Free 60 second phone calls

Remember Jahjah's free phone service offer. Well, James Bowman over at for60secs.com has started a service with a different angle.They offer free calling to US48 phone numbers (all U.S. except Hawaii and Alaska), but for60secs.com records (with consent!) the calls and publishes them on their website. James said, "We hope to pay for the phone service with the ad revenue from people listening to the calls.It's as simple as that." To use the service, you simply dial their toll-free 866-460SECS number and then enter the number you wish to dial.

He explained that they envision a few different kinds of user for the phone service, such as people who are very price-conscious or shameless self-promoters who view it like reality TV - you give up privacy in exchange for fame.

The original plan was to limit calls to 60 seconds - hence the name, but during the beta period the limit is actually 6 minutes.

Here's a quick Q&A via email I had with James:

1) How are you terminating the calls? VoIP? Regular PSTN conference bridge?

VoIP. We're using Asterisk using SIP into Gafachi for termination and origination.

2) Aren't you losing quite a bit of money to terminate the call (not to mention paying for the 866 toll-free number? Even using VoIP would cost some money, albeit less. Can you explain the architecture?

Asterisk using SIP into Gafachi. Asterisk records the call into local disk then we encode the recordings and add them to the SQL database editors vet and tag the call web presentation uses Apache and mySQL with a Flash player for the calls.

[he didn't directly address the cost of termination]

4) What if they make a call and then realize they have to give personal info, like their social security number? Do you give them the option to delete the recording? I would assume there might be some liability risks for you if you don't - i.e. someone uses personal info for identity theft. Of course, they're idiots for knowing the call will go public, but in today's litigious society, better safe than sorry.

Yes, there is a disclaimer for both parties in the call itself, and calls are vetted by editors before they are published.

Conclusion:
An interesting "free" phone calls model. Do you have a bit of exhibitionism in you? Feel like sharing romantic calls to your significant other? Of course, I don't think your significant other would be pleased to hear their private moments shared on the net all to save a buck. If you're feeling the voyeuerism inside you can also check out some sample recorded calls that for60secs.com has already posted online. Of course, all this effort just to save a few cents? Assuming an average of $0.10/minute and you use free60secs.com for 30 minutes per day, you only save $3. But I suppose $3/day X 30 days is $30/month saved.

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Tags: asterisk, gafachi , voip
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Written by VoIP & Gadgets Blog on October 5th, 2006 with no comments.
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Fonality acquires trixbox

Fonality will announce tomorrow that they have acquired trixbox, formerly known as Asterisk@Home, and the the world's largest Asterisk-based community. Trixbox is a turn-key, bootable .iso CD image that can turn a PC with no OS into an Asterisk server with a variety of open source tools in just a few minutes. The trixbox application lets someone download a bootable .iso image that then automatically installs Linux, Asterisk, SugarCRM, MySQL, FreePBX, and a whole variety of other applications. Trixbox fully supports the Linux yum command and RPM ecosystem for performing updates and bug fixes.

Essentially, trixbox uses the latest and greatest version of Asterisk. Within 48 hours of a new Asterisk version, you see a new release of trixbox and they add their own host of patches they they put on top of that. These are basically patches of innovators inside the Asterisk community that didn't want to sign a double waiver. The double waiver extends the GPL agreement by stating the code has to also be given to Digium.

I spoke with Chris Lyman, the CEO of Fonality earlier this afternoon to talk about the acquisition of trixbox. He began by wondering how I figured out Fonality and trixbox were working closely together.

Chris: I don't know how you figured out on June 6th that we were getting involved with trixbox, but my hat goes off to you.

Tom: I recalled that I was wondering why Fonality would offer their hudlite, a real-time call control and presence management platform that works with the commercial (paid) Fonality PBXtra. I didn't understand why Fonality would want to make a "free" version of Asterisk (trixbox) more "feature-rich". Curious what the relationship was between Fonality and trixbox, I I did some detective work. Besides googling I also did registrar 'whois' lookups on hudlite.org (Fonality website) and trixbox.org and noticed that the IP addresses were the same - in other words - the same web server. Interesting to say the least, which is what sparked my June 6th post.

In my June 6th post, I pondered, "Did Fonality buy out the rights to Asterisk@Home and then change the name to Trixbox.org? What does this mean for the popular open-source Asterisk@Home distro (now Trixbox) considering Fonality is a for-profit Asterisk solutions provider? Is the plan to try and convert Trixbox users (generally novice Linux users) into paying Fonality users? Fonality certainly focuses on businesses that have little or no Linux experts, so there is certainly a potential synergy there. Well, the mystery continues... I'll post more when I hear back from Chris @Fonality." Chris Lyman and Andrew Gillis responded in a follow-up article.

In any event, it appears the seeds were sown for this acquisition back in June and that I was 4 months early in my pondering whether Fonality had acquired trixbox.

Tom: So what about the double waiver requirement to give the code back to Digium? Is this an advantage of trixbox since it doesn't have a double waiver?

Chris: There are a number of open source innovators that don't like that -- like faxing. The only reason why fax isn't in Asterisk, but they are using it in trixbox, is because the smart guy that made it (fax code) didn't want to sign that waiver over to Digium. So basically consider trixbox the latest greatest Asterisk plus a whole lot of innovation.

Tom: So what market are you going after with trixbox?

Chris: So in terms of what market we're going after with trixbox, we're not really going after a market. trixbox is really a community of Asterisk innovators and we're just going to be supporting that community. It's our way of supporting the platform that has been a big part of our success.

Tom: What is the value that trixbox brings to Fonality?

Chris: The value to Fonality is the community value.  The business value that trixbox brings is that there is probably a number of IT directors lurking in the trixbox community that are sort of trying for free, but really do want a commercial company to hold their hand when they roll out. And so we just want to make ourselves known that there is an option to go fully supported.

Tom: So by working within the community you hope to build brand awareness for your commercial-based Fonality PBXtra?

Chris: Yes, we want to build some brand awareness in the Asterisk community to let them know we are a serious player that has a 100% supported, 100% service model.

Chris: The trixbox forums has over 20,000 posts in the last 3 months. It has become the defacto place to get questions answered about Asterisk. Questions answered about rolling an open-source small business environment. And that's really the value we saw is - there are a lot of smart open-source people in that community.

Tom: What are the download numbers?

Chris: 1,500 people download trixbox every day, which is more than Digium. Mark was quoted in a Forbes article as saying 1,000 downloads per day and we were surprised since we averaged 50% more than that.

Tom: Any issues with people knowing about the trixbox brand and knowing that is the latest and greatest version of Asterisk?

Chris: I would say given our download numbers and given the fact that we get more downloads of Asterisk every day more than the rest of the world combined, I would say no, there is no brand problem.

Tom: So how is Fonality going to contribute to trixbox with this investment?

Chris: There's two things that are really really important for us to let the world and the community know. Number one is, trixbox was free, is free, and will always be free. And when I say, I mean pure GPL. It won't have a double waiver, you won't have to sign over any rights to Fonality, and we're not going to get into any of those complicated licensing schemes that you see with some other open source companies. Number two, we're contributing broad financial support to the trixbox platform to continue to improve that application. This is not just a community of that site that we're going to pay the bandwidth on. We actually have a host of engineers internally working on improving trixbox.

Tom: On a different note, any thoughts about integrating SugarCRM, MySQL, etc. onto the Fonality PBXtra hybrid-CPE-hosted solution for an "all in one box"?

Chris: Now that we are very much looking into. We've had talks with SugarCRM about it. Looking at ways of linking PBXtra and SugarCRM's contact center together on one box. That is a product you will probably see in the future from Fonality. I will say, my message to any business, be careful of how much load you point on a single server since it becomes a single point of failure for your business.

Tom: I know the Fonality code is a more secure and stable version of Asterisk but running an older Asterisk codebase, yet without sacrificing functionality. So I was wondering what percentage of code that is in trixbox is going to come back to Fonality?

Chris: Today, it is 0% because our version of Asterisk has been hardened aggressively over the last two and a half years. We think there may be a time if the Trixbox community requests it where we might give our version of our code to the community and call it you know, "stable". But really, more than anything the community wants the latest and greatest features and are willing to sacrifice a little bit of reliability to get there. And so unless we see a great need, we're not going to mix the two different flavors.

end interview...

One final point of note is that trixbox founder Andrew Gillis will join Fonality and continue to lead the trixbox community. The main takeways from this news is that Fonality will commit engineering resources and financial support to trixbox, and just as importantly, trixbox will continue to be 100% GPL without a double-waiver. Trixbox founder Andrew Gillis said, "Fonality shares my vision of making Asterisk free and easy for everyone. They have already proven to me how serious they are by committing a team of engineers to help create the next version of trixbox.

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Tags: asterisk, fonality, freepbx, linux, mysql, sugarcrm, trixbox, voip
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Written by VoIP & Gadgets Blog on October 3rd, 2006 with no comments.
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Becoming an Asterisk guru

So you wanna be an Asterisk guru do ya? I bet you wanna be just like this Asterisk Master in the picture. Not the good looking guy on the left - I'm talking about the Asterisk guru himself, Mark Spencer - on the right.


Well, if you're not the self-taught type, or even if you are and just want some expert Asterisk training to "brush up your skills" -- then you may want to check out the TMC University program going on at Internet Telephony Conference & Expo which features Asterisk training. Our last show we actually had a full day dedicated to Asterisk training. According to Rich, the room was at capacity and we had rave reviews on how the session went.

Rich has a detailed synopsis of the Asterisk training sessions going on at the show. So go check it out.

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Asterisk replaces Cisco CallManager

If you recall, back in March, I talked about how Cisco was embracing the SIP standard, i.e. "Cisco is fully embracing the SIP standard on their desktop phones. I interviewed Cisco last week and they told me that I was the first journalist or analyst to have a "first look" at this major announcement." The March article explains in detail about Cisco's Unified Communications strategy and the impact embracing SIP will have on Cisco and the VoIP industry as a whole.

I wrote back in March, "Perhaps I misread the tone of the Cisco representatives during my call, but they initially seemed to downplay the significance of Cisco's embracing of SIP in favor of focusing on the entire Unified Communications platform. But in my opinion, Cisco embracing SIP is just as big news as their Unified Communications system..." This "big news" and impact on the VoIP industry will be explained in just a bit, but you probably figured it out already from the title of this article.

When I spoke to Barry O'Sullivan vice president and general manager of Cisco's IP Communications Business Unit back in March, he told me, "Our strategy is to put as much intelligence onto the network to allow applications whether our applications or others applications to take advantage of that intelligence. So call processing intelligence, presence intelligence, and rich-media applications. Our strategy is to be open and extensible. In other words, we have embraced SIP and built in natively into our platform and we'll make these capabilities available to third party applications and phones as well as our own applications and phones".

The "SIP support" exclusive that Cisco gave me caused my eyes to open wide with excitement since open standards will help drive the VoIP industry's growth. I knew what this news meant. Cisco, now offering SIP firmware for their phones and support for 3rd party SIP phones to connect to CallManager was huge.

But it also got me thinking how this could hurt Cisco. Cisco IP phones aren't cheap, and they run Cisco's proprietary Skinny Call Control Protocol (SCCP, or "Skinny"), so you are stuck with these Cisco phones and using Cisco CallManager once you buy them. Before Cisco embraced SIP back in March, you are locked into being a "Cisco shop". Now the beauty of SIP phones is that if you want to upgrade to a different IP-PBX that supports the SIP standard, it's a relatively painless upgrade to swap out the one and in with the new. Of course, it doesn't even have to be the same IP-PBX vendor. Sick of Cisco CallManager? Go with Asterisk. Sick of Asterisk? Go with Pingtel or another IP-PBX vendor. Point being, you get to keep your investment in the SIP IP phones without having to shell out ~$200-$300 per VoIP phone per workstation. Are you seeing where I am going with this? Nobody was ever fired for "going with Cisco", but IT managers are now seeing that they are paying a premium to use Cisco gear when they can just as easily use a less expensive open-source solution such as Asterisk.

So it came as no surprise to read my first application story today from Network World about somebody that was using Cisco CallManager and Cisco IP phones running the Skinny protocol that decided to switch the firmware to SIP and then dump CallManager in favor of a third-party IP-PBX - in this case, an open-source IP-PBX from Asterisk. Sam Houston State University (SHSU) is moving 6,000 students and faculty off Cisco to the open source Asterisk IP-PBX.

The main reason for this migration was cost, according to Aaron Daniel, senior voice analyst at Sam Houston State University. "We thought that it will be more cost effective in the long run to go with an open source solution, because of the massive amounts of licensing fees required to keep the Cisco CallManager network up and running," says Daniel. According to the article, each phone attached to the CallManager required a separate annual licensing fee to operate. I'm not sure that's entirely accurate. I could have sworn you weren't required to pay annual licensing fees for the phones, but you did have to pay an optional support cost based on the number of phones. But assuming this is true, this could become yet another strong driver to cause Cisco CallManager shops to jump ship to save on TCO (Total Cost of Ownership). In SHSU's Asterisk/Cisco setup, they will keep their existing Cisco phones but attach them to Asterisk servers on the back end, thus eliminating the phone licensing costs.

It's ridiculously easy to switch phones. You simply swap out the firmware on the Cisco phones from Skinny to SIP, reboot the phone, and the phone will automatically register with the Asterisk server. There are millions of Cisco IP phones and CallManager shops out there that now have a choice and as I said in my March article, that is a good thing.

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Tags: asterisk, callmanager, cisco, unified communications, voip
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Written by VoIP & Gadgets Blog on September 20th, 2006 with no comments.
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Gizmo Brings Free Calls to Small Businesses

Gizmo, created by SIPphone is one of my favorite VoIP clients and not just because they have one of the prettiest & slickest looking interfaces. Nope, Gizmo has some of the best features in any softphone client you will find and unlike Skype which uses a proprietary protocol, Gizmo uses the SIP standard. Back in July, Gizmo announced their "All Calls Free" program which allows you to call all of your Gizmo friends for FREE on their landline or mobile phones in 60 Countries. SIPphone said today that it is extending its All Calls Free calling plan to business users world wide.

The only catch is that the person you are dialing must also be a Gizmo user with their landline and/or mobile phone registered within the Gizmo client. According to SIPphone, "Callers are encouraged to make free PC-to-PC calls whenever possible. The All Calls Free calling plan applies when both call participants are active Gizmo Project users making a few phone calls per week with Gizmo Project. Free calls may originate from anywhere in the world, but must be to a qualifying number in one of the 60 countries for which the plan is offered. Calls must be made from the caller's contact list to either the "home phone" or "mobile phone" number the call recipient included in his or her profile, and both parties must have shared each others profiles with one another." If the call doesn't qualify under the All Calls Free plan, then you simply pay the low rates offered by SIPphone.



The beauty of the SIPPhone client is that it supports dual SIP logons. Thus, you can register with Gizmo's SIP server, as well as any second SIP server you wish, such as an Asterisk IP-PBX, Switchvox, epygi, Mediatrix, etc. Now you can simply deploy Gizmo Project to all your employees (which integrates nicely with Asterisk and other PBXs) and get free calls between all your employees. With this dual SIP configuration, incoming calls from your PBX will be clearly identified, as they are when a call comes from Gizmo. In addition you can choose which outgoing line to use (i.e. Asterisk or Gizmo).

 

If I have some time I'll play around with the latest version of Gizmo Project and report back here.

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Written by VoIP & Gadgets Blog on September 19th, 2006 with no comments.
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Asterisk 1.4 unveiled

Digium Inc., the Asterisk company, today announced Asterisk 1.4. According to Digium, the new version isn't available for download until October however. Although I know Mark Spencer, I have no inside information, however, my guess is that Asterisk will launch the software at TMC's Internet Telephony Conference & Expo in San Diego. Seems like the perfect venue to launch this major software release.

Asterisk 1.4 is the first major release of Asterisk since the release of Asterisk 1.2 in November 2005. It includes over 20 new functionality additions including IPFAX compatibility, unified messaging capabilities and Jabber/Jingle/GoogleTalk protocol compatibilities. Asterisk 1.4 features overall quality and performance improvements, as well as increased scalability and interoperability.

"This is by far the best version of Asterisk to date,” said Mark Spencer, president of Digium and creator of Asterisk.  "With the support of the Asterisk community, we have been able to develop an advanced platform that will make it even easier for users to migrate to VoIP, especially those in the enterprise community.”

Specific enhancements featured in Asterisk 1.4 include:
•    Generic Jitter Buffer- improves the quality of a call during network congestion.
•    Asterisk Extension Language Version 2- simplifies programming and dial plan configuration.
•    T.38- allows IP FAXes to pass through the server.
•    Jabber/Jingle/GoogleTalk- supports compatibility with all of these networks.
•    Increased language capabilities- offers new language capabilities in English, Spanish and French as well as new sounds and improved sentence structure support.
•    Unified Messaging- integrates voicemail, email, and fax into a central mailbox where users can send, retrieve and manage all of their messages using any communication device.
•    Whisper Paging- allows for selective, pre-programmed call interruption with controlled volume levels and muting capabilities.

Additionally, Asterisk 1.4 now includes variable length DTMF support (touch-tone signaling for IVR applications), the option for programming shared line appearance, centralized RADIUS storage for call detail records, a built-in web manager interface and a simplified, single user configuration for SOHO/SMB users. Asterisk 1.4 also offers increased memory usage and performance improvements such as improved interoperability of SIP call transfers, IAX2 scalability improvements, enhanced IAX2 media stream capabilities (enabling direct audio communication between IAX devices while eliminating server involvement and maintaining billing and control functionalities), Cisco® SCCP support, SNMP monitoring, and RTP native bridging capabilities.

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Tags: asterisk, googletalk, jabber, jingle, voip
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Written by VoIP & Gadgets Blog on September 12th, 2006 with no comments.
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Fonality PBXtra the world’s largest commercial Asterisk deployment

Chris Lyman, my pal over at Fonality (and their Founder/CEO) told me that Fonality will soon announce its position as "the world's largest commercial Asterisk deployment." Them's some mighty BIG claims!  But I have no doubts it's true. Well, maybe a few doubts. I would have thought the Linux gurus that download and install Asterisk for free, would as a whole be the #1 commercial Asterisk deployment. Chris stated, "It is important to distinguish between free Asterisk downloads and the 1,000 companies that have actually purchased PBXtra to run their business," Chris Lyman said."There is no secret why Fonality is leading the Asterisk market: we've extended Asterisk far beyond its base feature set, added tons of stability code and provide our customers with 24-hour support."

Chris told me that more than 1,000 SMBs with 15,000 employees have purchased and deployed Fonality's PBXtra, an Asterisk-based IP-PBX that began shipping in October 2004 and which is now averaging more than one million calls per week across its PBXtra platform.
 
PBXtra is less expensive and easier to deploy than all other major PBX offerings on the market today. The price advantage comes from open source technology and the deployment ease comes from an intuitive, web-based interface designed for the average business owner. PBXtra combines simplicity of installation with depth of features, providing SMBs with an enterprise-class PBX for 40 percent to 80 percent less than competitive offerings.

Fonality's PBXtra product line includes enterprise-class features such as telecommuting, branch office support, voicemail-to-email, click-to-call, VoIP, softphones, support for IP and analog phones, call recording, conferencing and advanced call center functionality. In addition to PBXtra Standard Edition and Call Center Edition, Fonality also offers PBXtra customers HUDpro, (Heads-up Display Professional) its real-time call control and presence management application.

PBXtra is available direct from Fonality or through a Fonality reseller. Pricing starts at $995 for the PBXtra Standard Edition server with unlimited licenses, or $2,935 with ten phones. PBXtra's Call Center Edition, which allows for distributed call centers, is $1,995. HUDpro is $995 for an unlimited seat licenses with support for Windows, Mac and Linux desktops. For more information about PBXtra, HUD or becoming a Fonality reseller, visit www.fonality.com.

See Also:
Fonality, an Asterisk solution passes 20 million calls
Fonality Asterisk-based IP-PBX breaks out of stealth mode

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Windows-based open source Yate IP-PBX launches stable version

Yate is an open-source IP-PBX solution similiar to the open-source Asterisk solution, but unlike Asterisk which is primarily run on Linux, Yate was wrriten in C++ and therefore can be compiled to run both on Windows and Linux. I wrote about YATE in my Sangoma telephony cards article, which is worth checking out. Today, Sangoma Technologies Corporation (TSXV: STC), a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure and the Null Team Company, the primary developer of Yate, have released a stable native Windows-based version of the GPL-licensed Yate telephony project.

Yate has a powerful telephony engine which can be easily extended to include voice, video, data and instant messaging all unified under Yate's flexible routing engine.

“Most commercial IVR applications such as call centers are Windows-based. This open source initiative will free developers from reliance on hard-to- use, proprietary solutions based on expensive hardware,” says Sangoma Technologies President and CEO David Mandelstam. “Recognizing the inherent strengths of the Yate model, we have offered our technical and financial assistance to offer this project to the large market that is dominated by Windows. We expect to take a leadership role as we support the project for the developer community.”

“Yate plus Sangoma are a natural fit which allow integrators to build inexpensive, robust and flexible applications for telephony on the Windows platform,” adds Diana Cionoiu, Null Team CEO. “Since Yate supports widely-used VoIP protocols like H.323 and SIP, integrators can build IP call centers, IVRs or any other telephony applications using technologies available under Windows platform. Yate can be used with the new YateClient 1 and Mozilla Firefox embedded browser that allows integration with various databases.”

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Asterisk Receives VC Funding

On Wednesday, Digium, founder of the open-source Asterisk IP-PBX will announce its first round of VC funding – receiving $13.8 million from Matrix Partners. Matrix Partners was JBoss' initial investor, also an open-source solution - a Java application server. Matrix's expertise in open source was one of the deciding factors for deciding to invest in Digium. Asterisk is now at 1 million Asterisk users compared to 1/2 million at the beginning of the year - a phenomenal growth spurt and with over 1,000 downloads per day.

The funding will help the continued growth of Digium and Asterisk, especially in launching new offerings for the enterprise communications market, ranging from small to large businesses. Profitable since 2002, Digium has experienced 100% growth in each of the last several years.

"We believe Digium has the potential to become one of our most successful open source companies, as every company in the world relies on telephony and the use of PBXs in order to run their businesses,” said David Skok, a general partner at Matrix Partners and JBoss board member. "As companies continue to be attracted to the cost savings and powerful new capabilities of Voice over IP, the opportunity for Digium becomes massive. Digium is definitely in a position to become the next big open source company, behind Red Hat, JBoss and MySQL. Their current revenues, profitability, and growth rates are extraordinary.”

Update: 11:44am Wednesday
I just had one other important thought with regards to this VC funding. Mark Spencer, the founder of the open-source Asterisk movement and Digium wasn't beholden to anyone else due to any sort of major financial investment. While I am pretty certain that Adtran gave some funding to Digium, I don't think it was on the scale of this multi-million dollar funding. I hope that Mark, the Asterisk guru and "telecom rebel extraordinaire" won't have to change himself or the company too much to accomodate Matrix Partners' wishes. Let's hope Matrix Partners has a pretty much "hands off" policy, except to help grow Digium and Asterisk.

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Another IP-PBX company bites the dust?

After the news that Zultsys was going out of business, only to hear that they are being resurrected, word from two sources is that a well-known IP-PBX company may be on its last legs. I don't want to disclose who it is at this point without some further investigation. No point causing a company harm from what is just rumor at this point, but I will keep you posted.

This got me thinking though. What happened to the days when there were dozens of PBX manufacturers? Sure there are still many around, but many are hurting, and some have gone belly-up, such as Comdial, Praxon, and others. You have inexpensive open-source IP-PBXs such as as Pingtel and Asterisk that are just as feature-rich as the "big boys" (Nortel, Toshiba, Avaya, Cisco) at 1/8th the cost or less. How can a large company with hundreds of employees and with vastly larger overhead compete with a small nimble company like Digium, the founder of the Asterisk open-source movement?

Will open-source communications systems inevitably kill the major PBX manufacturers? Hard to say, but open-source sure didn't do SCO UNIX any favors when the "free" Linux O/S came on the scene. The days of proprietary communications are over, which also means more competition and smaller margins. In telecom it's SIP that is opening the doors for small start-ups to innovate without being blocked by proprietary and predatory tactics. Only the nimble with the best features, best value, best marketing, and best support will survive the long haul.

On a related note I recently discovered PostPath, a Microsoft Exchange Server alternative, which is the first to implement Exchange network protocols on a Linux email server and the first to let you use your existing Outlook clients with no disruption. According to this article, benefits of selecting the PostPath Server include avoiding vendor lock-in, saving money, increasing performance by 5x, improving resilience, and increasing flexibility and innovation. According to the article, by moving to PostPath you can slash software, storage and infrastructure costs by 75%. We have Exchange Server at TMC and have experienced our share of Exchange Server failures resulting in email loss. Disaster recovery for Exchange Server is just that - a disaster. We've had some outages that took 2 days to entirely fix. Postpath, while not open-source or free, is a Linux-based solution that is less expensive and they claim more reliable with quicker disaster recovery.

Now if only I could have a 100% open-source, IP-PBX, with Exchange Server functionality, built-in web server, Jabber/IM server, collaboration capabilities, mobile phone email synching (e.g. Blackberry), and just about any other communications method, all on a turn-key platform with each component interoperating/integrating - then life would be good.

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Tags: asterisk, digium, exchange server, ip-pbx, microsoft, pingtel, postpath, voip
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I’d like a pizza with pepperoni and a sprinkle of VoIP and some GPS too

Recently, I received an email from a pizza delivery solutions provider seeking my assistance in an interesting application that involves VoIP, GPS coordinates, Bluetooth, and of course pizza. This is not the first time I have linked pizza with VoIP. No siree bob! In fact, my Vonage VoIP line resulted in me getting a cold pizza. Almost sued Vonage over my damn cold pizza too.

I know what you're thinking. What kind of VoIP application does a pizza delivery company need, right? Well, the best way to explain the application that ties VoIP, pizza, Bluetooth, and GPS together is to include the request they sent to me.

We sell software services to pizza chains.  Our system tracks drivers as they drive around town.  Since we know where they are and where they are going, when they are about 4 minutes from the customers’ door, we want our computer to initiate a telephone call to the customer to say the driver will arrive momentarily.  We imagine some kind of a softphone application running on the PC connected to a VoIP access point to the phone system.  Our software already has the .wav files prepared to “speak” to the customers.

Any suggestions on how to configure this or who we should be talking to that might sell the necessary hardware software?  Is open source stuff available?

PS  Our system is written in C#, so is pretty flexible.

Pizza Pilot recently announced the completion of a multi-store test in Boston-area Domino's Pizza Franchises.  Pizza Pilot was successful in reducing labor, mileage theft and lost drivers.  New tools for measuring each driver's "Smart Hustle" factor as well as data mining techniques to identify and prioritize target customers with Platinum, Gold or Silver service levels were also introduced.

Pizza Pilot is a software-based product that works with any POS-including Pulse and TMS/National Systems.  In combination with Bluetooth and GPS-enabled cell phones, it tracks the location of each delivery driver every 60 seconds.  Pizza Pilot mapping software determines the optimal dispatching and automatically assigns orders to drivers-allowing managers to focus on inside operations.

Due to its ability to track each driver's progress on their way to each destination, Pizza Pilot can determine the moment actual deliveries are made and to update drivers' estimated return times as they return to the shop-making subsequent dispatches more accurate.  Maps or on-board navigation is also available which reduce lost drivers, while real-time tracking identifies and discourages unauthorized stops.

I'm most impressed with the fact that they want to call you when the pizza is 4 minutes away. Imagine that -- a pizza company that is courteous enough to call you when they are about to arrive. How many times have you been told the pizza will take 30 minutes only to see 90 minutes roll by; then when you call to check on your pizza they simply tell you the pizza guy is on the way. Really they have no idea where he is and they're simply patronizing you. Well, with this pizza delivery company, they track all their vehicles so they can tell you exactly where the vehicle is and on top of that they plan to call you using an automated dialer using a VoIP line.

Now if that isn't a hot application for VoIP, I don't know what is. Mmmm. hot pizza... Ahhhh. GPS & Bluetooth gadgetry... Ooooo... VoIP coolness... Ahhhhh...  All I need now is some cold beer & ESPN -- and I'll be in heaven. Maybe I'll launch SightSpeed 5.0 and watch some ESPN from work.

I suggested to Pizza Pilot that there is C# code out there to initiate SIP calls and they he may want to consider looking at Asterisk, the open-source IP-PBX. I also referred him to Erik Lagerway and Ward Mundy over @Nerd Vittles, an Asterisk blog. Erik suggested that since he was using C#, that something in Microsoft LCS might do the trick - or paid kits like those offered by Counterpath. Ward Mundy said he could code something for him and asked for more details on the size and scope of the project.

I'll keep you posted if the pizza/GPS/Bluetooth/VoIP application ever goes live. GPS VoIP pizza delivery just might be coming to a neighborhood near you!

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Tags: asterisk, bluetooth, c#, gps, open source, pizza, sip, voip
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Written by VoIP & Gadgets Blog on July 25th, 2006 with no comments.
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Microsoft and Nortel to co-develop IP-PBX

Microsoft and Nortel announced an alliance as part of Microsoft's overall unified communications push. Nortel said it expects more than $1 billion in new revenue over the life of the four-year pact, under which the companies will work together on research and development and partner on sales and marketing. I was on their streaming "virtual press conference", which hosted both Microsoft's and Nortel's CEOs. Both CEOs participating certainly demonstrated their committment to this major partnership.

Their goal is to combine Nortel's network quality and reliability with Microsoft software's ease of use and to accelerate the availability of unified communications. "Nortel and Microsoft have each led fundamental transformations in their own market - Nortel's digital innovation and Microsoft's software on every desktop," said Mike Zafirovski, president and CEO of Nortel. "By combining our unique strengths, Microsoft and Nortel will accelerate the delivery of unified communications - delivering to our customers a higher-quality user experience, with greater reliability and lower total cost of ownership. That's where we can make a real difference."

We are investing together because the communications industry is at an inflection point," said Steve Ballmer, CEO of Microsoft. "We will have deep collaboration in product development with Nortel, allowing us to rapidly deliver high-quality, highly reliable solutions that will support mission-critical communications. The opportunity for our customers is fantastic. We will enable them to realize tremendous economic and business benefits from unified communications."

"This is a gutsy play for Nortel - accelerating the move of our voice technology into software and working with the world's software leader as part of our broader business strategy to transform the company into a software and services leader," Zafirovski said. "From this transaction, we believe we can capture well beyond $1 billion in new revenue, ramping up with increased momentum through 2009 via professional services, voice products and applications, as well as data pull-through in the enterprise."

"Unified communications will drive the next major advance in individual, team and organizational productivity in today's 24x7, always-connected and increasingly mobile work environment," said Jeff Raikes, president of the Business Division at Microsoft. "Our software-based approach puts people at the center of communications through a single identity across e-mail, voice mail, voice over Internet protocol (VoIP) call processing, instant messaging and video, and intuitively embeds communications capabilities into people's everyday work processes, including the Microsoft Office system and third-party software applications."

Microsoft and Nortel said that under the deal, which has an option to be extended, that they will jointly sell the advanced unified communications solution and integration services. The plan is to develop a training and incentive program for the companies' sales teams. Microsoft and Nortel will build a joint channel ecosystem using both companies' systems integrator, reseller, and service provider relationships.

Microsoft has been looking for a hardware partner to go up against Cisco and their unified communications platform. Certainly Microsoft has chosen a strong hardware partner to offer a comprehensive unified communications suite that includes VoIP, presence, mobile, and other functionality. It's worth mentioning however that Nortel has suffered from an accounting scandal in 2004 and the telecom downturn that started in 2001. It remains to be seen if Nortel's Zafirovski can turn around one of the largest telecom equipment manufacturers in the world when cheaper, open-source solutions such as Asterisk are nipping at the heels of all the IP-PBX manufacturers.

Shares of Nortel on the New York Stock Exchange rose >5% to above $2.

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Inc.com Names Mark Spencer to 30 Under 30: America’s Coolest Young Entrepreneurs

Digium, creator of Asterisk and pioneer of open source telephony, today announced that Mark Spencer, president of Digium, has been named to Inc.com's "30 Under 30: America's Coolest Young Entrepreneurs".

Inc. selected the top 30 entrepreneurs based on their proven ability to run a successful company, manage a company with a novel approach, create a successful or innovative product, and/or otherwise demonstrate their innovative idea in the world of entrepreneurship. Inc.'s article can be found at www.inc.com/30under30.

"I am honored to be included in Inc.'s 30 under 30," said Mark Spencer, president of Digium and creator of Asterisk. "Work has become quite a passion for me and it is very rewarding to receive such recognition."

Congratulations, Mark!

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Microsoft Office Communications Server 2007

On May 30th,  I speculated that Windows Live Meeting 2006 was coming - the next version after Windows Live Meeting 2005. I was close -- Microsoft actually "skipped a year" and released Microsoft Office Communicator 2007, a unified communications client that works in tandem with Office Communications Server 2007, which was also announced today. This solution delivers a presence-based, enterprise VoIP “softphone” for secure, enterprise-grade instant messaging that allows for intercompany federation and connectivity to public instant messaging networks such as MSN, AOL and Yahoo! It also enables one-to-one and multiparty videoconferencing, audioconferencing, and webconferencing. Office Communicator 2007 will be available in desktop, browser-based and Windows Mobile-based versions.

Microsoft Office Communications Server 2007 uses Session Initiation Protocol (SIP) standards-based protocol to enable presence-based VoIP call management, as well as VoIP communication. Unfortunately, it appears as though this solution is strictly targetting the enterprise and completely ignoring the consumer market. Although it does support SIP, it will not support all SIP based VoIP networks, but instead only connect to Microsoft's proprietary (and commercial) Microsoft Office Communications Server 2007 platform.

Sure, Microsoft has partnered with public instant messaging networks such as AOL and Yahoo! to offer IM connectivity, but what if I want to have my employees use my own SIP registrar server or SIP-based headsets in combination with just the Microsoft Office Communicator 2007 client? Unfortunately, you can't. Stupid Microsoft does it again… When will they get it that with so many open-source solutions out there you can't get away with this proprietary stuff - proprietary solutions are so 1990s. Perhaps Microsoft should go read my recent perfect unified communications client article?

Although the Microsoft Office Communicator 2007 client CAN connect to an IP-PBX, it has to FIRST go through Microsoft Office Communications Server 2007 as an intermediary. I should point out that Microsoft's smartphones (Windows Mobile 5.0) has still been fairly slow to take off as compared to Treo and other smartphones, which are often used for business applications such as email access. If Microsoft wanted to give its smartphones a shot in the arm, Microsoft should have included support in the Microsoft Office Communicator 2007 client client for ANY SIP-based IP-PBX without the need for the Microsoft Office Communications Server 2007 platform. Sure, it's nice to have all the tight integration and ease of management, but hasn't Microsoft realized that most organizations are not 100% Microsoft shops? What if I want to use Microsoft's client with the popular Asterisk IP-PBX and without the commercial Microsoft Office Communications Server 2007 software? Can't do it.

But if you are a 100% Microsoft shop - or at least your communications servers are - then here's what you get with today's announcement. First, Microsoft Exchange Server 2007 unified messaging will make it possible to view voicemail from traditonal PBXs and IP-PBXs in an Outlook inbox. Microsoft demonstrated an application where a user late for a meeting that is scheduled in an Outlook calendar can phone the Exchange server and tell the system to notify other participants that he or she is running late. The system, using voice recognition to interpret the message generated an e-mail notification. Another application is TTS (text-to-speech), which will enable users to have e-mail read to them by telephone. Of course, this is nothing new to the Asterisk community, which can even have the weather read to you.

Another application is Microsoft Office RoundTable, an audio-video collaboration device with a unique 360-degree camera. When combined with Office Communications Server 2007, according to Microsoft, "RoundTable delivers an immersive conferencing experience that extends the meeting environment across multiple locations. Meeting participants on site and in remote locations gain a panoramic view of everyone in the conference room as well as close-up views of individual participants as they take turns speaking."

Microsoft launched joint ventures with Motorola Inc. and Germany's Siemens AG. Microsoft will supply its Microsoft Office Communications Server 2007 for use in Motorola HC700 series mobile computing devices and the new sexy Motorola Q smart phone. Also, Siemens HiPath 8000 softswitch real-time telephony will be integrated with Microsoft Exchange Server and Microsoft Office Live Communications Server.

Microsoft Exchange Server 2007 is scheduled to be released in late 2006 or early 2007. Microsoft Speech Server 2007 will be available in late 2006. Communications Server 2007, Communicator 2007, Communicator phone experience, Live Meeting, RoundTable and the IP-enabled business desktop phones featuring Communicator phone experience will complete Microsoft’s unified communications solutions and are scheduled to be available in the second quarter of 2007.

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HUDLite for Asterisk Launches

Fonality today announced the immediate availability of HUDlite, a free application for Asterisk that provides businesses with a real-time call control and presence management platform, along with other features such as chat/IM and call recording. Based on HUDPro (paid version), HUDlite is available for download at www.hudlite.org, a web site that also provides a user forum and documentation. Fonality's CEO/founder Chris Lyman told me last week, "We have silently launched HUDlite (as you know) and already have almost 4,000 downloads." There is certainly a lot of "buzz" within the Asterisk community over the HUDLite application, which is probably the best call management, presence application (and other functionality) for the Asterisk platform.

"Free HUDlite helps bring Asterisk to the masses, making it much easier to use and even more powerful. Fonality has benefited so much from the Asterisk movement. We are excited to finally give back."

Through a user-friendly interface that runs on Windows XP, Mac or Linux desktops, users of HUDlite can see when other employees are on calls, to whom they are talking and whether calls are internal, external or in a queue-even if the employees are not in the office.

Mark Spencer, president of Digium and creator of Asterisk, made the following comment about HUD: "Fonality's new HUD application provides Asterisk users with an innovative and extremely productive way to improve their operations with call presence awareness and call management."

Features of HUDlite include:

Drag-and-Drop Calling - call external numbers, internal extensions and numbers listed on websites and in documents by dragging and dropping numbers on HUDlite

Real-Time Call Controls - use a mouse to quickly transfer calls to employees and to voicemail, place calls in a general parking area, even put calls on hold and tag them with notes

Call Monitor/Barge - barge or passively monitor inbound and outbound calls and better manage high call volumes, such as those in call centers

On-the-Fly Recording - allow employees and their managers to record calls with the press of a button - another required feature for call centers and for specific markets including legal and medical

HUDpro, the commercial version of HUD, is also available from Fonality ($995 unlimited seat license) and provides additional features, including advanced multi-hierarchical permission systems, enterprise-class secure instant messaging, mobile phone contact, Outlook integration, CRM integration with screen pops, and comes pre-installed and configured by Fonality's support team.

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Security Flaws In Asterix PBX, IAX VoIP Client

From Yahoo News:

Open source IP PBX application Asterisk PBX and the open source IAX VoIP client contain serious security vulnerabilities that could allow hackers to assault VoIP networks with denial-of-service (DoS) attacks, says Core Security Technologies, a security company that discovered the threat.

The good news is the open source Asterisk community has already released patches.

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Trixbox founder responds

Both Chris Lyman, CEO of Fonality, and Andrew Gillis, the founder of Asterisk@Home - now Trixbox, responded to my previous post pondering the relationship between Trixbox and Fonality. Long story short, it looks like the open-source Trixbox project is safe from any commercial intentions. Just a case of one Linux-based commercial company helping out a Linux hobbyist. You just gotta love the Linux community!

Also, interesting to read how the name "trixbox" came about:

Tom: So what’s the deal with Trixbox. I thought I read you were the registrar of this domain a few days ago, but now it’s registered to a registrar “anonymous” proxy service.

Chris: We helped him with the domain registration (we are old hosting guys, my CTO and I ran one of the largest hosting companies in the world before selling it to a telephone company in 2000.) Andrew may have changed the domain service when he got worried that it might appear that Fonality was trixbox.  (Actually I) checked about the domain. To be honest, I now do *not* think that my CTO helped him register the domain, but I could be wrong. But I do know that we are paying for the hosting of the site, and helped Andrew with a bunch of things including legal advice, etc.

Tom: Are you running the show with Trixbox – the replacement for Asterisk@Home?

Chris: No it is Andrew’s show. We provide him free hosting and bandwidth. He is very much the owner and leader of that community. Andrew and Fonality have begun to closely work together as of late because a) we wanted them to bake HUDlite into trixbox b) we wanted to get to know the Asterisk open source community a bit better. So far the relationship has been good. Andrew agreed to bake HUDlite into trixbox, which gives us access to all his users. And, he has taught us a lot about the mindset of the Asterisk hobbyist. In exchange we have done him some favors. We help sponsor him wherever we can, including: web-hosting, bandwidth, we sent him a PBXtra to play with, etc.

Tom: Anything behind the name change? Did Mark Spencer send a cease and desist to protect a trademark on the name “Asterisk”? (assuming he trademarked Asterisk)

Chris: Andrew told me that he got an email from Digium a long time ago, stating that Asterisk was their trademark. I didn’t know of any legal letter…so you may know more than I know.  I thought it was all pretty casual.

Tom: How much control (if any) do you have over the Trixbox development?

Chris: Very little. Unless Andrew really likes something I say to him I guess.

Tom: Is the plan to simply convert over Trixbox users to Fonality?

Chris: Trixbox is a free open source community – largely international. Fonality is a commercial paid product, largely domestic. We couldn’t be farther apart in communities, interest, or financial objectives. I guess our only real common ground is a usage and love of Asterisk.

And here's a response from Andrew from Trixbox regarding my blog post

Andrew: I created Asterisk@Home a year ago on a whim. I thought what a great idea to make Asterisk easy to install. It very quickly grew much bigger than I thought it would. One day Digium contacted me and told me that Asterisk is their trademark but I could use it as long as my project remained totally open source.

This sounded good from the start but it put restrictions of the product such as loosing the ability to use free (not open source) software. Then theres my user base. As it turns out most of my users were running Asterisk@Home for business and they didn’t like the name.

The Asterisk@Home name also pigeon holes the product into being an Asterisk distribution. I want to make it more than that and include other type of software. I want it to do more tricks. So the new name trixbox.

I do own the trixbox.org name. I register all my domains using anonymous servers. Hope this helps out. If you are interested in doing a review of trixbox or an interview with me. Just let me know.

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Trixbox 1.0 replaces Asterisk@Home

Apparently, development on Asterisk@Home, the "turnkey" Asterisk solution with many third-party add-ons pre-installed has ended. However, the developers have created a new product called Trixbox. Just like Asterisk@Home, Trixbox is a complete Asterisk PBX including, a Linux OS, Asterisk PBX software, a Web GUI, and many other useful add-ons. As with Asterisk@Home, Trixbox can be quickly installed in less than an hour.

Besides being reminded of that wacky Trix rabbit character, I'd be curious why the need for the name change, especially since Trixbox is based on the same exact source code as Asterisk@Home. I liked the Asterisk@Home name - it gave the product a connotation of being "easy" to use and install. (e.g. even a "home" user could install it, or it was designed for a "home installation" by a hobbyist or entrepreneur.)

I can speculate that my pal Mark Spencer over at Digium sent Asterisk@Home a "cease & desist letter" to protect his trademark. Although, I'm not sure if Mark even has a trademark on the brand "Asterisk". I'll have to shoot him off an email and ask. But if he does, then you can't blame him if he is indeed trying to protect his brand. If you don't vigorously protect your brand and it becomes watered down and used by everyone, then you can lose your rights to the trademark. This whole point may be moot since I believe you can't trademark GPL open-source code projects. Where's a trademark lawyer when you need one?

[Update: I received an email from Ward Mundy stating that Digium did give permission to use the Asterisk@Home name. He stated "Asterisk@Home was suitable for home and business use. A lot of customers were apparently put off by the @Home moniker." I suppose I can see the @Home moniker hurting its "business-class" image.]

In any event, the latest release of Trixbox 1.0 (technically Asterisk@Home v3.0) was released on May 31st (less than a week ago). Interestingly enough, I noticed that Fonality is offering their HUDLite softphone client for Trixbox. I just shot off an email to Chris Lyman over at Fonality to see what exactly the relationship is between Fonality and Trixbox, especially since I could have sworn that I saw Fonality as the one that registered the trixbox.org domain.

However, when I checked the whois database today, it's registered to one of those "anonymizing" proxy domain registrars with a date stating it just changed on 6/5/06 (just 2 days ago). I then did some more investigating to see if I could figure out who owned trixbox.org. I won't get into too many details on how I figured it out, but suffice to say I did a whois on www.hudlite.org, which is registered to Fonality, Inc. I then looked at the IP address for this whois record and it displayed "66.234.135.90". I then do a whois on trixbox.org and lo' and behold the IP addresses were the same! Apparently trixbox.org and www.hudlite.org are running on the same web server and using the host header to figure out which web page to display.

Did Fonality by out the rights to Asterisk@Home and then change the name to Trixbox.org? What does this mean for the popular open-source Asterisk@Home distro (now Trixbox) considering Fonality is a for-profit Asterisk solutions provider? Is the plan to try and convert Trixbox users (generally novice Linux users) into paying Fonality users? Fonality certainly focuses on businesses that have little or no Linux experts, so there is certainly a potential synergy there. Well, the mystery continues... I'll post more when I hear back from Chris @Fonality.

Finally, Nerd Vittles has an excellent three-part series on how to install Trixbox along with FreePBX, a web-based GUI add-on for making Asterisk easy to manage. If you ever wanted to try Asterisk, but find Linux a bit daunting, then head on over to Nerd Vittles and try the tutorial.

p.s. Silly Windows users, Trixbox is for Linux!

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Asterisk Business Edition B.1 Released

Digium today announced a major upgrade of its Asterisk Business Edition, the professional-grade version of Asterisk. Called Asterisk Business Edition B.1, the upgraded release includes enhanced security and scalability provided by Ranch Network's Asterisk security code, speech recognition capabilities through the LumenVox Speech Engine, text-to-speech applications through the Cepstral Text-to-Speech System and a customized Linux distribution to simplify installation. Asterisk Business Edition B.1 will also feature built-in support for Intel Dialogic Products and Aculab Prosody X cards.  

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Gizmo supports Asterisk

Some great news from SIPPhone, developers of the Gizmo Project announcing support for the open-source Asterisk PBX... The "Asterisk economy" continues to grow by leaps and bounds.

check it out:

SIPPhone today announced the delivery of Gizmo Project 2.0, highlighted by its support for Asterisk PBX software. Giving Gizmo Project users the ability to log into Asterisk means they can now be universally reachable via their Asterisk PBX or directly through the Gizmo Project network. The option to seamlessly receive calls from their office PBX while anywhere in the world leverages Gizmo Project's advanced NAT (“network address translation”), firewall and router traversal features and server infrastructure. More information about Gizmo Project 2.0 with Asterisk support may be found at www.gizmoproject.com/asterisk.

The benefits of using Asterisk to power an office PBX can be significant in terms of cost savings, efficiency and access to features previously only available to large businesses. Using Gizmo Project as an office softphone lets users easily place calls without the burden of special VoIP phones or the expense of traditional phone call charges. Incoming calls to an office PBX will reach mobile workers anywhere in the world. Gizmo Project also provides such high-end features as voicemail-to-email, free conference calling, call history, free access to millions of people on SIP-based networks, and built-in instant messaging (IM) capabilities.

“Whether a company is focused on a completely free Asterisk PBX installation or is running a premium version, Gizmo Project is now the ideal softphone for use with any size deployment,” said Michael Robertson, Chairman and CEO of SIPphone. “Our experience at routing millions of calls through almost any network setup means that mobile computer users can be reached anywhere as if they were physically in their office. Plus, companies save money with low domestic and International calling rates using Gizmo Project,” Robertson concluded.

Companies around the world deploying the Asterisk PBX software, premium PBXs developed by such companies as SwitchVox, Epigy, webFones, or any other SIP-based PBXs can now easily use Gizmo Project for making and receiving calls. Specific information about setting up Asterisk for use with Gizmo Project 2.0 may be found at www.gizmoproject.com/setupasterisk.

The free Gizmo Project software for Apple Macintosh, Microsoft Windows and Linux computers delivers crystal clear VoIP calls. Gizmo Project uses licensed, best-of-breed audio codecs such as GIPS and employs media relays around the world to route calls through the most efficient path. Gizmo Project also routes calls through one of several phone partners which provides for the lowest possible per minute International calling rates. PC-to-PC calls are always free.

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Comments on this Entry:

(michael on May 23, 2006 1:00 PM) I found it interesting that even though it is a generic SIP client implementation it is being marketed with Asterisk. Anyway, I am grateful for this, as now I can stick with Gizmo Project as my softphone client to use with my Asterisk server instead of using X-Lite. One thing I noticed is that it won't register with the server using a private IP address, it must be the public address of the server. Maybe not very many people have Asterisk behind a NAT, but I do so I noticed it.

(Tom Keating on May 23, 2006 1:59 PM) You can't use a private NAT'ed address? Really? Have you tried port forwarding? i.e. port 5060 although, you shouldn't have to since Gizmo claims to be able to traverse NAT firewalls. Haven't tested it with Asterisk, but last time I used Gizmo with SIPPhone service, it traversed the firewall just fine.

(michael on May 23, 2006 2:51 PM) I guess I wasn't clear in my statement. My Asterisk server is behind NAT, on the same local subnet as my PC. In the Gizmo Project Secondary Account settings, I must put the public IP address of the Asterisk server, as it doesn't work with the private IP address of the Asterisk server.

(michael on May 23, 2006 3:09 PM) As a follow up, Gizmo Project appears to my Asterisk server as being at IP address 198.65.166.XXX, so the Secondary Account is being proxied by sipphone as well. That explains why the Secondary Account server must be a public IP.

Written by VoIP & Gadgets Blog on May 23rd, 2006 with no comments.
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Fonality, an Asterisk solution passes 20 million calls

Some great news from Fonality, an Asterisk-based IP-PBX that earned a very positive review by TMC Labs in the April issue of Internet Telephony Magazine. Last time I checked, they were at 4 million calls last July. Some nice growth there Fonality! Congratulations!

Fonality, the leader in affordable, enterprise-class IP-PBX phone systems for small and medium businesses and the world’s largest distributed deployment of Asterisk™, today announced that users of PBXtra have placed 20 million calls since the product first became available in October 2004. Fonality has deployed PBXtra to more than 10,000 business users domestically and internationally and has a network of 1,200 resellers in 14 countries. Earlier this year, Business 2.0 magazine named Fonality one of “The Next Net 25” – 25 companies helping to reinvent the Web.

“Open Source is really helping us attack an overpriced telephony market with innovative products at the right price,” said Chris Lyman, Fonality’s founder and CEO. “Our growth over the past 18 months demonstrates how hungry SMB owners are for powerful, yet affordable, business phone solutions.”

Unlike most alternative Asterisk-based PBX offerings, PBXtra is more than a simple front-end designed to make it easier for Linux and Asterisk experts to install and manage Asterisk. Fonality has written more than 250,000 lines of code that dramatically streamline the complex tasks of PBX deployment and management. As a result, PBXtra is the world's first true PBX that can be installed and administered remotely using a Web browser, without any specialized training.

PBXtra's enterprise-class features include telecommuting, branch office support, voicemail-to-email, click-to-call, VoIP, softphones, support for IP and analog phones, call recording, distributed call center queuing and more. Businesses deploying PBXtra pay 40 to 80 percent less than they would for a traditional PBX purchase because the software runs on standard PC hardware and uses layers of Open Source Linux and Asterisk software.

PBXtra is available from Fonality or through a Fonality reseller. Pricing starts at $995 for the PBXtra Standard Edition server with unlimited licenses, or $2,935 with ten phones. PBXtra's Call Center Edition, which allows for distributed call centers, is $1,995.

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Written by VoIP & Gadgets Blog on May 14th, 2006 with no comments.
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Sangoma announces high-density 8 T1/E1 port card

Sangoma announced their A108 card which they claim provides the industry's highest density T1/E1 card for voice and data. Sangoma's cards work well on the Asterisk platform, as seen by my test drive of their hardware.

Here's the news they are announcing:

Sangoma's engineering uses a common printed circuit board to host hardware-based echo cancellation and voice enhancement

TORONTO – May 10, 2006 – Sangoma Technologies Corporation (TSXV: STC) (www.sangoma.com), a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure, has introduced its new A108 Series PCI cards to support high density voice and data applications. At 8 T1/E1 ports, the A108 has the highest T1/E1 density available on a PCI card.

As with other Sangoma PCI cards, the A108 is available in an A108d version especially built for the soft telephony industry. The A108d supports voice enhancement capabilities including G.168-2002 echo cancellation with 1024 tap/128ms tail per channel on all channel densities, DMF encoding/decoding and tone recognition, voice quality enhancement and adaptive noise reduction.

“We brought this high density product to market at the request of our enterprise and telco customers who require scalability and higher density solutions,” says Sangoma Technologies President and CEO David Mandelstam. “We continue to meet the flexibility and system needs of the industry with the launch of our new 8-port card.”

Distinctive A108d features include:

* Support for 1024 taps (128ms) of echo tail handling on every channel (DS0) meaning that troublesome delayed echoes are properly handled.
* Dynamic and selective activation of echo cancellation, making the system ideal for mixed voice/data applications.
* The same PCI interface, architecture and digital path as all of Sangoma's other AFT based analog and T1/E1 cards, meaning that the A108 has guaranteed stability, motherboard compatibility issues and proper interrupt handling.

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Written by VoIP & Gadgets Blog on May 10th, 2006 with no comments.
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Asterisk Flite text-to-speech program

Nerd Vittles has an excellent post on a new text-to-speech add-on for Asterisk called Flite that can read text in a MySQL database to a caller. While there already existed the Festival TTS, there were reports of performance problems when Asterisk 1.2 and Asterisk@Home 2.x were released often resulting in 5-7 second delays between when you hear the TTS engine say something. Not good if you don't want your callers hanging up.

The solution comes from Nerd Vittles:

And today our tip of the hat goes to Francois Aucamp from South Africa who has single-handedly reengineered Carnegie Mellon University's open source speech synthesis engine (Flite) to work with Asterisk@Home and restored voice synthesis to its rightful place as an indispensable component in the Asterisk@Home bundle. We'll show you how to install the necessary components in less than 15 minutes, and your Asterisk system will once again be speaking to callers whenever you need it to.

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Written by VoIP & Gadgets Blog on May 10th, 2006 with no comments.
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Asterisk forms Advisory Council

Looks like Asterisk's growth has resulted in the formation of the Asterisk Advisory Council. Somehow when I think of open-source I don't think of a "centralized" council. Isn't the whole point of the open-source model to allow for distributed input by your programming peers? Also, when I think of "Council", I think of the United Nations Security Council or a cabal of men and women plotting for worldwide domination. Didn't Dr. Evil also have a council of advisers - though he killed off a few.

Let's just hope Asterisk and the Asterisk Advisory Council doesn't have any plans on world domination.

Anyway, here's their news...

Digium, the creator of Asterisk and pioneer of open source telephony, today announced the formation of the Asterisk Advisory Council. The Council was developed to respond to the increased interest and participation in the Asterisk open source telephony project.

Composed of five experienced Asterisk community contributors, the Council will assist in the management of the Asterisk open source telephony project. Responsibilities of the council include the selection and supervision of community developers, management of release cycles, and maintenance of Asterisk contributions, among other duties.

"As the Asterisk market continues to grow rapidly on a daily basis, we saw the need to expand the team managing the open source project," said Kevin Fleming, co-maintainer of Asterisk and senior software engineer at Digium. "By identifying these key community members to participate in our council, we can ensure that the project continues to add innovations and improve without any delays."

The following members have been appointed to the council:

     Brian Capouch, Assistant Professor and Chair of the Computer Science Department at Saint Joseph's College: Capouch has integrated Asterisk with a number of other processors including home automation, network monitoring, camera-based security, and the openWRT distribution of Linux. He teaches a college course on VoIP, has presented at a number of industry conferences, and is working on a forthcoming book on Asterisk to be published by Addison-Wesley.

     Olle E. Johansson, Asterisk Developer, consultant and Evangelist, founder of Edvina AB, Sweden: Johansson has contributed to the SIP channel among other parts of Asterisk, worked as a bug marshal and has written documentation on the software and the Asterisk wiki. He is also one of the founders of Astricon - the Asterisk conference, and regularly performs Asterisk training sessions.

     Tilghman Lesher, Developer for VCCH, Inc., a leading provider of innovative solutions based on open source software: Lesher has contributed a large amount of code to the core of Asterisk and is the author of a number of applications and dialplan functions.  He has been programming for over twenty years, with eight years of professional experience.

     Jeremy McNamara, Founder of The NuFone Network, the first Asterisk-based Inter-Asterisk eXchange (IAX) provider: McNamara has been working in all aspects of the telecommunications industry for more than nine years and has extensive experience with the development, testing and deployment of Asterisk-based solutions.

     John Todd, Tello Corporation: Todd comes from an IP networking background, having worked in several large ISPs, ITSPs, and application service providers. He is currently developing next-generation network elements and systems, some of which involve integrating Asterisk with proprietary systems for customers and providers. Todd is also an active participant and speaker at various VoIP forums and conferences.

Details of the Council's organization, membership, management policies, decisions and current projects will be available on www.asterisk.org.

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Written by VoIP & Gadgets Blog on April 19th, 2006 with no comments.
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Iotum integrates with Asterisk

Iotum, winner of Internet TelephonyÂ’s Product of the Year Award and DEMOgod at DEMO 2006 has announced today the beta availability of the iotum relevance engine integration with the popular open-source Asterisk IP-PBX.

iotum's Asterisk integration module connects its call management applications to Asterisk IP-PBX's to assist users in prioritizing which calls are more important, and which can wait, based on whoÂ’s calling, and what the user is currently doing. This non-commercial beta will allow Asterisk users to filter, rank, and prioritize incoming calls using iotum as well as offer users the ability to easily schedule conference calls from within Microsoft Outlook using iotum's Pronto Conference Calling feature.

iotum's unique "Web 2.0" relevance engine which helps to prioritize calls has certainly garnered attention and awards from several media outlets. Certainly by integrating with Asterisk, iotum will garner plenty more attention, especially from the Asterisk community.

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Written by tkeating@tmcnet.com on April 3rd, 2006 with no comments.
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Digium Wildcard TDM2400 24 port analog card launches

Digium announced the Digium Wildcard TDM2400 which they claim provides the highest analog density available (24 ports) in a PCI card. Well, are they truly the highest analog density? Sangoma, their main competitor also offers a 24-port analog PCI card. So technically the Wildcard TDM2400P isn't the "highest" - it's equally "high" with the Sangoma card. I guess you could consider them both the "highest" analog density, but if both are the "highest" than who is lower? Anyway, certainly worth sharing the news...
 
Wildcard TDM2400P provides highest analog density available in a PCI card.

Digium Inc., the creator of Asterisk and pioneer of open source telephony, today launched the Digium Wildcard TDM2400, the most dense and scalable card available for building an Asterisk-based telephony system for SME and SOHO environments.

The 32-bit 33MHz PCI 2.2-compliant card combined with Digium’s patent pending VoiceBus™ technology supports quad-FXS and quad-FXO interfaces for connecting analog telephones and lines through a PC, without taking up numerous PCI slots.

With its flexible scalability features, our 24-port card is the best hardware card available for small and medium businesses looking to build an inexpensive, sophisticated VoIP telephony solution without compromising the use of multiple PCs.

The Wildcard TDM2400P replaces the requirement for a separate channel bank and T1 interface cards while offering superior echo cancellation on both FXO and FXS interfaces. The quad-FXO and quad-FXS modules are interchangeable allowing the combination of interfaces up to six slots for 4-port FXS or FXO modules. With this new card, small and medium businesses can benefit from features such as high density in fewer PCI slots, and an industry standard 50-pin Amphenol connector for easy installation.

Support and Availability
The Digium Wildcard TDM2400P is available from Digium and Asterisk resellers and distributors worldwide. For more information, please contact sales@digium.com or call +1-256-428-6262. All Digium products are backed with a two-year limited warranty, including installation and troubleshooting support. Users can also purchase an optional one-year extended warranty.

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Tags: Digium, TDM2400, Sangoma, Asterisk, VoIP
Related Tags: analog density, digium wildcard, highest analog, digium, Digium, wildcard

Written by tkeating@tmcnet.com on March 29th, 2006 with no comments.
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Sprint cell phone + Asterisk@Home = unlimited cell phone minutes

Nerd Vittles has a nice "dummy proof" tutorial explaining how to use Asterisk@Home in along with Sprint's plan to get unlimited wireless and unlimited wireline (using broadband VoIP) all for around $68/month. His recipe utilizes Asterisk@Home in combination with Sprints PCS unlimited FREE calls between your Sprint cellphone (or multiple PCS phones if youÂ’re on a shared plan) and your residential phone number regardless of the wireline carrier for an additional $5/month. According to Nerd Vittles, "HereÂ’s the math. A basic Sprint cellphone plan: $35. Sprint to Home service: $5. TelaSIPÂ’s unlimited US48 VoIP calls: $15 a month. Home phone number transferred to BYOD plan at BroadVoice (hereÂ’s how) or AxVoice (hereÂ’s how) for unlimited incoming calls: $9. Unlimited nationwide Sprint cellphone calls: Priceless ... and FREE. Total cost for residential home and cellphone service with unlimited nationwide calling: $64 a month plus about $4 in Sprint add-on fees. Remember this gets you unlimited nationwide calling BOTH from your cellphone and your home phones."

$68/month for unlimited landline and wireless services? WOW! I always knew it was possible to do with an Asterisk system and thought about setting up a home Asterisk box to do this exact thing, but Mr. Vittles beat me to it. Nicely done Mr. Vittles, nicely done!

Nerd Vittles sent me a nice summary of his recipe via email:

The Nerd Vittles Recipe:

• Cheapest Sprint Cell Phone Plan - $35
• PCS to Home Add-On Service - $5
• Asterisk@Home Server for Linux or Windows – FREE!
• Home Phone Service Switched to BroadVoice or AxVoice BYOD Plan - $9 TelaSIP VoIP Unlimited Residential U.S. Calling Plan - $15 Nerd Vittles DISA Script – FREE!
• Unlimited Monthly Calls from Home OR your Sprint Cellphone – PRICELESS!

He even explains how to edit the extensions_custom in Asterisk to detect the CallerID of an incoming call to automatically detect your cell phone. When you dial from your cellphone to the Asterisk@Home number you immediately get 2nd dialtone (DISA) so you can then dial out using an unlimited VoIP plan. Of course with CallerID spoofing, you may want to add a password authentication so people can't call your Asterisk box and start making outbound calls. Vittles explains how to easily add the authentication method to the extensions_custom file. But really, the odds of someone discovering your cellphone number and your Asterisk@Home's phone number are astronomical. As Vittles said, just don't advertise the fact.

Of course, now that Nerd Vittles has made these easy tut that anybody can follow, I wonder how long it will be before the carriers add some disclaimer to their Terms of Service (TOS) saying they'll cancel your service for using such a workaround. Then again, I doubt the average person such as my dad or even my grandma is going to install Asterisk@Home even if it is easy to install. So this "unlimited" wireless minutes recipe is more for geeks and hackers or those people just technical enough to be dangerous.

Check out the recipe.

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Tags: Asterisk@Home, unlimited minutes, cell phone, VoIP
Related Tags: sprint cellphone, phone number, unlimited nationwide, unlimited, asterisk, Asterisk

Written by randy@strategypluscommunications.com on March 22nd, 2006 with no comments.
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AsteriskBlog offers Asterisk classes

Chris over at AsteriskBlog is working on providing some free beginner's Asterisk classes in South Carolina at a local library. He told me, "I think it could be a great advent to the community here, at least for those interested in telecommunications."

Hey, even if you think you know your Asterisk and Linux, you might still want to swing by his free classes. After all, there's nothing like a bunch of Linux/Asterisk geeks getting together in a library to share their knowledge! Maybe afterwards he can throw a really cool party. You know, like a Linux Quake Party.

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Written by tkeating@tmcnet.com on March 21st, 2006 with no comments.
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GDS Voice Conferencing Solution for Asterisk

Today GDS released GDS Voice Conferencing Solution for the Asterisk platform. GDS Voice Conferencing is a feature-rich enterprise voice conferencing solution built on top of native Asterisk MeetMe application. Amazing how many third-party companies are now developing applications for Asterisk. Indeed, within the VoIP industry we have the Skype Economy, and now we can add the "Asterisk Economy".

Here is a short overview:

• Multiple conference types (scheduled, recurrence, reservation-less)
• Intuitive web interface for conference management, personal contact management, user management and system administration
• Manage conference attributes like announce user leave/join, wait for marked user and associate contacts and its roles within the conference (listen only, admin mode etc.)
• Monitor live conferences (mute/un-mute participant, kick out participant, lock conference, view on line participants and its attributes etc.)
• Integrated personal contact management for simple invitation and notification
• Import existing contacts
• User roles based privileges
• Port resources management (TDM and VoIP)
• Recurrence and conflict conferences management
• Automatic email notifications and reminders
• API for integration with third party applications and more

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Written by tkeating@tmcnet.com on March 21st, 2006 with no comments.
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Fonality HUD call management application launches

Fonality HUD softphone call managment application launched today, a userfriendly softphone call management application with presence information for specified extensions giving every user attendant console-type capabilities. Chris Lyman, Fonality CEO actually gave me a heads-up on HUD (Heads Up Display) - pun-intended several weeks ago. I played around with the beta version but wasn't able to get it entirely to work due to some firewall and configuration issues and I didn't have time to troubleshoot. In any event, here's the news in all its glory and a pretty screenshot to boot...



Fonality, the leader in affordable IP-PBX phone systems for small businesses and the world’s largest distributed deployment of Asterisk, today announced HUD, a new call management application that provides businesses with real-time, easy-to-use call control and management features. HUD comes in two versions – HUDlite, a free call management application for the Asterisk Open Source PBX, and HUDpro, an advanced call management application that enhances PBXtra, Fonality’s award-winning IP-PBX platform.



“The Digium-Fonality relationship is an important one to us,” said Mark Spencer, president of Digium and creator of Asterisk. “Fonality’s new HUD application provides Asterisk users with an innovative and extremely productive way to improve their operations with call presence awareness and call management.”



Through an intuitive, color-coded desktop interface, HUD, an acronym for Heads-Up Display, lets employees see when others in the office are on a call, to whom they are talking and whether calls are internal, external or in a queue. HUDlite, available next month, provides drag-and-drop calling and call controls, call monitoring and barging and on-the-fly recording. HUDpro, available immediately, provides additional features, including advanced multihierarchical permission systems, enterprise-class secure instant messaging, complete integration with PBXtra and configuration and support from Fonality.



Features in both HUDlite and HUDpro include:



Drag-and-Drop Calling – call external numbers, internal extensions or numbers listed on websites or in documents, by dragging and dropping numbers on HUD



Real-Time Call Controls – transfer calls to employees and voicemail, place calls in a general parking area, even put calls on hold and tag them with reminder notes



Call Monitor/Barge – barge or passively monitor inbound and outbound calls, which helps to more effectively manage call centers



On-the-Fly Recording – allow employees, or their managers, to record calls with the press of a button – a required feature for call centers and certain markets such as legal and medical



“We are excited to offer HUDlite to the Asterisk ecosystem in an effort to make their deployments more powerful and user friendly,” said Chris Lyman, Fonality’s founder and CEO. “HUDpro provides PBXtra customers with a fully supported version that gives them even greater call management and communications capabilities.”



HUDpro Delivers Advanced Permissions and Professional Instant Messaging

HUDpro, which integrates with the advanced features of PBXtra, provides businesses and call centers with all the features of HUDlite and the following advanced capabilities:



Sophisticated and Easy-to-Use Permissions System – grant rights to specific extensions and not to others, such as the ability to record their own or others’ calls, see who is on a call, transfer calls to and from extensions, monitor or barge calls and many more. PBXtra Call Center Edition customers can also easily build groups of extensions and grant specific rights to groups.



Secure Company-Wide Instant Messaging – automatic access to enterprise-wide instant messaging (IM) and the ability for all employees to chat. For call centers, supervisors can monitor an agent handling a customer dispute and seamlessly send instructions to the agent via IM without the customer hearing any chatter, being transferred or put on hold.



Pricing and Availability

HUDlite for the Asterisk Open Source PBX phone system will be available for free in April. It will support Windows XP, Macintosh and Linux desktop operating systems.



HUDpro for PBXtra is available immediately for $995 for an unlimited seat license that includes full support and configuration. It will be available for Windows XP desktops.

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Written by tkeating@tmcnet.com on March 14th, 2006 with no comments.
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Sangoma Asterisk cards test drive

Sangoma Technologies, a publicly traded on the TSX Venture Exchange (TSXV:STC ), has been designing WAN and telecom/telephony hardware for over 20 years. While Sangoma is well-known as a world leader in support of ATM, Frame Relay, SS7, X.25, PPP, BiSync, HDLC, and SDLC, all popular WAN-related protocols, Sangoma has been quietly making some inroads in providing analog and T1/E1 hardware to open-source PBX solutions such as Asterisk, Yate, and FreeSwitch. They manufacture a range of PCI based cards with T3/E3, T1/E1 TDM, Analog voice and data, ADSL and serial interfaces. Used as a TDM voice gateway, wanpipe router or with their APIs, their cards can turn a server into voice or data gateway. Their voice and data solutions and communications toolkits are available for all popular operating systems including Linux, Windows, Netware, FreeBSD, OpenBSD , NetBSD, SCO Unix and Sun Solaris.

Today, Asterisk is one of the most well-known open-source telephony solutions, which has a strong any loyal community following and is supported by the for-profit organization Digium, which sells telephony hardware to run on the Asterisk platform. Since Digium provides telephony hardware for the open-source Asterisk platform, the obvious next question is "How does the Digium hardware and Sangoma hardware stack up?"

Well, to answer that question, I needed to meet and visit with both Digium and Sangoma. I visited Digium in Huntsville, Alabama back in August 2005 and met with Asterisk guru Mark Spencer, so it was only fair that I also visit with Sangoma Technologies, which is DigiumÂ’s primary competitor. So I headed up to Toronto, Canada to meet with SangomaÂ’s CEO, David Mandelstam, as well as Doug Vilim, Vice President of Sales and Marketing, and Nenad Corbic, Software Manager, Senior Linux Engineer to find out just how their hardware stacked up against DigiumÂ’s product line.



Sangoma's offices

I should actually amend that statement that my sole purpose of visiting Sangoma was to compare/contrast them with DigiumÂ’s hardware since SangomaÂ’s hardware is also designed to work on other open-source telephony platforms, including YATE, and FreeSwitch. So this article isnÂ’t simply designed to compare Digium vs. Sangoma running on the Asterisk platform. The Sangoma hardware has some interesting applications on other open-source platforms that merit analysis within this article.

YATE is an interesting open-source solution (coded in C++) that for quite some time was solely developed by Diana Cionoiu, a Romanian programmer who I heard is a "perfectionist" when it comes to coding. For instance, IÂ’ve heard stories about people in the YATE community suggest she simply use OpenH323 or other open-source telephony components and sheÂ’d often respond with something to the effect that "theyÂ’re crap" and that sheÂ’d rather code it herself. Wow, now thatÂ’s one confident programmer! Also, since YATE is written in C++ you can compile it in both Linux and Windows. It has some distinct advantages over Asterisk including a built-in prepaid calling card platform, billing (including RADIUS) with full CDR.

Sangoma got the ball rolling by first "fueling me up" with some caffeine from their renowned espresso machine.

Sangoma's Espresso Machine

I say renowned, because the night before Sangoma and I went out for dinner and for 5 minutes all they could argue about was who made the best espresso.

Anyway, with some Sangoma engineers and my cup of espresso in tow, we headed to their lab area. With the Sangoma engineers watching, I installed YATE on a Windows XP PC to test SangomaÂ’s echo cancellation capabilities and to test YATEÂ’s ability to detect and use the Sangoma hardware. After installing YATE (a simple Windows installer), it asked us what kind of hardware we were using and we simply checked the box for Sangoma. Similar to Asterisk, there is text-based config file that had to be edited to map the extensions to the Sangoma hardware. I launched YATE and was immediately able to start making calls. I was pretty impressed with the hardware compression built into the Sangoma cards, which feature a 128ms tail for echo cancellation using algorithms from Octasic. I first disabled echo cancellation on the cards and made a test call from a YATE server to an Asterisk box over a PRI connection and there was noticeable echo on the call. I repeated the call after enabling the echo cancellation and the echo was gone.


Sangoma's A104d Qual T1/E1

For my next test, I wanted to see how easy it was to install Sangoma hardware on an Asterisk server. Sangoma provides an install Wiki that takes you step-by-step through the installation process, which I used as my install guide. The Wiki explains that I can do an automatic or manual.configuration of the Sangoma wanpipe drivers which "tie into" the Zaptel driver.

For automatic, it was simply doing these steps:

The wancfg utility is able to create wanpipe config files based on /etc/zaptel.conf. Once zaptel.conf is created wanpipe config files will automatically be generated.

1. Create your zaptel.conf file based on the hardware in your machine.
2. Create your wanpipe configuraton files based on zaptel.conf
/usr/sbin/wancfg zaptel

Wancfg will create wanpipe config files based on spans chans defined in zaptel and update your /etc/wanpipe/wanrouter.rc startup file.

3. To start wanpipe:
wanrouter start
4. Start zaptel:
ztcfg -vvv
5. Start asterisk run:
asterisk -p

ThatÂ’s it. You now have an Asterisk server with Sangoma hardware!

The manual configuration can also be done. YouÂ’ll need to do a manual configuration if for instance you define the Sangoma analog trunks before the Sangoma PRI or T1/E1 trunk lines in the zaptel.conf file. The automatic configuration utility is expecting a certain order of hardware that messes it up. I believe the PRI or T1/E1 trunks have to be defined before the analog trunks or the automatic configuration utility canÂ’t figure out the settings. You can either edit the zaptel.conf file and correct the order or you can do a manual installation to manually configure the settings for the installed Sangoma hardware. Actually, I did do a manual installation and selected each board just to see how hard it would be. It was surprisingly easy to do.

Hardware Overview
Sangoma's Voice drivers take advantage of their AFT (Advanced Flexible Telecommunication) technology to substantially reduce the processing required to handle voice calls by the host CPU. This reduces the CPU's workload; resulting in fewer dropped calls, less jitter, and better voice quality.

Sangoma also stressed a key competitive advantage in that their AFT technology enables them to not only field upgrade the device driver code, but also the card's FPGA firmware. Digium hardware is not firmware upgradeable. In addition, Sangoma's cards are self sensing for 3.3v and 5v PCI slots and software configurable for T1/E1 or J1. According to Sangoma, they share interrupts properly between themselves and other PCI compatible devices, supporting unlimited numbers of cards per PC chassis.

I know I have heard about some horror stories getting Digium boards to work on certain PC hardware due to hardware interrupt issues. Digium is very picky what kind of PC hardware you use. A lot of issues are reported with motherboards that share interrupts on a single PCI slot, with the Digium hardware. Better quality motherboards allow BIOS specification of the IRQ to a PCI slot to avoid interrupt issues but it certainly requires you do your "homework" before going out to buy Digium boards. Sangoma on the other hand told me they do not have any interrupt issues, their boards will work on any PC hardware and even offer a guarantee that they can get their board to work or they will refund your money.

The AFT series cards all conform to the 2U form factor specification, both in height and length, allowing you to install many cards in a 2U chassis so as to maximize your server capacity. Relatedly, Sangoma ships a 1U bracket free of charge with each board in case you want to mount the card in a 1U chassis. SangomaÂ’s boards are not full-length, so they unlike some cards they donÂ’t have issues of being blocked by the processor, heat sink or other motherboard components.



Optional Bracket is included

One nice feature with their analog cards is that they support removable FXO and FXS daughter cards that are completely interchangeable. Thus say you have one of SangomaÂ’s four port analog cards - you can start off with the analog card having 4 FXS ports, then change one daughter card to FXO and now you have an analog card with three FXS ports and one FXO port. This definitely enables a lot of flexibility and lets you retain your existing investment in the analog cards you purchased. You'll notice from this picture of a 24-port analog FXO/FXS card that you don't see the PCI pins. That's because it only uses one PCI slot for data communications and all the other cards interconnect and share the data connectivity via a connector in the back.



Sangoma 24 port FXS or FXO - Front

Below is a better photo demonstrating this. You'll also notice that since one single PCI slot cannot handle so many cards that Sangoma actually has a power connector in the back that uses the power supply to meet the cards' power needs. The benefit of this architecture is that you only need a single IRQ shared by all of the boards.
Sangoma 24 port FXS or FXO - Back

SangomaÂ’s original core strength was in the WAN market, so I should point out that in mixed voice/data installations, the data streams are separated out in hardware and handled completely independently of the voice channels by SangomaÂ’s standard Wanpipe data handlers. Only the voice related traffic gets diverted to the Asterisk system.

Comparison
HereÂ’s a comparison chart between Sangoma and Digium hardware. Note that sometimes Digium is more expensive and sometimes Sangoma is more expensive. YouÂ’ll have to weigh not only cost, but features, warranty, and "brand loyalty" before buying either brand. Digium certainly has built a lot of brand loyalty since their founder Mark Spencer founded the Asterisk open source PBX. Nevertheless, this chart should be a good guide to use when picking hardware for the Asterisk platform.

	Digital T1/E1 hardware
	Sangoma	Digium
model #	A101	TE110P
description	single port T1/E1/J1	single port T1/E1
compliance	3.3V/5V	3.3V/5V
warranty	5 year	2 year
price	$599	$595
model #	A102	TE210P/TE205P
description	dual port T1/E1/J1	dual port T1/E1
compliance	3.3V/5V	TE210P=3.3V, TE205P=5V
warranty	5 year	2 year
price	$999	$990
model #	A104	TE410P/TE405P
description	four port T1/E1/J1	four port T1/E1
compliance	3.3V/5V	TE410P=3.3V, TE405P=5V
warranty	5 year	2 year
price	$1,699	$1,495
model #	A104d	TE411P/TE406P
description	four port T1/E1/J1 with HW EC	four port T1/E1 with HW EC
HW echo canceller	128ms echo tail across all DS0's	64ms echo tail on 1st 32 DS0's, 16ms after 32 DS0's
compliance	3.3V/5V	TE411P=3.3V, TE406P=5V
warranty	5 year	2 year
price	$2,699	$2,495
	analog FXO/FXS hardware
	Sangoma	Digium
model #	A20002	TDM400P
description	4 port FXO	4 port FXO
compliance	3.3V/5V	3.3V/5V
warranty	5 year	2 year
price	$360	$421
model #	A21200d	TDM2460E
description	24 port FXS with HW EC	24 port FXS with HW EC
compliance	3.3V/5V	3.3V/5V
warranty	5 year	2 year
price	$2,034	$2,095
model #	A20606d	TDM2433E
description	12FXS/12FXO	12FXS/12FXO
compliance	3.3V/5V	3.3V/5V

Sangoma has some other interesting tricks up its sleeve. In addition to supporting open-source PBX systems, they also support open-source routers. Yep, that's right, a home-brewed DIY (Do It Yourself) router that you can build using a PC. For instance, users can build a Vyatta router by downloading a CD image from their website and installing it on a PC with the appropriate hardware. XORP, or extensible open router platform, is their flexible open-source software that is just as adept at being your home WiFi router as it is your enterprise business router. Vyatta works Sangoma T1 and T3 WAN hardware, but they plan to add other partners as well. Om Malik has an excellent overview of Vyatta which he wrote in Business 2.0 and repurposed for his blog, making note of the fact that Vyatta can be 1/5th the cost of a comparable Cisco router. Imagine if Vyatta does to the data router network what Asterisk has done - and is doing - to the voice switching network!

The one thing that I admired about my visits to both Digium and Sangoma is their corporate atmosphere and culture. Both were very laid back with more engineers that you could shack a stick at. My kind of people. Both companies seem to really enjoy technology and have made it their passion.

I was also impressed with Sangoma when they told me that many of their engineers, including their Senior Linux Engineer, Nenad Corbic, actually make themselves readily available to their customers via Microsoft MSN Messenger, a popular instant messaging (IM) client. Putting the irony of a company with strong Linux roots using MicrosoftÂ’s IM client aside, Nenad told me his buddy list is hundreds long filled with customers and that he is always eager to help. Giving your personal IM username to customers? - Now thatÂ’s customer service!

Finally, as IÂ’ve already stated, I was impressed with the audio quality of the Sangoma boards installed in both YATE and Asterisk, especially with hardware echo cancellation turned on. If you are evaluating YATE or Asterisk, you should give careful consideration to Sangoma hardware.

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Tags: Sangoma, Asterisk, Digium, VoIP, TDM, T1, E1, analog
Related Tags: sangoma hardware, asterisk platform, digium hardware, yearprice model, hardware sangoma, sangoma

Written by randy@strategypluscommunications.com on March 3rd, 2006 with no comments.
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Asterikast video podcasts to teach you Asterisk

Asterikast is a podcast that teaches and explains about the VoIP-capable Asterisk PBX by Digium. According to Asterikast, "We also plan on having videos that can help step you through the process of setting up your very own Asterisk PBX."

They already have two episodes available for download. Episode I, "the very first episode of Asterikast" they cover how to compile Asterisk and setup one SIP phone. Episode II has much higher quality video, audio and on-screen footage. They cover voicemail, conference bridges, macros in Asterisk and TDM/TDM cards. Looks like they're also offering a computer pre-installed and pre-configured with Slackware and Asterisk for $800.00 if your a bit intimidated to install Linux + Asterisk. Both are actually pretty easy to install, and in fact it's worth checking out the videos just to see how easy it truly is.

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Tags: Asterisk, PBX, Asterikast, Slackware, VoIP, Linux
Related Tags: Asterisk, asterisk, asterikast, Asterikast, episode

Written by tkeating@tmcnet.com on February 21st, 2006 with no comments.
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Asterisk@Home 2.5 released

Looks like a new version of Asterisk@Home is out - version 2.5. I checked Asterisk @Home on SourceForge and they're still listing v2.1, so I'll take Nerd Vittles word for it. Regardless of whether v2.5 is ready for immediate download, Nerd Vittles put Asterisk@Home in great perspective when he said in a post today:

Want a rock-solid PBX at a rock-bottom price: free! Well, it's been two days since our tutorial on AAH 2.4 but here we go again! Asterisk@Home 2.5 has hit the street because of another serious bug-fix release of Asterisk. Now you get version 1.2.4 of Asterisk, and you also get the latest and greatest version of Linux, CentOS 4.2; the latest Festival Speech Engine (1.96); the latest version of the Asterisk Management Portal (1.10.010); the Flash Operator Panel (version 0.24); Open A2Billing; Digium card auto-configuration; NVfaxdetect support; loads of AGI scripts including weather forecasts and wakeup calls; xPL support; the SugarCRM Contact Management System with the Cisco XML Services interface and Click-to-Dial support; plus lots more. And, yes, it still fits on a single CD! more...

Well said Nerd Vittles, I couldn't have said it better myself! And oh yeah, did I mention Asterisk@Home is free? Asterisk@Home's name is a misnomer - sounds like Windows Professional "Home Edition" - a dummied down version of Windows XP Professional that had Remote Desktop removed as well as the ability to join domains. So sure the name sounds like they "dummied down" the features to make Asterisk easy to use, but in fact, Asterisk@Home is like Asterisk on steroids. It has the complete functionality you get with Asterisk and then some. This version isn't simply for Linux noobs. Heck, I may try it myself just because it's a nice turnkey version of Asterisk with lots of scripts and add-ons.

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Written by tkeating@tmcnet.com on February 3rd, 2006 with no comments.
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Sangoma offers new analog cards

Sangoma Technologies Corporation announced at Internet Telephony Conference & Expo that it is now shipping its A200 analog FXO/FXS PCI cards for use in the popular Asterisk open-source IP-PBX..

“There is still a huge demand and need for FXO/FXS technologies in many countries around the world,” says Sangoma Technologies President and CEO David Mandelstam. “Not all telephone systems require the latest VoIP technologies, but still require robust analog capabilities to maintain quality of service. We understand this need and are taking an industry lead by providing a price/performance solution that our competitors will now follow.”

The A200 solution supports any combination of up to 24 FXO or FXS connections. A single PCI slot host connection for all ports ensures common synchronous clocking for all channels. The base AFT architecture is shared with Sangoma's A101, A201 and A104 and soon to be released A108 cards ensuring common 3.3V/5V, high performance PCI compatibility. The A200 is firmware upgradeable and they offer an optional low density hardware-based echo canceller for the A200 series.

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Tags: Sangoma, analog, PCI, Asterisk
Related Tags: sangoma technologies, Sangoma, technologies, sangoma, analog, cards

Written by randy@strategypluscommunications.com on January 26th, 2006 with no comments.
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Chit-chat with Mark Spencer

I met with Digium's founder, Mark Spencer yesterday to discuss what was happening with Asterisk. Mark told me that they are working very hard on Jingle support, an open set of extensions to the IETF's Extensible Messaging and Presence Protocol (XMPP) for use in VoIP, video, and other peer-to-peer multimedia
sessions. This will enable federation with Google Talk and other Jingle supporters. Mark said that Jingle support will be coming very soon. If Jingle is successful, this is yet another reason to be skeptical over Tello, which I discussed yesterday.

Mark also told me they are participating in more SIP interoperability bakeoffs and tests. He also reiterated the cool Bluetooth integration thast Asterisk has and offered to give me a technology demo at some point.

Mark noticed the Nokia N90 that I have as part of the blogger's review program and asked how I liked it. I told him it was pretty feature-rich but I still hadn't figured out how to increase the handset volume or how to switch to vibrate mode. (assuming it supports vibrate). I hate reading manuals, but looks like I may have to for the N90. Anyway, Mark noticed the built-in camera and he decided to take a photo. Actually, I found out later that Mark took a video of himself, not a photo. I was going to share the video, but I can't figure out how to email the damn thing from the Nokia phone. The N90 keeps telling me the file format isn't supported when I try to email it.

We also talked about a cool new videophone from Grandstream Networks called the GXV3000 which is launching at Internet Telephony Conference & Expo and which Mark said works with Asterisk. I'll post a new entry when I find a photo of the GXV3000. from I have a printed photo of it, but no scanner handy.

I know there was some other interesting stuff Mark told me, but I must have forgotten. Getting up at 5am to fly down to Florida will do that to your memory recall.

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Tags: Asterisk, Nokia N90
Related Tags: jingle support, photo, Jingle, jingle, asterisk, video

Written by tkeating@tmcnet.com on January 25th, 2006 with no comments.
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Signate announces SigPRO scalable Asterisk solution

Today, at ITEXPO, Signate is announcing SigPRO. SigPRO is a a hosted telephony system for telephone service providers with 5,000 to 500,000 customer extensions that Signate claims "sets a new price/performance standard".

I have heard various issues surrounding scalability issues when it comes to the Linux-based Asterisk IP-PBX solution, so I contacted one of my sources and asked about Signate's scalability claims. My source responded, "Signate is in for a shock when they really try to scale this stuff. They are using GPL'd code from Asterisk, and have not licenced the code. This means that they are likely not changing core Asterisk source code to implement scale. This means they are in for a bit of a surprise when they actually start getting hundreds, let alone hundreds of thousands concurrent users on one box."

He added, "It takes significant engineering to truly achieve scale, not just shoving it on a box with more horsepower. Under the GNU-GPL of Asterisk, any changes to the source code must be given back to the community. Therein lies the rub, and therein lies Digium's best business model yet."

I pointed out to my source how Signate was achieving scalability by showing him a portion of their news release: "SigPRO achieves further efficiencies because it works in concert with SignateÂ’s Telephony Server 5000 softswitches, which provide better price/performance than PC-based or proprietary switches. In an installation, the SigPRO server provides applications, services, and feature-functionality. Call routing, trunking, translations, and most call-control/management services are provided by Signate's Telephony Server 5000 softswitches. Softswitches are load balanced and deployed in an n+1 configuration for redundancy."

He replied to this by saying, "It takes significant engineering to truly achieve scale, not just shoving it on a box with more horsepower. Under the GNU-GPL of Asterisk, any changes to the source code must be given back to the community. Therein lies the rub, and therein lies Digium's best business model yet."

He then theorized that Signate is using one Asterisk box for SIP routing, this is the "softswitch box", and another Asterisk box for their PBX feature-set, etc. Thus, they may achieving some scalability by distributing the load across multiple servers. The problem is that they would need to actually get into that source code and make significant changes, and hence make community contributions, so long as they are using the GPL'd version of Asterisk.

How Signate achieves their scalability without running afoul of the GPL is something I will be sure to investigate at Internet Telephony Conference & Expo and report back here. I'll keep you posted. In the meantime, here is the release which will go out on the newswires in a few hours...

Signate Announces SigPRO Hosted Telephony Application That Sets New Price Performance Standard

Signate, the leading provider of VoIP telephony solutions based on industry standard hardware and open source software, today announced the production release of SigPRO, its state-of-the-art hosted telephony system for telephone service providers with 5,000 to 500,000 customer extensions. Targeted at providers who are moving into the IP voice market, SigPRO enables providers to increase revenue, reduce expenses and deliver feature-rich solutions to individual consumers, small businesses, and large enterprises.

"Signate's SigPRO is enabling our market entry with a capital investment of less than $5 per potential customer," said Jason Cohen, CEO of FutureLink, a service provider based in Coram, New York. "That means we can offer very competitive prices and richer service offerings than TDM-based carriers at the same time," he said.

"Until SigPRO, hosted systems for small and medium-sized service providers have either been prohibitively expensive or limited by low calling capacities and minimal feature sets," said William Boehlke, CEO of Signate. "SigPRO enables a new generation of nimbler IP carriers to deliver services to new markets, such as business PBX replacement," he said, "with the same features as the very best PBXs but without the capital investment."

About SigPRO
SigPRO is both feature-rich and cost-effective because it sits atop a foundation of industry standards such as SIP and open source components like Linux, Apache, MySQL, php, SIP Express Router and Asterisk.

SigPRO achieves further efficiencies because it works in concert with SignateÂ’s Telephony Server 5000 softswitches, which provide better price/performance than PC-based or proprietary switches. In an installation, the SigPRO server provides applications, services, and feature-functionality. Call routing, trunking, translations, and most call-control/management services are provided by Signate's Telephony Server 5000 softswitches. Softswitches are load balanced and deployed in an n+1 configuration for redundancy.

Up to eight Telephony Server 5000 softswitches may be deployed in a stack with a call set-up capacity of up to 350 calls per second on seven servers. The eighth server is back-up capacity should one of the active servers fail.

SigPRO includes one interface to a credit card provider and templates for popular customer premise devices such as Linksys adapters and Cisco and Polycom SIP telephones. Customer detail records at the system, reseller, enterprise and consumer levels may be interfaced to external billing systems.

Functionality is delivered through five web browser interfaces that can be branded with reseller or enterprise customer logos and colors.

SigPRO's Administration interface provides access control, DID inventory management, call rating, customer premise device management, and other administrative features.
Service provider contact centers and resellers manage customer care through SigPRO's ITSP interface. Each reseller has its own interface where the reseller services its own customers independently of other resellers, under parameters set by the service provider for that reseller.
Each business customer uses a SigPRO Enterprise panel to manage their own extensions, within the limits of the service plan they have contracted for.

From the Enterprise panel, customers can configure business features such as:

• Auto Attendant / IVR – Enterprises manage inbound calls and deliver them to the intended destination through interactions that can include recorded answers, extension dialing, and time of day pathing
• Conferencing – Conferencing lets enterprises coordinate dispersed teams
• Complex IVR – Enterprises can quickly build their own voice response systems that take DTMF tones or record responses
• The Self-Service interface lets consumers sign themselves up for service without human intervention. SigPRO automatically provisions their customer premise device and charges their credit card to begin service. Each reseller has its own unique Self-Service interface.

The SigPRO user panel gives users control over their own telephone service functions such as:

• Follow Me – Users can set the system to ring numbers such as mobile or home phones in sequence when someone calls their desk phone, so they never miss an important call
• Call blocking and Screening – Each user decides who reaches them and when
• Email Integration – Users easily integrate voicemail with their Microsoft Outlook mailbox
• Voice Messaging – Feature-rich voice messaging provides users with the flexibility to use and manage their messaging service from anywhere

Service and Support
SigPRO is offered with 24/7 technical support 365 days a year by Signate engineers in the U.K., U.S. and New Zealand. On site hardware service is provided by the global SGI support organization. Four hour and 30 minute service level agreements are optional.

Pricing and Availability
A redundant calling configuration for 1,000 simultaneous SIP calls begins at $125,000. That configuration includes a SigPRO server and two Signate Telephony Server 5000 softswitches.

An English language North American version of SigPRO is available immediately. An English version with support for currencies other than dollars will be available in the second quarter of 2006. User interfaces for Spanish and other languages will be available in the second half of 2006.

About Signate
Signate is a leading global provider of design, installation, configuration, training and management services for open source VoIP telephony systems. For more information, visit Signate at www.signate.com

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Written by tkeating@tmcnet.com on January 23rd, 2006 with no comments.
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Asterisk and Ranch Networks make Asterisk secure and more scalable

Today at Internet Telephony Conference & Expo, Digium and Ranch Networks have announced that they are teaming up to make Asterisk a secure and scalable VoIP solution. I have a meeting with Mark Spencer, Mr. Asterisk himself at 4pm today to discuss this new relationship, but I wanted to share this news before it hit the news wires.

Digium, the original creator of Asterisk and pioneer of open source telephony, and Ranch Networks, the first IP telephony network appliance provider to integrate security and bandwidth control for IP-based applications, today introduced its security code for the Asterisk open source IP telephony platform. When combining Asterisk with Ranch Networks new RN series appliances, the solution provides unprecedented security and scalability to the open source telephony industry. The code is available for download at www.ranchnetworks.com and ftp.digium.com

Ranch Networks solves the problems associated with VoIP implementations through both its MIDCOM integration with the Asterisk platform, and several of its Patent-Pending technologies. The MIDCOM integration provides for dynamic per-call firewall control, bandwidth management, NAT traversal, and RTP traffic bridging - all supporting encrypted signaling streams, while the Patent-Pending technologies separate voice, video, and data traffic into multiple, secure zones without having to reconfigure IP addresses.

"Security is potentially an issue for SIP deployments since ports need to remain open at all times in order to enable voice traffic-leaving potential open doors for intruders," said Mark Spencer, president of Digium and creator of Asterisk. "In providing this technology, we are proactively addressing the security of VoIP before it becomes a major concern, while also providing quality of service and increased scalability."

Ranch Networks' technology sends instructions to the firewall to open the ports on an as needed basis. Media streams can flow from one phone to another, then close ports when the call is terminated. Instructions are also sent to the appliance determining the amount of bandwidth needed. As calls come in, data is shrunk to allow necessary bandwidth on a protocol basis.Ranch Networks guarantees that voice is given priority over data. This technology supports multiple phones behind any number of firewalls.

"We expect this will change the mindset of large enterprises and service providers that were previously skeptical of VoIP and open source technologies, to look more closely at the customization, flexibility and huge costs savings that a platform like Asterisk provides," said Ram Ayyakad, founder and CEO of Ranch Networks. "We have eliminated any security and scalability issues that could be argued by proprietary vendors."

The Ranch Networks security code has received the highest certification through ICSA.The security code is available by the download of Asterisk 1.2.2 and is available for download from www.ranchnetworks.com and ftp.digium.com

Further, Ranch Networks today introduced the RN series of appliances which are the first to integrate security and bandwidth control for VoIP applications. The company's IP telephony network appliances secure, manage and scale VoIP traffic beyond existing firewall technologies. Ranch Networks' appliances separate voice, video, and data traffic into multiple secure zones without having to reconfigure IP addresses. Several patent-pending technologies, allow service providers to deliver VoIP services to enterprises without affecting existing data networks.

Ranch Networks' technology is designed to work with leading IP PBXs and supports all sizes of enterprise and carrier deployments. Instead of requiring labor intensive access lists or protocol specific application layer gateways, Ranch Networks' IP PBX controlled appliances provide dynamic, protocol independent, per-call authenticated network access. This unique approach both simplifies and increases network security, while allowing encrypted signaling streams.

Starting with Asterisk, the open source PBX, Ranch Networks' offers the RN300, RN20, RN40 and RN41 to provide dynamic, protocol independent, per-call authenticated network access. These products will supply unprecedented VoIP security, bandwidth management, VPN, accounting and switching capabilities to small to mid-size enterprises, service providers and carriers.

"Asterisk was a logical starting point for us as it allows us to leverage a vast growing user base, a focused and enterprising reseller channel and avoid the red tape of the proprietary PBX vendors," said Ram Ayyakad, founder and CEO of Ranch Networks. "We believe now that this puts Asterisk in line with the leading proprietary IP PBX vendors."

The RN300, RN20, RN40, and RN41 are available through leading Asterisk resellers worldwide ranging in price from $750 to $20K.

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Written by tkeating@tmcnet.com on January 23rd, 2006 with no comments.
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Mark Spencer podcast interview

Ronald Lewis interviewed Mark Spencer, Asterisk founder/guru this morning. I listened to the podcast while simultaneously trying to read and respond to email, surf the Web, and prepare for Internet Telephony Expo next week. Unfortunately, my multitasking skills aren't what they used to be since I only absorbed probably 30% of the entire interview. I could tell there were some good questions posed by Ronald and interesting answers by Mark making it worth checking out the podcast. Maybe I'll go listen to it again. I know Mark discussed the history of Asterisk, the release of Asterisk v1.2, the whole SIP vs. IAX protocol debate, and some other stuff.

Ronald Lewis and Mark Spencer no doubt used VoIP to record the podcast using a combination of Asterisk and Gabcast, which supports IAX Asterisk trunking and which I just blogged about yesterday. In fact, I detect one little "blip" in the sound quality in one part due to some dropped packets in one part of the interview. But overall, great voice quality and some good content. Go check out the Gabcast recorded interview.

Update: I called this a "podcast" interview, when technically it doesn't play on your iPod but rather on the web. You can of course view the HTML source to figure out the MP3 filename and then manually download to your iPod, but Gabcast doesn't feature the typical automatic seemlessly "podcast" integration. At least, not that I can see from the website.

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Written by tkeating@tmcnet.com on January 19th, 2006 with no comments.
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5 Days until Internet Telephony Expo plus Asterisk certification

I can't believe it's just 5 days till Internet Telephony Expo. While I'm excited to be leaving behind cold Connecticut for warm Ft. Lauderdale, Florida there is so much going on at this next show that I don't know how I'm going to visit with all the vendors I want to on the show floor, moderate my "Managing Your Network for High-Quality Voice" panel, attend a few of the interesting conference sessions, and attend all the pre-briefings I have scheduled. On top of it all, I still haven't replaced my cell phone that I lost in a taxi at the last Internet Telephony Expo, so it will be tough to juggle or change my schedule on the fly. Being the gadget lover that I am I just can't decide whether to get a Treo 650, Treo 700, HP iPAQ hw6515, or some other smart phone. I'm usually pretty decisive about these things, and just buy whatever the hell I want, but with a baby on the way, my gadget shopping-spree days are over.

In any event, I just wanted to let everyone know that the TMC team is continually trying to improve the education that goes on at Internet Telephony Conference & Expo. Even with the show just 5 days away, TMC has added an exciting new educational session to Internet Telephony Conference & Expo. According to Rich Tehrani's blog post, "Asterisk certification" has been added to TMC university. Rich writes, "Be sure to check out the Asterisk certification at Internet Telephony Conference & Expo next week. We are very proud to be expanding TMC UniversityÂ’s certifications into a new area. If you are interested in getting certification in open-source VoIP be sure to come to Internet Telephony Conference & Expo next week and partake in the learning and certifying!"

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Written by randy@strategypluscommunications.com on January 19th, 2006 with no comments.
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Packet Island debuts VoIP Lifecycle Management Solution for Asterisk

Packet Island debuted their VoIP Lifecycle Management Solution for Asterisk at INTERNET TELEPHONY Conference & EXPO West 2006. For those deploying Asterisk, this seems like a nifty tool to perform performance metrics, VoIP network assesments, and ensure that the Asterisk IP-PBX is performing up to snuff.

Here’s the news announced at the show…

Santa Clara, CA — Packet Island today announced that it will debut it’s VoIP Lifecycle Management Solution for Asterisk at TMC’s INTERNET TELEPHONY® Conference & EXPO West 2006, October 10-13, 2006 at the San Diego Convention Center in San Diego, California (www.itexpo.com).

Packet Island’s ‘PacketSmart for Asterisk’ is a Software-as-a-Service (SaaS) solution that has been designed to target the VoIP lifecycle management needs of VARs and SMBs deploying Asterisk. The solution is based on Asterisk software agents and purpose-built micro-appliances that work with the highly scalable PacketSmart SaaS platform, hosted at a Tier-1 data center. The Asterisk software agents are highly optimized applications installed on the Asterisk PBX and collect detailed quality metrics on all VoIP calls terminating at the PBX. The 4”x4” micro-appliances are complementary to the software agents, in that unlike the software agents, they can be easily moved around the network to isolate and troubleshoot network issues affecting VoIP quality. The “PacketSmart for Asterisk” solution has three distinct functions: Network assessment for VoIP, VoIP troubleshooting, and ongoing VoIP SLA monitoring. During the deployment planning phase, Asterisk VARs can use the micro-appliances to simulate live VoIP traffic on the SMB network, to identify and fix problems before VoIP deployment. After Asterisk deployment, the Packet Island software agent that is left installed on the Asterisk PBX continues to collect detailed quality metrics that can be used by the SME and VAR for SLA monitoring and to isolate and troubleshoot transient VoIP quality issues. For multi-site deployments with multiple Asterisk PBXs, the software agents can be employed to do periodic network assessments to ensure that the inter-Asterisk connectivity continues to support good VoIP quality.

“Packet Island’s participation at INTERNET TELEPHONY® Conference & EXPO is a primary reason why this show is widely recognized as the #1 venue where service providers, enterprises, government agencies, developers and resellers learn about the benefits inherent in IP Communications products and services,” said Rich Tehrani, TMC President.  “The evolution in this industry has been monumental and it still continues to grow. We are proud to have Packet Island and its ‘PacketSmart for Asterisk’ solution as a key part of this year’s program.”

“Open source Asterisk is creating a huge disruption in the PBX marketplace. However, one of the big challenges in deploying Asterisk with VoIP today is in ensuring that the customer’s network is ready for VoIP before the Asterisk deployment, and to stay on top of transient VoIP quality issues that invariably crop up as the network evolves. The problem with current VoIP assessment and monitoring solutions in the market is that they are priced in such a way that they exceed the cost of the entire Asterisk PBX. With ‘PacketSmart for Asterisk’ we are offering the industry’s first SaaS solution that has been specifically created for the VoIP lifecycle management needs of the Asterisk market. Our customers love the fact that instead of paying thousands of dollars up front, they pay us an annual service fee that works out to tens of dollars a month“ said Praveen Kumar, Co-founder & President of Packet Island Inc.

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Tags: asterisk, network assessment, packet island, testing, voip
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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trixbox 2.0 released

trixbox 2.0 beta will be available for download on Wednesday. This release will be Fonality’s first big contribution to the trixbox/Asterisk community after the recent Fonality acquisition of trixbox. which certainly caused a stir within the Asterisk community. I spoke with Chris Lyman, CEO of Fonality, to find out more about this major new release of trixbox.

First, I should point out that while previous version of trixbox have always been the easiest way to get Asterisk up and running in just minutes, trixbox 2.0 is much more than that.  First and foremost, trixbox 2.0 includes a new ‘overall’ web GUI to make the whole process "point and click". From this new web GUI you can simply select the modules you want (HUDLite, FreePBX, PHP, lame, etc.) and the web interface will automatically install them. Some of the packages are directly related to Asterisk such as HUDlite or FreePBX, while other options are ancillary, such as SugarCRM. The idea is you shouldn’t have to know anything about the command line interface (CLI). In addition, many users wishing to install trixbox want to keep the server as unbloated as possible and not add any unnecessary modules/packages.

Continue reading trixbox 2.0 released…

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Tags: asterisk, laamp, laamps, lamp, mysql, sugarcrm, trixbox, trixbox 2.0, voip
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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Sangoma shipping the A102d dual port E1/T1/PRI Asterisk card

Sangoma Technologies Corporation, announced today that the A102d, a dual port E1/T1/PRI card with carrier-grade echo cancellation is now shipping. Although Sangoma's cards work on a variety of applications, the most popular is using Sangoma hardware on Asterisk PBX servers. (See my Asterisk Sangoma test drive article)

The A102d includes the same digital processing and PCI/PCI Express technology as the four-port A104d card and is intended to provide the benefits of hardware-based echo cancellation and voice enhancement for smaller, two-port T1/E1 installations at a lower cost. The A102d also supports DMF encoding/decoding and tone recognition, voice quality enhancement and adaptive noise reduction.

"The A102d was developed in response to the continuing demand for a small capacity, low-cost E1/T1 card with telco-grade echo cancellation," says Sangoma Technologies president and CEO David Mandelstam. “As an integral part of Sangoma's AFT design family, the A102d inherits the compatibility, enhanced performance and reliability of its siblings."

The A102d supports PBX, IVR and VoIP applications, such as Asterisk, Yate, FreeSwitch and many proprietary telephony projects.

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Tags: a102d, asterisk, freeswitch, pbx, sangoma, voip, yate
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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Fonality launches new PBXtra Professional Edition

Fonality, provider of Asterisk-based IP telephony solutions (including trixbox 2.0), today announced the release of its new PBXtra Professional Edition IP-PBX and HUD (Heads-Up Display) Team application. I spoke with Chris Lyman, CEO of Fonality to get the scoop on this news. The short take is that PBXtra Professional Edition is a notch up from their Standard Edition targetting larger businesses that aren't necessarily call centers, scaling from 50 to 500 seats with enterprise-class features. Similarly, HUD Pro is launching, which is a big upgrade of HUD, offering more features that are interesting to larger companies - more on that later. The most interesting thing Chris said to me was this, "The whole point of this launch is to start taking on the PBX giants. This isn't like 'Are we competitive in the Asterisk space?' We know that we're the leaders of that space, ok? This is about 'can we take on Avaya, Cisco, and Nortel - with feature-parity for 50/60/70% price reduction.' "

If you think that statement is controversial, you should see the firestorm started by Voxilla's Marcelo Rodriguez in his inaccurate portrayal of Fonality as "unsecure" and open to an unscrupulous Fonality employee "spying" on their customer's networks. Marcelo even goes as far as to say customers are better of using Digium over Fonality when he says, "But those concerned about keeping company secrets are probably better served by Digium's offering. " I'm going to stay out of that battle, but it's worth checking out to see Chris Lyman's point-by-point rebuttal to Marcelo's assertions.

Getting back to today's news on their new PBXtra Professional  Edition, Chris said, "On the PBXtra Pro side, this news brings is more scalability in terms of supporting more larger groups of extensions, faster - more concurrent calls - faster reporting - all the things that you need in a bigger office."

Tom: What did you do to make Asterisk more scalable?
Chris: What we've done on the web admin site is we re-engineered our entire back-end API to handle many more concurrent connections and easier management of multiple systems. We originally built that web interface to handle systems under 100 seats. So very difficult to manage as you got up into the hundreds of seats, it was hard to navigate, it was slow."

The newly re-engineered web interface lets you easily hotswitch from one branch office to another for configuring users, making adds/changes/etc.

Tom: You're using MySQL on the back-end, correct?

Continue reading Fonality launches new PBXtra Professional Edition...

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Tags: asterisk, fonality, hud team, pbxtra, pbxtra professional edition, trixbox, voip
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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VoIP Call Recording using Port Mirroring

Remember my VoIP call recording round-up of various VoIP call recording solutions? Well, it's time to update that post with a new solution I found from Arcosoft Inc. Arcosoft today announced the release of VONaLink TeamRecord, which works with any VoIP phone system based on the open SIP standard, such as Asterisk or Vonage, to centrally record all phone calls. What is interesting about this solution is that it unobtrusively "sniffs" for voice packets on your data network by leveraging the port mirroring capability of a network switch. By implementing port mirroring, TeamRecord can see all of your network traffic, and then using intelligence packet capturing technology it can find the RTP (audio) stream without the need for any recording software at each workstation. Sure, you could do port mirroring even without Acrosoft and then use Ethereal or some other packet sniffer to decode the RTP streams, but it's more of a kludge. Besides, most network packet sniffers weren't designed with VoIP in mind, so their features are limited.

VONaLink has some interesting features. For instance, in addition to recording the RTP audio streams as a stereo WAV or MP3, you can add an inaudible watermark for later verification that the file has not been changed. In addition, users can listen to recordings of their own calls from any web browser.

VONaLink TeamRecord runs on Windows XP Pro, 2003, and 2000. Price starts at $500 USD for 5 phone licenses. Each additional phone license is $100 USD. You can download an evaluation copy from www.vonalink.com

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Tags: arcosoft, asterisk, call recording, port mirroring, voip, vonage, vonalink
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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VONaLink softphone uses SIP to capture CallerID

I just wrote about Arcosoft’s VoIP call recording software this week and now just a few days later they launched another interesting product called VONaLink ScreenPop. ScreenPop works with any VoIP phone system based on the SIP standard, such as Vonage or Asterisk, to provide screen pops and even the ability to reject calls. Buh-bye telemarketers!

Typically Caller ID is read by a computer using an analog modem or a proprietary CallerID box. VONaLink uses open standards in the VoIP world to simply extract CallerID info by monitoring the network packets.

More important than the ‘techno-speak’ on how it works, using the Caller ID of the incoming call, ScreenPop searches for the caller in Microsoft Outlook contacts, or launches custom applications to search the web or company database. If the caller is found, the information is popped on the screen.

Unwanted callers can be added to the reject list. Integration with Vonage Click2Call allows outbound calls to be placed by clicking in the call log within ScreenPop. Sweet! Now if only I was still a Vonage customer.

VONaLink ScreenPop runs on Windows XP, 2003, and 2000. Priced at $29 USD

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Tags: asterisk, caller id, sip, voip, vonage
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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Who is redirecting trixbox/Asterisk@Home web traffic?

Have you noticed how http://www.asteriskathome.com points to http://www.asterisknow.org/, a site that promotes AsteriskNOW, a new software distribution that includes a customized version of Linux, Asterisk, the Asterisk GUI, and all other software needed for an Asterisk system. AsteriskNOW is promoted as an easy to install turnkey installation when it says, "AsteriskNOW includes all the Linux components necessary to run, debug and build Asterisk, and only those components, so installation is easy." Doesn't this turnkey Asterisk sounds eerily similar to the old Asterisk@Home project which was recently renamed to trixbox?

The site www.asteriskathome.com is offering a competing product to trixbox (Asterisk@Home), which is now owned by Fonality. Fonality is a company that sells PBXtra and is a competitor to Digium.. www.asterisknow.org, where asteriskathome.com redirects to, is registered to you guessed it -- Digium Inc. which now offers an Asterisk distro that "aims" to be as easy to install as trixbox/Asterisk@home. Quite a nice rivalry, eh? While I applaud Digium for adding a GUI and making AsteriskNOW an easy to install Asterisk distro, why take a domain name associated with a competing Asterisk distro and competing company? My first thoughts on this were:

Why even go there? Is Digium’s goal to rank highly on the search engines for the search term “asterisk at home” or “asterisk@home” so that they direct people to their competing solution?. Seems a bit childish.

Personally, I doubt most Asterisk fans aren’t already aware that Asterisk@Home is now trixbox, so I doubt Digium will get any search engine referral traffic even if they did rank highly for the keywords. I’ve been to Digium’s offices and I know they are filled with many smart Linux people, so surely they know they won’t get any SEO benefit from registering www.asteriskathome.com. So is the motive simply because they could do it or should we view this as a childish prank?

When I visited Digium's offices I heard a group of employees playing some Linux shoot-em up game and shouting (gotta love that corporate environment). Now, I can only picture a group of Linux gurus huddled around, as they clicked the Submit button to register this asteriskathome.com domain followed by a few chuckles. But then it occurred to me that perhaps Digium wasn't involved with this at all. I checked the whois directory and saw that the domain was registered to Mitchel Constantin from a company called Dirty Clothing. I googled his name and see that he is indeed on Asterisk's mailing lists and seems very involved with the community - however he doesn't appear to be a Digium employee.

After some further investigation I discovered he works for or wrote the Asterisk application called Snap, that includes a dialer and call popup application for Asterisk with Outlook integration and a Firefox plugin. Actually, this seems like a pretty cool app that I'll have to check out.

Obviously, Mitchel is a huge Digium fan, the company that founded the Asterisk movement, and which apparently has spawned at least one 'domain turf battle' by a devoted (zealous?) Digium fan.

Lame to take this domain & direct it to Digium's website? Or pretty funny?
You make the call.

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Tags: asterisk, digium, fonality, trixbox, voip
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Comments on this Entry:

(roderickm on Dec 5, 2006 6:54 PM) Asterisk is a trademark of Digium. The project formerly known as Asterisk@Home was renamed “Trixbox.” How is it childish to protect legitimate trademarks?

(John Ferrel on Dec 6, 2006 9:39 AM) >>Asterisk is a trademark of Digium. The project formerly known as Asterisk@Home was renamed “Trixbox.” How is it childish to protect legitimate trademarks? A valid assertion, except there are plenty of cases of people using the name "Asterisk" within their domain name. Asterisk VoIP Blog - http://asteriskvoip.blogspot.com/ is one example. Even the infamous Apple Blog (http://theappleblog.com) uses the trademarked Apple in its domain name. I'm sure there are hundreds of other cases. I don't believe Digium sued Asterisk@Home to try and get them to reliquish the rights to the Asterisk@Home name. In fact, I remember reading Tom's blog post stating that Digium had given Asterisk@Home permission to use the name. Trixbox 1.0 replaces Asterisk@Home Here's 2 important snippets: [snip 1] "I can speculate that my pal Mark Spencer over at Digium sent Asterisk@Home a "cease & desist letter" to protect his trademark. Although, I'm not sure if Mark even has a trademark on the brand "Asterisk". I'll have to shoot him off an email and ask. But if he does, then you can't blame him if he is indeed trying to protect his brand. If you don't vigorously protect your brand and it becomes watered down and used by everyone, then you can lose your rights to the trademark. This whole point may be moot since I believe you can't trademark GPL open-source code projects. Where's a trademark lawyer when you need one?" [snip 2] "Update: I received an email from Ward Mundy stating that Digium did give permission to use the Asterisk@Home name. He stated "Asterisk@Home was suitable for home and business use. A lot of customers were apparently put off by the @Home moniker." I suppose I can see the @Home moniker hurting its "business-class" image." Taking the asteriskathome.com domain does seem to be a bit infantile, especially since the Asterisk@Home project no longer exists. It's now Trixbox. Of course I'm sure there are hundreds if not thousands of old webpages that talk about Asterisk@Home that aren't updated. So any Asterisk newbies that see Asterisk@Home mention might be tempted to google it and come across Digium's website instead. A form of stealth marketing I suppose. Still, like Tom said, the chances of asteriskathome.com getting any search engine traffic is small. Currently this domain is not even on the first page of Google. In fact, http://asteriskathome.sourceforge.net/ is #1 and it points to the trixbox Asterisk distribution. I won't classify this tactic "lame", especially since it doesn't appear to be done by a direct Digium employee. It is somewhat amusing to us in the Asterisk community to see two different factions play some relatively friendly war games. Just as long as they don't go hacking each other's websites, I'm cool with it.

Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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Signate, an Asterisk provider, bites the dust?

Signate is a provider of Asterisk-based solutions that apparently is going through major problems. I've been getting lots of emails asking if I know what is going on with them, so I did some research. Word through the grapevine is that they are about to go belly-up - but this is still unconfirmed. Their former CEO left and when I tried to reach their new CEO Paul Mahler and I got an automated voice message telling me to email Paul@paulmahler.com. It also mentioned that if I left a voicemail he would never get it. His work email also bounces back so I am guessing he is no longer with the company.

I then tried to reach an operator and was immediately put into voicemail. I talked with Joe Fabiano, one of our sales reps, and Joe emailed my pal Garrett Smith (SmithonVoIP) since Garrett's VoIPSupply.com company is a distributor of Signate. If anyone would know, a distributor would. Well, Garrett did some research of his sources and he told Joe, "According to my sources they are toast."

Is this a case of the Asterisk ecosystem growing too fast, with too many players trying to get into the Asterisk game? With free Asterisk offerings such as AsteriskNOW, trixbox, and inexpensive solutions from Fonality, Digium, and other providers - some hosted - some CPE, it will be interesting to see who survives and who doesn't. Considering Asterisk is an open-source solution, adding enough value and margin to Asterisk is a tricky business to remain profitable, especially when you consider that open-source advocates tend be very thrifty (cheap?) when it comes to paying for software.

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Tags: asterisk, asterisknow, digium, fonality, ip-pbx, signate, voip
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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Install Asterisk Business Edition on Fedora Core 5 or 6

Apparently there are some problems getting Asterisk Business Edition to install on some RAID configurations leading to some headaches. It's related to the RAID driver support and the install is looking for an older compiled driver. Dal over at Asterisk VoIP News has the solution to get it up and running on both Fedore Core 5 or Fedora Core 6. Go check it out.

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Tags: Asterisk Business Edition, Fedora Core 5, Fedora Core 6, VoIP, RAID
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Comments on this Entry:

(John Smith on Jan 18, 2007 3:29 PM) Cool! Thanks for the heads-up on this. Was having trouble with my Fedora/Asterisk install.

Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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Worldwide weather forecasts on Asterisk-based trixbox

Nerd Vittles is not only an Asterisk junkie, he's a weather junkie too. Nerd Vittles is at it again with a tutorial on how to receive worldwide weather forecasts on the Asterisk-based trixbox IP-PBX software platform. It retrieves the weather info from a website and then allows you to dial into the trixbox server to retrieve weather information using text-to-speech.

Apparently, it is one of the most requested features that Nerd Vittles receives. Personally, I just head over to weather.com via any browser. Or if I'm not on a PC or browser-capable cellphone, I can use WAP - available on just about any cellphone to retrieve weather from Yahoo! or other Internet sites. But who am I to judge people who need their weather fix by any means necessary? Go check out the tut.

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Tags: Asterisk, forecasts, Nerd Vittles, trixbox, weather
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Fonality gets financial boost from Intel

Fonality, makers of Asterisk-based PBXtra and the free trixbox Asterisk-based distribution, has secured
$7 million in a Series C round led by Intel Capital with participation from existing investor, Azure Capital Partners. It's a little known fact, but Intel Capital is the largest VC funding company in the world and of course if owned by well-known Intel, Inc.

I spoke with Chris Lyman, CEO of Fonality, on Monday discuss this news. Chris said, "Fonality didn't need another round of capital, but we were intested in a strategic round if we felt that it was a business that could take us to the next level - given that we are already profitable and we still have money in the bank from the last round. It has to be very interesting to get us to move and Intel is for a lot of reasons a very interesting company." Chris also mentioned that when approached with the opportunity to leverage Intel's worldwide distribution channel as part of the funding deal, it was something Fonality couldn't pass up.

In fact, Chris stated, "What most people don't realize is that Intel is 150,000 resellers. So we're more than a little excited to get exposed to that channel." When I asked, "So as part of this announcement you'll be able to go through Intel's channel?", Chris responded, "Well, I wouldn't say that's a part of this announcement, because it's a very young relationship, but that's certainly one of the primary reasons we were interested in them as a funding partner."

I responded "So it opens the door for you to use their channel" and Chris said, "It opens the door into their channel, it opens the door potentially to a closer hardware relationship with Intel. You know, Intel has always been interested in telephony, as proven by their Dialogic investment and then recent divestiture. When they divested Dialogic it doesn't mean they aren't interested in telephony. In my opinion, they are more interested in the future of telephony versus the past."

Chris also mentioned that the VC funding would be used to increasing product innovation for both the trixbox open source platform, and the PBXtra commercial product line. Further, the funding would be used to aggressively grow its channel presence, with a strong focus on international distribution, with support from Intel.

"Pairing industry standard hardware, such as Intel server and communication platforms, with open source telephony software can create a unique ecosystem that results in lower cost, high-end features, better ease-of-use and the potential for richer telephony environments down the road," said Lisa Lambert, managing director, Software and Solutions Group, Intel Capital. "Fonality is positioned to deliver this solution to the global mid-market and increase PC penetration in the emerging market for open source telephony."

In addition, Fonality told me they will announce the trixbox Open Communications Certification (FtOCC) workshop, the first in a series of training and certification courses for the trixbox application platform. The course will be held in Los Angeles on March 5th and 6th and will focus on VoIP, PBX deployment, network assessment, telephony troubleshooting, T1/PRI training, and IP handset education. The goal of the FtOCC is to arm data VARs, system integrators and telephony professionals with the knowledge needed to deploy and manage PBX installations for businesses from 1 to 1,000 employees. For more information visit www.trixbox.org

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Tags: Asterisk, Fonality, FtOCC, Intel Capital, PBXtra, trixbox, VoIP
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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