VoIP & Gadgets Blog

VoIP Lowdown has the lowdown on distributing VoIP thought your house. He covers all the bases and all the stuff you will need to consider if you want to deploy VoIP thoughout your home using your existing wiring. When I had Vonage, I spliced my Vonage line (analog port from the Cisco ATA-186) into my house"s wiring, so all of my RJ11 phone jacks would have an analog signal. This way all of my corded phones could make and receive calls using my VoIP line

In order to do this, I had to disconnect my outside wiring from the telco since I didn"t want any voltage coming in via the telephone company. Even if you cancel your traditional telco analog PSTN line, you can still have voltage on that line, which will conflict with the voltage provided by your ATA device. Could even fry it...

So best to be safe and simply disconnect it at the junction box on the side of most houses. Sometimes it"s in the basement.

Later, I switched to a Uniden 5.8Ghz TRU8885-2, which is a 100% wireless/cordless multi-handset phone system. Ok, technically not 100% cordless/wireless since it does require 1 wired connection to the analog port on the ATA device. Obviously, I connected the Uniden base unit (the transmitter to the other handsets) to the ATA"s analog port and then I used the other 3 cordless phones throughout the house. This is a much simpler solution for those looking to have multiple phones throughout the house without messing with your house"s phone wiring. Plus you get cordless functionality to boot!

There are other multi-handset cordless models as well, including the Click for Amazon price:
Panasonic KX-TG6052B 5.8 GHz Cordless Telephone w/Digital Answering machine and 2 Handsets
Buy Now", STICKY, TIMEOUT, 6000);" onmouseout="return nd();">Panasonic KX-TG6052B 5.8 GHz Cordless Telephone.

One cool model is the Click for Amazon price:
Vtech I5871 - Expandable System w/ digital Answering Device, Color Handset Display & Dual Caller ID
Buy Now", STICKY, TIMEOUT, 6000);" onmouseout="return nd();">Vtech I5871 , which features a 65k color handset display and animated color picture ID. You can even customize the ringer using sounds from your PC! Cool!

Here"s a picture of this cool multi-handset cordless phone system:

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Tags: cordless telephone, home, house, I5871, multi-handset, Panasonic KX-TG602B, VoIP, VoIP Lowdown, Vtech
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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.

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Tags: Aculab, Dialogic, Digium, OpenVox, PIKA, Rhino, Sangoma, ZAPMICRO
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I missed this bit of Skype news from a couple days ago that announced Skype had added Call Transfer functionality, which allows you to transfer a call to another Skype user OR to a PSTN number. At first glance, many of you may be thinking, "so what?"  Well, it is big news and I should have blogged about this earlier, but I was busy playing with Microsoft RoundTable, a 360-degree videoconferencing system. (Stay tuned for a full-fledged review on that.)

The big deal about call transfer is that now businesses can use Skype as a practical business phone solution. Operating your business"s voice communications without call transfer, is like tying the proverbially hand behind your back. Several Skype third parties can now leverage Skype"s new call transfer functionality. OnState Communications emailed me a couple days ago and announced their alliance with Seamless Development, a creator of business eServices, web portals, etc.

According to On-State, "The partnership will integrate OnState ACD for Skype 3.0, a certified Skype for Business Extra that delivers easy-to-use, low-cost call center solutions with Seamless Development’s comprehensive eService offerings -- spanning eBay-specific software, website design and programming, data management, IT and networking support."

So in other words using On-State, a business can setup a call center WITH ACD functionality and WITH Call Transfer functionality and with all the other typical Skype features (voicemail, IM, etc.). What else do you need to run your enterprise voice network? If any company is running a company with >10 employees and is exclusively using Skype, let me know and I"ll profile your organization here on my blog.

Alec mentioned Iotum had been waiting for this call transfer feature for quite some time, but that the features comes a bit late as they have changed their development focus.

Now Skype just needs multiple-party conferencing to both Skype users and PSTN numbers and Skype can be a truly killer business voice app instead of mostly used as a "consumer" voice app.

You can get the Skype 3.5 beta here.

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Tags: call transfer, Microsoft RoundTable, Skype, VoIP
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Out-of-date maps suck. As a huge GPS fan I"ve encountered my fair share of inaccurate GPS map data. But TomTom’s new Map Share software combines cartography with community features that allow users to update maps on the go and then later share the map data with other TomTom users. This still isn"t direct P2P GPS functionality with automatic traffic data sharing, but it"s a start.

So when is this useful? Suppose you are driving and the GPS takes you to a closed road, a detour, or an entirely new road. You can enter changes to the route on the TomTom’s screen - assuming you don"t rear-end the car in front of you while trying to key in the data!

The new data can be uploaded to the Internet with the TomTom Home software. Then other TomTom users can choose to download all map updates from the community to their own TomTom units, or just those verified by TomTom. I can think of some nice pranks to pull with this. Like say reprogramming Boston Red Sox"s Fenway Park to take them to Yankee Stadium instead.

The TomTom Map Share technology is free and included with the new TomTom GO 720 GPS receiver. The GO 720 has a 4.3-inch color screen, a built-in FM transmitter to play music stored on the device or from MP3 players through the car’s radio, and the ability to record your own audio driving instructions.

The Map Share software will eventually be released to users of older TomTom products.

Source: NY Times

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Tags: GPS, maps, Map Share, TomTom
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Aepona announced today their new "Voice Call Continuity" for wireline and broadband service providers. Their solution integrates with the service provider"s existing services, leveraging standard such as SIP, IMS, and SS7 to deliver intelligent rating and charging. Aepona goes beyond mere call handoff to provide a VCC solution that integrates with the service provider"s existing services and OSS/BSS systems. Its VCC solution provides wireline and broadband IP service providers with the capability to seamlessly implement their fixed mobile convergence strategies by assuring scalability, service interaction, and intelligent rating and charging.

A unique aspect of Aepona"s approach to VCC is the company"s emphasis on the real-world challenges that are faced by service providers deploying fixed mobile convergence and triple or even quadruple play. Aepona"s VCC solution addresses the process of continuing a voice (or video) call as a user moves between an IP-based access network that supports VoIP and a mobile network. As wireline and broadband service providers migrate to IP Multimedia Subsystem (IMS) and offer their customers wireless service (either directly or as MVNOs), Aepona enables them to use VCC to offer the use of a single phone number (or SIP identity) as well as seamless transfer of services between a cellular network and an IMS/IP network.
The service provider then has the ability to deliver a truly converged service offering, rather than just being a single source of billing.

"The ability of subscribers to optimally move to and from any network based on the VCC specifications will be one of the main initial drivers for service providers to implement IMS in their networks," said Jean-Charles Doineau, Service Infrastructure Practice Leader at leading research firm, Ovum. "Many wireline service providers are taking a service-led approach towards adopting IMS, and our research indicates that they view VCC as being a strategically important element of their fixed mobile convergence strategies."

Call handover is the current focus of standards activity, however VCC does not exist in isolation and there are significant issues in its deployment. For example, as part of the drive towards the single phone concept with convergence, it is generally accepted that users will need to be to be charged differentially when using multiple networks. Aepona provides the call supervision needed to create the necessary charging records and crucially inter-work with real-time charging engines. It also ensures that charging is properly handled both for pre- and post- paid subscribers.

Aepona"s VCC solution provides service interaction management to resolve conflicts between VCC applications that use both IN and SIP signalling with the same trigger points as other services and enable the services to interwork. This enables service providers to deploy existing value-added services using legacy technology and the ability to offer new ones will emerge as IMS/SIP takes hold. It also provides the intelligence and additional information, such as location and customer preferences, to ensure that unnecessary resources are not used by calls tromboning in and out of the IMS overlay.

"VCC functionality is native to our Universal Service Platform, including the required connectors to both SIP/IMS and SS7 networks, multi-party call control, user status and charging capabilities", said Kieran Dalton, Aepona"s Chief Technical Officer. "This means that our VCC solution can address the necessary details that will make VCC a practical reality for wireline and broadband service providers."

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Tags: Aepona, Fixed Mobile Convergence, IMS, quadruple play, SIP, triple play, VoIP
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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.

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Tags: Asterisk, IP-PBX, open source, PIKA Technologies, predictive dialer, predictive dialing, VoIP
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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.

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Tags: Asterisk, ENUM, IAX, SIP, termination, VoIP
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The ultra low-power Wibree wireless standard founded by Nokia will become part of the Bluetooth specification. Once the spec is integrated into Bluetooth, low-power PAN (Personal Area Network) devices such as watches, toys, consumer health care devices and sensors will benefit from the extra battery life.

Wibree makes minor changes in the media access control layer of Bluetooth to deliver data rates of up to 1Mbit/second while requiring just 10 to 40 percent of Bluetooth"s power consumption. Both Bluetooth and Wibree serve a similar range of around three meters, however Bluetooth can deliver up to 3 Mbits/s. 

Later it will follow up with a dual-mode spec that will require minor hardware modifications to the classic Bluetooth spec to support the lower-power Wibree. The changes will require less than a square millimeter of silicon. Not sure if mobile manufacturers will let you retrofit your existing Bluetooth devices by sending it back in. I doubt it.

Source: EE Times

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Tags: Bluetooth, BT, Nokia, PAN, Wibree, wireless
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I noticed some interesting Google Adsense ads on my blog. It piqued my interest, but I"m not supposed to click on my own ads. Fortunately, I noticed it elsewhere and clicked through. The website shows two traditional PBXs complaining how they are going to be replaced by voice over IP. I tried to embed the .swf Flash file, but the file apparently has to reside on the original source website. I was able to capture the sound however as 4 separate .mp3 files, which are pretty damn funny!

I"ve included them here: (first one is set to AutoPlay)





For the full effect of the two PBXs talking you should check out the website. You can hear the background "hum" of the PBXs in the background even when the PBXs stop speaking. The background noise sounds like your typical IT room. But before I share the website URL, try and guess who created this VoIP promotion.

Whoever can guess it first, I"ll send a free Internet Telephony Conference & Expo "Conference Superpass" - valued at $2,195.  This will get you into all the excellent educational conferences -and of course the Exhibit Hall to see the cool VoIP products displayed in the booths. Of course, the Exhibit Hall is always free, just as long as you pre-register, but access to the paid conference tracks is pretty cool, no? So take your guesses and post a comment.

I"ll give it a few hours and if no one can find out who is doing this promo, I will update this post with the answer and the link to the source website. Good luck!

Update: Doh! The ad is appearing on almost every single impression. Last night I couldn"t get it to display again. It looks like it"s not only in the Adsense ads, but also the 24x7 OAS ads as well. The clues were in the audio files. Well this contest should be an EASY one.

Update 2: We have a winner! David Burr from Converged Network was first to guess Microsoft. i"ll send you details on your free conference pass shortly.

The link to the hilarious Microsoft VoIP promotion is here:
www.microsoft.com/uc/voipasyouare/default.mspx?WT.mc_id=PBX

It starts with an intro, but then to get the other clips to play, you have to click on the various menu options in the orange ticket tape. Do that and close it, and the other 3 clips will play.

Here"s a screenshot of the 2 PBXs talking. Note the colored "talking" indicators. Pretty funny!

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Tags: humor, Internet Telephony Expo, ITEXPO, VoIP
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Comments on this Entry:

(David Burr on Jun 8, 2007 10:58 AM) Microsoft

(Andrew on Jun 8, 2007 11:36 AM) Microsoft !!

(David Burr on Jun 8, 2007 12:08 PM) waa hoo thanks!

PIKA Technologies today announced a reseller partnership with Slacht, a division of Irish software and telecoms services company, Tuxenergy. Slacht will utilize PIKA boards in its Asterisk-based PBX (as opposed to Digium, Rhino, Sangoma, or other Asterisk compatible hardware).

Slacht spent over two years developing a wall-mounted Asterisk-based IP-PBX phone system  PIKA worked closely with Slacht to assist with integrating the InLine, PIKA"s low-density analog board, and the company"s Digital Gateway Board, which provide Slacht"s customers a reliable integration to the PSTN, with solid echo cancellation.

"We could see right away that PIKA offered solid technology," said Kevin Buckley, Managing Director, Slacht. "It supported our open-source technologies of choice, Asterisk and Mandriva Linux, and it offered echo cancellation, something the other vendors we looked at did not feature. Best of all, the price was right."

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...

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Tags: Asterisk, PIKA, Slacht, VoIP
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