissioned the research, said: "Commuting to work everyday can be a major strain on people's finances, and considering the current economic climate it's not surprising that the mood of the country is more credit crunch than Christmas lunch. Commuters equipped with the right, readily-available technology, could save money and lower their stress levels by taking advantage of flexible working practices. With many workers able to do their job equally well, if not better, from home, it's a wonder why more companies are not encouraging home working."

"It's not just commuters that could see real economic benefits from flexible working practices, many companies could benefit from lower real estate and energy costs, higher morale, and increased staff retention. Flexible working technologies such as Voice over Internet Protocol phones can even reduce the cost of calls, while making corporate communications more effective."

If commuters with 50-mile round trips by car worked from home one day a week they could save enough money in time for Christmas to buy:

VoIP security is often overlooked by IT administrators as well as VARs and resellers that deploy VoIP in the enterprise. They do so at their own peril, however. One of the main factors behind using VoIP is to save money. Well imagine your IP-PBX has been hacked and you don't notice anything wrong until you receive the next phone bill with hundreds or thousands of dollars in phone charges. There goes all the savings you anticipated when you decided to install VoIP!



This laissez faire towards defensive security reminds me of Star Trek, where for whatever reason the Enterprise flies through space with danger lurking around every corner but they keep their defensive deflector shields off and often turn them on when it's too late. The Enterprise has a fusion reactor with nearly limitless power, so why not keep the deflector shields on all the time? Maybe they're just being "green" and shooting for five nines (99.999%)  of efficiency.

In any event, security in general is often overlooked, whether it's securing your web server or your email server, or confidential database servers. But in most cases when these particular systems are hacked it's usually just an inconvenience (defaced web pages, spamming through your email server) with minimal financial impact. Not so when it comes to VoIP. The financial impacts of a hacked SIP server or VoIP gateway could be tremendous. This is especially true for larger organizations which already have hundreds or thousands of calls per month, including international business calls. How does accounting find the fraudulent calls in the phone bills which are 4 inches thick? It's like finding the proverbial needle in a haystack.

If the hackers are smart, they will limit the amount of traffic they route through a hacked gateway as not to set off any red flags. It could be months or possibly even years before anyone notices anything is amiss. I'm reminded of an old PBX technology called DISA (Dialed In Switch Access) which was one of hackers first tricks to get free calling. DISA was designed to allow employees to remotely call into the PBX and get second dial-tone. With this second dial-tone using touchtones they could logon to ACD queues, monitor agents calls, and of course initiate outbound calls.

In fact, many years ago, TMC was hit with a DISA-like attack on our Comdial PBX resulting in quite a few international calls. If I recall, there was a vulnerability in the Keyvoice voicemail system which allowed someone to make outbound calls. Needless to say, I was able to shut it down pretty quickly.

Part of the attack also involved using a scripted dialer which accessed the voicemail system by automatically sending the # key, then sending a chosen extension (say 100), and then iterating through all the various PINs (0000 - 9999). Since TMC has a toll-free 800 number, the attacker only has to make at most 10,000 calls to find a PIN to a particular extension. Obviously, chances are they'd find the script in much less than 10,000 calls and you get 3 tries before the voicemail hangs up. Once the PIN is found, not only does the attacker have access to the the user's voicemail, they also have access to any DISA capabilities of the voicemail system. More reason why IP-PBXs today need to have a PIN expiration feature just like Active Directory supports password expiration. No matter how many times IT staff reminds employees to change their passwords/PINs, they just don't do it unless the system forces them to. I don't believe any of the Asterisk systems I've tested have password expiration - so my open source Asterisk fans, if you're listening, add it to the code, will ya?

With all this in mind, I was fascinated to read an article by an Austrian company IPCom titled "Analysis of a VoIP Attack". It's an excellent read. Let me give you the abstract:

Here's a teaser:

1 The Attack
1.1 Analysis






Max-Forwards: 100

4985269162167113266321671132663213.130.74.70

Call-ID: 83764811100011101110010010110101101100111001001011
0101111110111000100213.130.74.701.2.3.41863480914f
df23881052555169000021671132663-

CSeq: 1 INVITE

Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 394 Contact: <sip:fdf238@213.130.74.70:3808;transport=udp> 4509759162167113266321671132663213.130.74.70174046 6380 To: <sip:810525551690000@1.2.3.4> From: <sip:5199362832664@1.2.3.4>;tag=21671132663- 00100213.130.74.701.2.3.41863480914;rport Via: SIP/2.0/UDP 213.130.74.70:3808;branch=100100101101011111101110 INVITE sip:810525551690000@1.2.3.4;transport=udp SIP/2.002:12:42 SIP_TR> [GW] < Stack: from 213.130.74.70:3808

Let's have a look at this SIP message. The funny thing is that absolutely nothing in this SIP message is trustworthy: Probably the SIP message has been received via UDP and the source IP address could be easily spoofed. Further, every data in the SIP message is user generated (in this case by the attacker) and does not necessarily reflect real data. Nevertheless, let us try to analyze the message: Ok, you've been thoroughly 'teased', now go read the full article (PDF). Good stuff!
On 23.09.2008, heise.de reported an attack against VoIP devices of German VoIP users [heise]. This article references a thread in the IP-Phone-Forum [ipphone] in which people report that their VoIP phones started ringing in the middle of the night and displayed incoming calls from the phone number 5199362832664. One of the users presented a log file of a Patton SIP device which captured the suspect INVITE request:

Fonality's CEO Chris Lyman spoke with me earlier today to talk about the strong inroads Fonality has been making in the call center market. Chris said, "Fonality has become a big player in what I like to call the micro call center market. We launched our call center product in 2005 and we have almost 3,000 deployed call centers now."

When asked to define "call center" since many people have a different definition, Chris responded, "Anybody who purchases Call Center Edition plus HUD Agent. We can't imagine someone buying barging, monitoring, recording, and queue reporting if you weren't doing call center activity. Since these products effectively add in some cases double the costs to the phone system, you're pretty serious if you're buying those."

When asked what percentage were call center deployments versus regular enterprise deployments, Chris said that 40-50% of all their solutions sold are Call Center Edition.

Chris stated that there is a vastly under served and untapped call center market when he explained, "The 5-50 agent market where I have a regular business with 50 employees, but I've got 10 sales people who work the queues. And so there is this micro call center that has been ignored by Genesys and the expensive players out there for all these years because you cannot afford a $100,000 drop-in, bolt-on Avaya call center system. I think the low end of the market hasn't been able to afford that and we're enabling the micro call center market."

Chris explained that Fonality PBXtra Call Center saved Crusecom, a Michigan-based outsourcing call center, more than $250,000 annually. Art Cruse CEO at Crusecom explained he'd have to hire a full-time Avaya engineer on-site at $140,000 - $160,000 per year plus maintenance costs of the Avaya system. He also explained Avaya call center phones are more expensive than regular Avaya phones. They also have 94% call completion rate or an amazing 6% abandonment rate. Other call centers are coming to look at how they've achieved such a phenomenally low abandonment rate. Fonality's call center system has helped Crusecom attain rapid growth. The $250,000 savings enabled Crusecom to invest in a new 14,000 square foot call center facility that will house up to 150 call center agents.

"Fonality and PBXtra Call Center made our whole business model of providing cost effective, outsourced onshore call centers possible. They have delivered remarkable business benefits to our company," said Art Cruse CEO at Crusecom. "When we bought our Fonality system, it was 75 percent less than a comparable Avaya system. With Avaya, we would easily be paying an extra $250,000 per year in support, maintenance and hardware costs, which would directly impact our bottom line and limit our growth capabilities."

Crusecom provides 24x7x365 call center operations to state agencies and companies that want to keep call center operations in the U.S. but cannot afford the high costs of urban call centers. Since deploying PBXtra Call Center in 2007, the company has expanded its customer base, added 50 new call center agents and increased inbound call volumes.

"PBXtra Call Center is amazing - the more calls we get the more we save," Cruse continued. "We've been able to reinvest these cost savings in our company and grow our business very rapidly."

By leveraging web-based reporting capabilities in PBXtra Call Center and other technology developed in-house, Crusecom is the only electronic benefits transfer (EBT) call center in the country that is offering customers real-time, web-based service level agreement reporting. In addition to the $250,000 he is already saving with Fonality, Crusecom estimates that this real-time, self-service reporting functionality saves his company eight to 16 man hours every day, or $50,000 to $100,000 annually, while providing customers with better service and support.

"Gone are the days of the cumbersome, big-iron call center oligopoly. Crusecom is a perfect example of why there is a changing of the guard in the call center market as companies rapidly adopt more agile technologies like PBXtra Call Center," said Chris Lyman CEO of Fonality. "Small and mid-size call centers need affordable, flexible phone systems with solid service agreements that allow them to grow rapidly. Fonality is delivering on these requirements and is constantly innovating with newer advanced technologies."

Chris also explained that Fonality's flat rate support was also a key advantage over other call center IP-PBX competitors. Lastly, he explained the hybrid-hosted approach enables call centers to have home agents without the need for VPNs. The hybrid-hosted approach resolves pesky VoIP over NAT firewall issues making telecommuting a much easier approach with lower TCO.

Registration to the upcoming Digium|Asterisk World conference is now open, according to an announcement today from Technology Marketing Corporation (TMC) and Digium, the open source Asterisk Company. So you early bird types can now register and add it to your Calendar now, lest you forget. TMC and Digium also announced the launch of the new event Web site at www.digiumasteriskworld.com.
 
Digium|Asterisk World is collocated with TMC's INTERNET TELEPHONY Conference & EXPO East 2009 and is a 3-day event commencing on February 2, 2009 in Miami, Florida.
 
According to TMCnet reporter Michelle Robart: Entering its third year, Digium|Asterisk World is the conference that educates business users, resellers and executive decision-makers on "Everything Asterisk."

Michelle also gives more interesting details about Digium|Asterisk World:
 

The Aastra 57i is one of my favorite VoIP phones. The 57i and its sister, the 57i CT (cordless telephone adjunct), offers some unique features and is undoubtedly one of the most flexible IP phones you will find. The 57i and 57i CT sport a large 144 x 128 pixel graphical backlit LCD display and 6 dynamic context-sensitive softkeys. Although the resolution isn't designed for photos, it's a very large LCD, one of the largest I've seen making it very easy to read the number of voicemail messages, the CallerID of an inbound call, and the one touch feature keys you've programmed. The 57i is of course SIP-based making it fully interoperable with IP-PBX platforms such as Asterisk. The 57i and 57i CT offer advanced XML capability to access custom applications and support for up to 9 calls simultaneously.


                                        57i CT Web Interface Preferences

The 57i CT is exactly the same as the 57i except it has a built-in wireless transmitter in the base unit and it comes with an integrated WDCT cordless handset with a range of up to 300,000 sq ft. The cordless phone sports 10 previous number redials, a mute button, on hold, Callers List, transfer,4 ringtones, and more. My only complaint is that there isn't a dedicated transfer button. While on a call using the wireless handset, you have to press the F (Function) button, scroll to Xfer, and then enter the extension number. The mobility the cordless handset gives you is perfectly suited for executives, mobile warehouse personnel, as well as retail staff. Here's a photo of the 57i CT on my desk along with the cordless handset:


The Aastra 57i has excellent XML browser capabilities. You can for instance install an Aastra developed and freely available XML application that gives you Visual Voicemail on trixbox CE, an Asterisk-based IP-PBX. This allows you to scroll through your messages on the phone, see the CallerID, and pick the one you want to play. Here's a screenshot of visual voicemail in action:


Another freely available app is meet-me conference with the ability to see attendees along with the ability to kick or mute participants from the LCD. Still another XML app lets you play .mp3 files using the LCD and buttons and it can even read the ID3 tags in the MP3 files to browse by album, artists, or entire song list, as seen here:


One useful business app is a SugarCRM XML app that lets you look up names and numbers in your SugarCRM database simply by typing the name into the LCD using the numeric keypad. After fining the proper record you can instantly dial the contact.


Other freely available apps include Area code lookup, Ask Google, CNN News, ESPN News, Horoscope, Local Weather, Netflix, Movies, Stock, World Clock, and Yahtzee. Here are some screenshots of many of these free XML applications.




The 57i supports directories, including a shared corporate-wide directory, which is easily created simply by creating a CSV file called directory.csv file and uploading it to the TFTP server.  Then you edit the aastra.cfg file on the TFTP server and add:

directory 1: directory.csv

When you next reboot the phone, the corporate directory is automaticly downloaded. 

The phone also supports callers logs, a redial with up to the last 100 phone numbers, and 12 programmable keys located next to a large LCD. The 6 top keys are static softkeys with up to 10 programmable functions. The 6 bottom keys are state-based softkeys with up to 20 programmable functions. All top and bottom keys can be configured for specific functions, such as Intercom, Callers List, speed dials, etc. When you program one of the keys, it displays its function within the LCD. For instance, if you look at my phone you'll see I have China King set as one of my speed dials. One press and I can instantly call my favorite Chinese restaurant!


The 57i phones support shared call and bridged line appearances (SCA & BLA). In fact, Aastra's line of IP phones do a good job of emulating key system functionality. It fully supports BLF (busy lamp fields) allowing you to see when a person/extension is on the phone. From the web interface you simply pick BLF, enter the extension and the name of the person. When the monitored user is idle, there is a small telephone icon shown with the handset on-hook. When the monitored user is on an active call then a small telephone icon is shown with the handset off-hook.

The BLF also acts a one-click speed dial, allowing you to simply click a BLF and instantly ring that extension. Another BLF capability is the BLF List feature. The BLF List feature on the Aastra IP phones is specifically designed to support the BroadSoft Broadworks Rel 13 Busy Lamp Field feature simply by entering the BLF List URI. The BLF List URI is the name of the BLF list defined on the BroadSoft BroadWorks Busy Lamp field page for your particular user. For example, my480i-blf-list@as.broadworks.com.

                                        57i CT Web Interface Preferences

Some other features of note. First, in addition to shared call and bridged line appearances, typical features such as call forward, call transfer, call waiting, intercom and local 3-way conference are available. Second, the Aastra 57i and 57i CT support stuttered tone letting you know that you have new voicemail waiting. Third, the IP phones support several different languages. You can have the IP Phone UI and the Aastra Web UI display in English, French, Spanish, German, and Italian.


                                        57i CT Web Interface Preferences

It's also worth mentioning that the Aastra 57i offers two different module extension options (images to right). The 57i supports up to three Aastra 536M modules, each offering 36 keys with LED indicators to create a feature rich attendant console. Or, it will also support up to three of the advanced Aastra 560M modules, each offering 60 keys with a screen based LCD display and LED system.

The 57i Aastra IP telephones have a full-duplex speaker phone with excellent voice quality. I've used an Aastra phone's speaker phone for a long time and have never had any complaints. The phones sport dual auto-sensing switched Ethernet ports along with integrated IEEE 802.3af Power-over-Ethernet.

Configuring Aastra phones are done via configuration files which can then be uploaded to a TFTP server. The syntax of Aastra config files are very straightforward. However, due to the flexibility of Aastra phones and the amount of features, it can get a bit complicated. In fact, Aastra phones are so flexible and configurable, that someone even built an Aastra Configuration Generator, leveraging Microsoft .NET to make it easer.

The stated goal is: "This utility will hopefully take some of the pain out of creating the configuration files for the Aastra range of IP phones." The website adds, "With it you can create the base config, the buttons (programmable, top softkey and softkey) config, the console button config, the IP phone directory and encrypt the file prior to uploading it. You can even link in with NMap to perform network browsing for Aastra MAC addresses. You should be able to quickly and easily create phone configurations without having a vast knowledge of the options required. I have tried, where possible, to display tooltips for options and restrict the choice of input to what the option expects."

Conclusion
The Aastra 57i and 57i CT are two of the most flexible VoIP phones you will find on the market today. The large LCD and plethora of configurable buttons make this phone a must have for executives and other enterprise users that require maximum flexibility for increased productivity. My only suggestion to Aastra would be to offer a color LCD version with a higher resolution for displaying Internet graphics pulled via RSS using an XML application. Perhaps the next series, most likely called the Aastra 67i series, will have color? I love the Aastra 57i and the Aastra 57i CT and in fact have a 57i CT as my primary desk phone. What more can I say? Two enthusiastic thumbs up!

Price: 57i: $219.95 57i CT: $319.95

57i and 57i CT Resources:
57i
Aastra XML Scripts for trixbox CE (PDF)
57i Installation Guide (PDF)
57i User Guide v2.1 (PDF)
Latest firmware downloads

57i CT
Aastra XML Scripts for trixbox CE (PDF)
57i CT Installation Guide (PDF)
57i CT User Guide v2.1 (PDF)
Latest firmware downloads

In August, I visited Digium's headquarters and excitedly wrote, "the next release of AsteriskNOW (1.5) will bundle the very popular FreePBX front-end GUI. This is big news! FreePBX has become the defacto standard for web-based GUI administration of Asterisk, so now Asterisk fans can download AsteriskNOW, boot of the self-install CD and not only getting a working copy of AsteriskNOW, but FreePBX as well!" I also explained that the forthcoming AsteriskNOW which previously ran on rPath will now be using the very popular CentOS (5.2) distribution

Well my Asterisk friends, that day has come! AsteriskNOW just came out in beta. Now go grab yourself a copy!