Digium announced the details of their annual AstriCon Event, the official conference for Asterisk. For the past three years it has been held in Phoenix. This year the East Coast will play host. The seventh iteration will be held on October 26-28, 2010, at the Gaylord National Resort and Convention Center. Located just minutes outside of Washington, DC, the Gaylord National is located in National Harbor, MD. For more details you can read the official press release here.
Digium Announces AstriCon 2010 Details
March 4th, 2010Enter to Win an Extreme Phone Makeover from Digium, Polycom, and Chromis Technology
February 19th, 2010
Is your phone system driving you crazy? Do you wish you had a more effective communications solution for your business? Digium and Polycom are looking for companies that have had it up to here with their key systems and PBXs! Tell them your story by clicking here and you could win an EXTREME PHONE MAKEOVER! Enter to win a Digium Switchvox system complete with Polycom telephones – and if you tell them Chromis Technology sent you, we’ll install it for FREE! But you must click the referral link here or enter “Chromis Technology” in the “Referred by Company” box at www.extremephonemakeover.com.
Don’t delay, enter today!
Contest entry deadline is March 12, 2010!
Sponsored by:
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Important Tests For Measuring VoIP Quality
February 16th, 2010
You say you have been living under a rock and you are not familiar with broadband Speed Testing? Allow me to explain… Speed Testing is a important tool for measuring bandwidth speeds. This is especially important when preparing for a VoIP migration, or any other broadband dependent application… My preferred speed test site is Speedtest.net. This broadband speed analysis tool allows anyone to test their Internet connection. Ookla Solutions provides this service for free to anyone curious about the performance of their connection. Their technology is used to perform over one million speed tests every day.
The Speed Test site allows you to test multiple sites to and from hundreds of locations around the world. For the test I ran for this example, pictured above, I chose the closest, recommended server, which happens to be located in Phoenix, where Chromis Technology is based. As you can see, the result from my location is quite good, I get almost 20MB downstream, and 6.5MB upstream with reasonable Ping times. Very healthy for a VoIP connection.
But as we have all learned over the years, a solid connection with good bandwidth is only part of the VoIP puzzle… So that’s why Ookla created Pingtest.net. This test takes speed testing a step further by testing Ping, Jitter and Packet Loss – each critical to understanding the true quality of a broadband connection. But what do these mean? Let’s define these terms and relate them to a real time application such as VoIP.
Packet Loss – The percentage of packets sent to a server that never arrive. If I am speaking the sentence: “Chromis Technology is a Phoenix, Arizona VoIP Company” over a VoIP connection and the packets that contain “Phoenix, Arizona” never arrive, my sentence might play out as “Chromis Technology is a {silence} VoIP Company.”
Ping - The time it takes for a packet to travel from your computer to a server and back. Ping tests measure for Latency by determining the time it takes a given network packet to travel from source to destination and back. Physical distance, the number of router hops, encryption, and voice/data conversion all impact latency. A good broadband Internet connection typically results in ping test latency of less than 100 ms, often less than 30 ms. A satellite Internet connection normally suffers from latency above 500 ms. In our VoIP scenario, this is typically a SIP endpoint such as a Polycom Telephone and server you are connected to, such as Switchvox. So the greater the ping time and latency, the more likely you are to have over-talking (when one caller talks before the other caller is finished).
Jitter – Jitter is the variance in measuring successive ping tests. Zero jitter means the results were exactly the same every time, and anything above zero is the amount by which they varied. Like the other quality measurements, a lower jitter value is better. And while some jitter should be expected over the Internet, having it be a small fraction of the ping result is ideal. Most VOIP endpoint devices have jitter buffers to compensate for network jitter. Callers experiencing jitter on a VoIP call will notice delays in the conversations.
So while these tests are important, it’s also critical for me to point out that they are not the end all/be all solution to be used when designing your VoIP setup. But they are very powerful place to start yourself out towards a successful VoIP implementation. Chromis Technology, using BroadSoft Packetsmart VoIP testing equipment, can provide a detailed analysis of your LAN/WAN to help assess and resolve potential network issues before deploying VoIP services. For more information about how Chromis Technology can help you mount a successful VoIP implementation using our network assessment tools, please contact us at 602.357.8070.
Polycom and Switchvox Phone Feature Packs, Let’s Talk Compatibility
February 8th, 2010New to Switchvox version 4.5, Phone Feature Packs give you the ability to easily setup your Polycom IP phones to work seamlessly with your Switchvox PBX. They also deliver many of the Switchvox Applications right to your handset, such as recording a call, browsing your voicemail, picking up parked calls, and more! We’ve received several inquiries from customers asking about which phones are compatible with the Phone feature packs… So which phones will, and will not work?
Below is a list of supported Polycom phones:
The new Applications view on a Polycom phone. The color display of an IP670 is not much different than the b/w display of an IP650 using Phone Feature Packs.
- SoundPoint IP 320
- SoundPoint IP 330
- SoundPoint IP 321
- SoundPoint IP 331
- SoundPoint IP 335
- SoundPoint IP 430
- SoundPoint IP 450
- SoundPoint IP 501
- SoundPoint IP 550
- SoundPoint IP 560
- SoundPoint IP 650
- SoundPoint IP 670
- SoundStation IP 6000
- SoundStation IP 7000
The SoundPont IP 301, 600, 601, and 4000 are partially supported: all of the Phone Setup functions are supported but some of the features such as Profiles and Applications are not available. The SoundPoint IP 300 and 500 are not supported at all. Switchvox Phone Feature Packs provision telephones with Polycom firmware version: 3.2.2.xxxx.
**Certain snom phones are also supported by Phone Feature Packs. Stay tuned for a list of compatible snom devices.**
Switchvox VoIP Security
February 1st, 2010
Security is always a concern when installing any new devices on a network, and Switchvox certainly is no exception. We frequently get asked about what steps Digium has taken to ensure security on their Switchvox SMB appliances. Chromis Technology addresses four main concerns that our customers ask us about: 1) access to the web interface, 2) access to the manufacturer console and asterisk core, 3) SIP authentication security, and 4) RTP session security.
Following are descriptions of Switchvox security relating to each concern:
1. Switchvox uses the web standard Secure Socket Layer (SSL) to access the web configuration interface. SSL is the same technology that banks use to conduct online banking. Switchvox allows for the creation of multiple administrators, each with different service level access to the administration web interface. Thus administrators may create different administrators so that powerful controls are limited according to the requirements for specific administrators. Users also may be granted or denied access to the user web interface. Users also use SSL to access their individual settings and must use a password to log in.
2. Switchvox runs using the secure operating system Linux. The server is locked down by default so that only PBX and web interface functions are permitted. The administrator may optionally enable or disable remote console access so that the manufacturer can access the server for advanced diagnostics or repair. The protocol used is Secure SHell (SSH), a high-security system for console access to servers. SSH is disabled by default and the manufacturer (Digium) only has access to the backend if an administrator chooses to grant a technician access. This will then allow a Digium support engineer to access the asterisk core.
3. Session Initiation Protocol (SIP) is the VoIP protocol that Switchvox uses to connect to telephones. SIP uses a password in a method that is analogous to the way that an email account uses a password to access an email server. Switchvox automatically configures Polycom and snom telephones with secure SIP passwords. This ensures that if a SIP device attempts to connect to Switchvox it will be prevented access unless it has the correct secure SIP password for that individual extension it is trying to register. This password is hidden and cannot be discovered. If a user or administrator forgets their password, they must reset with a new pasword. Switchvox employs complicated, secure passwords for SIP devices and informs the administrator if he or she attempts to create an insecure password.
4. Real Time Transfer (RTP) is the VoIP protocol that Switchvox uses to transport the audio calls between telephones and the server. RTP by itself does not prevent someone from collecting the data from the Switchvox network and listening to pieces of a call. Enhancements to RTP in the future will allow the audio data to be encrypted so that the data captured will not be audible. It is difficult to intercept RTP traffic and requires direct access to the telephone system network. However, until Digium incorporates a secure RTP technology, Chromis recommends that administrators not allow public access to the Local Area Network if they can help it. In other words, standard security procedures such as firewalls, building access control to network equipment, WiFi passwords, etc. should be employed for your network. If VoIP calls are placed over public Internet they may optionally be made through a VPN connection to secure the conversation.
So we want to know, what are your concerns with Switchvox VoIP security? Drop us a line in the comments section or email us at the address in the header above. We love hearing from you and I’m sure you have your own opinions on what I’ve left out…
Digium Releases Switchvox SMB 4.5
January 20th, 2010Switchvox SMB 4.5 (Release number 21669) has been announced today by Digium at IT Expo East in Miami. The new version is available for immediate download. Go to ‘Machine Admin > Upgrades‘ to upgrade your SMB server.
The new version contains the following enhancements:
Phone Feature Packs – In this release, ‘Phone Provisioning Tokens’ are updated to ‘Phone Feature Packs.’ You must reboot each of your Configured phones if you want to take advantage of the new features:
- Phone Setup
Now you can set an Alternate Host and up to 3 extensions on each phone, and set several new administration options across all phones. (See ‘System Setup > Phone Setup‘) - Distinctive Ringtones: Administration
Admins can upload system-wide ringtones for use by Configured Polycom phones. Also, two new IVR actions let admins set and remove a ‘hint’ so that extension-owners can set a distinctive ring based on a caller’s IVR selections. (See ‘PBX Features > Distinctive Ringtones‘ and ‘PBX Features > IVR Editor‘) - Distinctive Ringtones for Extensions (Polycom Only)
Extensions can use system-wide ringtones, or upload their own. They can also create ‘Ring Rules’ that cause their phone to ring differently based on the caller or call-type. (See ‘Settings > Phone Features‘) - Phone Features (Polycom Only)
Extension-owners can show Extension Profiles on their phone, set the number of line keys that the extension uses, and interact with PBX applications right on the handset. (See ‘Settings > Phone Features‘) - Phone Features (Polycom and snom)
Extension-owners can customize line labels, auto-answer Switchboard-initiated calls, and disable the missed calls notification. (See ‘Settings > Phone Features‘)
Extension Profiles with Pictures – Phone-type extensions now include a picture, title, and location. This can be edited by the extension-owner (with permission). Profiles are shown in a new Switchboard Panel, and on Polycom phones that have a Phone Feature Pack. (See ‘Extensions > Manage Extensions‘ or ‘Settings > Modify Account‘)
Language Support – The ‘admin’ user, sub-admin users, and extension-owners can each select a language for the PBX’s User Interface. Related sound packs are available. (See ‘Machine Admin > Manage Admins‘ or ‘Settings > Modify Account‘)
PBX Monitoring – A set of SNMP OIDs is now published to monitor the PBX server, phone status, current calls, VOIP providers, and more. (See ‘Machine Admin > Network Settings‘)
Updated Polycom firmware – Most models: SIP 3.2.2 and bootrom 4.2.1 – Discontinued phones (301,501,600,601,4000) do not receive new firmware
Updated snom firmware – 3xx phones: 7.3.30, 820 phone: 8.2.11, 870 phone: 8.3.6
Video Calling - As a result of changes that Polycom, Inc. has made to their VVX 1500 phone, video calling is now available on that phone.
Stay tuned to Chromis.com for a detailed review once we’ve tested this new version on our demo server.
Buy a GN 9330E headset, get a FREE Polycom EHS cable!
January 14th, 2010
Chromis Technology has partnered with Polycom and Jabra to offer a FREE Polycom EHS cable with every GN 9330E headset purchased. Buy one for everyone in the office, get a FREE EHS cable with each headset! There is no limit to how many you can get, and no rebate forms to fill out, it’s really that simple! This is a $30.00 value! The more you buy, the more you save! This promotion is valid from January 5th to March 31, 2010. Click here to buy from the iChromis.com store or call 602.357.8070 to speak with a Chromis sales representative.
Equipped with new DECT 6.0 technology, this stylish, ultra light headset delivers exceptional sound up to 330 feet from your phone. Plus, it’s Wi-Fi friendly and secure, letting you talk without interference. With Up to 9 hours of talk time and 43 hours of standby time ensures your GN 9330E is always ready for action. The Noise-canceling microphone transmits your voice clearly, even in the noisiest environments. Two wearing styles let you choose from over-the-head or over-the-ear to suit your particular preference.
Polycom Electronic Hook Switch (EHS) enables remote operation of compatible Jabra wireless headsets with Polycom SoundPoint IP phones. The EHS adapter allows you to hear ring tones, answer and end calls, adjust the volume or even mute the call from the controls integrated into the headset. All quite convenient, when you are 300 ft. or more away from your desk.
The following Polycom Soundpoint IP telephones have been tested and support Electronic Hook Switch: 320/321/330/331 (These phones require a 2.5mm to RJ-9 adapter), 335, 430, 450, 550, 560, 650 and 670 desktop phones. Check the Jabra-Polycom compatibility matrix to make sure your phone and headset are compatible with each other.
Click here for detailed instructions on how to setup your Polycom phone with your Jabra headset.
Polycom Releases a new entry level phone: Introducing the SoundPoint IP 335
November 2nd, 2009
Polycom introduced another phone to round out their entry-level line up this past weekend: The SoundPoint IP 335. I’ve been beta testing this phone for the past 6 weeks and have put it through the paces. And now that it’s been formally introduced, I can tell you my thoughts. (Spoiler Alert: This phone is spot on… They took the IP330/331, and made it right.)
The IP 335 contains all the features of the Polycom SoundPoint IP family that we’ve come to know and love: Polycom quality and looks, Integrated Power over Ethernet (PoE) support, interoperability with leading IP PBX (i.e. Switchvox) and Softswitch platforms, etc, all in an easy to configure package. But wait there’s more…
The IP 335 looks identical to the IP 330/331 but there are some very distinct differences. The first very noticeable difference is the high-resolution backlit display. It’s still not very big (102 x 33 pixels) but for the price (est. $199 MSRP) it’s respectable. The back lighting makes for a display that is much easier to read.
The second thing you’ll notice is the headset port. I have always had a huge hangup about the 2.5mm jacks that come on the IP 320/321/330/331. It really annoyed me that I could outfit an organization with IP450’s, 550’s, and 650’s and if we paired them with a headset, we would have to sell a completely different headset for the low end phones. IT managers hated it as well. Problem solved with the IP 335, it comes with a dedicated RJ-9 headset port with Electronic Hook Switch (EHS) support. To accommodate the smaller form factor the headset and EHS connections are made towards the top of the chassis as seen in the image below.
A rear view look at the connections of the Polycom SoundPoint IP 335.
And lastly, Polycom has made the barrier of entry into the HD Voice arena very low. At the $199 estimated MSRP, there is very little reason not to future proof your telephones on the low end with the IP 335’s support of the G.722 wideband codec commonly known as High Definition Voice.
To really beta test a new piece of hardware like this, I knew I needed to go all the way, so I temporarily retired my IP 650 and replaced it with the IP 335. At first I was a little nervous to lose all my “buttons”. But that quickly subsided once I really started making and receiving calls on the smaller phone. I have a Jabra GN9350e with EHS which I also used to connect to the IP 335. It worked like a gem. EHS connectivity was Polycom/Jabra flawless. HD Voice calls were crisp and clear on our Switchvox server. The only thing I missed was my real time buddy status that I get with my IP 650. But bottom line here is this: I could use this phone everyday. And with a low price tag, a lot of people will do just that.
Here are the highlights of the Polycom SoundPoint IP 335:
- High-resolution backlit, graphical display
- Two-line entry-level phone
- Easy to configure and use
- Integrated Power over Ethernet (PoE) support
- Interoperability with leading IP PBX and Softswitch platforms
- HD Voice support
- XML microbrowser
- Backlit 102 x 33-pixel, grayscale graphical LCD
- Two port 10/100 Ethernet Switch
- Dedicated RJ-9 headset port
Voice Over the Grand Canyon: A Switchvox Case Study
October 27th, 2009Grand Canyon Resort Corp (GCR) is the company that oversees Grand Canyon West. Grand Canyon West comprises the Western part of the Grand Canyon in Arizona and is contained within the Hualapai Nation. GCR chose Chromis Technology to install Digium’s Turnkey Asterisk PBX, Switchvox SMB, to create a VoIP solution to connect to remote locations that do not have traditional telephone facilities.
Grand Canyon West is an amazing canyon land that is very close to Laughlin, NV; Kingman, AZ; and Flagstaff, AZ. It is also a reasonably short drive from Las Vegas, NV; Sedona, AZ; and Phoenix, AZ. GCR has multiple attractions in Grand Canyon West including the increasingly famous Skywalk that takes you 70 feet from the rim of the Canyon and suspends you 4,000 feet above the Canyon floor! (Click on any of the images below for the high resolution version in a new window.)
Grand Canyon West Skywalk.
GCR has one Digium AA350 server at their headquarters in Peach Springs, AZ and three AA60 servers at three remote locations. The AA350 has a TE122 card that connects to a local PRI circuit. A satellite connection to each of the remote locations connects each site back to headquarters via IAX trunks and provides a connection to the public telephone network. The G.729 compression codec is used for calls between the servers and G.722 (HD Voice) and G.711 is used for calls internal to each server. In addition to the Digium Switchvox servers, GCR chose Polycom SoundPoint IP telephones.
A big challenge for GCR was connecting those remote facilities. A recently installed a satellite data network from HughesNet is serving up data and voice services to the remote locations. As is common with Satellite data links, excessive latency can wreak havoc on VoIP and make it difficult to have a normal, duplex conversation. GCR appreciates this fact and primarily focuses on providing telephone service to the edge of the Grand Canyon where there is no terrestrial connection to the rest of the world.
View of the Guano Cafe at Guano Point.
Indeed, latency proved to be high, but was often as low as 600ms. Unfortunately the latency results still cause over-talking (when one caller talks before the other caller is finished). The greater problem with the satellite link is jitter, or variation in the delay. During one test the latency varied from 600ms to 1100ms. GCR installed accelerators from Expand Networks to improve the performance over the satellite links.
Soar through the canyon on a Helicopter tour and enjoy spectacular views.
A beautiful view of Grand Canyon West and the Colorado River below from Guano Point.
The latency and jitter fall just outside the range of acceptable, and the delivery of calls is certainly not what we’re used to back in civilization, but by using Digium’s VoIP technology, calls are now being made in and out of Grand Canyon West like they never have before. For more information on how Chromis Technology can help your business leap canyons, give us a call at 602.357.8070 or email us at info@chromisinc.com.
dCAP: The nitty gritty…
October 26th, 2009
Our resident Digium Certified Asterisk Professional (dCAP), Jonathan Rusk, was asked about the certification process in the wake of Astricon. I thought his response was too detailed and too good not to publish as its own entry. So if you’ve ever wondered what the dCAP is all about, here you go:
“There are two parts to the dCAP: 1) practical, 2) written. The practical requires you to configure a fully functional but tiny PBX. You start with Linux and must compile and install Asterisk, etc. Then you configure two or three phones (my test included one Polycom, one X-lite SIP phone, and one analog phone connected to an FXS port on a TDM410). You must also configure trunking (SIP or DAHDI), a simple dialplan, and a simple menu (auto attendant). Overall the practical test was easy for someone with Asterisk experience but the limited time (90 minutes) proves to be the largest challenge. The best best approach is quick and dirty configuration. There was no Asterisk GUI for my test; all configuration was done using text files (extensions.conf, sip.conf, etc.).
The written test was about 114 multiple choice questions. The time limit is also 90 minutes which was more than enough for me. The challenge for the written test is the breadth of subject matter. For example, I don’t typically work with H.323 but there are H.323 questions on the test. The Asterisk Advanced class helps very much with the practical test and helps with some of the the written test but does not cover everything seen on the written test. If you know typical Asterisk PBX configuration and design, and you read the O’Reilly Asterisk book, then the written should be passable.
You need to pass the practical and written parts of the test with 70% or better and each part stands on each own. In other words you can potentially pass the practical with 100% but get a 69% on the written and thus not receive the dCAP certification. My understanding is that you can retake just the practical or written if you only pass one of them.”
So there you have it. Please feel free to reach out to Jonathan with any Asterisk related questions. He can be reached at:
jrusk@chromis.com
602.357.8071 – direct
Skype: ChromisJonathan
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