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Red Hat, makers of the popular Red Hat Linux distribution, is expanding their footprint in telecommunications with the acquisition of Mobicents technology and its new membership to the SCOPE alliance, an alliance dedicated towards building carrier-grade platforms.  Red Hat will work to provide the infrastructure software, platforms and tools to enable a next-generation service creation environment, and advanced delivery and management within the telecommunications industry.

“As the telecommunications industry undergoes fundamental transformation, traditional Telcos must prepare for a new competitive environment that demands greater innovation, faster time to market, responsible investment and new ways of thinking,” said Joanne Rohde, Executive Vice President of Verticals Marketing at Red Hat.

“Open Source is becoming strategic in Telecom, not only as a means to reduce costs, but as a model to inspire collaboration, inclusion and reuse – and to drive innovation.  Today, Red Hat software is routinely deployed in mission-critical Telco IT and Network environments around the world.  Through our investment in Mobicents, our alignment with the SCOPE Alliance and strong commitment to standards, we look forward to establishing Red Hat and open source software as fundamental building blocks for the successful Telcom companies of the future.”

Mobicents adds a Service Logic Execution Environment (SLEE) to Red Hat's technology portfolio. It complements J2EE to enable convergence of voice, video and data in next-generation intelligent applications.  In the telecommunications industry with call setup transactions as many as ten times the number of web data transactions per second, Mobicents enables high throughput with low latency.  

Mobicents is a highly scalable event-driven application server with a robust component model and fault tolerant execution environment that is effective beyond the telecommunications industry. It is the first and only Open Source Platform certified for JSLEE 1.0 compliance.  Web and SIP can be combined together to achieve a more sophisticated and natural user experience.  In fact, Mobicents claims to be the most popular Open Source SIP Application Server for the Java platform.

Mobicents also provides a network abstraction layer that insulates developers from the underlying complexities of legacy telecommunications network protocols and enables rapid development of new services and seamless deployment across IP and legacy Telco networks.

Red Hat intends to develop a communications platform offering that integrates Mobicents with the broader Red Hat middleware offerings.  More information about subscription product offerings will be available later this year.  In the interim, Red Hat will offer a pilot/beta program and a “bridge support” offering for development and deployment of production applications and solutions that incorporate Mobicents technology.
 
"JSLEE is an important emerging industry standard for even-driven architecture that is gaining its early traction in telecommunications industry," said Yefim Natis, vice president and distinguished analyst, Gartner. " An open source offering in this space extends the available options for users looking into event-driven computing and architecture."

Last week, I posted a blog entry about how Slacht, an Irish company offering a wall-mounted Asterisk-based PBX chose PIKA’s hardware. I also posted a 'teaser' when I stated, "I have some further thoughts on this news, which I can't get into right now, but I will post a follow-up blog post hopefully later today. Trust me, it will be interesting..." Well, it took me longer than I thought to write about my "further thoughts" but finally I found some time. What I find interesting about this news is that there are now several hardware choices when deploying an Asterisk-based PBX. Sure, PIKA has provided Asterisk-supported boards for some time, but it got me thinking how many brands of cards are supported and how this affects Digium. Let's see -- you have Digium cards, Aculab cards, Dialogic cards, PIKA cards, Rhino, and Sangoma cards that all work on Asterisk-based systems. (You also have ZAPMICRO and OpenVox which are Digium-cloned cards.)



Some cards work more seamlessly and have better driver support than others. For instance, I've heard it is difficult to get Dialogic channel drivers to work on Asterisk. I recall hearing that the Dialogic driver was licensed such that it could only be used with Asterisk Business Edition. In any event, with so much hardware competition for the Asterisk platform, how does this affect Digium, the corporation behind the open source Asterisk movement? A lot of their revenue comes from their hardware business, so with so many choices will this leave Digium "high and dry"?

Case in point, Fonality was Digium's largest customer, buying more Digium cards than anyone else. However, Fonality made the decision to go with Sangoma hardware over Digium because Sangoma hardware was less expensive and until recently, only Sangoma hardware supported the Octasic echo-cancellation for superior VoIP sound quality. I recall Fonality's CEO Chris Lyman a year or two ago mentioning they went with Sangoma hardware because they were sick and tired of all the support issues with Digium hardware. The trixbox appliance, another Fonality product, by default comes with Sangoma hardware, though you can get it pre-installed with Digium cards. There are also many horror stories of Asterisk users trying to get Digium hardware to install properly due to hardware interrupt issues. I compared/contrasted Digium hardware vs. Sangoma hardware last year. It's a bit out of date now since Digium now supports Octasic echo cancellation. Nevertheless, it's worth a look.

As SmithonVoIP points out, Sangoma's stock has been going like gangbusters when he points out, "Sangoma posted their Q3 earnings today, which showed a 24% increase in revenues over the previous quarter of this year, a 68% year over year increase in sales revenues, a 69% year over year increase in net income, and a 56% year over year increase in Net earnings." Relatedly, Rich Tehrani and I were discussing Sangoma's phenomenal stock growth a few weeks ago and both of us planned on writing about it. I believe Rich has an article planned for Internet Telephony Magazine highlighting Sangoma. Obviously, Sangoma has been riding the "hockey stick curve" of Asterisk, which has been dramatically boosting Sangoma's revenue. (they sell other hardware as well)

Then you have OpenVox, a company based in China offering "Digium-cloned" hardware. They use the same hardware reference design that Digium uses. In fact, they look nearly identical. While they also suffer from the same hardware interrupt issues as Digium hardware, they're 20% cheaper - or more. OpenVox was probably the first Digium clone and I believe is the largest. Similarly, another Chinese-based company, ZAPMICRO is also offering Digium-cloned hardware. Then you have cyLogistics, a great online VoIP store, offering OpenVox hardware, as well as Sangoma, Aculab, PIKA, and Digium. I heard through the open source grapevine that Digium is refusing to allow any distributors to carry Digium hardware if they sell OpenVox cloned hardware. But apparently cyLogistics either has a "pass" from Digium or they're skirting the "ban" by purchasing Digum hardware through other resellers. I've heard from my other sources as well that they aren't happy that Digium is forcing distributors to carry Digium Asterisk hardware exclusively.

I say all this to ponder Digium's future. Will the open source Asterisk community have "brand loyalty" to Digium, since Mark Spencer founded the whole Asterisk movement? Or will the open source community, which is notoriously "fickle" when it comes to price choose the least expensive hardware that just plain works? T1/E1 and analog cards that work on Asterisk are becoming commoditized, so if Digium doesn't sell their telephony cards, where does that leave them? They can make revenue on the Asterisk Appliance once it ships, but their core revenue right now is from their telephony cards.

Let me say that I personally like Digium and especially Mark Spencer. I'm met Mark a few times and he even took me out for "linner" (lunch/dinner) at TMC's Internet Telephony Conference & Expo. I want Digium to succeed because it will only help further grow the Asterisk community. Even though a healthy ecosystem of third-party Asterisk-based PBXs now exists and there is still a strong open source community helping to drive the open source Asterisk code, Digium is still Asterisk's champion. Losing Digium to under-priced Chinese cloned hardware or even to tough competition from Rhino or Sangoma would be a tough pill for me to swallow.

If you were to ask me which hardware I would use in an Asterisk solution, well if I think with my "open source loving heart", my choice would be Digium. However if I think with my technical CTO brain, my recommendation would have to be to install Sangoma hardware. It has some key advantages over Digium & Digium cloned hardware, including lower cost, better scalability, and better interrupt handling for a trouble-free Asterisk installation. In fact, you can install a single PCI or PCIx card and then attach daughter cards to it that don’t use a PCI slot and share the same interrupt. And so I ask you open source community and Asterisk-based dealers/resellers/end-users, etc., what are your thoughts on Digium's future? The comment lines are open.

With PIKA Technologies's latest version of MonteCarlo SDK, PIKA Technologies new software development kit speeds development of predictive dialer applications. Since PIKA's boards work on Asterisk, in theory, this SDK could be used to develop predictive dialer applications on the open source Asterisk platform. Having predictive dialer functionality on Asterisk is one of the features sorely lacking in Asterisk, which makes Asterisk not suited to call center environments that require this functionality.

So in theory, PIKA's SDK could add predictive dialer functionality to Asterisk - but more on that later - first the news. Today, PIKA Technologies announced it can help application developers more speedily build predictive calling applications for call centers thanks to today's release of its latest MonteCarlo software development kit. The SDK includes updates to GrandPrix, PIKA's high-level application programming interface, that decreases time to market for developers by making the design of their application quicker and easier.

The GrandPrix API is designed to work directly with low-level APIs and provides mechanisms that allow applications to access the finer control provided by these low-level APIs. In addition to the call-progress analysis function included in this latest release, GrandPrix provides an abstraction of call signaling in analog, digital and IP, and call control for SIP, ISDN, CAS and analog with Caller-ID.

Many modern call centers rely on predictive dialers to place calls. Predictive dialers are computerized systems that automatically dial batches of numbers while using algorithms to determine availability of agents and calls answered, then adjusting dialing patterns based on real-time data.

Call-progress analysis is of vital importance to contact centers using predictive dialers. If a call placed by a predictive dialer is answered, the determination of whether it is a live person or an answering machine on the other end of the call must happen swiftly so that the appropriate action may be taken - a call is transferred to an agent, a message is left on the answering machine or the call is disconnected and redialed at a later time.

PIKA's customers know the value that call progress analysis adds to their solutions. "We wanted to create a predictive dialer that was easy to install and to use," said Richard Hardgrave, President, Electronic Voice Services. "While there are a lot of predictive dialers on the market, most are very complex to deal with. The problem is that most predictive dialers have a long lag time. A recipient picks up the phone, says "hello," but doesn't hear anything for a few seconds. This gets people's guards up because they suspect they're being called by another telemarketer."

Call-progress analysis can solve this problem and provides distinct competitive advantage for application developers.

In Frost and Sullivan's "World Outbound Dialing Markets," analyst Seema Lall predicted the market will reach $204.9 million by 2011 and emphasized the competitive advantage offered by top-tier dialing solutions. "Essentially, the outbound dialing products will afford a means for providing superb customer care, which becomes a competitive differentiator for the service-oriented culture," said Lall.

Predictive-dialer developers see PIKA solutions as a competitive advantage as they are assured excellent call-progress detection and call-analysis features in their applications.

"PIKA allows developers to modify certain parameters, such as the number of words in a greeting, which enable predictive dialer applications to achieve a higher accuracy," said PIKA Technologies field application engineer Cindy Xu. "We are continuously improving DSP and host-based algorithms so that developers have the proper tools to create the best solutions for their call center customers. We have performed significant and extensive testing in-house and feel that the tools that we provide predictive dialing system developers are equal to or better than others on the market."

Getting back to my point about adding predictive dialing functionality to Asterisk, I asked PIKA about this possibility. I asked, "Can this SDK be used on Asterisk to build a predictive dialer application on Asterisk? I know that is one thing lacking in Asterisk."

PIKA's representative responded, "It looks like that will indeed be the case with the next release of PIKA Connect for Asterisk, slated for the Fall." The PIKA representative continued, "The SDK, MonteCarlo GrandPrix, works with Asterisk but this particular feature hasn't been implemented in our channel drivers yet. We need to take the call progress analysis in GrandPrix and "plug in" to Asterisk's call progress analysis for this to work."

So this is great news for Asterisk! Not sure how easy it will be to develop a predictive dialer application using PIKA's SDK, or if the open source community will embrace it, but certainly this is good start. Traditional predictive dialer companies must be quaking in their boots at the thought that Asterisk, known as an open source IP-PBX, could one day be an open source predictive dialer as well.

I discovered this interesting Asterisk termination post saved as "unpublished" dated 6-20-2005. I must have put it aside to work on some other projects. I thought I would publish it now since it still has some pertinent thoughts. Reading my article, I couldn't help but notice that it is two years later and there still isn't a sizable intra-enterprise VoIP peering network that I am aware of (with one exception - VPF). As I discuss in my thoughts from 2 years ago, I was hoping for a sort of P2P IP-PBX model where an IP-PBX from Company A communications with IP-PBX from Company B to initiate an outbound call at Company B's local calling rates. The other scenario is that you could simply initiate a call from Company A to an extension at Company B which travels over IP. In either scenario you can bypass ITSPs or the need for VoIP gateways entirely. One option is for Company A to "peer" directly with Company B, by contacting them and configuring some call routing settings. However, Company A would have to contact several companies to peer with before realizing any cost savings.

The other option as I mention below, you would require some sort of trusted third-party to act as a go-between and to centrally organize all the various peers to reach the critical mass needed for real phone cost savings. ENUM is supposed to help with that, but the carriers aren't exactly rushing to offer ENUM and certainly not "free" ENUM services.. One interesting ENUM registry is from the Voice Peering Fabric (VPF). The ENUM registry is based on the IETF (RFC 3761) standard which maps telephone numbers to Internet (URL) addresses and uses a look-up architecture based on DNS They built their own ENUM registry, which is a multilateral peering service that allows organizations to send and receive calls among members directly, IP end-to-end, for no termination fee, including no cost to register numbers or querying the registry. It's free. Let me repeat that - it's FREE! Querying the ENUM registry is free and so is terminating a call to another VPF customer. Thus, as the VPF adds more corporations to their customer list, this also increases the number of FREE calls you can make. Kudos to the VPF which isn't waiting around for public ENUM to finally take off. I expounded the benefits of ENUM in my "ENUM ENUM ENUM!" post, which is a good refresher on ENUM and I compare public ENUM registries versus private ENUM registries.

Ok, without further adieu here's the post I started 2 years ago. Enjoy...

Asterisk has had the ability to call other Asterisk PBXs for terminating calls over an IP connection for quite some time. Thus, if you have multiple branch offices all with Asterisk PBXs, you can terminate calls over the IP connection for free.

Hunter Newby over at Telx and I discussed how it would be very easy for Asterisk PBX users to join in a massive Asterisk community and "share" their connection and barter/exchange minutes. Let's call it "enterprise peering". It's actually a form of "peer-to-peer enterprise telephony" actually. In theory, you can get a "cut" of the revenue for the PSTN minutes that call out of your Asterisk PBX. This is a scary concept since eventually all enterprises can "peer" with other enterprises and essentially negate the need for the PSTN altogether - another nightmare for the phone companies caused by VoIP.

There are two scenarios when peering between two corporate IP-PBXs. The first scenario is where an IP-PBX from Company A communications with IP-PBX from Company B to initiate an outbound PSTN call at Company B's local calling rates. Company B charges Company A for terminating the call. The second scenario is that you could simply initiate a call from Company A to an extension at Company B which travels over IP. Since it's all over IP, the call is free.

Of course, the Asterisk system uses its proprietary IAX protocol for inter-Asterisk communication and not standard SIP, so unless IAX is supported by all IP-PBXs (not going to happen), this particular "doomsday scenario" for the phone companies may just be a dream. It also requires that each Asterisk IP-PBX trust other Asterisk IP-PBXs to not abuse or max out their limited outbound PSTN resources.

Just imagine if the SIP protocol matures to the point where you can securely "peer" with other SIP-based IP-PBXs and some sort of clearinghouse takes care of bartering minutes, revenue exchange, etc. Of course, you could bypass a clearinghouse altogether and just let outside SIP IP-PBXs dial out over IP through your PSTN connection free of charge, but there would have to be some level of "trust" to allow this to prevent abuse.

In theory, you could charge outside IP-PBXs only after a certain usage criteria has been met, but that opens the door to fraudulent billing practices. Company A could say that Company B owes it $500 for making SIP calls over its PSTN lines. That's why an independent third-party clearinghouse would be needed to prevent billing fraud. In theory, with a large enough "trusted" clearinghouse you can join this clearinghouse network and it would provide the least cost routing and calculate what revenue you are owed by those that terminate calls on your PSTN lines. Essentially, using your corporate IP-PBX you become your own little phone company making your corporate IP-PBX a revenue source instead of a liability.

In a very similar fashion to this idea, it appears one company called AsteriskOut is using multiple Asterisk PBXs for termination. I was perusing the VoIP Forums and came across this Asterisk thread in the VoIP Forums where a company is leveraging Asterisk to terminate VoIP calls, but it doesn't appear they are building any sort of peering model. AsteriskOut has some decent rates, including just $0.016 per minute for U.S. termination.

Is enterprise-to-enterprise IP-PBX peering just a dream? With the help of a large enough and "free" ENUM database it is certainly possible. Or perhaps with Asterisk's continued growth, the open source community will create a popular Asterisk ENUM (AENUM?) registry of their own which will reach critical mass and even cause other IP-PBXs to join. Only time will tell.

What happens when a VoIP blog (yours truly) writes about the fact that a former Nortel subsidiary (Blade Network Technologies) went looking for a new phone system, chose an open-source Asterisk-based solution from Fonality instead of using Nortel's own PBX and then agreed to go on record on the VoIP & Gadgets blog about why they made such a shocking decision?

A) Nothing - it's a VoIP blog - who cares? Nortel is an $11 billion dollar company that certainly doesn't read blogs for their news.
B) Nortel reads the blog post, is a little peeved, but other than some emails sent internally, no one outside Nortel would ever know they were annoyed.
C) A Nortel Board Member flips out over the article, contacts Blade and then pressures Blade to return the Fonality system and have Fonality print a retraction to the blog article (and the subsequent press release).

If you answered C) congratulations, we have a winner! Yes, it's true - and in true David (Fonality) vs. Goliath (Nortel) fashion it would appear that we have Nortel peeved that one of their former subsidiaries chose an open-source IP-PBX (PBXtra from Fonality) and who had the audacity to speak to the press about why they made such a decision. Why, the nerve!

Although they are a former subsidiary of Nortel, I should point out that according to InternetNews.com, Nortel still has a minority interest in Blade. Interestingly, according to Blade's website, Eric Schoch, the Vice President of Business Development for Nortel, serves on Blade's board of directors.

As if we needed any more proof of the power that the blogosphere holds, the fact that a Nortel executive took exception to my blog post and contacted Blade to apply pressure is almost beyond belief. I spoke with Fonality's CEO, Chris Lyman to get the exact facts from his perspective. Here is my phone conversation with him transcribed:

Chris explained, "Fonality sells it's PBXtra IP PBX to Blade. During the sale, Blade's Director of IT, Amon Prasad agrees to go on record in a Fonality press release about why he made this decision against his parent company. Then you (Tom) decided to do a story about such an interesting customer win."

Chris continues, "Here is where it gets interesting. The day after your story hits the net, Fonality gets a panicked call from Stefan Zuckut, Blade’s Vice President of Corporate Development. Stefan tells us that a board member from Nortel read your (Tom Keating) blog and hit the roof. "

Chris explains, “Well, Stefan must have been telling the truth, because 30 minutes later, the CEO of Blade, Vikram Mehta, is demanding to speak with me personally. I acquiesce. When Vikram gets me on the phone he immediately tells me that Blade is going to return their brand-new, still-in-the-box phone system. I ask, 'Why are you returning?'" To which Vikram replies: “We changed our mind.’

“You can’t change your mind. That’s not how our terms and conditions work.” says Chris, and then continued, "The next thing out of the Blade executive’s mouth astonishes me “We also want a retraction of your press release about how the Fonality system was more affordable and easier to use than Nortel.”

Stunned, Chris asks, “Why?”

“Because you didn’t follow our internal process for authorizing a press release.”

“But it is *your* internal process, and we spoke, with permission, to your own Director of IT, who personally signed off on the release. All that press release said and all Tom Keating's article said was that we're less expensive and easier to use than Nortel. What you want me to publish a document that we're more expensive than Nortel and harder to use? How the heck do you expect me to print a retraction for something that is a) true and b) out of my control now that it is in the blogosphere?”

I interrupted Chris's retelling of the conversation with Vikram and asked Chris, "How long have they had PBXtra for?"

Incredulously, Chris responds, "They haven't even installed it yet. It's still in the box."

I responded, "Right. I recall when speaking to Blade's Director of IT last week that he explained Blade hadn't deployed it yet, but were excited to do so."

After Vikram pressed Chris for a retraction, Chris in an irate mood, issued an ultimatum - "You have 60 seconds to change your mind. If within the next 60s you don't change your mind, you're going to discover how an incident turns into a scandal. Because either way, any reporter worth his salt is going to find the fact that Nortel's board of directors is putting pressure on you to get us to print a retraction about the comparative price of our systems much more juicy than the stupid article in the first place. This is getting better all the time and it's good for us and bad for you."

Chris told me they got so upset they hung up on him before he got five seconds into his count. I asked who was on the call and Chris said it was Vikram Mehta, Blade’s CEO, Stefan Zuckut, their VP of Corporate Development and Jim Sladek, VP of Finance.

Chris told me, "That's when I called you Tom on Friday to tell you about what was happening."

Chris, "Then I told Arnold, my SVP to play good cop. Call them back and tell Blade that my CEO (that’s me) is chain dialing the press. All they have to do is call off the dogs about us printing a retraction and returning the system and everything will be fine. We won't even talk about the little incident."

According to Chris, "Blade came back within an hour and changed their mind about returning - agreeing to keep it and be friends and we're sorry. Everything was lovely. That was all on Friday. Well, I came in this Monday morning and they've again decided to return the unopened system. The reason quoted in the RMA ticket is: “Poor customer service, Arnold Waldstein.” Chris laughed, "Arnold is our SVP of Marketing. Ha!"

I asked Chris, "So they let me get this straight. They changed their mind a second time? They flipped on the flip-flop?"

Chris acknowledged, "Yes, they re-changed their mind again. This is a bit of a watershed event when big big iron like Nortel calls Blade to pressure little ole' Fonality into printing a retraction of what is absolutely the truth from inside their own company. And that company goes crazy at the executive level and makes 3 decisions within 24 hours - to return it, then to not return it, then to return it again."

I contacted Blade's Stefan Zuckut their VP of Corporate Development, to get their perspective. I explained who I was and said, "I was the one who wrote the Blade & Fonality story and I was wondering if I could have your comments on what transpired between yourself, Fonality and Nortel?"

Stefan replied, "I cannot comment on that, but I'd be happy to have our CEO give you a call." I then asked, "What's you're CEO's name?" and Stefan said, "Vikram Mehta, but he's in a meeting right now, is there a way of reaching you?"

I gave my contact information, but no one called me back so I called back a few hours later and this was my conversation with the Blade's CEO:

Tom:  Hi this is Tom Keating with Technology Marketing Corporation. I was the one who wrote the Blade & Fonality story and I was wondering if I could have your comments on what transpired between yourself, Fonality and Nortel?

Vikram: <pause> I'm not quite sure I know what you're talking about.

Tom: I spoke with Chris Lyman and he mentioned that your Director of IT was pretty pleased with the Fonality system and I was wondering what changed over the weekend as far as far as why you are no longer using their solution?

Vikram: <long pause. sighs> I'm not... First of all, I'm a little concerned about getting into details about what we do on a day-to-day basis and our commercial relationships with Fonality and Nortel are in confidence, so...

Tom: So basically you're answer as to why you decided to return Fonality is -- "no comment"?

Vikram: I didn't say that. You said that.

Tom: Well I'm just trying to understand why your Director of IT very much loved the Fonality solution and then as the result of my article resulted in you guys changing your mind.

Vikram: <again repeating himself> Like I said, our relationship with our suppliers and customers are commercial-in-confidence and I don't want to be getting into any details about what we're doing with Nortel and likewise I don't want to be getting into what we are doing with Fonality. That is my comment.

Tom: Well Fonality gave me the green light to talk to you. They were very open and honest as far as their position so I'm just trying to get what your take on it is.

Vikram: I can't speak for what Fonality told you and what they did not, but like I said my position is that our relationships with our customers and suppliers are commercial-in-confidence.

Tom: Chris gave me the green light to speak about the relationship with you. He mentioned you spoke with a Nortel board member. If you let me know the contact information at Nortel, I can speak with them and find out if they would be willing to give me the green light as well.

Vikram: Our relationships, like I said, with customers and suppliers are commercial-in-confidence. I am not at liberty to divulge what we are doing with anybody.

Tom: That's fine. So if you like I can contact Nortel directly and get their perspective and they can either comment or no comment. If you just give me the person to speak with I can get their position.

Vikram: Don't know that I can point to... uhhh. Don't you have public relations contacts at Nortel?

Tom: Yes I do, but it's a big company, but I don't know who exactly you spoke with at Nortel, so I wouldn't know who to refer to. I would need a specific person for me to contact their PR firm about. So is there a specific person on the board at Nortel you spoke with?

Vikram: Like I said, our relationships with our customers and suppliers are commercial-in-confidence. I cannot divulge anything about our commercial relationship with our customers and suppliers.

Tom: But I'm not asking you to divulge the relationship, I'm just asking to speak to Nortel.

Vikram: Go right ahead.

Tom: But who do I speak with?

Vikram: I wish I could help you.

Tom: Ok, so did Nortel offer anything in exchange for you returning Fonality's PBX. Was there a quid pro quo?

Vikram: I don't even know what you're talking about.

Tom: According to Chris Lyman you said you were returning the Fonality system.

Vikram: Like I said, our relationships with our customers and suppliers are commercial-in-confidence and I'm not in a position to divulge anything about our commercial relationships with anybody. Whether that is Fonality, Nortel, other customers, suppliers who we do business with. I'm not in position to speak about any of that stuff with the press.

We spoke a little more, but as you can tell, I was getting nowhere with Vikram. However what "wasn't said" spoke volumes -- both from his demeanor and his avoiding answering my questions, in my mind confirmed what Chris said was accurate. I then contacted Nortel to get their perspective. I spoke with a Nortel employee who wishes to remain anonymous. He stated that Eric Schoch, the Nortel board member was travelling and therefore wasn't able to get him to respond.

The employee did however admit that he was aware that Eric sent Vikram (CEO of Blade) a note about the Fonality press release where it simply stated "I would appreciate seeing copies of any news releases that have our name 'Nortel' in it before they go out." The Nortel official explained, "Anything that uses our trademark name we like to take a look at it." The employee added that he was not aware of any pressure applied by Nortel to have Blade reverse their decision on selecting Fonality or forcing a retraction.

Let's tally the score to try and figure out who is lying and who is telling the truth. Chris Lyman and Arnold Waldstein from Fonality claim that Vikram called them and stated unequivocally that a Nortel board member was very upset over the blog article and press release and as a result Blade demanded a retraction and a return. When I contacted Vikram at Blade, he was evasive, wouldn't set the record straight, and simply hid behind "customer-in-confidence".

Finally, I contacted Nortel, and although the board member in question was not available for comment, the Nortel representative admitted that an email from Nortel's Eric Schoch was sent to Vikram. Nortel claims however they were simply upset that Nortel was not consulted for approval in the issuance of the press release. So let me get this straight - this whole story is simply about not giving Nortel a heads-up on a press release wherein Nortel doesn't exactly come out smelling like a rose resulting in Blade demanding a retraction and returning Fonality's PBXtra? C'mon!

Just to put this all in perspective, according to the latest research I've seen, Nortel's market share has declined from 29% in the fourth quarter of 2004 to 17% in the third quarter of 2005, according to Merrill. With Nortel's slipping marketshare is the new way of retaining marketshare by strong-arming or pressuring open source PBX rivals from telling their successful stories to the press/media including blogs? While I cannot confirm this is the case with 100% certainty, it certainly is quite suspicious. As Chris stated, this is indeed a watershed event for open source telephony.

Seems like Digium, makers of the Linux-based open source Asterisk IP-PBX, draws a huge crowd at every tradeshow, every keynote, and every exhibit booth. I remember the last ITEXPO they were demoing their new AsteriskNOW in the booth using a large TV flat-screen monitor and it was like 5 people deep by 12 people wide. So I wasn't surprised when Rich Tehrani shared a video and some photos from Communications Developer showing Digium's well-attended keynote. Once again, Digium proves itself as a real headliner attraction on the tradeshow circuit!

Rich Tehrani said, "Digium/Asterisk's Kevin Fleming is keynoting TMC's Communications Developer Conference and kicked off his presentation by dispelling the myth that Asterisk is only an open-source PBX. In fact he says it is much much more. Contributors to Asterisk do not come from Greenland or Antarctica but according to Kevin they come from everywhere else."

Check out the video and the photos taken from Rich's blog entry...


click Play button above to play video

Here's an exclusive sneak peek of a TMC Labs review of the SIP-based pbxnsip CS 410 IP-PBX appliance, which is scheduled for review within Internet Telephony Magazine. As hinted at within the review, the trend towards low-cost IP-PBX appliances <$1000 is quite apparent, which is good news for the SMB market looking for a cost-effective phone system replacement that has advanced VoIP functionality. Enjoy the review! And be sure to check out my full review of the pbxnsip standalone software, which runs on Windows and Linux.

pbxnsip 1600 Osgood St Bldg 20 Suite 223 North Andover MA 01845 Ph: 978-746-2777 Web site: http://www.pbxnsip.com Price: $999

RATINGS (0–5) Installation: 5 Documentation: 4.5 Features: 4.75 GUI: 4.75 Overall: A

TMC Labs got an exclusive peek at pbxnsip's CS 410 IP-PBX "all-in-one" appliance, which features a “mini” Session Border Controller(SBC) , built-in 4 analog (FXO) PSTN ports, voicemail & auto attendant, as well as support for up to 10 SIP-based IP stations (hardphones, softphones) and supports up to 10 simultaneous calls. The CS 410 actually includes all available features from the other pbxnsip PBX editions which run on Windows or Linux boxes. This includes standard features like voicemail, auto attendant or conferencing, but also advanced features like call barge in and call forking to cell phones. Like other pbxnsip versions, the CS 410 also supports advanced routing functions such as paging groups, hunt groups, and agent groups.

The pbxnsip appliance reminds us of two other popular Linux-based IP-PBX appliances in the VoIP industry, namely Digium's Asterisk Appliance and Fonality's trixbox appliance. Just comparing the appliances based on price alone, the trixbox appliance (4 FXO ports) is $1499, and the Asterisk Appliance (4 FXS, 2 FXO) is $2195, while the CS 410 comes in at just $999. Of course, each of the appliances has features the other two don't, so it really depends on what features you need. Even with the lowest cost, the CS 410 doesn't skimp on features and unlike the other two, it features integration with Microsoft Exchange Server 2007 UM.

Targeting small-to-medium businesses (SMBs), the CS 410 is a solid-state appliance device with no moving parts, no fans, which results in very minimal heat to ensure long-term reliability of this phone system. In fact, we noticed very minimal heat when we touched the unit's plastic casing. TMC Labs took the CS 410 for a spin and were very impressed with its easy plug-and-play installation and ease-to-use web admin.

In lieu of a hard drive, the CS 410 appliance sports 256MB of Flash and 128MB of RAM running Debian Linux. This only gives you about 1 hour of voicemail storage, however, you can easily get around this 1 hour voicemail storage limit by leveraging Microsoft's Unified Messaging capabilities in Exchange 2007. The CS 410 can send all the voicemail to the Exchange 2007 Server. The CS 410 is unique in the industry in that it is the first IP-PBX appliance to directly integrate with Microsoft Exchange Server 2007 leveraging SIP. The latest version of Microsoft's Exchange Server features built-in unified messaging capabilities, including voicemail storage/playback, text-to-speech reading of email/calendar, and more. Used in conjunction with each other, you can achieve a rich user experience. For example, we were able to dial into the CS 410 and then enter our extension, followed by * and then our PIN to logon to our personal mailbox. From that point we could playback voicemail, but also email using text-to-speech (TTS). Similarly, we also had remote access to our calendar which also leverages TTS.

We setup some Exchange 2007 extensions and configured the CS 410 to transfer these extensions to the Exchange Server (screenshot above). We simply had to configure a new trunk on the CS 410, set the trunk to “SIP Gateway” ,and point the IP address to connect to the Exchange Server. On the CS 410 we also had to define a new dial plan for routing incoming calls to the Exchange Server's built-in SIP gateway, which would then accept the call and play the extension's outgoing message. On the Exchange Server side we also had to create a new dial plan and configure the new Unified Messaging IP gateway (the CS 410's IP address or FQDN). Finally, we had to active Unified Messaging for each Exchange mailbox and assign an extension number and a PIN. Voila! We now had fully integrated unified messaging for each of the CS 410 extensions that we created.

Some other nice features of the CS 410 include paging and music on hold audio connectors. In fact, the system can also send RTP multicast traffic and use multicast-enabled devices for office audio paging, such as the snom 370 IP phone. Another security-related feature is full support for TLS and SRTP to secure the voice. pbxnsip mentioned to TMC Labs that they are thinking about adding a built-in firewall/router so they could support TOS tagging to ensure QoS for time-sensitive voice packets. Since the unit runs on Linux, they stated it wouldn't be too hard to add that functionality. The CS 410 also supports a built-in conference bridge, which is great for SMBs looking to save money by avoiding paid hosted conferencing services.

Another feature worth noticing is the agent waiting queues. You can record up to ten announcements and have the music on hold mix in these announcements. The queues also feature agent recovery time, call pickup from queue, call escalation, day/night mode, holidays, and web-based queue status display.

Other features:
- Plug and Play of popular IP phones including Polycom, snom, Aastra, and Cisco/Linksys.
- Web admin as well as SSH access.
- SNMP support.
- Voicemail triggers call to cell phone
- ENUM
- DID
- Dial by name
- Message Waiting Indication (MWI) support
- Call park, call pickup, call retrieve
- Last call return, redial
- Caller-ID blocking
- CDR export through SOAP interface
- Call Supervision: Call barge in, Training mode, Listen in

Conclusion
We really liked the Exchange 2007 voicemail integration. Even without Exchange Server 2007 integration, you can still have the system send you voicemail notification via email. We liked how this appliance integrates the analog telephone lines within the appliance without the need for a separate PC with telephony cards or a separate PSTN gateway. The dial plans are especially powerful and easy to configure. The dial plans allow SMBs to take advantage of VoIP’s low per-minute costs, while also giving SMBs the option to utilize their existing phone lines. For instance, local calls can be terminated through the FXO lines while international calls can be sent to an ITSP using VoIP. It's important to note that it has built-in session border controller functionality for connecting with remote offices. The CS 410 which supports 10 IP endpoints can also be easily upgraded to a CS 425 (simply by changing the license key), which supports 25 registrations and 15 concurrent calls. Overall, TMC Labs was quite impressed with pbxnsip's CS 410, which was easy to configure, a breeze to administer, and sports more features than IP-PBXs that cost four times as much.

TMCnet recently launched an Open Source PBX community site where you can check out the latest news in the open source PBX industry. It also features whitepapers, tutorials, open source PBX resources, and more. Disclaimer - this community site is sponsored by Sangoma, a provider of Asterisk-based telephony hardware, so some of the content is specifically geared towards Sangoma. However, there is lots of useful content that is not Sangoma-specific, including open source news and TMCnet featured articles. Click here to check it out. Also, if you are interested in other open-source community sites, check out TMCnet's Asterisk forum as well as Digiums' own well-attended forums.

Garrett Smith informed me that Sayers Media Group tomorrow will launch PBX Prompts. What is PBX Prompts? It's pretty much what it sounds like. PBX Prompts offers a variety of standard voice prompt packages for Asterisk and other open-source systems such as Fonality, SwitchVox, and Pingtel, as well as custom voice prompts for IVR or voicemail. So why would you need prompts for Asterisk and these other solutions when they already come pre-loaded with prompts? Well, according to PBX Prompts, "Based on our experience working with a number of Asterisk resellers, integrators, and consultants, we found that there was a need for highly quality direct replacements for the standard Asterisk voice prompts. Out of this feedback, we have created PBX Prompts."

Recorded in professional sound studios, PBX Prompts’ standard, advanced, and custom voice prompts has no minimums and a quick 72 hour turnaround. "We are currently launching with full sets of voice prompts for Asterisk based phone systems in Spanish and English (NA & UK), in both male and female voices. We will soon be launching prompts in French, and over the next few weeks, we will be launching prompts packages specifically for Fonality, SwitchVox, Pingtel, and other phone systems. In addition to our standard packages, we also offer custom packages, for companies in need of non-standard prompt recordings, such as auto-attendant and voice mail greetings."

Wow, high-quality voice prompts on open-source solutions like Asterisk? First the Asterisk gurus groaned over adding a nice professional GUI to Asterisk (instead of staying with the geeky command line) and now professional prompts for Asterisk? Before you know it, installing a professional-grade version of Asterisk will be so simple anyone can do it. Heck, soon Asterisk will be so simple and easy to install, you can even install it on just about anything - say an on an Apple TV or on a Linksys router. Oh wait, that's been done already.

“Over the past two years, we have heard over and over again about the difficulties many small medium businesses and value added resellers have had finding high quality professional voice prompts for Asterisk Open Source PBX systems,” said Garrett Smith, Director of Sales and Marketing for Sayers Media Group. “Based on these experiences we have created PBX Prompts with the help of these very same companies in order to deliver on a simple, easy to use, ordering interface and installation process for those who want an alternative to the default Asterisk voice for their phone system.”

PBX Prompts is currently offering voice prompts packages for Asterisk systems featuring male and female voice talents in English (NA), English (UK), and Spanish languages. PBX Prompts plans on launching additional languages, such as French, German, and Japanese, in the coming weeks. Prices for the current voice prompt packages for Asterisk systems range from $49.99 for select standard voice prompt packages for Asterisk systems featuring over 500 voice prompts, to $129.99 for advanced voice prompt sets for Asterisk systems that feature over 600 voice prompts. From now until May 1st, PBX Prompts is also offering FREE voice prompt packages for Asterisk systems that contain 100 of the most popular voice prompts for Asterisk.

My Apple on Asterisk blog post immediately drew a response from several Asterisk fans, including the person that "hacked" Apple TV to run Asterisk (l0rdrock). I emailed l0rdr0ck and Steven Sokol about how to get Asterisk running on Apple TV and they provided me with the information on how to do this. The Asterisk 1.4.2 on AppleTV "soup to nuts" tutorial is courtesy of Steven Sokol from Sokol & Associates, Inc., who was the initiator behind this project when he proposed a "bounty" on loading Asterisk on Apple TV and which successfully won by Jeff Gambera (aka l0rdr0ck). The information below is a combination of material from www.appletvhacks.net (for the SSH stuff) and emailed information from l0rdr0ck and Steven Sokol for the main tutorial on loading Asterisk on Apple TV.

Ok, let the fun commence.

Installing Asterisk on Apple TV Steps:

First, get Apple TV. (obviously)

Enable ssh by using this tutorial:

http://www.appletvhacks.net/2007/03/24/enable-ssh-and-afp-on-your-apple-tv/
Since AppleTVHacks has been "dugg" to death, I am including the SSH instructions below taken from the Google Cache. The images weren't cached, so they may not appear below until AppleTVHacks go back online. Credit for the original information used in making this how-to goes to TylerL82 over at the SomethingAwful.com forums. Written up here by Jonathan Bare.

So while you have your Apple TV open and you’re installing a new hard drive or just following along with the people over at SomethingAwful.com forums to get Xvid working, you might as well un-break SSH so you can access the Apple TV remotely.

Opening the Apple TV and connecting the hard drive to your Intel Mac are covered in the hard drive upgrade and elsewhere, so we’ll skip those steps and jump right to the point where you have the OSBoot and Media volumes mounted on your Mac.

This process assumes using an Intel Mac because the sshd binary may or may not be the same in the PowerPC version of Mac OS X; we haven’t checked. If someone would like to try using the PowerPC binary and let us know if it works, that would be great!

There are 2 ways to go about this; using the finder, or via the terminal. The finder method is probably best if you aren’t sure what is going on, whereas the Terminal method is quicker if you are confident.

Step 1 - Using the Finder
Note: ignore any missing images since the Google Cache didn't have them

Once you have OSBoot mounted in the Finder, double click on its icon. You’ll see something like this:



From the Finder’s Go menu, select Go To Folder, or press Command-Shift-G. Type in /Volumes/OSBoot/usr/sbin/ and click Go.



Open a new Finder window and do the same thing to go to a folder, this time, however, select /usr/sbin/. This is the same folder located on your Mac’s hard drive.



From the Mac’s sbin folder, drag the file “sshd” to the Apple TV’s sbin folder. Your cursor will change in to the green plus sign to indicate that you are making a copy of the file.

Note: If you have accidentally ended up in /usr/bin instead of /usr/sbin and you accidentally copy the “ssh” file instead of “sshd”, you’ve made a big mistake and ssh will not work on your Apple TV. Be sure you are working to and from the respective /usr/sbin directories.



You now have sshd installed on your OSBoot volume. Now we need sshd to start when the Apple TV boots up.

In the OSBoot window, choose Go To Folder again and this time, enter /Volumes/OSBoot/System/Library/LaunchDaemons/.



If you open the Apple TV’s ssh.plist file from the LaunchDaemons directory, you’ll see why we need to replace it. Apple left a dummy plist file to throw us off the trail.



You could copy the ssh.plist file from the same location in your Mac’s /System/Library/LaunchDaemons/, but for simplicity, we’ve included the contents of the plist here. Using TextEdit, BBEdit, or your favorite text editor, copy and paste this text, replacing the original ssh.plist contents:

<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE plist PUBLIC "-//Apple Computer//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
<plist version="1.0">
<dict>
        <key>Label</key>
        <string>com.openssh.sshd</string>
        <key>Program</key>
        <string>/usr/libexec/sshd-keygen-wrapper</string>
        <key>ProgramArguments</key>
        <array>
                <string>/usr/sbin/sshd</string>
                <string>-i</string>
        </array>
        <key>SessionCreate</key>
        <true/>
        <key>Sockets</key>
        <dict>
                <key>Listeners</key>
                <dict>
                        <key>Bonjour</key>
                        <array>
                                <string>ssh</string>
                                <string>sftp-ssh</string>
                        </array>
                        <key>SockServiceName</key>
                        <string>ssh</string>
                </dict>
        </dict>
        <key>StandardErrorPath</key>
        <string>/dev/null</string>
        <key>inetdCompatibility</key>
        <dict>
                <key>Wait</key>
                <false/>
        </dict>
</dict>
</plist>

Close the file and save it. In a nut shell, the ssh daemon is activated by launchd whenever an incoming connection on port 22 is detected.

Step 1 - Using the Terminal
Firstly, copy the sshd binary from your Mac to the AppleTV drive:

cp /usr/sbin/sshd /Volumes/OSBoot/usr/sbin/.

Next, copy the ssh.plist file over, so launchd knows to start sshd on boot:

cp /System/Library/LaunchDaemons/ssh.plist
/Volumes/OSBoot/System/Library/LaunchDaemons/ssh.plist

(The above is all on 1 line - I hit enter so the above line didn't wrap into the middle column of my blog)

Alternatively, you could use the contents of the file from above to create an ssh.plist file.

Step 2
Unmount the OSBoot and Media volumes, reassemble your Apple TV, and power it on. You need to get the IP address of the Apple TV from the Settings menu, once you know that, open a Terminal window from your Mac (or any OS that has ssh installed) and type:

ssh frontrow@your.apple.tv.ip.address

Press return. Type “yes” when it asks if you want to permanently store the key. Then enter “frontrow” as the password.

You’re in.

Now What?
SSH opens up a lot of doors to accessing the Apple TV. For example, you can now type sudo /usr/sbin/AppleFileServer and cause the built-in Apple File Protocol (AFP) server to start. Then you can connect to the Apple TV from your Mac by using the Connect to Server command in the Finder. AFP, SCP or SFTP can now be used to copy files to the Apple TV!!

You can type ps auxww to see a list of running processes on your Apple TV. (Hey, what is /usr/bin/ripstop and why is it running under the frontrow user?)

One thing we’ve already learned from SSH is that the root filesystem on the Apple TV (aka OSBoot) is apparently mounted as a read-only filesystem. That means it’s not initially possible to copy any files to the root mount point (/etc, /usr, /var, for example) while it is running. TylerL82 has commented to point out you can remedy this with the sudo mount -uw / command.

It’s also important to note that any changes you make to the OSBoot volume will be wiped out by a Factory Restore of the Apple TV. It appears that a disk image of the stock Apple TV operating system exists on the unlabeled partition on the drive and is used to restore the OSBoot volume when Factory Restore is selected.
 ---
 ssh into the appletv using the tutorial above. everything below
 will be through ssh:

 ----
 Download Bison 2.3 from:
 http://ftp.gnu.org/gnu/bison/bison-2.3.tar.gz

 (use curl via ssh)

 Decompress, open terminal, do:

 ./configure
 make
 sudo make install

 ------

 get wget:

 http://ftp.wayne.edu/pub/gnu/wget/wget-1.10.2.tar.gz

 decompress, terminal:

 ./configure
 make
 sudo make install


 close terminal.app and reopen
 ----

 do i need mysql here? is it installed?

 asterisk 1.4.2

 get http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.2.tar.gz

 decompress, open terminal, type:

 ./configure
 make
 sudo make install
 sudo make samples

 done. everything installed.

 --

 install a launchd service to start Asterisk.

 /Library/LaunchDaemons/com.asterisk.org.asterisk

 with contents:

 <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE plist PUBLIC
 "-//Apple Computer//DTD PLIST 1.0//EN"
 "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
 <plist version="1.0">
 <dict>
 <key>Disabled</key>
 <false/>
 <key>Label</key>
 <string>com.asterisk.org.asterisk</string>
 <key>ProgramArguments</key>
 <array>
 <string>/usr/sbin/asterisk</string>
 <string>-f</string>
 </array>
 <key>UserName</key>
 <string>frontrow</string>
 <key>GroupName</key>
 <string>frontrow </string>
 <key>OnDemand</key>
 <false/>
 <key>ServiceDescription</key>
 <string>Asterisk PBX</string> </dict> </plist>

 then do in terminal via ssh:

 sudo launchctl load -w /Library/LaunchDaemons/
 com.asterisk.org.asterisk

 ----
 GUI:

 svn checkout http://svn.digium.com/svn/asterisk-gui/trunk
 asterisk-gui
 cd in asterisk-gui
 ./configure (might not be needed) make sudo make install make
 samples sudo install samples

 mod the config files:
 cd /etc/asterisk

 manager.conf:

 enabled = yes
 webenabled = yes

 We will have to add a new user to `manager.conf`:
 [administrator]
 secret = wrxiur
 read = system,call,log,verbose,command,agent,user,config
 write = system,call,log,verbose,command,agent,user,config

 `http.conf`
 enabled=yes
 enablestatic=yes
 bindaddr=0.0.0.0

 make checkconfig to see if you messed up anywhere
 --
 change any other asterisk config files and reboot appletv. all
 Done!
 --

 gui can be accessed at:
 http://appletv.local:8088/asterisk/static/config/cfgbasic.html
 http://appletv.local:8088/asterisk/static/config/cfgadvanced.html

 If appletv.local is not the host, then change to the ip of the appletv.

You now should have a running version of Asterisk on Apple TV that should look like this:



Finally, if you want a PDF version of these instructions which has some additional interesting information (courtesy of Steven Sokol), click here to download. For instance, in the PDF, it states, "There is alot of testing needed and some call applications may not work yet. It's also unfortunate that the installation requires cracking the case and voiding the warranty. Hopefully in the near future somebody will come up with a reliable way to boot from USB and the SSH daemon, obviating the need for a hardware hacking."

Update
Sokol has also published a tutorial that is worth a look.