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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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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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Tags: asterisknowasterisknow 1.5.0, voip, asterisk, yum, linux, centos 5.3
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Digium Launches Support Services for Asterisk
Some big news from Digium. Rich Tehrani met with them yesterday to get the inside scoop. Rich takes copious notes on his iPhone, which he sent off to me to try and write up this news. Alas, I’ve been pretty busy myself, but I wanted to share Rich’s notes below, since there are some good “nuggets” in there.
For instance, from Rich’s notes I see that Switchvox 4.0 is on the verge of shipping. But the really big news is that Asterisk has announced the general availability of technical support subscriptions for open source Asterisk. Before if you wanted support from Digium, you had to purchase Asterisk Business Edition. Well, no longer. Now, all of you Asterisk fans out there that try Asterisk and get stuck can now contact Digium and get some support. No more relying on the Asterisk community to answer you questions. Not that asking the Asterisk community is a bad thing, but if you phone system is down, you can’t wait hours for someone to respond to an online posting. This could be a huge revenue-generating opportunity for Digium, which can now monetize the open source version of Asterisk with support subscriptions. I’m surprised they didn’t offer it sooner. Maybe they were afraid it would upset channel partners?
Rich’s notes:
- Open Source Subscriptions
- 2 smb subs
- And 2 enterprise class
- Incident based
- Problem: up to today needed community support or consultant with hourly rate
- Now annual sub - 3 year 10% discount
- Can call Digium based
- Level one - support local hours 12 hours - starting at your 8:00 - 7:00
- For 5 days a week
- Buys sub online
- Available in a month through the channel
- Get a key, name contact and get details when you call
- Get incident/case handled
- Can open via we or phone
- Find a bug - gets entered in bug tracker
- Gets handled like any biz edition type of bug
- Not really SLA like a commercial licensed product
- Biz edition - now only available as OEM or commercially licensed product
- They want people to buy the open source - engineering opens up 1.4 and 1.6 - first time Digium provides support for open source asterisk
- Up till now consultants, etc
- Open source - people buying business edition for support reasons
- Now getting open source subs
- Can now support enterprise class apps
- In the past - anyone who built a large network - 2 levels of enterprise class support
- 24×7 - server based
- Unlimited users
- Up to 3 names contacts
- First foray into enterprise from server side
- Up to 24×7 support
- Switchvox 4.0 on the verge of shipping
The new Asterisk support services enable companies to leverage the power of open source Asterisk with the confidence that their system will be supported by the very founders of the Asterisk movement. According to the news release, “The support subscriptions provide technical support, hardware replacements and substantial discounts on training programs to enable users to take full advantage of the power of the Asterisk platform.”
“Digium’s new subscription services give Asterisk users the best of both worlds–they can download and use Asterisk free of charge, as always, and now they can also call on Digium for technical support when needed,” said Spencer. “We think the combo of free and open, with support, is going to appeal to many of our most technical users. The Asterisk community has long been a source of great expertise through online forums, and now we’re supplementing that with the ability to call us, 24×7, for access to our Asterisk experts.”
Danny Windham, CEO of Digium, said: “As Asterisk gains traction within large businesses, demand for professional support is on the rise. Our deep knowledge of open source Asterisk and total commitment to its development makes us ideally suited to offer these new services. Companies that purchase subscriptions will receive support from the most knowledgeable group of Asterisk experts in the industry. We see this offering as a substantial step forward for Asterisk in the enterprise and a valuable service for companies of all sizes.”
Asterisk support subscriptions are bundles of services sold on an annual basis. They include technical and engineering support, consultative services, advance hardware replacement, and discounts on Asterisk training and conference passes.
Asterisk support subscriptions are available immediately from the Digium webstore at http://store.digium.com and will be available through Digium channel partners in Q2. SMB pricing begins at U.S. $595 per year for support during the subscriber’s business hours (8:00 a.m.-5:00 p.m., Monday through Friday); 24×7 support for an SMB begins at U.S. $1,995 per year. Enterprise subscriptions, including 24×7 support, begin at U.S. $3,995 per year. Pricing includes a defined number of servers supported and cases opened per year.
You can read the official news announcement here.
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Tags: technical support, subscriptions, asterisk, open source, digium, ip-pbx, linux, switchvox 4.0, voip, mark spencer, danny windham
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Bring Out Your Dead! Wait! Skype for Asterisk is not dead!
Digium has an excellent post today titled The Rumors of Our Death discussing Skype for Asterisk (SFA) and the recently launched (beta) SkypeforSIP (SFS). There has been much discussion on the blogosphere, twitter, and elsewhere if SFS means the death of SFA. Some were even seen carting SkypeforAsterisk away into the trashbin of other failed software endeavors, as seen here:
It’s not a pretty sight when people write you off for dead when you’re really not. But wait just a second. Digium’s Steve Sokol explains late today that SFA is not dead. He writes:
With Skype’s recent announcement of Skype For SIP there has been a great deal of pontification on the impending death of the not-yet-released Skype For Asterisk. I’d like to take a moment to explain why Skype For SIP (SFS) does not spell the end for Skype For Asterisk (SFA), and why Skype For Asterisk is still in beta.
First, the key differences between Skype For SIP and Skype For Asterisk:
Steve lists more differences, but I don’t want to steal his thunder. Go read his post. I knew there were key advantages in SFA over SFS and there was so much confusion out there, I was tempted to write a comparative chart, but was too busy. In any event, it’s nice to see Digium clarifying the advantages of Skype for Asterisk. Any questions?
Tags: asterisk, sfa, sfs, skype, skype for asterisk, SkypeforSIP, voip
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trixbox 2.8 beta is out
In case you missed it, Andrew posted that trixbox 2.8 beta is available and is based on the latest version of Asterisk 1.6. Andrew, is even very complimentary over Asterisk 1.6 & Digum, which is nice to see coming from a Fonality employee:
We have been testing Asterisk 1.6 and DAHDI (the replacement for zaptel) since they came out last year. I am happy to say they are coming along quite nicely. Digium has worked hard on 1.6 with a lot of attention paid to reliability and scalability. I am happy to say our testing shows Asterisk 1.6 is ready for prime time!
You can grab trixbox 2.8 beta .iso image here
Tags: asterisk, asterisk 1.6, trixbox, trixbox 2.8 beta, 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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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 […]
MFC/R2 support added to Asterisk 1.6.2
Asterisk has just added ‘official’ MFC/R2 support for chan_dahdi. Here’s the commit. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva.
According to the commit:
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others.To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample.
Via Russell Bryant
Tags: asterisk, chan_dahdi, digium, LibOpenR2, MFC/R2, Moises Silva
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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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- trixbox Pro 2.0 review - Sep 04, 2008
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- Skype for Asterisk Launches - Sep 25, 2008
- ITEXPO West 2008 a Resounding Success - Sep 18, 2008
- Text-To-VoIP Plug-in for MorphVOX Pro - Aug 08, 2008
- Using Asterisk to Scam Credit Cards - Aug 04, 2008
- Asterisk Wake-Up calls and Web Scheduling - Feb 25, 2008
- PIKA for Asterisk boards now trixbox CE compatible - Jan 22, 2008
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PIKA WARP Appliance Adds BRI Support
PIKA Technologies announced today the release of a BRI expansion module for the PIKA WARP Appliance. The PIKA WARP Appliance is a very flexible hardware telecom appliance that can run various flavors of Asterisk, including native Asterisk, Schmooze, trixbox CE, and others. They even support FreePBX, the popular front-end GUI for Asterisk. They support FreeSwitch as well.
PIKA’s BRI module supports two ports and four channels, allowing the WARP Appliance to reach a total port density of four ports and eight channels when two BRI modules are installed. BRI is very popular in Europe and is very commonly used in the SMB space, making the WARP Appliance a suitable option.
Check out my PIKA WARP Appliance for Asterisk review for more details on this flexible piece of hardware.
Tags: asterisk, BRI, freepbx, PIKA, PIKA WARP Appliance, schmooze, voip
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- PIKA WARP Appliance adopted by Schmooze - Sep 16, 2008
- PIKA WARP Appliance for Asterisk Review - Sep 12, 2008
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- PIKA T1/E1 and analog boards now compatible with FreeSWITCH - Apr 15, 2008
- Sangoma Technologies Offers Lifetime Warranty - Mar 18, 2008
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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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- Digium Acquires Astricon - Jul 11, 2007
- Digium - The Showstopper! - May 18, 2007
- Digium podcast of Mark Spencer’s new role - Jan 30, 2007
- Signate, an Asterisk provider, bites the dust? - Dec 08, 2006
- Asterisk Receives VC Funding - Aug 08, 2006
- Another IP-PBX company bites the dust? - Aug 08, 2006
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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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- Skype for Asterisk Launches - Sep 25, 2008
- Microsoft Response Point Adds T1 Support and SIP Trunking Service Providers - Sep 17, 2008
- PIKA WARP Appliance adopted by Schmooze - Sep 16, 2008
- PIKA WARP Appliance for Asterisk Review - Sep 12, 2008
- Fonality’s trixbox Pro Unified Agent Edition integrates with Salesforce.com - Sep 05, 2008
- Digium Headquarters Tour - Aug 26, 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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- SIP to Skype gateway breaks Skype’s Great Wall of VoIP - Feb 06, 2006
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- SippySkype SIP-to-Skype Gateway - Mar 07, 2008
- Future of SIP to Skype Gateway in Doubt? - Feb 04, 2008
- Skype SIP Gateway (PE) 1.0 Released - Oct 30, 2007
- SkyStone bridges Skype and PBXs using only software - May 15, 2007
- VoIP Gateway clones an iPod - Nov 27, 2006
- OpenSky enables SIP-to-Skype Calls - Feb 10, 2009
- Dual Stack SIP and Skype IP Phone Coming - Jan 30, 2009
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Bootable Flash Drive creates 15-Minute Turnkey Asterisk Installs on Atom-based processors
How’d you like a bootable USB flash drive which can create turnkey, full-featured Asterisk PBX systems in 15 minutes or less? Nerd Vittles has a recipe specially designed for the new Atom-based motherboards found in most netbooks. Nerd says it also works just fine with Dell’s PowerEdge T100 servers.
He writes how this is perfect for on-site demos:
For those that demo systems for a living, no one will touch this presentation. Just show up at a customer site with a $300 Acer Aspire One NetBook and an Aastra 57i business phone. While the customer watches the Atomic Flash build a new PBX in a Flash server from the ruins of a Windows XP clunker, you can connect and configure the 57i and explain how simple VoIP networks can be.
When you finish your 10-minute slide show, your system will be operational. Dial any 800 number from your Aastra phone, and presto… instant, flawless communications!
Check out the article.
Tags: Acer Aspire One NetBook, asterisk, Atom, atom processor, bootable drive, flash, pbx in a flash, usb
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- Schmooze the Yiddish Asterisk - Feb 01, 2008
- Nerd Vittles Delivers some Asterisk Christmas Goodies - Dec 11, 2007
- PBX in a Flash Asterisk Distro Launches - Nov 13, 2007
- Iomega 120 GB Portable Hard Drive - Oct 20, 2005
- ContactQ Enhances Asterisk’s Call Center Functionality - Feb 04, 2009
- Switchvox SMB 4.0 launches at ITEXPO - Feb 02, 2009
- <a href=”http://blog.tmcnet.com/blog/tom-keating/voip/56-reasons-to-attend-digium-asterisk-world-in-miami.asp” title=”
56 Reasons to Attend Digium Asterisk World in Miami”>56 Reasons to Attend Digium Asterisk World in Miami - Jan 13, 2009
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ContactQ Enhances Asterisk’s Call Center Functionality
ContactQ is a new call center application server created by Braxtel Communications designed to run on Asterisk that brings advanced call center functionality to the Asterisk platform. Their aim is to handle any sort of contact method and put it into their advanced multi-media queue. For instance, they plan on queuing video calls, text messages, web callbacks, and of course regular calls. ContactQ is a fully featured multi-media skills based routing ACD. In my meeting with Braxtel at ITEXPO, Lee McCabe, Director of sales said video call queuing brings up some interesting possibilities, such as the ability to play corporate video promotions while the call is on hold. The concept is intriguing and it takes traditional music-on-hold to the next level with video-on-hold.
Lee mentioned that ContactQ is currently being developed as both a commercial product and also as an open source GPL project. It supports industry standards such as SIP, Voice XML(VXML), AJAX (Web 2.0), XMLRPC and designed for inter-working with VoIP softswitches like Asterisk, FreeSWITCH, PingTel and Cisco.
I asked Lee if they leverage FreePBX at all for the front-end GUI or if they use their own and Lee stated they use their own front-end interface for configuration as well as monitoring of call center queues and call center statistics. Importantly, ContactQ sports the ability for call center supervisors to listen in on agents using DTMF/touch tones on their phone.
Features include a powerful IVR with drag and drop programming tool and historical reporting delivered via the web browser. It features powerful dialing capabilities critical to call centers, including Outbound Preview, Progressive and Predictive dialing modes.
Other features include:
- Fully featured ACD supporting
- Multiple queue modes
- Call pull back on no answer
- Overflow to voicemail
- Queued voicemails
- Time / Day routing rules
- Agent unavailable types
- Skills based routing (9999 skill levels)
- Telephony based agent logon
- Web browser based system configuration
- Multi Language support (US English only in Version 1)
- LDAP integration
- Multiple partition configuration
- Role based login
- Supervisor Monitor / Listen-in
- SNMP support
- Real-time Supervisor Dashboard web application providing
- Agent and queue performance statistics
- Agent and queue drill down statistics
- Real-time Agent Dashboard
- Agent performance statistics web application
Lee said their software always uses the latest version of Asterisk with nightly builds available. Installation is via a bootable .iso image which will automatically format and install ContactQ in just minutes. I tried to get some screenshots of the admin from their website, but the website seems a bit of a work in progress. But the feature-set seems pretty powerful and I hope to check it out soon.
Tags: asterisk, call center, ContactQ, ip-pbx, voip
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- PIKA WARP Appliance for Asterisk Review - Sep 12, 2008
- Digium Headquarters Tour - Aug 26, 2008
- trixbox Pro 2.0 review - Sep 04, 2008
- Sangoma Acquires Paraxip Technologies - Jul 07, 2008
- SIP-based VoIP Wake Up Call Service - May 16, 2008
- Shoretel Rumors - May 01, 2008
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Quicknet: What Might Have Been
The technology industry is full of alternate histories, tales of what might have been had things unfolded just a bit differently. We all know about Skype, but did you know that the likes of Skype could have emerged a decade sooner, and almost did, thanks to a scrappy little San Francisco company named Quicknet Technologies?
Stacey […]
Switchvox SMB 4.0 launches at ITEXPO
I just got into Miami for ITEXPO and to the convention center around 12pm and I’ve already learned some exciting news launching at the show. Digium has released Switchvox 4.0 with some new Unified Communications features. Matthew Nickasch, a writer for Considering Convergence on NetworkWorld.com covers the news extensively in his ITExpo East Kicks Off with Big Switchvox News article.
Off the bat I noticed the video phone support with support for H.263 and H.264 codecs. Another interesting new feature is XMPP support for a private chat server. Fonality’s trixbox, a competitor to Switchvox actually also uses a XMPP server for their chat/presence server as well. Fax has never been Asterisk’s strong suit, so the fax integration sounds very intriguing. Check out some of the new features below.
- Fax integration-Users can send and receive faxes quickly and easily via Switchvox.
- Video calling-Switchvox supports video phones that use the codec standards H.263 and H.264.
- Instant messaging-Switchvox includes a private chat server that uses the widely adopted, open XMPP protocol. The Switchvox Switchboard offers a Chat Panel, or users can select their favorite XMPP-based client.
- Centralized presence-Presence and status details for call and chat activity are visible across multiple peered Switchvox PBXs.
- Web-aware interactive voice response (IVR) tools-Switchvox includes many new IVR functions for building custom applications. These sophisticated tools include exchanging sound files with web applications, setting system-wide variables and more.
- Unified messaging enhancements-IMAP integration provides a standards-based solution for voicemail and faxes. Also, users can customize multiple greetings and e-mail notifications optimized for display on a wide range of devices.
- Organized phonebooks-Employees can organize their contacts into groups and keep multiple phone numbers for each contact. Also, a company directory panel uses type-to-find to help users find extensions quickly for any of their coworkers.
- Call queue improvements-Small businesses and call centers alike will benefit from the ability to log into, log out of and pause a member’s status on each queue with a single click, and even add comments that are displayed to supervisors or other queue members.
- Switchvox Notifier-A Windows desktop client provides interaction with MS Office applications. Pop-up notifications show incoming caller details and call history, and one-click options let users quickly add Outlook contacts and dial phone numbers.
- Switchvox Extend-An XML-based API lets administrators create new extensions and access call logs, reports and extension lists.
- Auto-provisioning snom IP phones-Switchvox offers automatic configuration of VoIP phones manufactured by snom technology and Polycom.
- More call options-Switchvox supports BRI-an important step toward expanded international sales, and HD Voice-high-quality wideband audio delivered via the G.722 protocol.
According to the new release, “Switchvox SMB 4.0 is available free of charge to customers with a current Switchvox SMB software subscription. For new customers, pricing for Switchvox SMB remains unchanged and begins at U.S. $3,390 for a 10 user system, including hardware, software, a one-year subscription and warranty. The entire line of Switchvox SMB appliances, the largest of which scales to serve 4″
ITEXPO has barely begun and already some exciting news in the VoIP & Asterisk space. Stay tuned for more news updates…
Tags: asterisk, switchvox, Switchvox SMB 4.0, uc, unifed communications, voip
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- Switchvox SMB 3.5 and AA60 Switchvox Appliance - Mar 18, 2008
- Digium reaches 1 million download milestone - Dec 19, 2007
- Digium launches Switchvox Free Edition - Oct 31, 2007
- Digium Acquires Switchvox - Sep 27, 2007
- Avaya #1 VoIP PBX Vendor? - Aug 21, 2007
- PBX Prompts for Asterisk and other open source PBXs launches tomorrow - Apr 03, 2007
- Running into fellow VoIP bloggers and some VoIP news - Oct 10, 2006
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56 Reasons to Attend Digium Asterisk World in Miami
Bill Miller over at Digium today blogged 5 reasons why you should attend Digium Asterisk World in Miami taking place just a few short weeks away from February 2nd-4th, which is colocated with TMC’s Internet Telephony Conference & Expo (ITEXPO).
Left-to-right clockwise: Steve Sokol, John Todd, Russell Bryant, Dave Rodriguez (TMC), Greg Galitzine (TMC), Jane Brooks, Tom Keating (TMC), Bill Miller
Bill forgot to mention another important reason - all the networking and meeting with industry insiders at ITEXPO, which also includes eating at great restaurants and fun dinner reception parties at the show. Now, the picture above isn’t from ITEXPO - it’s my visit to Digium’s offices - but it may as well have been from ITEXPO since we always go out to fantastic restaurants with major VoIP vendors to discuss the IP communications industry and meet the people who design/engineer or market VoIP products and services.
While there is plenty of important business to be had at ITEXPO, I find that I acquire some of the best information at these dinner parties where I meet industry heavyweights in a more relaxed atmosphere. I’ve pried some NDA information from some vendors after they’ve had a few beers.
I’ll have to remember to snap a few photos at this upcoming ITEXPO at the various after-show functions & parties. I’ll try and make sure to snaps some photos at the ITEXPO Bloggers’ Dinner that Andy Abramson organizes.
I’m very excited to attend since this is the first time Asterisk World will be at ITEXPO. With the global slowing economy, it’ll be interesting to see how well attended the show will be. The pre-show attendance numbers look good, so I’m crossing my fingers.
Tags: Andy Abramson, asterisk, Asterisk World, Bill Miller, digium, ITEXPO, voip
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- TMC & Digium Announce Educational program for Digium|Asterisk World - Dec 15, 2008
- TMC & Digium Announce Registrations open for Digium|Asterisk World - Oct 20, 2008
- TMC & Digium Partner for Digium|Asterisk World at ITEXPO - Oct 06, 2008
- Skype for Asterisk Launches - Sep 25, 2008
- Phonevite API released at ITEXPO - Sep 16, 2008
- Interviews with Top 60 VoIP Movers and Shakers… - Sep 05, 2008
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CNAM (CallerID with Name) on Asterisk using Reverse Phone number lookup
CallerID is cool, but CallerID with Name (CNAM), — also known as Calling NAMe — is where it’s at. When you go with a traditional phone provider (non-VoIP) they’ll often offer you CallerID with Name for an additional fee. But penny-pinching Asterisk & VoIP fans want CallerID with Name too. Unfortunately, too often the SIP trunks or traditional PSTN trunks (analog, T1/E1) connected to your Asterisk IP-PBX don’t provide CallerID with Name - just regular CallerID (number only). So how do we solve this dilemma?
Well, Asterisk sits on an IP network, which of course means it can access the Internet. With access to the Internet, “in theory” a special Asterisk script can take the CallerID number, perform a reverse lookup on AnyWho, Google, and 411.com and then change the CallerID data string before it gets passed to the Asterisk extension.
Well, theory is all well and good, but has it been done? Oh yes it has!
Check out this Perl script I found on Team Forrest’s website that leverages AGI (Asterisk Gateway Interface), a powerful interface that lets your programmatically control Asterisk.
Calling the CallerID with Name Perl script (calleridname.pl) is as simple as calling this single line of code:
exten => s,n(getname),AGI(calleridname.pl,${CALLERID(NUM)})
Here’s the calleridname.pl script:
#!/usr/bin/perl -w
use strict;
use LWP::UserAgent;
$|=1;
sub trim($);
my %AGI; my $tests = 0; my $fail = 0; my $pass = 0; my $result = “”; my $cidnum = “”; my $cidname = “”;
my $npa = “”; my $nxx = “”; my $station = “”; my $name = “”;
$cidnum = $ARGV[0];
while(<STDIN>) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
my $AnyWho = ‘1′ ;
my $Google = ‘1′ ;
my $www411 = ‘1′ ;
if(substr($cidnum,0,1) eq ‘1′){
$cidnum=substr($cidnum,1);
}
if(substr($cidnum,0,2) eq ‘+1′){
$cidnum=substr($cidnum,2);
}
if ($cidnum =~ /^(\d{3})(\d{3})(\d{4})$/) {
$npa = $1;
$nxx = $2;
$station = $3;
}
elsif($cidnum =~/\<(\d{3})(\d{3})(\d{4})\>/){
$npa = $1;
$nxx = $2;
$station = $3;
}
else {
print qq(VERBOSE “ERROR: unable to parse caller id” 2\n);
exit(0);
}
if ($AnyWho > ‘0′) {
print qq(VERBOSE “STATUS: checking AnyWho for name lookup” 2\n);
if ($name = &anywho_lookup ($npa, $nxx, $station)) {
$cidname = $name;
print qq(SET VARIABLE CALLERID\(name\) “$cidname”\n);
print qq(VERBOSE “STATUS: AnyWho said name was $cidname ” 2\n);
exit(0);
}
else {
print qq(VERBOSE “STATUS: unable to find name with AnyWho” 2\n);
}
}
else {
print qq(VERBOSE “STATUS: AnyWho lookup disabled” 2\n);
}
if ($Google > ‘0′) {
print qq(VERBOSE “STATUS: checking Google for name lookup” 2\n);
if ($name = &google_lookup ($npa, $nxx, $station)) {
$cidname = $name;
print qq(SET VARIABLE CALLERID\(name\) “$cidname”\n);
print qq(VERBOSE “STATUS: Google said name was $cidname ” 2\n);
exit(0);
}
else {
print qq(VERBOSE “STATUS: unable to find name with Google” 2\n);
}
}
else {
print qq(VERBOSE “STATUS: Google lookup disabled” 2\n);
}
if ($www411 > ‘0′) {
print qq(VERBOSE “STATUS: checking www411 for name lookup” 2\n);
if ($name = &www411_lookup ($npa, $nxx, $station)) {
$cidname = $name;
print qq(SET VARIABLE CALLERID\(name\) “$cidname”\n);
print qq(VERBOSE “STATUS: www411 said name was $cidname ” 2\n);
exit(0);
}
else {
print qq(VERBOSE “STATUS: unable to find name with www411″ 2\n);
}
}
else {
print qq(VERBOSE “STATUS: www411 lookup disabled” 2\n);
}
print qq(SET VARIABLE CALLERID\(name\) “$cidnum”\n);
print qq(VERBOSE “STATUS: Unknown name for $cidnum ” 2\n);
exit(0);
sub anywho_lookup {
my ($npa, $nxx, $station) = @_;
my $ua = LWP::UserAgent->new( timeout => 45);
my $URL = ‘http://www.anywho.com/qry/wp_rl’;
$URL .= ‘?npa=’ . $npa . ‘&telephone=’ . $nxx . $station;
$ua->agent(’AsteriskAGIQuery/1′);
my $req = new HTTP::Request GET => $URL;
my $res = $ua->request($req);
if ($res->is_success()) {
if ($res->content =~ /<!– listing –>(.*)<!– \/listing –>/s) {
my $listing = $1;
if ($listing =~ /<B>(.*)<\/B>/) {
my $clidname = $1;
return $clidname;
}
}
}
return “”;
}
sub google_lookup {
my ($npa, $nxx, $station) = @_;
my $ua = LWP::UserAgent->new( timeout => 45);
my $URL = ‘http://www.google.com/search?rls=en&q=phonebook:’ . $npa . $nxx . $station . ‘&ie=UTF-8&oe=UTF-8′;
$ua->agent(’AsteriskAGIQuery/1′);
my $req = new HTTP::Request GET => $URL;
my $res = $ua->request($req);
if ($res->is_success()) {
if ($res->content =~ /<font size=-2><br><\/font><font size=-1>(.+)<font color=green>/) {
my $temp = $1;
my $clidname = “”;
if ( $temp =~ /(.+)<font color=green>/o ) {
$clidname = substr($1, 0, -3);
} else {
$clidname = substr($temp, 0, -3);
}
if ($clidname =~ /<a href(.+)\//) {
$clidname = $1 ;
if ($clidname =~ />(.+)</) {
$clidname = $1 ;
}
}
return $clidname;
}
}
return “”;
}
sub www411_lookup {
my ($npa, $nxx, $station) = @_;
my $ua = LWP::UserAgent->new( timeout => 45);
my $URL = ‘http://www.411.com/search/Reverse_Phone?phone=’ . $npa . $nxx . $station;
$ua->agent(’AsteriskAGIQuery/1′);
my $req = new HTTP::Request GET => $URL;
my $res = $ua->request($req);
if ($res->is_success()) {
if ($res->content =~ /Location: <strong>(.*)<\/strong>/s) {
my $temp = $1;
my $clidname = “”;
$temp =~ s/&\;/&/g;
$temp =~ s/%20/ /g;
$clidname = $temp;
return $clidname;
}
}
return “”;
}
For more details, head on over here.
Tags: AGI, asterisk, CallerID, CNAM, perl, reverse phone number lookup
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- trixbox CE vs. Asterisk Downloads - Dec 23, 2008
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- Fonality Launches PBXtra Unified Agent on Salesforce.com’s Force.com AppExchange - Dec 09, 2008
- Digium Responds to FBI Vhishing Security Warning about Asterisk - Dec 09, 2008
- Asterisk-based VPN in a Flash Mobile Telephony Appliance - Dec 08, 2008
- Sangoma Launches B600 Series of Analog Voice Cards - Dec 04, 2008
- PIKA WARP Appliance Adds FreePBX Support - Nov 18, 2008
- Bandwidth.com invests in FreePBX - Nov 14, 2008
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trixbox CE vs. Asterisk Downloads
As you may have read, Digium announced a sharp rise in Asterisk downloads for 2008 registering in a whopping 1.5 million downloads for 2008. That got me thinking just how does Asterisk & AsteriskNOW compare with Fonality’s trixbox CE distro? Arguably trixbox CE is the either the #1 or the #2 downloaded Asterisk-based distribution. But are they #1 or #2? Let the Fonality vs. Digium download battle commence…
First, we head to Sourceforge, which tracks trixbox CE downloads and see how trixbox CE stacks up against Digium’s claimed 1.5 million downloads.
| Date (UTC) | Downloads | Bytes Served |
| Dec 2008 * | 18,050 | 6.8 TB |
| Nov 2008 | 31,000 | 12.2 TB |
| Oct 2008 | 35,187 | 15.6 TB |
| Sep 2008 | 19,449 | 9.6 TB |
| Aug 2008 | 6,948 | 3.3 TB |
| Jul 2008 | 8,329 | 3.7 TB |
| Jun 2008 | 19,993 | 9.3 TB |
| May 2008 | 21,822 | 13.3 TB |
| Apr 2008 | 18,238 | 11.0 TB |
| Mar 2008 | 41,085 | 25.2 TB |
| Feb 2008 | 27,725 | 16.9 TB |
| Jan 2008 | 43,942 | 26.4 TB |
| Total | 291,768 | 153.8 TB |
* Partial data: End of month not yet reached.
Taking a daily average, the final 2008 total downloads should be ~292,587.
That’s 292,587 (year-end estimate) vs. 1.5 million (so far) for Asterisk. So it would appear Digium’s Asterisk beats Fonality trixbox CE by over 5X!
Not so fast though. I was pretty sure the Digium download numbers were for Asterisk and AsteriskNOW combined. AsteriskNOW and trixbox CE are more comparable to each other, since both have FreePBX installed, both are .iso bootable CDs, they’re both nearly plug-and-play distros, etc.
So to be fair to Fonality’s trixbox CE, I should compare AsteriskNOW to trixbox CE. I contacted Digium’s Bill Miller, VP, Product Management, and asked him for the breakdown in their 1.5 million downloads.
I told Bill my intentions of making a side-by-side download comparison, but also pointed out I had to wear my skepticism hat:
“The only caveat to any comparison I make between Asterisk/AsteriskNOW & trixbox CE is that you host your own downloads - not a 3rd party like Sourceforge. So there I have to have a healthy level of skepticism for any numbers Digium cites. Nevertheless, I think it’ll be an interesting discussion if I post these comparative stats in an article.”
Bill explained, “19.8% are AsteriskNOW. We are close to par with them.”. Indeed at 19.8% of 1.5 million equates to 297,000 downloads or roughly 6500 more than trixbox CE. Although, I said above I had to wear my skepticism hat, I think the numbers Bill gave me sound credible.
Bill Miller ads, “Due to the other open source packages they have integrated, and the head start they had, we suspect that they have more in trixbox than AsteriskNOW in production but we are taking downloads and share from them.”
I would agree with Bill that there are indeed more trixbox CE boxes in production, but it does appear from these numbers that Digium’s AsteriskNOW is closing fast in production numbers and already exceeding Fonality’s trixbox CE in downloads.
To be perfectly frank, I’m a little bit surprised AsteriskNOW has surpassed trixbox CE, since I’m sure even Digium would admit that trixbox CE, which was formerly Asterisk@Home, had a huge head start and developed a strong community (see trixbox forums and the sheer number of posts & threads). In fact, Bill Miller mentioned the head start in our conversation. Still, the number of downloads between the two is very close and they are neck-in-neck.
Of course, if you add in regular Asterisk downloads it’s no contest. Digium wins by a landslide.
Tags: asterisk, asterisknow, Bill Miller, digium, downloads, fonality, trixbox ce, voip
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- Schmooze the Yiddish Asterisk - Feb 01, 2008
- Digium Acquires Switchvox - Sep 27, 2007
- Avaya #1 VoIP PBX Vendor? - Aug 21, 2007
- Top 10 Reasons Why I love Asterisk - Aug 03, 2007
- AsteriskNow Now Has 1-Click Features - Jan 24, 2007
- Digium releases AsteriskNOW - Jan 03, 2007
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TMC & Digium Announce Educational program for Digium|Asterisk World
Some good news from my company TMC and Digium, the founder of Asterisk that I thought I’d share. If you’re an Asterisk fan, you need to be at Asterisk World, which is co-located with the #1 VoIP tradeshow in the U.S., namely TMC’s ITEXPO.
Technology Marketing Corporation (TMC) and Digium, the Asterisk Company, today announced the educational program for Digium|Asterisk World, scheduled to take place February 2-4, 2009 in Miami. The event is collocated with TMC’s ITEXPO East 2009.
TMC and Digium also announced that although the conference content and exhibiting opportunities are more than 90 percent complete, additional opportunities remain for interested Digium partners.
Digium|Asterisk World at ITEXPO is the conference that addresses “Everything Asterisk” for business users, resellers and executive decision-makers. The initial conference sessions featured in the program include:
1. Web-Aware Unified Communications with Switchvox
2. Asterisk for Enterprise
3. Enterprise Pitfalls: Lessons Learned
4. The Case for Asterisk: Call Center App Integration with .NET
5. Multi-Site Open Source Call Center Deployment: A European Case Study
6. Asterisk as a Regulatory Compliance Toolkit
7. Website Identity Management and Authentication Using Asterisk
8. VoIP Transparency: Asterisk and the Economics of Monitoring
9. The Asterisk VoIP Conversion and the Opportunity for Substantial ROI
10. Druid: Case Study for Selling UC Solutions
11. Ingredients for Successful Asterisk PBX Sales
Limited booth space and sponsorship opportunities are still available for companies to showcase their offerings and provide training to Digium|Asterisk World and ITEXPO attendees. Companies exhibiting in the showcase include: Asteria Solutions; Camrivox; Freeside Internet Services; High Powered Help; Interlink Communications Systems; OpenLine Networks; Orecx; TransNexus; Voice Pulse; and Xorcom.
Registration for Digium|Asterisk World and ITEXPO is now open.
“Now in our third year, Digium|Asterisk World’s content is evolving to reflect the maturity of the world’s most widely used open source telephony software,” said Mark Spencer, CTO and founder of Digium and the creator of Asterisk. “Our three-day event at ITEXPO will provide the hands-on training and education that Asterisk developers need to build more advanced and more flexible telephony solutions.”
“We’re proud of several new and innovative programs that will make the 19th iteration of ITEXPO one of the best ever,” said Rich Tehrani, TMC president and ITEXPO East 2009 conference chairman. “Digium|Asterisk World bolsters an already strong developer focus, and other programs like TMC University, the 4GWireless Evolution event, TMC Editors’ Week and the balance of the conference content make ITEXPO the one-and-only ‘must-attend’ event in the communications industry.”
ITEXPO East 2009 is the world’s largest and most significant communications technology event, featuring hundreds of companies exhibiting on the EXPO floor and hundreds of sessions led by the industry’s most prominent thought leaders. The show helps attendees identify the issues and challenges affecting the deployment of communications technologies. It provides a comprehensive forum for evaluating the latest products and services and delivers a face-to-face networking opportunity that service providers, carriers, resellers, distributors, equipment manufacturers and IT executives from enterprise and SMB companies need to cultivate new business relationships. For more information on ITEXPO East, please visit: http://www.tmcnet.com/voip/conference/.
Tags: asterisk, Digium|Asterisk World, ITEXPO, tmc, voip
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- Astricon, Win $100, and Party Hardy at ITEXPO - Jul 29, 2008
- Asterisk Wake-Up calls and Web Scheduling - Feb 25, 2008
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- Asterisk-based FreePBX clones Microsoft Response Point’s Easy Button - Jan 31, 2008
- Cool Stuff at ITEXPO - Jan 30, 2008
- ITExpo Prizes - including a Mustang Convertible! - Jan 18, 2008
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Fonality Launches PBXtra Unified Agent on Salesforce.com’s Force.com AppExchange
Back in September I wrote about Fonality’s trixbox Unified Agent Edition (UAE) and how it can automatically match all inbound and outbound calls with the corresponding record in salesforce.com’s AppExchange, and call data is captured and logged. Apparently, this was still a yet-to-be-announced product I wrote about that resulted in a call from Fonality’s CEO Chris Lyman asking how I found out about it. Woops, my bad, Chris.
Well, today, Fonality has officially launched UAE. I spoke with Chris Lyman to learn more about this product. Chris told me, “What we’ve done is allow managers and agents using CRM and a phone system to make some pretty serious decisions about how they run their business and what we consider some major cost savings.”
Chris explained that they pumped the CDR (call data records) from Fonality’s PBXtra into Salesforce.com. Once the CDR data is housed in Salesforce.com obvious business benefits can be realized. You can for instance know when customers called you or when you last called a particular customer. Importantly, with the reporting mechanisms you can understand how long it takes to call a lead or opportunity and how that affects your close rate. The reports allow you to for example easily see “neglected opportunities” such as ‘opportunities that have not been contacted in XX minutes’. This lets you see how long does it takes an agent to make a first contact for a lead/customer/opportunity that been created inside the CRM system. Here’s a screenshot of some of the reports:
[click for larger image]
Fonality UAE brings a whole new level of CRM integration with PBXtra, an Asterisk-based system. This tight-level of integration with an Asterisk-based system with the popular Salesforce.com hosted CRM platform is great news for the Asterisk community. While Asterisk-based systems have made in-roads in the contact center space, heavyweights like Nortel and Avaya still win a huge share of the contact center space because of their advanced CRM integration. So the more Asterisk-based solutions offer advanced CRM applications, the better for the Asterisk ecosystem. We’ll see more Asterisk deployments in contact centers as a result.
It’s worth mentioning that Digium’s Switchvox IP-PBX also offers integration with CRM apps, including Salesforce.com and SugarCRM. I saw a demo of Switchvox’s web-based panels down in Huntsville, Alabama. I remember it supported querying Salesforce.com on an inbound call (screenpop), but don’t recall if it supported querying Salesforce.com on an outbound call. I’m also not sure to what level Switchvox offers reporting on inbound & outbound calls tied to the CRM contact records. I don’t believe Switchvox offers contact-based reporting based on CDR information. I’ve actually been meaning to do a full-review of Switchvox.
In any event, getting back to UAE, I should point out that Fonality queries a local cache of the Salesforce.com database for it’s instant when it performs a contact record lookup and is not affected by any sort of Internet outage or Salesforce.com outage. Importantly, inbound calls can be automatically directed to the contact or opportunity owner. This prevents sales personnel from ’stealing’ a lead.
“The Frankenstein, bolt-on products from other telephony vendors provide only basic screen pops or click-to-call features. Fonality has a much more sophisticated approach for delivering feature-rich, end-to-end contact center solutions,” said Corey Brundage, vice president of marketing and product management at Fonality. “PBXtra Unified Agent is the only offering available today that immediately impacts a contact center’s top and bottom line without the need for professional services or a six-figure capital expenditure.”
Overview:
- PBXtra Unified Agent from Fonality combines phone system, call center, and CRM capabilities to provide a complete view of contact center operations. With Salesforce CRM and PBXtra Unified Agent, companies can increase close rates, revenue and profits, while providing much better customer service and dramatically reducing sales and support costs.
- For less than $20/user per month, PBXtra Unified Agent delivers an immediate return on investment. Agents benefit from productivity enhancements and timely access to more accurate and complete information so they can do a better job selling to and supporting customers. Managers gain deep visibility into real-time and historical agent call activity within Salesforce CRM so they can make more intelligent business decisions based on accurate data.
“Today, contact center managers are often faced with inaccurate or missing data and agents struggle with cumbersome CTI applications, resulting in a poor customer experience,” said Richard Gonzales, president of Buvelo Solutions, a salesforce.com consulting partner. “PBXtra Unified Agent solves these problems with a unified solution that helps managers and agents use Salesforce CRM more effectively than ever before.”
With PBXtra Unified Agent managers can easily measure, test, and improve sales and support processes, view the effectiveness of human resources, salvage opportunities or identify lapses in service level agreements. PBXtra Unified Agent includes the following features for managers:
- Track and record all call data in Salesforce CRM automatically
- View customer call histories
- Understand how call follow-up times impact profitability
- Find out the number of calls it takes to convert a lead to a sale
- Determine which agents are the most responsive
- Visually rank agent calling frequency and duration
- Instantly report all leads not called within a specified period
With HUD, Fonality’s communications application, you can view desktop alerts with inbound caller name, number, even deal size. In the sample below, the sales agent can see an inbound $120,000 opportunity:
“PBXtra Unified Agent’s deep integration with Salesforce gives our customers even more value from their CRM installations,” said Clarence So, CMO, salesforce.com. “Thousands of customers have installed applications from the Force.com AppExchange because partners like Fonality are offering innovative tools that expand the capabilities of Salesforce CRM.”
Chris explained that one of reasons why they called it Unified Agent Edition is because they created a unified agent creation and logon. When you make an agent on PBXtra, it displays a drop-down box of your Salesforce.com agents allowing you to map your extension on Salesforce.com account. So it matches username and password across the two platforms. That’s how you know the name of the person making a call as opposed to the extension number.
For current PBXtra customers it’s a $4,000 upgrade for unlimited agents. New customers the pricing starts at $6,995 for server hardware and software. Interested parties may also participate in a Fonality-hosted informational webinar taking place on Friday, December 12 at 10 a.m. PT. To register for this webinar, head to www.fonality.com/UAEwebinar.
Tags: AppExchange, asterisk, Chris Lyman, crm, fonality, PBXtra, salesforce.com, switchvox, UAE, Unified Agent, voip
- Related Entries
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- trixbox CE 2.4 Released - Jan 04, 2008
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- Avaya #1 VoIP PBX Vendor? - Aug 21, 2007
- PBX Prompts for Asterisk and other open source PBXs launches tomorrow - Apr 03, 2007
- Fonality gets financial boost from Intel - Feb 06, 2007
- Running into fellow VoIP bloggers and some VoIP news - Oct 10, 2006
- Fonality Beats Avaya and Boasts 3,000 Call Centers - Oct 22, 2008
- Fonality Targets Call Centers with Advanced Call Center Features - Sep 18, 2008
- ITEXPO West 2008 a Resounding Success - Sep 18, 2008
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Digium Responds to FBI Vhishing Security Warning about Asterisk
A few days ago the FBI’s Internet Crimes Complaint Center (IC3) issued an unclear warning that says versions of Asterisk software are vulnerable to vhishing (voice phishing) attacks, but didn’t say which versions, but causing a flurry of news activity on VoIP news sites, tech sites, and blogs.
It all started with this warning from the IC3:
New Technique Utilizing Private Branch Exchange (PBX) Systems To Conduct Vishing AttacksThe FBI has received information concerning a new technique used to conduct vishing attacks. The recent attacks were conducted by hackers exploiting a security vulnerability in Asterisk software.
My esteemed colleagues Rich Tehrani and Greg Galitzine did some research to find out what the story was, including contacting Digium’s John Todd.
Here’s Rich’s take:
Before commenting I waited to hear back from Digium’s John Todd who explained that there were some methodology and editorial process issues in this alert - basically no one checked with Digium before going public. As it turns out, after checking with Digium, the FBI quickly revised their statement and everything is fine.
The details are that there was a bug which Digium found in March of 2008 and subsequently patched in version 1.2 and 1.4. Version 1.6 is not affected. Besides, according to Todd, the security issue would arise if system administrators basically disregarded logical security measures like using numerals in passwords.
Greg Galitzine also writes about the FBI’s warning about Asterisk in an article titled Digium Defends Asterisk Against Fed Warning: “Tempest in a Teapot”
In it, Greg writes:
Todd writes in a blog entry titled SIP Security and Asterisk:That bug allowed in some cases unauthorized callers to make calls through an unprotected “context” in Asterisk. Due to the nature of the bug there was fairly limited exposure - it would have required a fairly unusual set of configurations to permit fraud, and there was both a simple config file change that would provide protection, as well as an actual patch to the code which we have every reason to believe has been widely implemented by the very proactive Open-Source community using Asterisk in production environments. The bug didn’t allow arbitrary setting of caller ID, and would only work in a limited set of circumstances that personally I think would be unusual, though possible.
Early on, Todd had a sense that this might just be a misunderstanding: Sorry for the fuss, and I suspect this is just a tempest in a teapot. Use good passwords, keep your packet filters up, and I’ll update things here as we hear more.
Digium’s John Todd wrote an excellent blog post describing what happened after he was able to contact the FBI in charge of the security warning. While there was indeed a security vulnerability in Asterisk, it was patched in 1.2 and 1.4 and doesn’t exist in 1.6. Thus, someone would have to be using a very old version of Asterisk. And as for the security vulnerability itself which seemed to enable vhishing attacks, John indicates that it was a relatively obscure exploit and “an administrator would have to consciously configure their system in what I believe to be an extremely unusual way in order to be victimized by this particular vulnerability.” So indeed in John’s own words, it seems to be a tempest in a teapot after all.
John Todd wrote:
As we had surmised, the warning from the IC3/FBI on Friday was just a re-hash of a bug that was fixed back in March of this year. I was in touch with the agent in charge of this release this morning (after contact attempts on Friday failed) and he understood quickly that the wording was lacking in ways that created questions in the minds of readers, and this was being amplified by bloggers who more clearly outlined the set of questions raised by the advisory/release. To his credit, the IC3 agent quickly pushed through a set of changes today to the posting which more specifically describes the issue, which indeed is the AST-2008-003 SIP guest permissions problem.
John Todd also wrote:
This bug was discovered and patched for 1.2 and 1.4 versions of the software, and 1.6 releases were not vulnerable. Simple changes to site-specific configurations typically would be all that would be required even on systems that did not get patched or upgraded. The bug that is described is relatively obscure, and was found by Jason Parker here at Digium. We didn’t know of any “in the wild” exploits back then, though of course there may be some now. I’m still somewhat surprised that anyone has been able to use this bug to the extent that they were able to mount “vishing” attacks. While I won’t get into the details of configuration specifics, I would say that an administrator would have to consciously configure their system in what I believe to be an extremely unusual way in order to be victimized by this particular vulnerability.
John Todd complains about the “vagueness” of the warning but in an update after speaking to the IC3 agemt, John Todd says “To his credit, the IC3 agent quickly pushed through a set of changes today to the posting which more specifically describes the issue, which indeed is the AST-2008-003 SIP guest permissions problem.” (an old issue)
So my Asterisk-loving friends. If you are indeed running patched1.2/1.4 Asterisk or v1.6 you have nothing to worry about. And if you aren’t running these versions, what the heck is wrong with you? And you call yourself an Asterisk fan. Per shame!
P.S. As Rich said, “I am sure by the time Asterisk World rolls around in a few months in Miami, we will all be laughing about this incident and marveling at the opportunity that is open source communications.”
Tags: Asterisk, digium, FBI, IC3, Internet Crimes Complaint Center, John Todd, security, vhishing
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- Rev B of Astfin’s BRI (ISDN) Asterisk Appliance Arrives - Apr 29, 2008
- Podcast Interview with Digium CEO Danny Windham - Apr 17, 2008
- Hulk Smash Asterisk 1.6! - Apr 16, 2008
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Asterisk-based VPN in a Flash Mobile Telephony Appliance
Nerd Vittles (Ward Mundy) is the man! He just combined the popular Acer Aspire One NetBook featuring the powerful Intel Atom N270(1.60GHz) processor along with Fedora 10 and his Asterisk-based distro. Nerd just may have cooked up the perfect low-cost, powerful, GUI-rich (low overhead KDE), and portable Asterisk-based appliance! Now you can take your phone system on the road while driving, on the plane, etc. Imagine sticking a 3G/EVDO card in the system while driving and your Asterisk-based PBX is still able to make/receive VoIP over EVDO calls and route them accordingly. He writes about the challenges of building a portable Asterisk-based platform:
Continue reading Asterisk-based VPN in a Flash Mobile Telephony Appliance…
Tags: Acer, aspire one, asterisk, FreePBX, Intel Atom, netbook, PBX in a Flash 1.3, voip, VPN in a Flash
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Oct 06, 2008
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Sangoma Launches B600 Series of Analog Voice Cards
Rich has a great interview with Sangoma’s VP Sales and Marketing, Doug Vilim about their new B600 Series of Analog voice cards, which can be used in Asterisk-based systems. Apparently, Sangoma’s über customizable & modular cards can be a slight cost detriment for those that don’t need such modularity.
B600D PCI model.
Here’s a teaser from Rich’s blog:
What is the reason for these new cards to be introduced?
Some of our OEM customers; those who are building IP PBX appliances Empowered by Sangoma, often do not require the modularity, port customization and single slot expandability that our A-Series analog cards provide. Also, four PSTN lines and one fax connection is a common configuration that creates economies of scale, which can lead to a substantial cost savings for these customers when purchasing in larger volumes.
Also, here are the specs:
- 4 ports of FXO and 1 Port of FXS.
- Support for Asterisk®, FreeSWITCH™, Yate™, trixbox™, and PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications.
- Single synchronous PCI interface for all ports.
- Dimensions: 2U Form factor: 140 mm x 55 mm for use in restricted chassis.
- Includes both standard and short half-height compatible mounting clips for installation in 2U rack-mount servers.
- 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention.
- Autosense compatibility with 5 V and 3.3 V PCI busses.
- Fully PCI 2.2 compliant, compatible with all commercially available motherboards, and shares PCI interrupts.
- Power: 3.6 W (0.3A @ 12 V) and 1.65 W (0.5A @ 3.3 V).
- Temperature range: 0 - 50°C.
- Cables are included.
- Suports Fax: Single FXS Port designed for optimum fax support.
- Sangoma B600 cards synchronize perfectly with Sangoma’s T1/E1 cards and the PSTN clock for error-free fax and modem passthrough over T1 lines.
-
Optional: DSP Echo Canceller
- G.168-2002 echo cancellation in the hardware.
- 1024 taps/128 ms tail per channel on all channel densities.
- DTMF decoding and tone recognition.
- Voice quality enhancement: music protection, acoustic echo control and adaptive noise reduction.
- No CPU load as a result of echo cancellation.
- No additional slot is required.
Tags: analog, asterisk, B600, pci, voip
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Oct 06, 2008
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PIKA WARP Appliance Adds FreePBX Support
When I met with PIKA Technologies at ITEXPO they told me support for FreePBX was coming. Well today, PIKA Technologies announced that PIKA WARP the Appliance is now compatible with the Asterisk-based FreePBX GUI (Graphical User Interface) application. I reviewed the PIKA Appliance recently and was pretty impressed with it. Having FreePBX support is a huge milestone for the PIKA Appliance. FreePBX is a popular user-friendly web application that makes it easy to setup and configure Asterisk.
According to PIKA, “While customers often develop their own GUIs, many have told PIKA that if WARP were compatible with industry-standard GUIs like FreePBX, they would be much more inclined to adopt the portfolio. With today’s announcement, PIKA has once again demonstrated its responsiveness to the needs of its user base.”
“We are very pleased to have supported the effort to adopt the FreePBX application to run in the PIKA Warp environment,” said Terry Atwood, vice president of sales, marketing and customer care at PIKA Technologies. “Used in many Asterisk implementations around the world, including Trixbox, FreePBX has proven its value, time and again. When the FreePBX team expressed their willingness to work with us to port to the Warp Appliance, we jumped on the opportunity.”
“FreePBX has become the de facto standard for enterprise grade PBX functionality delivered to the SMB business and includes a very rich set of functionality and customization potential,” said Philippe Lindheimer, open source community director of Bandwidth.com and leader of the FreePBX project. “But no GUI is complete without a wide range of hardware options to complete the package. We are delighted that PIKA can now include FreePBX in the PIKA WARP and bring our two eco-systems together.”
Today’s announcement from PIKA follows news of a new partnership between FreePBX and Bandwidth.com, a complete business communications provider offering advanced VoIP, Internet services and managed network services to small and medium businesses. Bandwidth.com will devote significant resources to expand the scope of FreePBX while protecting its charter to remain open source and free.
“The partnership with Bandwidth.com is great news as it gives FreePBX the support it needs to grow while ensuring it remains a free GUI for the entire open source eco-system,” said David Clarke, business development manager at PIKA and director of the PIKA Warp Community. “I know the choice of Bandwidth.com was a decision that Philippe made only after months of consideration and sound input from the key developers and contributors to the FreePBX project.”
Out of the box, FreePBX provides a long list of features including many typically found only in an enterprise-grade PBX, some examples are:
• Unlimited number of voicemail boxes
• “Follow me” functionality
• Ring groups and call queues
• Unlimited number of conference bridges
• Paging and intercom functionality
• and much more
The PIKA WARP Appliance product portfolio is ideal for deploying small- to medium-sized IP-PBX systems, IVR self-service systems, predictive dialling systems, fax servers and many other features typical of a traditional, purpose-built business telephone system that are often lacking in a computerized system. Compatible with a variety of open-source development platforms, including Asterisk and Linux, the Appliance offers a cost-effective alternative to traditional off-the-shelf computers and plug-in-card network connectivity in a smaller footprint.
Tags: Asterisk, Bandwidth.com, David Clarke, FreePBX, Philippe Lindheimer, PIKA WARP Appliance, Terry Atwood, voip
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Oct 03, 2008 - Skype for Asterisk Launches -
Sep 25, 2008 - ITEXPO West 2008 a Resounding Success -
Sep 18, 2008
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Bandwidth.com invests in FreePBX
Bandwidth.com has just made an investment in FreePBX, the popular front-end interface to Asterisk-based distros. I discussed this news with Philippe Lindheimer just a couple hours ago. One of the questions I asked was if Bandwidth.com would get “preferred treatment” within the FreePBX interface, since Bandwidth.com offers SIP trunking. Obviously, if FreePBX gives Bandwidth.com a prominent position in the GUI or they make it “easier” to configure FreePBX (i.e. plug-n-play) that could be a huge boon to Bandwidth.com Philippe said that that isn’t part of the investment announcement being made today, however, that is something they are looking at.
As for the purpose of the investment, Philippe said it was mostly due to Bandwidth.com’s desire to grow the market and help build the FreePBX community. The idea is that the more IP-PBXs out there, the more SIP trunks, and hence more revenue for Bandwidth.com. I have some further thoughts on this, but I’m pretty busy today and wanted to share the news.
Philippe Lindheimer said, “Part of assuring the success of FreePBX is to make sure that we continue to have strong leadership, community participation and a thriving eco-system of users and partners. I would like to announce a new partnership that will help the project tremendously. I have joined forces with Bandwidth.com as their Open Source Community Director, where we will be devoting significant resources and effort to expand the scope of FreePBX while protecting its charter to make sure it remains open and strong.”
One significant piece of news is that Bandwidth.com helped protect the FreePBX’s project several months ago when the FreePBX trademark (which FreePBX.org nor Phillipe never owned) was “being shopped around to parties that did not have this project’s best interest in mind” according to Phillipe. Thus, Bandwidth.com preemptively purchased the trademark with Phillipe’s blessing in order to assure FreePBX was not jeopardized.
You can read Phillipe’s blog post about this here which has more details.
Tags: asterisk, Bandwidth, com, FreePBX, Philippe Lindheimer, voip
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Sep 18, 2008
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AudioCodes enters IP Phone arena with 300HD Series
AudioCodes has now entered the IP phone arena with a phone that supports wideband codecs (HD) for superior sound quality. Seems a bit odd for a company that makes VoIP PCI and cPCI communication boards and VoIP media gateway modules (PMC form factor) and Analog Media Gateways (2/4/8/24 ports) to be entering the crowded VoIP arena, but enterprise IP phone market is expected to quadruple, from $2.1 billion in 2007 to $8.4 billion in 2001, with some 63 million endpoints being shipped by that time, according the Synergy Research Group. So there is a pretty big pie to go after. Traditional network hardware vendor Adtran has also recently entered the IP phone arena with their Adtran IP700 series (see Adtran IP706 review).
The AudioCodes 300HD Series includes three models: The 310 HD entry level phone with a basic display and interface; the 320HD premium endpoint with a larger screen, and the 350HD executive phone with a color LCD screen. All three models are based on AudioCodes newly announced VoIPerfect software, and include the most commonly used wideband codecs. Of course they are SIP-based so they should work on standards-based IP-PBXs such as Asterisk and these phones are also PoE (Power over Ethernet) compliant.
For more details, check out TMCnet reporter Erik Linask’s article. One interesting take Erik has is when he writes, “First and foremost, it owns the DSPs that enable the higher quality, which means it can provide its HD VOIP-enabled handsets at a price point comparable to other high-end non-HD devices”.
Now I just have to get my hands on on to review. Stay tuned…
Tags: 300HD, 320HD, AudioCodes, Erik Linask, HD, ip phone, PoE, voip, wideband codec
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New Blackfin BF51x Processor Launches
Analog Devices unveiled the new Blackfin BF51x series, the newest members of their convergent-processor family. Blackfin processors are very popular when building Asterisk-based appliances, including the Digium Asterisk Appliance AA50 and Astfin. The Blackfin convergent-processor architecture offers reduced cost, power consumption, and software complexity. Although the processor is popular in creating Asterisk appliances, it can be used for a variety of low-cost, low-power consumption required applications.
The new Blackfin processors are the BF512, BF514, BF516 and BF518. According to Analog Devices, “All are single-core convergent processors that surpass outdated, heterogenous MCU+DSP approaches in reducing part-count, system cost, board space, and power consumption. Like traditional DSPs, the BF51x processors feature high clock rates and low power dissipation per unit of processing (MMACs/mW), and like traditional MCUs, these convergent processors are OS and compiler-friendly.”
All four of the new 16-/32-bit BF51x processors are available at clock speeds up to 400 MHz (800 MMACS) and include 116 kBytes of RAM plus an optional 4 Mbits of serial (SPI) flash memory. Each also integrates Lockbox™ security for code and content protection.
The Blackfin processors on-chip integration assures easy connection to a variety of audio, video, imaging and communications peripherals and memory types. Integrated features include support for sixteen stereo I2S digital-audio channels, twelve peripheral DMA channels, and an advanced memory controller for glueless connection to multiple banks of external SDRAM, SRAM, Flash, or ROM. Each processor includes two dual-channel synchronous serial communication ports (SPORTs), a high-speed parallel peripheral interface (PPI), an I2C compatible two-wire interface (TWI), dual PC-compatible UARTs, and 2 SPI-compatible serial peripheral interface ports.
“System solutions ultimately determine how much power any particular application will consume,” said Jerry McGuire, vice president, General Purpose DSP, Analog Devices, Inc. “It’s quite intuitive that a single convergent processor with the right mix of integrated peripherals is always going to lead to lower BOM costs and power consumption than an inelegant combination of disparate processors and parts can possibly achieve. Many companies today talk about the lowest power or the highest performance. But what is important for today’s applications is the highest levels of performance at low power.”
All of the new Blackfin processors, delivering 8.5 MMACs/mW (100 MHz), include dynamic power management (DPM) functionality that lets developers match the processor’s power consumption to processing requirements during program execution. ADI pioneered the application of DPM more than seven years ago with the release of the first Blackfin processors.
The BF512 is the new low-cost entry point in the Blackfin processor family. The device balances performance, peripheral integration, and price, and is well suited for the most cost-sensitive applications including portable test equipment, embedded modems, biometrics, and consumer audio. All members of the BF51x family also include a new 3-phase PWM generation unit for inductive motor control applications and a quadrature interface for rotary encoders.
The BF514, BF516, and BF518 all extend the convergent processor family further into the portable application space with on-chip removable-storage interfaces. All three devices include Secure Digital Input Output (SDIO) for connectivity to standard flash memory and Wi-Fi cards; a power-optimized CE-ATA storage interface for small form-factor handheld and consumer electronics applications; and an embedded multimedia card (eMMC) interface for integrating mass-storage flash memory in a wide range of consumer electronics, wireless, navigation, and industrial applications.
For developers of network-connected industrial and instrumentation applications, the BF516 adds an Ethernet 10/100 MAC with Media Independent Interface (MII) and Reduced Media Independent Interface (RMII). Highly integrated for industrial, portable and VoIP applications, the BF518 Ethernet MAC supports the IEEE-1588 clock synchronization protocol for networked measurement and control systems.
An increasingly wide variety of applications are viewing the contemporary convergent-processor approach as the soundest choice for cost- and power-sensitive designs. For example, some voice-over-IP (VoIP) telephony system developers have designed in separate DSP and microcontroller chips to implement the required media and control functionality. With BF51x Blackfin processors, however, a single architecture enables full VoIP telephony functionality in a unified software development environment with faster system debugging and deployment, lower overall system cost, and the lowest possible system power demand.
“GIPS VoiceEngine media processing capabilities meet the highest requirements of VoIP equipment manufacturers and paired with Analog Devices’ Blackfin processors we can assure customers a consistently high quality VoIP experience. The performance, power and functionality profile of Blackfin is a superb fit for VoIP technology,” said Larry Golob, Senior Director Business Development, Global IP Solutions.
With the Global IP Solutions (GIPS) VoiceEngine package of VoIP software components available for Blackfin processors, and a VoIP reference platform available on uClinux, the feature-rich Blackfin family has driven down the price required to easily design and deploy a fully scalable range of VoIP telephony designs across multiple market spaces.
Pricing and Availability
The BF51x family includes the BF512 at $4.95, the BF514 at $7.75, the BF516 at $8.75 and the BF518 at $11.85. Processors are sampling immediately. All prices are based on 25,000-unit quantities.
Tags: appliance, asterisk, Astfin, BF512, BF514, BF516, BF518, Blackfin, gips, processor, uClinux, voiceengine, voip
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Use VoIP to Telecommute for a Merry Christmas!
How do save money in this worldwide tight economy so that you can have a Merry Christmas with lots of gift giving? (Not that gift giving is the main point of Christmas) Well, one way is by using VoIP to telecommute. Research done by Aastra found that commuters driving into the UK’s largest cities could potentially save enough money by Christmas to buy more than half a kilometer of wrapping paper if they worked from home just one day a week. Based on commuters with 50-mile round trips, the average transit time soars in London to 111 hours - almost fourteen working days a year. Researchers found that London was the most expensive and time-consuming city to commute into, followed by Leeds and Bristol.
Aastra 57i CT phone
Telecommuting just once per week could save on average, could save £19.36 per day of telecommuting or £174 ($271 U.S. dollars) in the nine weeks running up ’til Christmas - enough to buy 17 turkey crowns, with change to spare for cranberry sauce - if they were equipped to work from home one day a week. In London, this figure soars to £41.90 a day - more than twice the national average due to parking costs, traffic delays, and petrol.
I wonder if this study was done since the recent petrol/oil price jobs? In any event, the research commissioned by Aastra revealed that commuters making 50-mile round trips by car, on average, could save £19.36 per day by working from home. In London, this figure soars to £41.90 a day - more than twice the national average.
I have an Aastra 57i (see review) at home that I use to telecommute myself. One nice thing about the Aastra VoIP phones is that they licensed Packet8’s NAT technology for their firmware, which solves those pesky VoIP-over-NAT issues.
Working from home one day a week could also save penny-pinching parents with young children more than £460 in day care (£286) and travel costs (£174) in the build-up to Christmas - enough to buy all of this year’s top 5 most wanted presents, as predicted by the Toy Retailers Association, with change to spare for more than 130 bags of chocolate coins for their Christmas stockings.
According to Aastra, as Christmas looms and inflation hits a decade high, more people are looking to home working as a means to enjoy a better work/life balance and save money.
Michael Calvert, UK General Manager of Aastra, who commissioned the research, said: “Commuting to work everyday can be a major strain on people’s finances, and considering the current economic climate it’s not surprising that the mood of the country is more credit crunch than Christmas lunch. Commuters equipped with the right, readily-available technology, could save money and lower their stress levels by taking advantage of flexible working practices. With many workers able to do their job equally well, if not better, from home, it’s a wonder why more companies are not encouraging home working.”
“It’s not just commuters that could see real economic benefits from flexible working practices, many companies could benefit from lower real estate and energy costs, higher morale, and increased staff retention. Flexible working technologies such as Voice over Internet Protocol phones can even reduce the cost of calls, while making corporate communications more effective.”
If commuters with 50-mile round trips by car worked from home one day a week they could save enough money in time for Christmas to buy:
- 1 Xbox 360
- 3/4 of a Playstation 3
- 3,400 fairy lights
- 1,560 migraine tablets
- 828 Christmas crackers
- 207 mini Christmas puddings
- 58 pairs of men’s novelty socks
Tags: aastra, christmas, Michael Calvert, telecommute, telecommuting, voip
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Analysis of a VoIP Attack
VoIP security is often overlooked by IT administrators as well as VARs and resellers that deploy VoIP in the enterprise. They do so at their own peril, however. One of the main factors behind using VoIP is to save money. Well imagine your IP-PBX has been hacked and you don’t notice anything wrong until you receive the next phone bill with hundreds or thousands of dollars in phone charges. There goes all the savings you anticipated when you decided to install VoIP!
This laissez faire towards defensive security reminds me of Star Trek, where for whatever reason the Enterprise flies through space with danger lurking around every corner but they keep their defensive deflector shields off and often turn them on when it’s too late. The Enterprise has a fusion reactor with nearly limitless power, so why not keep the deflector shields on all the time? Maybe they’re just being “green” and shooting for five nines (99.999%) of efficiency.
In any event, security in general is often overlooked, whether it’s securing your web server or your email server, or confidential database servers. But in most cases when these particular systems are hacked it’s usually just an inconvenience (defaced web pages, spamming through your email server) with minimal financial impact. Not so when it comes to VoIP. The financial impacts of a hacked SIP server or VoIP gateway could be tremendous. This is especially true for larger organizations which already have hundreds or thousands of calls per month, including international business calls. How does accounting find the fraudulent calls in the phone bills which are 4 inches thick? It’s like finding the proverbial needle in a haystack.
If the hackers are smart, they will limit the amount of traffic they route through a hacked gateway as not to set off any red flags. It could be months or possibly even years before anyone notices anything is amiss. I’m reminded of an old PBX technology called DISA (Dialed In Switch Access) which was one of hackers first tricks to get free calling. DISA was designed to allow employees to remotely call into the PBX and get second dial-tone. With this second dial-tone using touchtones they could logon to ACD queues, monitor agents calls, and of course initiate outbound calls.
In fact, many years ago, TMC was hit with a DISA-like attack on our Comdial PBX resulting in quite a few international calls. If I recall, there was a vulnerability in the Keyvoice voicemail system which allowed someone to make outbound calls. Needless to say, I was able to shut it down pretty quickly.
Part of the attack also involved using a scripted dialer which accessed the voicemail system by automatically sending the # key, then sending a chosen extension (say 100), and then iterating through all the various PINs (0000 - 9999). Since TMC has a toll-free 800 number, the attacker only has to make at most 10,000 calls to find a PIN to a particular extension. Obviously, chances are they’d find the script in much less than 10,000 calls and you get 3 tries before the voicemail hangs up. Once the PIN is found, not only does the attacker have access to the the user’s voicemail, they also have access to any DISA capabilities of the voicemail system. More reason why IP-PBXs today need to have a PIN expiration feature just like Active Directory supports password expiration. No matter how many times IT staff reminds employees to change their passwords/PINs, they just don’t do it unless the system forces them to. I don’t believe any of the Asterisk systems I’ve tested have password expiration - so my open source Asterisk fans, if you’re listening, add it to the code, will ya?
With all this in mind, I was fascinated to read an article by an Austrian company IPCom titled “Analysis of a VoIP Attack”. It’s an excellent read. Let me give you the abstract:
Recently, several IT news websites reported VoIP attacks against home users, containing lots of myths and incorrect statements. Unfortunately, they also give wrong security advices. This article analyzes the attacks and describes the motivations behind. Further, it shows simple workarounds how “insecure” software can be used in a secure way.
Here’s a teaser:
1 The Attack
1.1 Analysis
On 23.09.2008, heise.de reported an attack against VoIP devices of German VoIP users [heise]. This article references a thread in the IP-Phone-Forum [ipphone] in which people report that their VoIP phones started ringing in the middle of the night and displayed incoming calls from the phone number 5199362832664. One of the users presented a log file of a Patton SIP device which captured the suspect INVITE request:
02:12:42 SIP_TR> [GW] < Stack: from 213.130.74.70:3808
INVITE sip:810525551690000@1.2.3.4;transport=udp SIP/2.0
Via: SIP/2.0/UDP 213.130.74.70:3808;branch=100100101101011111101110
00100213.130.74.701.2.3.41863480914;rport
Max-Forwards: 100
From: <sip:5199362832664@1.2.3.4>;tag=21671132663-
4985269162167113266321671132663213.130.74.70
To: <sip:810525551690000@1.2.3.4>
Call-ID: 83764811100011101110010010110101101100111001001011
0101111110111000100213.130.74.701.2.3.41863480914f
df23881052555169000021671132663-
4509759162167113266321671132663213.130.74.70174046 6380
CSeq: 1 INVITE
Contact: <sip:fdf238@213.130.74.70:3808;transport=udp>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 394
Let’s have a look at this SIP message. The funny thing is that absolutely nothing in this SIP message is trustworthy: Probably the SIP message has been received via UDP and the source IP address could be easily spoofed. Further, every data in the SIP message is user generated (in this case by the attacker) and does not necessarily reflect real data. Nevertheless, let us try to analyze the message:
- Source IP address 213.130.74.70 and source port 3808: Although the IP address could be easily spoofed, in this case it may be the real address of the attacker as the IP address is also present in the Via: header (used for sending back responses). Further, if the attacker wants to know the result of the attack, he has to receive the SIP responses meaning that he has to provide his real IP address.
- The Call-ID looks like a random string and contains the source IP address. As the Call-ID is invalid (per RFC 3261 the Call-ID must not contain spaces), it can be assumed that the attacker did not use a fullfledged SIP stack, but some scripts to generate the request.
- The User-Agent header displays “X-Lite” as client. However, if you compare the above request with an INVITE request sent by X-Lite you will find out that the random strings (call-id, tags, branch
Ok, you’ve been thoroughly ‘teased’, now go read the full article (PDF). Good stuff!
Tags: DISA, hack, security, sip, sip headers, spoofing, star trek, voip
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Fonality Beats Avaya and Boasts 3,000 Call Centers
Fonality’s CEO Chris Lyman spoke with me earlier today to talk about the strong inroads Fonality has been making in the call center market. Chris said, “Fonality has become a big player in what I like to call the micro call center market. We launched our call center product in 2005 and we have almost 3,000 deployed call centers now.”
When asked to define “call center” since many people have a different definition, Chris responded, “Anybody who purchases Call Center Edition plus HUD Agent. We can’t imagine someone buying barging, monitoring, recording, and queue reporting if you weren’t doing call center activity. Since these products effectively add in some cases double the costs to the phone system, you’re pretty serious if you’re buying those.”
When asked what percentage were call center deployments versus regular enterprise deployments, Chris said that 40-50% were off all their solutions sold are Call Center Edition.
Chris stated that there is a vastly under served and untapped call center market when he explained, “The 5-50 agent market where I have a regular business with 50 employees, but I’ve got 10 sales people who work the queues. And so there is this micro call center that has been ignored by Genesys and the expensive players out there for all these years because you cannot afford a $100,000 drop-in, bolt-on Avaya call center system. I think the low end of the market hasn’t been able to afford that and we’re enabling the micro call center market.”
Chris explained that Fonality PBXtra Call Center saved Crusecom, a Michigan-based outsourcing call center, more than $250,000 annually. Art Cruse CEO at Crusecom explained he’d have to hire a full-time Avaya engineer on-site at $140,000 - $160,000 per year plus maintenance costs of the Avaya system. He also explained Avaya call center phones are more expensive than regular Avaya phones. They also have 94% call completion rate or an amazing 6% abandonment rate. Other call centers are coming to look at how they’ve achieved such as phenomenally low abandonment rate. Fonality’s call center system has helped Crusecom attain rapid growth. The $250,000 savings enabled Crusecom to invest in a new 14,000 square foot call center facility that will house up to 150 call center agents.
“Fonality and PBXtra Call Center made our whole business model of providing cost effective, outsourced onshore call centers possible. They have delivered remarkable business benefits to our company,” said Art Cruse CEO at Crusecom. “When we bought our Fonality system, it was 75 percent less than a comparable Avaya system. With Avaya, we would easily be paying an extra $250,000 per year in support, maintenance and hardware costs, which would directly impact our bottom line and limit our growth capabilities.”
Crusecom provides 24×7x365 call center operations to state agencies and companies that want to keep call center operations in the U.S. but cannot afford the high costs of urban call centers. Since deploying PBXtra Call Center in 2007, the company has expanded its customer base, added 50 new call center agents and increased inbound call volumes.
“PBXtra Call Center is amazing - the more calls we get the better the system performs and the more we save,” Cruse continued. “We’ve been able to reinvest these cost savings in our company and grow our business very rapidly.”
By leveraging web-based reporting capabilities in PBXtra Call Center and other technology developed in-house, Crusecom is the only electronic benefits transfer (EBT) call center in the country that is offering customers real-time, web-based service level agreement reporting. In addition to the $250,000 he is already saving with Fonality, Crusecom estimates that this real-time, self-service reporting functionality saves his company eight to 16 man hours every day, or $50,000 to $100,000 annually, while providing customers with better service and support.
“Gone are the days of the cumbersome, big-iron call center oligopoly. Crusecom is a perfect example of why there is a changing of the guard in the call center market as companies rapidly adopt more agile technologies like PBXtra Call Center,” said Chris Lyman CEO of Fonality. “Small and mid-size call centers need affordable, flexible phone systems with solid service agreements that allow them to grow rapidly. Fonality is delivering on these requirements and is constantly innovating with newer advanced technologies.”
Chris also explained that Fonality’s flat rate support was also a key advantage over other call center IP-PBX competitors. Lastly, he explained the hybrid-hosted approach enables call centers to have home agents without the need for VPNs. The hybrid-hosted approach resolves pesky VoIP over NAT firewall issues making telecommuting a much easier approach with lower TCO.
Tags: avaya, call centers, chris lyman, Crusecom, fonality, voip
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TMC & Digium Announce Registrations open for Digium|Asterisk World
Registration to the upcoming Digium|Asterisk World conference is now open, according to an announcement today from Technology Marketing Corporation (TMC) and Digium, the open source Asterisk Company. So you early bird types can now register and add it to your Calendar now, lest you forget. TMC and Digium also announced the launch of the new event Web site at www.digiumasteriskworld.com.
Digium|Asterisk World is collocated with TMC’s INTERNET TELEPHONY Conference & EXPO East 2009 and is a 3-day event commencing on February 2, 2009 in Miami, Florida.
According to TMCnet reporter Michelle Robart: Entering its third year, Digium|Asterisk World is the conference that educates business users, resellers and executive decision-makers on “Everything Asterisk.”
Michelle also gives more interesting details about Digium|Asterisk World:
The conference will feature booth exhibition space and a Presentation Theatre on the EXPO floor where attendees can learn more about Digium’s open source communications solutions. In addition, TMC and Digium will join forces to create the conference track agenda, which will be revealed in the upcoming weeks.
ITEXPO East 2009 is the world’s largest and most significant communications technology event. It features more than 200 companies exhibiting on the EXPO floor and hundreds of sessions led by the industry’s most well-known thought leaders. The show provides a forum for assessing the latest products and services and offers numerous opportunities for face-to-face networking that service providers, carriers, resellers, distributors, equipment manufacturers and IT executives from enterprise and SMB companies need to create new business relationships.
Tags: asterisk, digium, Digium|Asterisk World, open source, tmc, voip
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Aastra 57i and 57i CT Review
The Aastra 57i is one of my favorite VoIP phones. The 57i and its sister, the 57i CT (cordless telephone adjunct), offers some unique features and is undoubtedly one of the most flexible IP phones you will find. The 57i and 57i CT sport a large 144 x 128 pixel graphical backlit LCD display and 6 dynamic context-sensitive softkeys. Although the resolution isn’t designed for photos, it’s a very large LCD, one of the largest I’ve seen making it very easy to read the number of voicemail messages, the CallerID of an inbound call, and the one touch feature keys you’ve programmed. The 57i is of course SIP-based making it fully interoperable with IP-PBX platforms such as Asterisk. The 57i and 57i CT offer advanced XML capability to access custom applications and support for up to 9 calls simultaneously.
57i CT Web Interface Preferences
The 57i CT is exactly the same as the 57i except it has a built-in wireless transmitter in the base unit and it comes with an integrated WDCT cordless handset with a range of up to 300,000 sq ft. The cordless phone sports 10 previous number redials, a mute button, on hold, Callers List, transfer,4 ringtones, and more. My only complaint is that there isn’t a dedicated transfer button. While on a call using the wireless handset, you have to press the F (Function) button, scroll to Xfer, and then enter the extension number. The mobility the cordless handset gives you is perfectly suited for executives, mobile warehouse personnel, as well as retail staff. Here’s a photo of the 57i CT on my desk along with the cordless handset:
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