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Military Suspension Plan from VoIP Providers - Who’s Going to Step Up?
I received an email from a U.S. military service person asking me if I was aware of any VoIP service providers offering a “military suspension plan” which allows U.S. military members to “suspend” your contract without paying monthly fees.
Mr. Tom Keating,I’m a current customer of Vonage, but have recently been disappointed by them. You see, being in the military, deployments do not allow us to use our regular phones, for obvious reasons. As far as I’m aware, every cell phone carrier has some form of “military suspension plan” which allows you to keep your contract without paying any monthly fees until you notify them that you want it reactivated (not sure about landlines, but most ISP’s seem to have a similar policy).
My question to you is: do you know of any VOIP companies that have a policy to accommodate military deployments, or even long vacations? At $35ish/month, I’d prefer not to pay this during a 6+ month tour in Iraq.
“Bitter cold, Bitter fight” a weary U.S. Marine in Korea 1950
Tags: iraq, korea, military, military suspension plan, packet8, service provider, troops, voip, vonage
Related tags: military suspension, service providers, contract paying, military deployments, military, suspension
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flaphone, the Flash Web-based SIP-to-SIP & SIP-to-Skype VoIP App, Adds New Features
Flaphone has done some updates to their Flash-based VoIP application. Back in 2007 I broke the story about the first Adobe Flash-based SIP VoIP app called Flashphone, later renamed Flaphone. Flaphone is a web-based SIP softphone that uses ubiquitous Flash (Mac, PC, Linux), to enable you to make or receive calls to/from all SIP phones and SIP services, including Yahoo! Messenger, MSN Messenger, and Google Talk. You can make free web-based Flash calls to Yahoo! Messenger, MSN Messenger, and Google Talk (gtalk) users. You can even make Flash-based SIP-to-Skype calls using Flaphone, which I tested back in February. You simply enter sip:skype_username@skype to make a call to a Skype username. Good stuff! 
They just announced some new features, including a new skin (white), emoticons in chat, and the ability to transfer files up to 5MB in size between Flaphone users if both have Flash player 10 installed. The file size is restricted for now, but Flaphone stated that when they add P2P support they will remove the restriction. I like Flaphone since I can run it from any PC and make SIP-to-SIP calls or SIP-to-Skype calls without installing anything. Definitely worth checking out.
Tags: adobe, chat, file transfer, flaphone, flash, google talk, im, msn messenger, p2p, sip, skype, voip
Related tags: flash based, messenger google, skype calls, skype username, yahoo messenger, flaphone
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sipgate enters U.S VoIP Broadband Market offering free calls
Today, sipgate is launching a new free VoIP broadband phone service called sipgate one. sipgate one is similar to Vonage, Packet8, and other broadband VoIP providers, but with some additional cool features and a fairly unique pricing plan. For instance, you get a free telephone number, no set-up costs and no monthly charges or minimums.
I spoke with sipgate CEO Thilo Salmon to find out more about sipgate one. First, unlike many VoIP broadband providers, sipgate one isn’t tied to any specific SIP hardware (locked ATAs, SIP phones, softphones, etc.), so you can use it with your favorite SIP device or use a SIP-based ATA and connect your favorite analog cordless phone. Want to use a softphone but don’t want to mess with SIP credentials? No problem - sipgate one has a free softphone app that will autoprovision for you. Those that want to use their favorite SIP device will be pleased to see a helpful drop-down list of many popular SIP devices with corresponding screenshots on how to configure the device to work with sipgate one.
What does it cost?
When using a VoIP phone, outbound calls to other sipgate users are completely free, just like Skype. However, inbound calls on the free U.S. phone number are also free, unlike Skype which I know charges a monthly or yearly SkypeIn subscription. I was a bit surprised sipgate was offering the first phone number for free, so I contacted Thilo a second time to confirm. He responded, “The first U.S. number is indeed completely free. So we are inviting everyone to sign up for a free number and as long as one only uses sipgate to receive calls on a SIP phone the service is free in its entirety. We do not even ask for a credit card. That does set us apart from Skype. While they do not charge per incoming minute, Skype charges a yearly (monthly?) fee as you have pointed out.”
When dialing other phone numbers in the U.S. and Canada are just 1.9¢ per minute and calls to toll-free numbers are free. Other rates apply when calling other countries. Thus, sipgate only charges for outgoing calls and E911 ($1.90 monthly), if activated. No other charges or fees are leveraged on a sipgate one subscriber. Thus, to get started you can initially charge your account with a minimum of $10 with automatic crediting if the balance falls below a user-specified amount ($5, $10, $20, $50). Thilo told me that they aim to keep their international rates lower than Skype’s to make them a very compelling option to potential customers. I then asked Thilo who was comparable feature-wise and he responded, “Google Voice. They don’t allow you to connect a SIP phone, but the features are very similar.”
The true power behind sipgate is its Web-based interface which gives you full access to your voicemail messages, recordings, and faxes. I tested a preview version of sipgate one and was pretty impressed with the features and ease of use. My experiences in my mini-review of sipgate follow below…
Thilo told me that they designed the Web-based interface with Google’s Gmail in mind. For instance, you can search, star/unstar a recording, as well as label recordings, which are very similar to Gmail. Additionally, the interface allows users to call someone back with a single click, as well as divert calls to other landline or mobile phones. Starting a call is as simple as clicking the New Call button and then choosing which phone device you wish to use. Below you’ll see I have 3 options for a new call - an arbitary connection (specify any phone number ad hoc), phone of Tom Keating, and Mobile or Landline of Tom Keating.
As seen by the screenshot above, you can bridge a call to any of your phone devices by initiating a call via the web. This is very similar to Jajah, who made the web-based calling method famous. This interface can be used on your web-enabled smartphone - Windows Mobile, Apple iPhone, etc. if the device doesn’t have an embedded SIP stack or 3rd party SIP softphone installed for making outbound calls.
Mobility features are just as powerful. You can have your home, office and mobile phone ring in parallel. If you are using a mobile phone with a SIP stack, (many Nokia series have SIP) you can be in the car, receive a VoIP call, and then simply press *6 on your mobile phone to record the call. When you next gain access to a web browser you can retrieve the call recording. This is perfect for sales people on the go and who need to enter important call details into a CRM system.
Other features included the ability to customize your outgoing message by uploading an mp3 file, divert calls to another number, create an ad-hoc conference with the touch of a button, and the ability to view missed calls with CallerID info. I asked Thilo what they use on the back-end and he told me they use a fork of SIP Express Router, which enables Class 5 type features. Additionally, Thilo stated there is no maximum storage limit for voicemail and faxes.
Outbound faxing is a breeze. You can click some fields on the web interface and add the fax number, from/to info, add a signature, and of course body text. You can also attach a PDF and it will render it. I asked Thilo about Word support and he said that was in the works.
Inbound faxing does require a separate phone number, but there is no fee to receive faxes. There is a nominal $2.90 one-time fee per additional phone number. I asked Thilo about CNG autodetection of fax tones using just a single universal phone number, but he said “We have found that most people don’t really like that.” Personally, I’d rather just have one phone number on my business card, but it’s a minor complaint.
“There is simply no barrier to people disconnecting their old phone lines anymore. Phone and cable companies have long been pushing voice plans in the region of $25 to $40 per month–which end up being as much as $60 or more with extra charges–and that’s just ridiculous,” said Thilo Salmon, CEO of sipgate. “Even with calls to other landlines and mobile phones, most users will spend less than $5 a month using sipgate one. And for those people only receiving incoming calls on their VoIP phone, the service is completely free.”
sipgate is also readying a multi-user edition of the service aimed at small businesses, which will not only replace landlines, but also customer premise phone systems. sipgate is certainly setting a new low-price benchmark with bundled powerful features that should cause VoIP fans to seriously consider them. Free phone number, free toll-free calling, free inbound calling, free inbound faxes, what’s not to love? 
Tags: broadband voip, e911, free calls, iphone, packet8, sipgate, sipgate one, skype, voip, vonage, windows mobile
Related tags: phone number, asked thilo, phone service, mobile phone, based interface, phone
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Truphone 3.0 comes to the Apple iPod touch
Truphone today launched Truphone 3.0, a major new upgrade to its mobile VoIP application for the Apple iPod touch. Truphone 3.0 was already available for the Apple iPhone, so this release just brings the latest features to the popular iPhone touch.
IM services currently supported include Skype, MSN Messenger, AIM, Yahoo! Messenger and Google Talk. It also does free calls when in Wi-Fi to other Truphone users as well as free WiFi calls to Skype and Google Talk users. Though I would like to see 3G data support to enable VoIP over 3G. Yes I know Apple blocks VoIP over 3G apps, but if you jailbreak your iPhone, you should be able to make VoIP over 3G calls. (read my tutorial on how to do VoIP over 3G on jailbroken iPhones) Yet, there is no mention whether their truphone app will work over 3G on jailbroken iPhones. Ironic that in 2007 truphone was the first to demonstrate VoIP over WiFi on an Apple iPhone that they jailbreaked.
Of course, you could use Truphone Anywhere for free calls, but that uses the 3G voice channel not 3G data. It leverages a callback system that uses your bucket of cell minutes for ‘relatively’ free calling.
In any case, check out the news.
Tags: 3g, apple, google talk, iphone, ipod touch, skype, wifi
Related tags: apple iphone, jailbroken iphones, truphone, apple, calls, iphone
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50% of Mobile Voice Traffic Will Use End-to-End VoIP by 2019
Gartner, a respected research firm located just a stone’s throw from TMC said today, “mobile VoIP poses a huge challenge for traditional mobile voice providers.” You don’t say? Hmm, I would have never guessed such a thing. Ok, other than stating the obvious, the research does have some fascinating points, including claiming that ore than 50% of mobile voice traffic will be carried using end-to-end VoIP by 2019 - or basically 10 years from now.
“Mobile portal voice over Internet Protocol (VoIP) offered by third-party application-based providers poses a huge and direct challenge to the $692.6 billion global mobile voice market”, according to Gartner, Inc. Gartner predicts that over time traditional network-based mobile carriers face the real prospect of losing a major slice of their voice traffic and revenue to new non-infrastructure players that use VoIP.
But Gartner doesn’t paint an entirely rosy picture for VoIP. Gartner said “that despite this significant potential, conditions for the rapid expansion in the use of mobile VoIP are not yet right and are not likely to become right for at least five years and perhaps as long as eight years.”
Yeah, well when carriers like AT&T pull crap like forcing Apple to only allow VoIP over WiFi and not over a 3G data connection, it’s no wonder why mobile VoIP growth will be stunted by anti-competitive tactics. Then you have countries like Canada which outright block Skype on the iPhone. Fortunately, avid mobile phone users aren’t taking this lying down. For instance, you can jailbreak your iPhone and run VoIP over 3G no problem.
Why do we have to put up with this crippleware? I get that AT&T is a business and needs to make money. If they’re worried that flat-rate data plans that run VoIP over it will drastically hurt their voice revenue, then change your business model! Keep the flat-rate data plan, but install packet-inspection technology that detects voice packets and charge a few cents for VoIP calls. If the flat-rate data plan’s business model is outdated, which it seems to be, then change it.
Yeah, sure customers may not like the idea and certainly it seems that our culture today expects “something for nothing”, but hey, it’s business folks. I’d rather have the capability of making VoIP calls using a SIP provider or make Skype calls and pay a few pennies than not be able to use VoIP over 3G/4G at all. I’m sure if AT&T did try and charge money for VoIP calls running over a data connection that customer advocate groups, the ACLU, and people who think they should get something for nothing will run to the government and complain that AT&T is charging them money for using VoIP. Perish the thought that a business is charging you money for using their services!
Maybe we should just nationalize all the carriers like most countries have and then petition Congress to pass a law that will force the nationalized carriers to give us free mobile VoIP. Heck, with trillions of dollars being spent on the bailout package, why not free mobile VoIP at the expense of the big bad carriers? Free healthcare for everyone, free mobile VoIP for everyone, it’s all free baby! Socialism here we come!
Ok end rant. Back to Gartner…
“Mass-scale adoption of end-to-end mobile VoIP calling will not happen until fourth-generation (4G) networks are fully implemented in 2017,” said Tole Hart, research director at Gartner. “Once the basic market conditions are in place, transition to mobile portal VoIP should be fairly rapid because of the inherent convenience and end-user cost savings. In 10 years time we expect that 30 percent of mobile voice traffic will be carried out through third-party mobile portals, such as Google, Facebook, MySpace and Yahoo, which will adopt wireless VoIP service as a voice option to their current communications hub.”
A number of third parties, such as Skype, Truphone and fring, which carry VoIP traffic using a mobile phone, have cropped up in the past couple of years, offering access to voice services via Wi-Fi and/or the carriers’ wireless voice networks. This has been the most efficient way to offer the service to date because of the inconsistencies of voice services over third-generation (3G) data networks. However, with the advent of 4G networks (WiMAX and Long Term Evolution [LTE]), and increased use of smartphones with open operating systems, it is conceivable, perhaps even inevitable, that wireless voice services will be run completely over VoIP.
“Ten years from now, more than half of mobile voice traffic will be carried end-to-end using VoIP,” said Akshay Sharma, research director at Gartner. “Carriers will adopt voice services because of the increased capacity and reduced cost of delivering voice over 4G networks. Third parties will adopt a voice option for their communications hub.”
Gartner analysts warned that there will also be a number of factors that will inhibit the adoption of third-party, end-to-end VoIP services, including the delay in rolling out 4G networks because of current economic conditions and also the general plan to put 4G only in the main cities and build out from there. Nevertheless, in five to 10 years time, as 4G networks become common, mobile VoIP services will have a strong impact on the communications market.
Competing with mobile portal VoIP will be wireless carriers that offer circuit and VoIP voice and data services, and resellers and mobile virtual network operators (MVNOs) that also offer services off the carrier networks. Gartner expects this opening of the VoIP channels to spawn a number of voice services from companies that offer voice services to communities using voice as a communications link. This means that the biggest competitors to mobile VoIP may be text messaging and e-mail, as people may prefer to use these types of communication because of their non-intrusive, less emotional and less time-consuming nature.
Although the impact of the technology shift will be gradual as 4G networks roll out, Gartner advises carriers to start thinking now about how the transition will occur and how they might cooperate and partner with other types of service providers. Third-party providers, such as Google and Yahoo, should look to offer voice services today using the carriers’ networks and Wi-Fi to leverage their portfolio of services. Mobile social communities, such as Facebook and MySpace, which benefit from messaging traffic as it keeps eyeballs on their sites, should also have a voice option.
Tags: 4g, fring, gartner, long term evolution, lte, mobile voip, sip, skype, tole hard, truphone, voip, wimax
Related tags: voice services, mobile voice, voice traffic, third party, voice option, mobile
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Free Calls on Skype
3 UK is opening up its network to allow anyone with a 3 SIM and a compatible handset to have unlimited Skype-to-Skype calls and instant messages without ever having to pay. 3UK announced that on May 1st users with a 3UK SIM and a 3-provided Skype-enabled handset will be able to use Skype for free. Additionally, previous deals for free Skype required a monthly topup of £10 on pay-as-you-go and there will be no data charges. Thus, Skype will be 100% free, subject to their fair use policy of course. 
There is just a one time fee of £1.99 for a SIM card.
Anyone with a compatible unlocked 3G handset in the UK, can take advantage of free Skype calls, whether or not their phone is from 3. This is a huge barrier-busting move by 3 that changes the game. Forget about using AT&T on your iPhone who is preventing Apple from allowing Skype to run over the 3G data connection. Simply get an unlocked iPhone, stick in a 3 SIM and enjoy free Skype calls! Of course, this “free” calling deal only applies to the UK. Why do the Brits get all the fun? 
No worries, I’m sure this is just the first step before the U.S. gets in on the free Skype calling action. I wonder if there might be some synergies with the new Skype for SIP program (of which I’m a beta tester)?
According to 3 UK, “By removing these key barriers to Skype-to-Skype calls, 3 UK is creating a compelling reason for new customers to join 3 and to enjoy all the products and services available on the UK’s biggest mobile broadband network.”
According to 3 UK:
We’ve been working with Skype for over two years now and we’ve learned a lot about how our customers use the service. We know that a keen mobile Skype user is an instinctive, active communicator. They want to take full advantage of free mobile calls.Increasingly Skype use is linked to other internet communications activities, such as Facebook, Twitter and a host of other services for which our customers will happily pay a fixed fee for unlimited use.
While others have looked at Skype as a potential threat to voice and text revenues we see another advantage. Did you know when you call a friend on a different mobile network their network will charge your operator a fee for carrying the call? This is called a Mobile Termination Rate or MTR and is currently charged at around 4.7p or more for every minute of the call.
This fee is regulated by the industry regulator Ofcom but we think they’re still too high. Without these fees we’d be able to offer our customers much better value and that’s where Skype comes in. With Skype, MTRs don’t apply so we can give our customers all the minutes they like without over charging them.
At 3, we believe our customers should be able to choose how they communicate because that’s mobile as it should be; simple, useful and always good value.
Kevin Russell, Chief Executive Officer of 3 UK, said “Communication through the internet is exploding. Internet calling or VoIP, social networking, instant messaging and email are used by millions in the UK every single day. They are open to all on their PCs and laptops. We want people to be free to communicate from their mobiles in the same way as they do from their PCs.
“In future you will be able to buy a 3 SIM for unlimited Skype-to-Skype calls for less than the price of a cup of coffee and talk for as much as you want without ever paying us another penny. We won’t ask you for a top-up or a monthly commitment. If you want to talk on a mobile for free, just join us and give it a go. This is for everyone.”
Josh Silverman, President of Skype said, “Demand for mobile access from our users has never been higher. The introduction of unlimited Skype-to-Skype calls and instant messages across all 3 price plans is a really exciting move from a key partner. 3 UK clearly understands the desire for people to use Skype wherever and whenever they want. This is the first mobile network to show this kind of innovation to enable their customers to access Skype.
“We believe this is how the future looks for the Internet on mobile. With this bold move 3 UK has again shown their willingness to be the customer champion for mobile services in the UK.”
Currently, 3 UK’s growing Skype community enjoys 1.5 million minutes of free Skype-to-Skype calls every day. The launch of the first 3 Skypephone in October 2007 really kick-started the growth of free internet calling on the 3 network. With over 433 million people registered on Skype worldwide, the new free Skype-to-Skype offer from 3 opens up a world of free calling.
Two years experience of providing open access to Skype-to-Skype calling has enabled 3 and Skype to better understand the behaviour of mobile Skype users. Success with an easy-to-use Skype experience on more specialised internet-enabled handsets, such as the INQ1 and the 3 Skypephone collection; has proven to 3 that enabling customers to make free Skype calls to other Skype users on their mobiles or PCs is a real benefit.
3 UK has found that regular Skype users:
- Are less likely to churn than non-Skype users
- Use more traditional voice minutes than non-Skype users in addition to calling their Skype contacts
- Use Skype IM, but also send more SMS than non-Skype users
- Are more likely to browse the internet on their mobile
- Are higher margin customers
- Are twice as likely to access social networking sites as non-Skype customers
“Today we are moving in a clear direction towards making Skype-to-Skype calling available to all UK mobile consumers,” said Mr Russell. “We know that Skype users are instinctive communicators, keen social networkers and mobile internet users. They love the things that we are building the UK’s biggest mobile broadband network for.
“Our network is built to deliver the benefits of the internet to the mobile. That’s why we’re removing the conditions and restrictions from our current Skype offer and opening up the opportunity to try free internet calling to all UK mobile users, whether they are currently with us or a competitor network.”
Tags: 3, 3 UK, calling, free, josh silverman, kevin russell, SIM, skype, skype-to-skype, unlocked phone
Related tags: skype calls, skype skype, skype users, internet mobile, internet calling, skype
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Using monit Tool to Monitor Asterisk
Your IP-PBX is one of the most critical pieces of corporate infrastructure. It cannot afford any downtime, which is why the fives 9’s (99.999%) of reliability was coined. While Asterisk is a pretty stable open source IP-PBX platform, it it still in its infancy, so it hasn’t had the same time that the old ‘Big Iron PBXs’ have had to reach five 9s of reliability. Then again, many traditional PBX manufacturers have abandoned 100% proprietary hardware and use many of the same standard off the shelf components that are in Asterisk, including motherboards, memory, processors, etc. So the old wives tale that big iron PBXs are more reliable than PC-based PBXs no longer applies.
Still, Asterisk and all of its derivatives (trixbox CE/Pro, PBX in a Flash, etc.) have a cult following (of which I’m a member) — and like any cult, we like to do crazy things, like tweak Asterisk or trixbox in the middle of the work day to see if some newfangled text-to-speech feature will work. 
Well, with so much tweaking by some Asterisk cultists, something is bound to go wrong, usually at the end of the work day on a Friday when you’re driving home, forcing a return to the office or waiting to you get home and SSH into Asterisk to restart the service.
So how do we ensure a more reliable Asterisk platform using an automated tool? Surely there must be a way of monitoring the Asterisk service and if it crashes, automatically restart it, right? Ever second is precious when you’re trying to achieve 5 9s of reliability, which equates to 5 minutes, 15 seconds or less of downtime in a year. Or if you want to get really crazy, shoot for 6 nines of reliability (99.9999%) which is 31.536s of downtime per year!
Well, before we continue, you must remember that Asterisk runs on Linux and there are many great monitoring tools for Linux. In fact, for the blog web server you’re reading this article on, I’m running a free monitoring tool aptly called monit, which you can get here. This tool is so easy to use, it should be in any Linux admin’s arsenal. I use it to monitor various parameters of the blog server, and if certain conditions are met, it automatically restarts the apache web service.
It got me thinking, “Why not use monit to monitor Asterisk?” Well, here’s how to do it!
1) Install monit.
2) Simple way: Run ‘yum install monit’ or run ‘apt-get install monit’ Go to Step
3) Compile/Harder way: Go here: http://mmonit.com/monit/download/ and download the .tar file, currently called monit-5.0.tar.gz
4) Untar monit
# tar -zxvf monit-5.0.tar.gz Configure and compile monit:
# cd monit-5.0
# ./configure5) Install monit
# make
# make install
6) Copy monit configuration file to /etc/ folder
# cp monit.conf /etc/monit.conf (older versions used monitrc filename)
7) Edit monit.conf & put in your monitoring rules (see examples below)
Add monit service to the startup. Red Hat command follows:
# chkconfig --add monit
# chkconfig –level 2345 monit on
# {confirm the run levels}
# chkconfig –list|grep monit
It is super easy it to setup the mail server for notifications and to configure monitoring of processes, files, loads (CPU, memory), and ports. And of course, using monit you can monitor Asterisk, trixbox CE or Pro, PBX in a Flash, and other IP-PBXs that run on Linux.
Here’s a snippet from two monit.conf configuration files (one the blog server, the other Asterisk):
############################################################################### ## ## Start monit in background (run as daemon) and check the services at 2-minute ## intervals. # set daemon 120 # can set lower if want downtime <2min set mailserver mail.tmcnet.com # primary mailserver ## You can set the alert recipients here, which will receive the alert for ## each service. The event alerts may be restricted using the list. # set alert blogalerts@tmcnet.com # receive all alerts set alert anotheremailhere@somewhere.com check system blog.tmcnet.com if loadavg (1min) > 4 then alert if loadavg (5min) > 2 then alert if memory usage > 75% then alert if cpu usage (user) > 70% then alert if cpu usage (system) > 30% then alert if cpu usage (wait) > 20% then alert check process apache with pidfile /var/run/httpd.pid start program = "/etc/init.d/httpd start" stop program = "/etc/init.d/httpd stop" if cpu > 60% for 2 cycles then alert if cpu > 80% for 25 cycles then restart if totalmem > 1300.0 MB for 5 cycles then restart if children > 250 then restart if loadavg(5min) greater than 10 for 8 cycles then stop if failed host blog.tmcnet.com port 80 protocol http and request "/monit/doc/next.php" then restart if failed port 443 type tcpssl protocol http with timeout 15 seconds then restart if 3 restarts within 5 cycles then timeout depends on apache_bin group server # Asterisk Monitoring rule set daemon 30 # Check every 30s set logfile syslog facility log_daemon set alert asteriskalerts@yourdomain.com check process asterisk with pidfile /var/run/asterisk/asterisk.pid group asterisk start program = "/etc/init.d/asterisk start" stop program = "/etc/init.d/asterisk stop" # Check uptime via Asterisk Manager Interface (AMI) port 5038 if failed host 127.0.0.1 port 5038 then restart if 5 restarts within 5 cycles then timeout #Check Veritas BackupExec Agent check host blog.domain.com with address 192.0.0.6 start program = "/etc/init.d/VRTSralus.init start" #stop program = "/etc/init.d/VRTSralus.init stop" if failed port 10000 with timeout 35 seconds then restart
Further, you can even test the SIP protocol, which uses port 5060. The SIP test is similar to other protocol tests that monit supports, however, it allows extra optional parameters.
IF FAILED [host] [port] [type] PROTOCOL sip [AND] [TARGET valid@uri] [AND] [MAXFORWARD n] THEN action [ELSE IF SUCCEEDED [[<X>] <Y> CYCLES] THEN action]
TARGET : you may specify an alternative recipient for the message, by adding a valid sip uri after this keyword.
MAXFORWARD : Limit the number of proxies or gateways that can forward the request to the next server. It’s value is an integer in the range 0-255, set by default to 70. If max-forward = 0, the next server may respond 200 OK (test succeeded) or send a 483 Too Many Hops (test failed)
SIP examples:
check host openser_all with address 127.0.0.1
if failed port 5060 type udp protocol sip
with target “localhost:5060″ and maxforward 6
then alert
check host sip.broadvoice.com with address sip.broadvoice.com
if failed port 5060 type tcp protocol SIP
and target 1234@sip.broadvoice.com maxforward 10
then alert
Now that you know how to automatically monitor Asterisk, trixbox, PBX in a Flash, etc. those five nines (6?) of reliability are just around the corner. As the PBX administrator / telecom manager, you will be worshiped by your sales team 
and boss for keeping the phone system up all the time. 
They will think you an Asterisk God, who will be adored and who shall command great respect and admiration. And none shall mourn for any Asterisk outages.
Tags: asterisk, five nines, monit, monit.conf, monitoring, pbx in a flash, sip, trixbox, voip, Who Mourns for Adonais?
Related tags: monitor asterisk, install monit, start program, asterisk trixbox, asterisk start, monit
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KIRK 2010 WiFi VoIP phone
The KIRK 2010 WiFi VoIP handset launched today. This is the latest edition to the Polycom KIRK Series. I wonder if Captain Kirk is aware that Polycom is trying to copy his copyrighted ‘communicator’?
Then again, Kirk’s communicator didn’t have a numeric keypad. Just turn the dial and instantly talk to the Star Trek bridge with seemingly no latency either. You can compare the look of it here:
Or this closely resembling USB-based Star Trek communicator:
They definitely kept it simple, by going with a black & white screen, but they no doubt kept the costs down. The pricing for its brethren is certainly pretty expensive, i.e.:
The list price for the Polycom KWS 300 is U.S. $360. The KWS 6000 list price is U.S. $1,200 and includes a server and one base station, which supports up to 30 users. With the scalable nature of the KWS6000 it can also be set up for more users. The KIRK 5040 handset sells at a list price of U.S. $310.
I couldn’t find pricing info online for the KIRK 2010, but certainly businesses are looking for affordable WiFi VoIP phones.
Features and Benefits of the KIRK 2010
- Black & white LCD screen (3 lines of text/icons)
- Internal/external ring pattern, volume control and silent modes
- Telephone book with room for 40 numbers
- Speech/stand by time > 12/150 hours
- Weight incl. battery: 120g
- Size (LxWxH): 124×47x31mm
Tags: captain kirk, KIRK 2010, polycom, voip, wifi, wireless
Related tags: black white, communicator, polycom, price
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OCS 2007 R2 PIC problem with AOL Fixed
Microsoft OCS 2007 R2 users were having communications issues with AOL’s AIM when federating using PIC (Public IM Connectivity) and using a Windows Server 2008 (x64) Edge role server - Windows Server 2003 (x64) is unaffected by this problem. Microsoft’s Scott Oseychik just issued a fix that solves the problem. The fix involves changing the Windows Server 2008 Edge role to initially establish the SSL dialog using the TLS_RSA_WITH_RC4_128_MD5 cipher suite.
It’s pretty easy to fix via Group Policy (gpedit.msc). Once you make the fix you should be able to successfully communicate with AOL AIM clients using Office Communicator 2007 R2 via PIC.
Click here for the resolution.
Tags: LS_RSA_WITH_RC4_128_MD5 cipher suite, Microsoft OCS 2007 R2, pic, Public IM Connectivity, Scott Oseychik
Related tags: windows server, server, using, windows, problem
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AsteriskNOW 1.5.0 Released
AsteriskNOW 1.5.0, which launched as a beta in October 2008, is now available for download at http://www.asterisknow.org/downloads. Of course, existing AsteriskNOW users can simply run “yum update” to update to the latest release. I love ‘yum’ for Linux systems - it’s like Windows Update on steroids, but without the Internet Explorer GUI.
According to AsteriskNOW, here are the notable changes since beta2:
* Updated several packages to latest versions (Asterisk, DAHDI, etc)
* Fixed more permissions issues between Asterisk and httpd/FreePBX.
* Updated to CentOS 5.3 (http://lists.centos.org/pipermail/centos-announce/2009-April/015711.html)
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Microsoft Roundtable is now Polycom CX5000 Unified Conference Station
Polycom and Microsoft today announced that “as part of Microsoft’s vision to broaden the availability of Microsoft RoundTable, Polycom has licensed the right to distribute RoundTable, effective April 13, 2009.” The product, renamed the Polycom CX5000 Unified Conference Station, will be available through Polycom and its channel network.
Polycom has ‘exclusive rights’, to sell the CX5000, which is a huge win for them. Although I have been a huge fan of the Microsoft Roundtable with it’s cool 360 panoramic video, my guess is that Microsoft has had difficulty selling this expensive ($4300) videoconferencing equipment.
The CX5000 when used in conjunction with Office Live Meeting service, or as part of Office Communications Server 2007, it combines content, a panoramic 360-degree view of the entire meeting room, and a separate view of the active speaker for a unique and engaging voice and video experience.
The Polycom CX5000 will be available beginning April 13, 2009, at a list price of U.S. $4,300. The CX5000 will be available in 27 countries through Polycom’s extensive channel partner network and will be available for shipment in late April. Once the Polycom CX5000 is available, RoundTable will no longer be sold. Microsoft will continue to support all RoundTable devices already sold, while Polycom will provide front-line customer support for CX5000 units sold beginning April13. To learn more about the Polycom CX5000, visit www.polycom.com/go/polycomcx5000.
You can check out my review of the Microsoft Roundtable, now called the Polycom CX5000 for more details on this product.
Tags: cx5000, microsoft roundtable, polycom cx5000 unified conference station, video conferencing, voip
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Finally! New Windows Mobile App AudioRoute Enables Earpiece for VoIP Apps
Finally a software tool called AudioRoute that can be used to route Windows Mobile audio from the earpiece speaker to the backspeaker and vice-versa. This is especially needed for VoIP applications on Windows Mobile phones.
I’ve tested several VoIP apps (SIP clients, Skype, etc.) on my Windows Mobile XV6700 phone and other Windows Mobiles and from what I understand the carrier forced the hardware manufacturers to block VoIP applications from using the earpiece for listening to the remote caller. You couldn’t even use speakerphone. Instead, you were forced to use the backspeaker, a tiny low-quality speaker located on the back of the phone, which made phone quality horrendous when making VoIP calls. I’d have to flip the phone over when the person was talking due to low volume & quality, and then flip it back over to talk into the microphone. It was all but unusable.
Well glory glory hallelujah! 
I never thought the day would come when someone would come up with a solution. According to Teksoft, “After several years of tests and many questions in the development forum, we’ve finally did it: a tool to route the audio to the earpiece speaker is available, and we’ve released it as freeware.” Woohoo! Now I can register my SIP client on my Windows Mobile to my Asterisk-based IP-PBX and make/receive VoIP calls.
Features:
- Routes the audio output to earpiece or backspeaker
- VoIP compatible
- Easy to use User Interface
- Command line support
- Uses Teksoft’s DynRIL library
It’s compatible with Pocket PC and Smartphone Windows Mobile 5.0 / WM6.0 and above
Usage (via forums)
Install the CAB and use the titlebar icon to open the user interface.
The first icon routes the audio to the earpiece speaker.
The second blue icon, can be used to route the audio to the backspeaker.
The orange icon, routes the audio to the speakerphone, while in a phone call.
You can also use the bottom slider to move the taskbar icon, or the about button to show this page.
The top-right square hides the user interface.
Command line
This tool can be executed by command line with parameters.
You can execute /program files/teksoft/audioRoute/audioRoute.exe with the following:
-earpiece , routes the audio to the earpiece
-backspeaker , routes the audio to the backspeaker
-speakerphone , while in a phone call, activates the speakerphone
-switch , toggles between earpiece and backspeaker
| Code: |
| audioroute.exe -earpiece audioroute.exe -backspeaker etc. |
Download
The CAB file is available in the freeware section of www.teksoftco.com, direct link here.
Tags: asterisk, audio, audioroute, backspeaker, earpiece, sip, skype, speakerphone, teksoft, voip, windows mobile, wm5, wm6
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Michael Robertson Responds to OpenSky vs. SkypeforSIP article
I would be remiss if I did not point out that Gizmo5’s CEO Michael Robertson responded to me regarding yesterday’s OpenSky vs. SkypeforSIP article. I had tried to reach Michael before the article went live but was not able to reach him.
Michael emailed me his thoughts, which I have now included into my original article prefaced with the word “Update” in front and with his quotes italicized.
For instance, Michael disagrees with the Skype representative on how Skype will price SkypeforSIP. He still thinks it will be expensive. He wrote me to say, “I stand by my price and delivery estimates. Skype for Asterisk and Skype for SIP will be expensive options as ebay tries to get money from business customers.”
I guess we shall see.
There’s some other additional info and commentary from Michael definitely worth reading. I wanted to present both sides of this.
I report, you decide.
Tags: gizmo5, Michael Robertson, OpenSky, sip, skype, SkypeforSIP, voip
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OpenSky vs. SkypeforSIP - It’s On Baby!
It’s on! It’s Gizmo5’s OpenSky vs. Skype. Michael Robertson, the CEO of Gizmo5 has a post up where he compares Gizmo5’s OpenSky to Skype’s SkypeforSIP. It’s interesting how Michael talks about open standards, mentioning .mp3, a standard which Michael was a proponent of when he ran MP3.com and then backslaps Skype upside the head basically saying “What took you so long? I’ve been telling you all along Skype that you need to embrace the SIP standard.”
He continues the blog post by comparing OpenSky, a SIP-to-Skype service that launched last month to Skype’s just launched SkypeforSIP. It’s a worthy endeavor on Michael’s part to provide such a comparison, even if he is biased in favor of his company’s offering.
Check out Michael’s comparison of the two SIP for Skype services:
| SIP for Skype Solutions | ||
| OpenSky (Gizmo5) | SkypeforSIP (ebay) | |
| Receive Skype calls on SIP software and hardware | Yes | Yes |
| Place Skype calls to SIP devices | Yes | Yes |
| Answer/Call Skype from browser | Yes | No |
| Includes free voicemail | Yes | No |
| Free trial | Up to 5 minute calls to Skype | No |
| Single user cost | $20/per year | No |
| 50 User Pricing | $800/per year | $2499 purchase plus annual fee* |
| Per call PSTN connection fee | None | 3.9 cents |
| Availability | Now | Dec. 2009* |
| Supported codecs | g729, ilbc, pcmu/pcma, gsm, speex, and custom | g729 |
| * Estimated | ||
At first glance, it’s easy to look at this pretty chart and declare OpenSky the clear winner. But let’s take a closer look. He mentioned OpenSky offers free voicemail. While that’s nice to have, SkypeforSIP is targeted towards IP-PBXs which have their own auto-attendant and fancy voicemail systems. Most people with IP-PBXs would rather keep their existing voicemail system rather than switch to a hosted one. He lists ‘Answer/Call Skype from a browser’ as a feature that SkypeforSIP doesn’t have. I spoke with a Skype representative just yesterday and he mentioned inbound calling capabilities from the Web (click-to-call buttons) to your corporate IP-PBX, though he didn’t indicate if this feature would be part of the initial beta. So I’m not sure exactly what Michael is referring to here.
Another thing to consider is how OpenSky and SkypeforSIP connect to the Skype cloud and if Skype has a competitive advantage there. While both support SIP on the front-end, the back-end conversion for OpenSky sits on Gizmo5’s network and the back-end conversion for SkypeforSIP sits on Skype’s network. So is there any better call quality or QoS that Skype can offer over Gizmo5? Without knowing the exact technical details of each, I don’t think Skype would be better than OpenSky. However, it’s possible that without net neutrality legislation, that Skype could detect OpenSky connections and “throttle” the packets, inject latency, etc. Doubt they’d do that, especially after the FCC slapped down Comcast for messing with user’s Internet traffic, but you never know.
The SkypeforSIP $2499 purchase price plus annual fee seemed way too high, however Michael has an asterisk with the word ‘Estimated’ at the bottom. Later on I confirmed with a Skype representative that this is indeed inaccurate. Also, the 3.9 cent connection charge per call vs. free for OpenSky seemed like a huge pricing advantage for OpenSky. As an example, if the average SMB makes 500 outbound calls per day, that’s $19.50/day in connection fees. With 20 work days per month, that’s $390/month more for SkypeforSIP. Skype would have to be dramatically lower in per-minute pricing to catch up to OpenSky’s lower pricing. If you make mostly domestic (U.S.) calls, and not a lot of international calls, then there’s no way SkypeforSIP could catch up to OpenSky’s pricing. This pricing comparison assumes Michael’s chart was indeed accurate, but I had my skeptic hat on - and rightfully so once I talked to a Skype rep.
As I mentioned, the $2499 purchase price seemed too high and I didn’t think the 3.9 cent connection charge was entirely accurate so I contacted a Skype representative to respond to Michael’s claims. I said, “I haven’t seen any pricing info for SkypeforSIP, so I’m not sure where Michael got his estimated pricing”. The unnamed Skype representative responded, “Skype has not settled on final pricing. I can tell you that we are going to be following the typical Skype disruption to existing business models when we do announce our pricing.”
With regards to Michael’s chart where he lists Skype as having a $0.03 cent connection fee, the Skype rep responded, “Nowhere have we said there is a connection fee. Essentially, the only time there is a connection fee is when they are using pay-as-you-go - they’re just using Skype credits. And actually the connection fee does not apply in the U.S. Ultimately, someone who is using this for business is going to have a subscription anyway. So they’re not going to be paying a connection fee to start with.”
With regards to Michael’s knocking SkypeforSIP’s “availability” (December 2009), he responded, “The product is available now. It’s using the same Skype technology that has been tested & deployed for the last 5 years. He’s saying December 2009 - that’s not the case.”
With regards to the chart comparing codecs supported, he told me that G.711 will be rolled into the beta in the coming weeks and it’s already available to internal testers. As for the other codecs which OpenSky supports and SkypeforSIP does not, he responded, “Most global termination providers only support G.729 and G.711, so the other codecs Michael listed are redundant until SIP endpoints deploy a high-quality wideband codec like G.722 or SILK. “
Finally, the Skype representative said, “Some other interesting things he has not raised, but we are happy to raise, is the Skype business control panel. People can administer their accounts, setup billing, resolve call logging histories, and other features for businesses. I don’t think OpenSky does that. The key thing here is that SkypeforSIP was built by the same engineering team that is at the heart of Skype. They’ve been dealing with SkypeIn and SkypeOut for years, so we understand scalability, usability, and high performance core network design. Gizmo is running on consumer clients running on large number of servers which questions scalability and efficiency to a business.”
The battle has commenced. Your move Michael. Wonder if I should get Mark Spencer from Digium/Asterisk involved in this fight since Digium recently launched Skype for Asterisk? May as well make it a battle royal! 
Update:
Skype Journal has a nice comparison of Skype for Asterisk vs. SkypeforSIP.
Tags: G.711, G.729, gizmo5, Michael Robertson, opensky, SILK, SIP, skype, skype for asterisk, SkypeforSIP, voip, wideband codec
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Google Voice Meet Asterisk
Nerd Vittles has another cool Asterisk recipe that combines Google Voice, voicemail transcription (via Google Voice), free calling, and of course Asterisk. Nerd does some packet sniffing and determines that Google Voice, powered by Grandcentral, is using SIP. What’s most interesting is that Nerd determine that your SIP connection and your Google Voice phone bill is only protected by a 4-digit PIN. Yikes! That’s not good.
Anyway, here’s a teaser of Nerd’s awesome recipe:
what we want to do is examine some ways to integrate the Google Voice feature set into our existing Asterisk implementations. The potential benefits are enormous. There’s free calling in the U.S., free distribution of inbound calls to multiple phone numbers scattered around the country, free SMS messaging and delivery by email, free transcription of voicemail messages into text-based emails, free conferencing, and free GOOG-411, a voice-activated service that let’s you find nearby businesses by saying where you are and what you’re looking for. For today, we’ve set our sights on the Google Voice feature set which is easiest to integrate into existing Asterisk systems: free voicemail message transcription, free calling in the United States, and free GOOG-411 directory assistance. For lack of a better term, we call it… Googlified Messaging™.
Well, what are you waiting for? Go read the entire recipe and tutorial. Great stuff!
Tags: asterisk, google, Google Voice, nerd vittles, voicemail transcription, voip
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Skype For SIP Marries Skype and IP-PBXs
Today, Skype announced it is enabling SIP-based IP-PBX to connect to the Skype network, which will allow low-cost SkypeOut calling, receiving calls from Skype users, and receiving calls from regular PSTN phone lines. Outbound calls from IP-PBX SIP handsets to Skype phones is not part of this news announcement. Skype commented it is too difficult to dial Skype usernames from a desktop handset.
Features:
- Receive and manage inbound calls from the 405 million Skype users worldwide on SIP-enabled PBX systems, connecting the company website to the PBX system using Skype click-to-call buttons
- Place calls via Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX, saving your business money with Skype’s low rates
- Purchase Skype online numbers to receive calls to the corporate PBX from landlines or mobile phones
- Manage Skype calls using your existing hardware and system applications such as call routing, conferencing, phone menus, voicemail and call recording and logging - no additional downloads or training are required
Skype For SIP is perfectly suited to businesses that already have IP-PBXs and want to connect to Skype’s network which offers low-cost calling. Skype for SIP is being launched as a closed beta program, but you can register and try to be part of the beta. Interestingly, Asterisk for Skype, a SIP-based (and IAX) open-source IP-PBX was the first to offer some tight integration with Skype.
If Skype continues to open their proprietary doors (they recently announced they were giving away the SILK codec), and now they’ve finally enabled SIP-to-Skype integration, then we can see a momentous shift from SIP trunking to Skype trunking. Customers will not only choose the lowest cost IP trunk, but also the one with the best quality. Skype is renowned for their high-quality due to their P2P architecture, so companies will look very closely at Skype trunking instead of SIP trunking.
And of course if customers shift to purchasing Skype trunks, the SIP trunks will have to follow suit, which means a win-win for businesses looking for low-cost calling.
Via Skype blog
Tags: ip-pbx, open source, sip, Skype, Skype for SIP, voip
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Skype Now Means Business, Friends The SIP World
Skype, a division of beleaguered eBay, is going corporate. The company today announced that it will play nice with corporate PBX systems that use Session Initiation Protocol (SIP). According to The Wall Street Journal, Skype for SIP product will be introduced as a beta product and will be tested by a limited number of companies.
The […]
Luca’s Top 30 VoIP Leaders on Twitter list
Luca Filigheddu has a Top 30 VoIP Leaders On Twitter post worth checking out. TMC’s Rich Tehrani and I are on the list. Honestly, I haven’t seriously started using twitter until the beginning of this month, so I’m just ramping up my followers and who I follow. So I’m grateful to Luca for still considering me for the list considering my twitter nascency.
Many good people worth following are on Luca’s list which is in alphabetical order. Make sure to check out the list and follow them if you are interested in VoIP, including such topics as Skype, Vonage, SIP, Packet8, Asterisk, FCC regulation on VoIP, etc.
Tags: follow, followers, Luca Filigheddu, twitter, voip
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New Nimbuzz VoIP app for the iPhone and iPod touch
Nimbuzz just released their new iPhone version of Nimbuzz which also supports 3G VoIP “dial up” calling and can turn the iPod touch into an iPhone. The old version was just released into the Apple iTunes store in November, so Nimbuzz is cranking out new version pretty quickly!
The new version features a full dial-pad, and the ability to make VoIP calls to PSTN numbers using SkypeOut, as well as via their 10 VoIP partners including Gizmo5, Vyke, sipgate and A1 by leveraging SIP. You can now add individual buddies from AIM, Google Talk, Windows Live Messenger (MSN), MySpace, Yahoo!, and Nimbuzz.
If Wi-Fi is unavailable you can make VoIP calls to Nimbuzz buddies using what Nimbuzz calls “Dial-Up VoIP”, which is available in over 50 countries.
Dial-Up VoIP simply means that Nimbuzz dials a local access number that your iPhone dials and then Nimbuzz’s VoIP servers terminate the call. Jajah and others have this feature as well.
According to the Nimbuzz blog post, “We are experimenting with Twitter, and you can post to Twitter via the Personal Message feature! Try it. Your comments are always welcome, so please feel free to give feedback.” Wow! Twitter integration with a VoIP app. Gotta love it!
Fixes:
• Facebook names are displayed
• Mobile Me usernames with a dot are now supported
• Improved stability
Tags: 3G, apple, iphone, nimbuzz, skype, skypeout, voip, wifi
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- Apple Large screen iPod Touch in Future Plus Skype & Apple Partnership? - Dec 31, 2008
- fring Adds VoIP to iPhone - Oct 04, 2008
- Om Interviews Skype CEO - Sep 12, 2008
- T-Mobile VoIP Shenanigans - AGAIN! - Jul 18, 2008
- Truphone on iPhone Apple Store enables VoIP - Jul 11, 2008
- sipgate adds SIP-based VoIP support to iPhone - Jun 04, 2008
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Dotcom-Monitor announces new SIP Monitoring tool
Today, Dotcom-Monitor announced a new SIP monitoring tool to add to its portfolio of external monitoring services. It’s similar to other web-based Monitoring-as-a-Service (MaaS) services which monitor the uptime of web servers and notify when a problem occurs. In this case, Dotcom-Monitor’s SIP Monitoring service monitors on-premise or hosted IP-PBXs.
How’s it work? Dotcom-Monitor’s SIP monitoring service makes live intermittent SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity. When a problem is detected, the SIP monitoring notification feature sends an alert via phone, pager, email, or SMS. Basically ,it acts as a SIP end client, placing an actual telephone call to a specified number, and checking the results of that call. The expected result of the call is setup as “Answer”, “No Answer”, “Busy”, or an Error Condition (if there is an unexpected result).
According to their representative, “real-time connectivity status reports are provided via an intuitive online Dashboard interface offering sufficient detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for managing VoIP systems and components, including Service Level Agreement (SLA) compliance issues.”
I’m going to talk to then next week to find out more. For now, check out the news release…
Dotcom-Monitor Enhances Unified Suite of Monitoring Services with SIP Monitoring for VoIP Systems
Easy-to-Use, Cost-Effective External Service Monitors and Analyzes SIP Systems or Infrastructure for Uptime and Performance
Minneapolis, Minn. − March 18, 2009 − Dotcom-Monitor, (www.Dotcom-Monitor.com), a leading provider of externally-hosted network monitoring services, today announced the addition of a cost-saving SIP monitoring service to the company’s unified suite of monitoring capabilities. Today’s announcement adds another critical tool to Dotcom-Monitor’s portfolio of external monitoring services, which includes uptime and performance monitoring of websites, web applications, and Internet network infrastructure.
Dotcom-Monitor’s new SIP monitoring service makes live periodic SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity. When a problem is detected, the SIP monitoring notification feature sends an alert via phone, pager, email, or SMS. Additionally, real-time connectivity status reports are provided via an intuitive online Dashboard interface offering sufficient detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for managing VoIP systems and components, including Service Level Agreement (SLA) compliance issues.
“Due to SLA requirements and hybrid VoIP traffic routes, it is important for VoIP monitoring to proactively mimic the end-user’s perspective from external locations, rather than only relying on passive internal network analysis systems,” said Vadim Mazo, founder and chief technical officer of Dotcom-Monitor. “Many organizations’ VoIP monitoring and uptime needs are best addressed by a simple, cost-effective external system, rather than a large, expensive in-house system. Dotcom-Monitor’s SIP monitoring service provides customers a unique, easy-to-use, targeted solution for quickly identifying and pinpointing VoIP connectivity error conditions,” noted Mazo.
The new SIP monitoring service can be configured and managed with little or no IT expertise, which is ideal for the growing number of small and mid-sized businesses (SMBs) with on-premise or hosted IP-PBXs. Its proactive monitoring ensures connectivity errors can be addressed before the errors become downtime problems for customers. Dotcom-Monitor’s SIP monitoring service ensures SMBs can rely on their VoIP systems, Service Providers can monitor their VoIP infrastructure, VoIP Wholesalers can monitor Service Provider connectivity and reliability, and VoIP VARs and managed service providers can count on client uptime and revenue.
“As the VoIP ecosystem continues to grow in scope and complexity the need for simple and affordable SIP monitoring has never been greater,” said Jonathan Fuld, CTO of SIP Print, the only provider of pure, affordable SIP call recording systems for SMBs. “In fact, SMBS and any cost-conscious organization that is dependent on SIP-based communications could benefit by investigating an externally hosted SIP monitoring provider like Dotcom-Monitor.”
Dotcom-Monitor’s SIP Monitoring is available immediately by visiting: www.dotcom-monitor.com
Tags: Dotcom-Monitor, phone, SIP Monitoring, sla, sms, voip
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How to make OCS 2007 R2 non-RFC 3966-compliant using RemovePlusFromRequestURI
Office Communications Server 2007 Mediation Server uses a plus sign (+) to prefix E.164 numbers in the Request Uniform Resource Identifier (URI) for outgoing calls. Unfortunately, some IP-PBXs don’t comply with RFC 3966 and do not accept numbers that are prefixed with a plus sign (+).
As the UCSpotting blog points out:
To make sure that OCS 2007 operates correctly with non-RFC 3966-compliant PBXs, Microsoft released an update for Mediation Server (R1), which is described in KB articles 952780 and 952785. After installing the update, it’s necessary to create a configuration file - MediationServerSvc.exe.config - with the following content:
"1.0" encoding=“utf-8″ ?> <configuration> <appSettings> <add key=“RemovePlusFromRequestURI” value=“Yes” /> </appSettings> </configuration>
In OCS 2007 R2, Microsoft changed this slightly negating the need for the above configuration file. There’s a new WMI setting, RemovePlusFromRequestURI , which is described in this TechNet article called Enterprise Voice Server-Side Components.
According to the TechNet article, Office Communications Server 2007 R2 introduces two new Windows Management Instrumentation (WMI) settings for Mediation Server. The first new setting specifies how Mediation Server processes E.164 numbers in outbound calls. The second new setting enables Quality of Service (QoS) marking on Mediation Server.
Handling E.164 Numbers in Outbound Calls (OCS 2007 R2)
By default, E.164 numbers in the Request Uniform Resource Identifier (URI) for outgoing calls are prefixed with a plus sign (+). Most Private Branch eXchanges (PBXs) process such numbers without problem. Certain PBXs, however, do not accept numbers that are prefixed with a plus sign.
To ensure interoperability with these PBXs, Mediation Server has a new WMI Boolean setting called RemovePlusFromRequestURI, which has two values: TRUE and FALSE. If your PBX does not accept numbers prefixed with a plus sign, the value for the WMI setting should be set to TRUE, which causes Mediation Server to strip the plus sign from a Request URI for outbound calls. The default is FALSE, which causes Mediation Server to pass the outgoing INVITE’s Request URI, To URI, and From URI unchanged.
The TechNet article also discusses compatibility with PBXs that do not support the plus (+) sign.
By default, E.164 numbers in the Request URI of outgoing calls from Office Communications Server 2007 R2 are prefixed with a plus sign. Most PBXs process such numbers without problem. Some PBXs, however, do not accept numbers that are prefixed with a plus sign and do not route those calls correctly.
Additionally, the From headers of inbound calls from some PBXs does not conform to RFC 3966 because they are not prefixed with a plus sign. Microsoft Office Communicator cannot resolve these numbers to the correct user.
To assure interoperability with these PBXs, Office Communications Server 2007 R2 has a new Mediation Server setting for WMI called RemovePlusFromRequestURI. This setting can be set to YES or NO. The default value is NO.
- If a PBX downstream from the Office Communications Server 2007 R2 Mediation server does not accept numbers prefixed with a plus sign, set the value of RemovePlusFromRequestURI to YES. This causes Mediation Server to remove the plus signs from the Request URIs of outgoing calls. It also causes the plus signs to be removed from the To and From URIs.
- If the downstream PBX accepts numbers prefixed with plus signs, leave the value of RemovePlusFromRequestURI set to its default value of NO. This causes Office Communications Server 2007 Mediation Server to pass Request URIs, To URIs, and From URIs unchanged (that is, with plus signs).
UCSpotting’s excellent article explains all this, and includes a nice VBscript for how to change the boolean value (true or false). Check it out!
Tags: Microsoft OCS 2007 R2, OCS 2007 R2, RemovePlusFromRequestURI, e.164, outbound calling, plus sign, ucspotting, unified communications
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- OCS 2007 R2 Online Labs - Feb 03, 2009
- Microsoft and IBM Announce Sametime and Microsoft OCS integration - Nov 13, 2008
- VoIPshield Finds Security Flaws in Microsoft Office Communications Server 2007, Office Communicator and Windows Live Messenger - Nov 12, 2008
- What’s New in Administration and Management with Office Communications Server 2007 R2 - Oct 31, 2008
- Microsoft Office Communication Server R2 ships in December plus Hosted OCS Coming - Sep 23, 2008
- Microsoft OCS 2007 R2 (next release) to be 64-bit Only - Aug 29, 2008
- Windows Server 2008 RDS Does VoIP - Mar 11, 2009
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Microsoft OCS 2010 Will Finally Eliminate the PBX
Well, Microsoft has let the cat out of the bag and leaked word that Microsoft OCS 2010 will “remove the need for PBX equipment within your organization”. I’m certainly not surprised. Let’s flash back to last year where I wrote and article titled Microsoft OCS 2007 R2 Heralds the Death of the IP-PBX. In it I wrote:
“Office Communications Server 2007 R2, debuting just one year after the Microsoft unified communications launch, highlights the pace of innovation that is possible with software,” said Stephen Elop, president of the Microsoft Business Division at Microsoft. “This new release puts Microsoft on a rapid path to deliver voice software that does much more than a network private branch exchange (PBX) and with much less cost.”
Interesting quote, eh? Does this not sound like Microsoft is sounding the death knell for the network PBX (IP-PBX)? This is an interesting turn of events. Microsoft hasn’t been pitching OCS 2007 as an IP-PBX replacement, but rather as something complementary. In fact, I remember talking with Microsoft about this last year (2007) and they went out of their way to explain that OCS 2007 is not an IP-PBX replacement. Also, Microsoft has many IP-PBX partners in the OCS 2007 arena, including Mitel, Nortel, and others. Slip of the tongue? Or is Microsoft going full-out into the IP-PBX arena? Certainly, the fear by many IP-PBX vendors is that one day Microsoft will offer a full-fledged software-based IP-PBX replacement, but I don’t think that day has come yet - even with the new features in OCS 2007 R2.
Now with OCS 2007 R2 fully launched and with added support for direct SIP trunking, the next logical step is a 100% Microsoft UC solution without the need for a PBX/IP-PBX at all. Of course, Microsoft OCS 2007 R2 is still currently very limited in the support it has for SIP IP phones. Most businesses aren’t ready to toss desktop phones for a 100% software-based softphone solution, i.e. Microsoft Communicator. So OCS 2010 will have to support SIP phones from popular SIP phone players such as Aastra, Polycom, and snom. Perhaps Microsoft will borrow or acquire the technology from SmartSIP, which recently launched an add-on for OCS 2007 R2 that enables any SIP phone to work with OCS.
So where did I hear that Microsoft was aiming to eliminate the need for a PBX in OCS? I discovered the information within a document on Microsoft’s website titled ‘Microsoft Unified Communications Business Value Tool’. On Page 24 it states:
You will deploy Office Communications Server 2010, which expands on the communications capabilities delivered in OCS 2007 R2. This release is designed to remove the need for PBX equipment within your organization and replace it with an integrated communications system that dramatically reduces management costs and gives end users innovative tools to communicate and collaborate across geographic boundaries from their office, home or on the road.
Not only do they state they will eliminate the PBX, but they declare the next version name of OCS (OCS 2010), which as far as I know Microsoft hadn’t announced yet. Many UC/VoIP experts predicted that eventually Microsoft would attack the IP-PBX space alone, but one has to wonder if alienating their IP-PBX partners is such a good idea. One of their strongest OCS partners is Nortel, who is experiencing financial difficulties and is probably not in a position to pressure Microsoft to back off. Mitel is another strong partner as well that could be impacted by Microsoft’s decision. Of course, Nortel and Mitel could still go after the SIP-based IP phone space within the OCS arena, but the IP phone market is much more of a commodity with a much lower margin than a full-fledged IP-PBX. Of course, there’s always the high-end media phone market with large margins. For instance, Polycom recently announced their VVX1500 media phone, which created some buzz.
I doubt OCS 2010 will have all the advanced call center functionality you get from Nortel, Avaya, Mitel, etc. After all, this will be Microsoft’s first release that doesn’t rely on the IP-PBX to do the intelligent call routing & handling. They’ll probably have some rudimentary call queues and skills-based routing, but not much else. Don’t expect predictive dialing in OCS 2010, a mainstay of the call center market. Still, a 100% software-based IP-PBX with unified communications capabilities will be a compelling choice for many businesses.
Tags: microsoft, Microsoft OCS, Microsoft OCS 2007 R2, mitel, nortel, sip, SmartSIP, unified communications, voip
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- Microsoft and IBM Announce Sametime and Microsoft OCS integration - Nov 13, 2008
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- Microsoft OCS 2007 R2 (next release) to be 64-bit Only - Aug 29, 2008
- Microsoft Drops VoIP from Windows Live Messenger - Jun 26, 2008
- Dell VoIP Products Analysis - Jan 23, 2008
- Microsoft OCS 2007 Review - Oct 16, 2007
- Microsoft and Nortel to co-develop IP-PBX - Jul 18, 2006
- Windows Server 2008 RDS Does VoIP - Mar 11, 2009
- Sametime and Microsoft OCS integration - Nov 10, 2008
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Windows Server 2008 RDS Does VoIP
Terminal Services allows you to remotely run applications as well as perform remote administrative duties on servers. It has allowed remote audio to be streamed over IP from the remote computer to your local computer (audio redirection) but has never allowed the microphone or line-in port to be redirected. If Microsoft did, you could do VoIP. Of course, you’d have to redirect from the local PC to the remote server and not the other way around. Well read on…
Continue reading Windows Server 2008 RDS Does VoIP…
Tags: audio recording, audio redirection, microphone, microsoft, skype, Terminal Services, unified communications, vdi, Virtual Desktop Infrastructure, voip, Windows Server 2008 RDS
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- Microsoft and IBM Announce Sametime and Microsoft OCS integration - Nov 13, 2008
- Sametime and Microsoft OCS integration - Nov 10, 2008
- VoxOx Skype killer? - Nov 03, 2008
- What’s New in Administration and Management with Office Communications Server 2007 R2 - Oct 31, 2008
- Microsoft OCS 2007 R2 Heralds the Death of the IP-PBX - Oct 14, 2008
- Microsoft Office Communication Server R2 ships in December plus Hosted OCS Coming - Sep 23, 2008
- ITEXPO West 2008 a Resounding Success - Sep 18, 2008
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Polycom VVX 1500 Media Phone Game Changer?
Today, Polycom has launched the Polycom VVX 1500 touch-screen business media phone, a new VoIP phone that combines IP telephony with business-class video and the ability to integrate with business applications. Recently, Verizon make a big splash with their consumer-class Verizon Hub, a multimedia phone that combines VoIP, Internet access, color screen, video streaming, and more. One could easily make the case that the Polycom VVX 1500 is the “business-class” version of the consumer-oriented Verizon Hub phone.
Although there are many similar features and both could be classified as “media phones“, the Verizon Hub does not do video conferencing, since it does not have an embedded camera. The Polycom VVX 1500 on the other hand does have a video camera embedded and is therefore more suited to video conferencing, which is more prevalent in the business world any way.
The Polycom VVX 1500 combines a personal video conferencing system with a fully featured voice over IP (VoIP) telephone along with Polycom HD Voice (wideband telephony) and an open application programming interface (API) and microbrowser for real-time delivery of personalized Web content. It also includes a color touch-screen interface making this a very unique business IP phone.
So is a business-class media phone with a color touch-screen, web browsing, and video conferencing capabilities a game changer in the VoIP space? Well, the VVX 1500 has a list price of U.S. $1,099, so this is not an IP phone for everyone’s desk in a corporate office. A decent IP phone for the every day worker can be had for $150-$300 which is much less expensive. However, for business executives, CEOs, VPs, and other high-level management, the VVX 1500 is a very attractive IP phone. Often times if a VP or CEO has to have a high-quality video conference, they have to reserve a high-quality video conferencing system located in a particular boardroom. With the VVX 1500 they can stay at their desk and have their meeting. Further, impromptu video conferencing with co-workers sporting a VVX 1500 on their desk can be had allowing for quick collaborative meetings.
In-Stat is very high on the prospects for business-class media phones. According to Keith Nissen, principal analyst at In-Stat, “We anticipate that within five years, nearly 10 million business media phones will be shipped worldwide, generating more than U.S. $3 billion in annual revenue. They are a key to the future of the IP PBX business.” He added, “With its rich heritage in voice and visual communications and content sharing, Polycom is well positioned to be a leader in this new world of communications. The company’s VVX 1500 is the first business media phone that enables customers to work more efficiently and effectively than ever before by tying together voice and visual communication with critical business processes.”
Polycom VVX 1500 Touch-Screen
“There is growing demand from our service providers and customers to help them configure video within our BroadWorks call control platform,” said Mike Tessler, CEO of BroadSoft. “We have a long history of teaming with Polycom to deliver high quality hosted VoIP solutions, and the VVX 1500 is especially compelling because it goes far beyond the functionality of a traditional video phone by combining rich telephony, business-class video and an applications platform that is all deeply integrated with the BroadWorks platform, and it is extremely easy-to-use.”
The VVX 1500 was also specifically designed for lower power consumption, using power over Ethernet (PoE) using IEEE 802.3af, and requiring less than half the power of similar competing products such as traditional video phones. The device’s cool smart-motion technology enables the screen to go into power-save mode when no one is in the office.
The VVX 1500 features an open API and microbrowser that enable third-party application developers to integrate VVX 1500 with business applications such as unified communications, customer relationship management (CRM), and appointment management systems. The always-on, touch-screen user interface of the VVX 1500 includes a menu screen on which developers can place icons for users to locate and start their applications.
Polycom VVX 1500 Profile View
The VVX 1500 comes bundled with several applications including the Polycom Productivity Suite, which enables users to initiate and control audio conference calls right from the device’s screen as well as record calls locally using a flash drive in the phone’s USB port. The VVX 1500 also features a free Web service called My Info Portal through which customers can select to receive content such as local weather reports and other personalized information on the screen when the device is not in a voice or video call.
Interoperability is not a problem since the VVX 1500 uses the same Session Initiation Protocol (SIP) software as incorporated in Polycom’s SoundPoint IP and SoundStation IP desktop and conference phone product lines to communicate with SIP based IP-PBXs and hosted SIP servers. The product is in the process of being SIP video-certified by Polycom’s ecosystem of more than 30 VIP and VoIP Field Verified call control partners, including BroadSoft, Deltapath, NEC Sphere, Objectworld, and Zultys.
“Our customers consistently seek better leverage of their communication systems to improve productivity and reduce costs. They also expect Polycom to continuously deliver innovative, intuitive products to market,” said Sunil Bhalla, senior vice president and general manager of Voice Communications Solutions at Polycom. “Our leadership and legacy in both voice and video communications enables us to develop a truly unique device. The VVX 1500 is the business media phone to combine a superior business-grade VoIP telephone that features our renowned HD Voice with one-touch video and access to key enterprise applications. We’re delighted propel collaborative communications to the next level with this ground-breaking device.”
The Polycom VVX 1500 will be available this month through Polycom’s channel partner network at a list price of U.S. $1,099. To learn more about the Polycom VVX 1500, visit www.polycom.com/vvx1500.
Tags: broadsoft, ip phone, ip-pbx, keith nissen, mike tessler, polycom, sip, SoundPoint IP, SoundStation IP, sunil bhalla, video, video conferencing, voice, voip, VVX 1500
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- BroadSoft, Fonality, and ITEXPO news all rolled in one - Jul 16, 2008
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SmartSIP Launches for OCS 2007 R2 Enabling Any SIP Phone & Any SIP Trunking Service Provider
OCS 2007 R2 won’t replace your PBX just yet. However, their latest R2 version adds the ability to do direct SIP trunking, thus bypassing the need for an IP-PBX.
One drawback however is that Microsoft only supports direct SIP trunking with two providers, namely Global Crossing and Sprint. Well that’s pretty lame, considering their are dozens of decent SIP trunking service providers and probably hundreds across the entire world.
Fortunately, Mike Stacy an OCS 2007 guru, over at Evangelyze Communications has some products that enhance OCS 2007 R2 functionality. One such product is SmartSIP which launches tomorrow. According to Mike, the first dot release due next month will add the capability to use standard SIP phones with OCS. Currently, you have limited options namely Tanjay or Snom phones, but with SmartSIP you can use a Polycom IP phone, an Aastra IP phone, or dare I say, a Cisco IP phone connected to OCS 2007 R2. With the Cisco SIP firmware load of course.
Tags: aastra, cisco, Evangelyze Communications, ip phone, microsoft, Microsoft OCS 2007 R3, mike stacy, ocs 2007 r2, polycom, service provider, sip, SIP trunking, smartsip, unified communications, voip
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- Microsoft Office Communication Server R2 ships in December plus Hosted OCS Coming - Sep 23, 2008
- Objectworld Offers Unified Communications for 7 Cents a Day - Jul 22, 2008
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- Hmmm… Buy an IP-PBX at Costco - May 29, 2008
- Cisco and Microsoft in Clash of the UC Titans - Apr 24, 2008
- Microsoft Response Point Announces SIP Trunking Service Providers - Apr 18, 2008
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SpinVox Transcribes Skype Voicemail & VoiceScribe does the same for Asterisk
Skype users can now have their voicemails converted into text via SpinVox. Today, SpinVox announced that your Skype voicemails transcribed and sent to you via SMS for €0.20/£0.17/25 cents plus the cost of the SMS. SimulScribe, now PhoneTag, is a similar service, that Rich Tehrani uses regularly. GotVoice is yet another one.
But how about another cool TTS app that is currently ‘free’ and works with the popular open source Asterisk platform? VoiceScribe is a beta web-service for Asterisk that converts your voicemail to text and delivers them to you via e-mail. What’s cool about this is how easy it is to integrate with Asterisk, trixbox CE, and trixbox Pro. I tested it with trixbox Pro and it worked flawlessly in just minutes. It uses the Nuance engine. The accuracy was OK, but I’m told by VoiceScribe’s Mitchel Constantin, “Quality will get much better.”
Simply edit /etc/asterisk/voicemail.conf, go to the [general] section and make sure wav49 is the default format. Also add a line with mailcmd that sends an email with your voicemail attachment to their hosted servers.
Here’s a sample of the 4 lines you need in voicemail.conf:
Continue reading SpinVox Transcribes Skype Voicemail & VoiceScribe does the same for Asterisk…
Tags: asterisk, Mitchel Constantin, skype, spinvox, text-to-speech, trixbox ce, trixbox pro, tts, voicemail
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SHSU Switches Back to Cisco CallManager from Asterisk
In 2006, I came across a Network World article, which espoused the fact that Sam Houston State University (SHSU) had switched from the Cisco CallManager IP-PBX to open source Asterisk. I wrote about this news since 6,000 students and faculty were moved off Cisco to the open source Asterisk IP-PBX, which was great news for the open source Asterisk community. This deployment demonstrated that Asterisk could scale and put to rest one of the main complaints against Asterisk.
Well, 3 years have passed, and according to this thread written by Jason Fuermann, who is responsible for SHSU’s IP phone system, SHSU has switched back to Cisco from Asterisk. Say what?
Continue reading SHSU Switches Back to Cisco CallManager from Asterisk…
Tags: asterisk, CCVP, cisco, cisco call manager, Cisco Certified Voice Professional, digium, ip-pbx, Jason Fuermann, tco, voip
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EZCallerID.com Hosted CNAM for Enhanced Caller-ID on any IP-PBX Launches
EZ Call, Inc. today announced the launch of EZCallerID.com, a new service that provides enhanced Caller ID, also known as CNAM, for VoIP calls. The hosted CNAM service gives you not just the phone number, but the name of the person calling.
Most SIP trunking providers do not provide the caller’s name with Caller ID on inbound calls. EZCallerID.com solves this issue by simply having you route your inbound calls to their server. They insert the caller’s name and send the call back to your IP-PBX.
How’s it work? Simply put, EZCallerID.com connects to the national databases that contain the name associated with each phone number, perform a reverse lookup, insert the CallerID info into the From SIP header and then send the call back to you.
This is similar in concept to my recent article, CNAM (CallerID with Name) on Asterisk using Reverse Phone number lookup, but in this case, no Asterisk PBX is required. It works with any SIP-based IP-PBX. It is a hosted offering, so there is a fee of course, but it’s not expensive. They only charge $0.015 per call (or $1.50 for every 100 calls). The service is available on a pay-as-you-go basis ($10 minimum initial charge), with no recurring charges or minimum monthly commitment.
Head on over to EZCallerID.com if you want to sign-up.
Hat tip to Eric Hernaez for the news tip
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Tags: Asterisk, Caller ID, CallerID, CNAM, EZCallerID.com, IP-PBX, Session Initiation Protocol, SIP, voip
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flaphone Enables Free Web-based SIP-to-Skype calls
Today, flaphone (formerly Flashphone) announced that users of their Flash VoIP application can now make a call from flaphone to skype. You simply need to enter sip:skype_username@skype after selecting “none”(global)” for the SIP account. I should mention that flaphone supports multiple SIP credentials, which is a really nice feature. I’ve been testing flaphone for several weeks now and have been meaning to write up their cool Flash-based VoIP application.
In any event, for my first test call I entered sip:tomkeating@skype and pressed the call button. The call was initiated and the call quality was superb!
You can also use this SIP-to-Skype feature for flaphone’s CallMe widgets that you place on your website.
Similarly, Gizmo5 recently launched OpenSky which also enables SIP-to-Skype dialing. However, Gizmo5 calls are free only up to 5 minutes long. For longer calls they are offering a paid service. There is no such restriction that I am aware of with flaphone.
By leveraging Flash, flaphone is cross-platform, has minimal download time, and you can run it from any browser. That and the fast that it supports SIP-to-PSTN calling, SIP URI dialing, and SIP-to-Skype calling, means this is one VoIP app you should check out!
Tags: flaphone, flashphone, gizmo5, sip, skype, voip
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- Flash-to-SIP Video Calls - May 27, 2008
- SippySkype SIP-to-Skype Gateway - Mar 07, 2008
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Build your own SIP-to-Skype gateway using Asterisk
While we wait for Digium’s official SIP-to-Skype gateway, Nerd Vittles today informed me about his very cool recipe that you can use today to build your own free SIP-to-Skype gateway enabling you to use your SIP-based desktop phones connected to Asterisk to make Skype inbound/outbound calls.
Part of the recipe uses SipToSis - SIP to Skype Gateway Bridge Proxy. SipToSis is a piece of software which Nerd Vittles points out “forms the lynchpin of Gizmo’s offering and which lets any Asterisk user create much the same gateway at no cost other than the expense of any Skype Out calls you may choose to make.”
Nerd Vittles explains in his tutorial:
When we’re finished, you’ll be able to call any Skype user in the world from any extension on your Asterisk server by entering either a Skype username or any 10-digit telephone number preceded by an 8 to take advantage of SkypeOut calling rates. You’ll also be able to receive incoming calls from any Skype user on any extension of your Asterisk system. In short, what you get is a transparent interface to several hundred million Skype users from your Asterisk server.
In summary, with this tutorial you’ll be able to dial Skype users, as well as receive incoming calls from any Skype user! Nerd Vittles’ recipe should work on just about any Asterisk-based system. I might have to try this recipe myself later on today. Good stuff!
Tags: asterisk, gateway, nerd vittles, sip, sip-to-skype gateway, SipToSis, skype, voip
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Want to Make Some Sick Money in VoIP?
Garrett Smith over at VoIP Supply has an interesting post titled ‘Sick of not making money on VoIP hardware?’ He’s seeing the glass half empty. I see it half-full, which is why my article is titled ‘Want to Make Some Sick Money in VoIP?’
Garrett writes:
Remember the good ‘ole days? When you actually made fat margins on the VoIP hardware products you sold.
It was great. You made 25, 30, even 35 points of margin (and that’s on top of any services you performed).
Unfortunately those days are long gone. Over the last few years vendors and distributors in the VoIP industry have commoditized hardware and driven margins to the brink of extinction.
Leaving many channel partners frustrated and scrambling to make ends meet. It’s not pretty, but you know it’s the truth.
Imagine how nice it would be if there was a VoIP hardware channel program out there that provided up to 50% margins, street price protection, marketing funds, rebates and a team of individuals dedicated to your success as a channel partner.
You’d jump at the opportunity to be a part of that right?
What if I told you that there IS a channel program that offers all of this and more. You’re not imagining this. This channel program really does exist.
The program? It’s from VoIP Supply. Together with QuickPhones we have put together a channel program for the hottest new wireless VoIP product on the market - the QuickPhones QA-342.
Wow, 50% margins on a Wi-Fi SIP phone? Sweet!
QuickPhones QA-342 features a 112 x 64 pixel monochrome display with backlight, 14 hours of talk time and 7 days standby.
Other features include:
- FCC & CE compliant, IEEE 802.11b/g, WEP/WPA/WPA2, WiFi Protected Setup, G.711
- Tested with Asterisk Open Source PBX, Trixbox, Elastix, FreeSwitch and other platforms
- Phonebook
- Call history
- WLAN signal strength level
- Battery level
- Caller ID
- Key Lock
- Contacts
- Clock
- Up to 6 languages
- Auto search & association upto 4 APs
So here’s the rest of the offer as explained by VoIP Supply’s Garrett Smith:
Today we are looking for 50 47 (three new partners joined today) who want to invest in a product line with a channel program built from the ground up by those who know what it is like to be in YOUR shoes. In exchange for your investment you’ll receive:
Go check out his full post here for more info and if interested in signing up to their channel program.
Tags: channel program, Garrett Smith, QuickPhones QA-342, voip, VoIP Supply, wi-fi phone
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