the VoIP Digest http://www.thevoipdigest.com VoIP News, VoIP Reviews, and VoIP Information Sun, 02 Sep 2007 18:18:02 +0000 http://wordpress.org/?v=2.0.7 en Free 60 second phone calls http://www.thevoipdigest.com/free-60-second-phone-calls.htm http://www.thevoipdigest.com/free-60-second-phone-calls.htm#comments Thu, 05 Oct 2006 20:55:00 +0000 VoIP & Gadgets Blog VoIP Asterisk 29371@http://blog.tmcnet.com/blog/tom-keating/ 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Free 60 second phone calls

Tags: , ,
Copyright VoIP & Gadgets Blog

]]> 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Free 60 second phone calls

Tags: , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/free-60-second-phone-calls.htm/feed/ Fonality acquires trixbox http://www.thevoipdigest.com/fonality-acquires-trixbox.htm http://www.thevoipdigest.com/fonality-acquires-trixbox.htm#comments Wed, 04 Oct 2006 03:00:00 +0000 VoIP & Gadgets Blog VoIP Asterisk Linux mysql fonality trixbox 29309@http://blog.tmcnet.com/blog/tom-keating/ 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Fonality acquires trixbox

Tags: , , , , , , ,
Copyright VoIP & Gadgets Blog

]]> 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Fonality acquires trixbox

Tags: , , , , , , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/fonality-acquires-trixbox.htm/feed/ Becoming an Asterisk guru http://www.thevoipdigest.com/becoming-an-asterisk-guru.htm http://www.thevoipdigest.com/becoming-an-asterisk-guru.htm#comments Fri, 29 Sep 2006 20:10:00 +0000 VoIP & Gadgets Blog VoIP Asterisk 28811@http://blog.tmcnet.com/blog/tom-keating/ - 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.

TrackBack (0) | Comments (1) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Becoming an Asterisk guru

Tags: , ,
Copyright VoIP & Gadgets Blog

]]> - 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.

TrackBack (0) | Comments (1) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Becoming an Asterisk guru

Tags: , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/becoming-an-asterisk-guru.htm/feed/ Asterisk replaces Cisco CallManager http://www.thevoipdigest.com/asterisk-replaces-cisco-callmanager.htm http://www.thevoipdigest.com/asterisk-replaces-cisco-callmanager.htm#comments Wed, 20 Sep 2006 14:26:00 +0000 VoIP & Gadgets Blog VoIP Cisco Asterisk unified communications 27229@http://blog.tmcnet.com/blog/tom-keating/ 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Asterisk replaces Cisco CallManager

Tags: , , , ,
Copyright VoIP & Gadgets Blog

]]> 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Asterisk replaces Cisco CallManager

Tags: , , , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/asterisk-replaces-cisco-callmanager.htm/feed/ Gizmo Brings Free Calls to Small Businesses http://www.thevoipdigest.com/gizmo-brings-free-calls-to-small-businesses.htm http://www.thevoipdigest.com/gizmo-brings-free-calls-to-small-businesses.htm#comments Tue, 19 Sep 2006 13:26:00 +0000 VoIP & Gadgets Blog VoIP Asterisk gizmo sipphone 27156@http://blog.tmcnet.com/blog/tom-keating/ 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.

Gizmo 2nd SIP account line

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).

 Gizmo choose outgoing line

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

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Gizmo Brings Free Calls to Small Businesses

Tags: , , ,
Copyright VoIP & Gadgets Blog

]]> 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.

Gizmo 2nd SIP account line

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).

 Gizmo choose outgoing line

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

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Gizmo Brings Free Calls to Small Businesses

Tags: , , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/gizmo-brings-free-calls-to-small-businesses.htm/feed/ Asterisk 1.4 unveiled http://www.thevoipdigest.com/asterisk-14-unveiled.htm http://www.thevoipdigest.com/asterisk-14-unveiled.htm#comments Tue, 12 Sep 2006 14:59:00 +0000 VoIP & Gadgets Blog VoIP Asterisk jabber 26970@http://blog.tmcnet.com/blog/tom-keating/ 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Asterisk 1.4 unveiled

Tags: , , , ,
Copyright VoIP & Gadgets Blog

]]> 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Asterisk 1.4 unveiled

Tags: , , , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/asterisk-14-unveiled.htm/feed/ Fonality PBXtra the world’s largest commercial Asterisk deployment http://www.thevoipdigest.com/fonality-pbxtra-the-worlds-largest-commercial-asterisk-deployment.htm http://www.thevoipdigest.com/fonality-pbxtra-the-worlds-largest-commercial-asterisk-deployment.htm#comments Thu, 17 Aug 2006 22:36:00 +0000 VoIP & Gadgets Blog VoIP Asterisk ip-pbx fonality 26737@http://blog.tmcnet.com/blog/tom-keating/ 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

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Fonality PBXtra the world's largest commercial Asterisk deployment

Tags: , , , ,
Copyright VoIP & Gadgets Blog

]]> 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

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Fonality PBXtra the world's largest commercial Asterisk deployment

Tags: , , , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/fonality-pbxtra-the-worlds-largest-commercial-asterisk-deployment.htm/feed/ Windows-based open source Yate IP-PBX launches stable version http://www.thevoipdigest.com/windows-based-open-source-yate-ip-pbx-launches-stable-version.htm http://www.thevoipdigest.com/windows-based-open-source-yate-ip-pbx-launches-stable-version.htm#comments Mon, 14 Aug 2006 13:17:00 +0000 VoIP & Gadgets Blog VoIP Asterisk sangoma yate 26697@http://blog.tmcnet.com/blog/tom-keating/ 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.”

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Windows-based open source Yate IP-PBX launches stable version

Tags: , , ,
Copyright VoIP & Gadgets Blog

]]> 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.”

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Windows-based open source Yate IP-PBX launches stable version

Tags: , , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/windows-based-open-source-yate-ip-pbx-launches-stable-version.htm/feed/ Asterisk Receives VC Funding http://www.thevoipdigest.com/asterisk-receives-vc-funding.htm http://www.thevoipdigest.com/asterisk-receives-vc-funding.htm#comments Wed, 09 Aug 2006 04:59:00 +0000 VoIP & Gadgets Blog VoIP Asterisk ip-pbx digium open source 26605@http://blog.tmcnet.com/blog/tom-keating/ 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.

TrackBack (1) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Asterisk Receives VC Funding

Tags: , , , ,
Copyright VoIP & Gadgets Blog

]]> 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.

TrackBack (1) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Asterisk Receives VC Funding

Tags: , , , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/asterisk-receives-vc-funding.htm/feed/ Another IP-PBX company bites the dust? http://www.thevoipdigest.com/another-ip-pbx-company-bites-the-dust.htm http://www.thevoipdigest.com/another-ip-pbx-company-bites-the-dust.htm#comments Tue, 08 Aug 2006 21:08:00 +0000 VoIP & Gadgets Blog VoIP Microsoft Asterisk ip-pbx digium exchange server pingtel postpath 26602@http://blog.tmcnet.com/blog/tom-keating/ 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Another IP-PBX company bites the dust?

Tags: , , , , , , ,
Copyright VoIP & Gadgets Blog

]]> 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.

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: Another IP-PBX company bites the dust?

Tags: , , , , , , ,
Copyright VoIP & Gadgets Blog

]]> http://www.thevoipdigest.com/another-ip-pbx-company-bites-the-dust.htm/feed/ I’d like a pizza with pepperoni and a sprinkle of VoIP and some GPS too http://www.thevoipdigest.com/id-like-a-pizza-with-pepperoni-and-a-sprinkle-of-voip-and-some-gps-too.htm http://www.thevoipdigest.com/id-like-a-pizza-with-pepperoni-and-a-sprinkle-of-voip-and-some-gps-too.htm#comments Tue, 25 Jul 2006 15:42:00 +0000 VoIP & Gadgets Blog VoIP GPS Asterisk SIP bluetooth pizza open source c# 26468@http://blog.tmcnet.com/blog/tom-keating/ 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.

Pizza PieI'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!

TrackBack (0) | Comments (0) | Tag with del.icio.us | VoIP & Gadget Blog | Permalink: I'd like a pizza with pepperoni and a sprinkle of VoIP and some GPS too

Tags: , , , , , , ,
Copyright VoIP & Gadgets Blog

]]> 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