How to configure hardware independent faxing using HylaFAX and IAXModem is well documented for Asterisk and trixbox, but not for Switchvox, so Chromis customer Jill Rouleau from Liberty Distribution created a Switchvox-specific walk-through. Jill used Debian 5.0, HylaFAX, IAXModem, a Xen virtual machine, and Switchvox. Read on for Jill’s how-to… Thanks Jill!
Posts Tagged ‘digium’
Configuration of hardware independent faxing
Friday, July 9th, 2010Configuring Mac OS X for Switchvox Faxing
Wednesday, July 7th, 2010
The Digium Switchvox User Manual contains general information for setting up your computer for outbound faxing. The User Manual gives detailed instructions for Microsoft Windows but not for Apple Mac OS X nor Linux. Outbound faxing with Switchvox is as easy as setting up a network printer but it turns out that the OS X configuration is not as straight forward. Click here to download a PDF document which contains complete instructions on configuring OS X Snow Leopard with Switchvox outbound faxing.
Panasonic Introduces DECT Wireless SIP Handsets
Thursday, June 10th, 2010
Panasonic KX-TPA50
Panasonic recently introduced a couple of new DECT 6.0 Wireless SIP products that I’ve been able to play with for the past few weeks before they were released. And now that they’re shipping, let me show you what I’m impressed with, here’s the short list:
- G722 Wideband Support
- Provisioning Simplicity
- Price
The support of G722 wideband codec, often referred to as “HD Voice” to use Polycom’s marketing term, is not very prevalent within the DECT wireless product portfolios of SIP telephone manufacturers. Nobody that I know supports it yet, not Polycom, not Aastra, so correct me if I’m wrong on this. But from my perspective this is the first DECT handset that I’ve used that supports the codec. And because I received my test units before the official product release, I didn’t have spec sheets or install guides to tell me that there was support for G722. So when I first provisioned it to my Switchvox SMB server and made a test call, I was very pleasantly surprised. Speaking of provisioning the handsets, it couldn’t be more easy. Simply enter a few bits of information in the easy to use web GUI and away you go.
And last but certainly not least is Price. The base-station and one handset MSRP is $199.95. The Panasonic KX-TGP500 and KX-TGP550 systems can be expanded up to a total of 6 cordless handsets by adding additional KX-TPA50 cordless handsets at $99. Each handset comes with charger and belt clip. Panasonic has made wireless SIP communications very affordable, while maintaining a robust feature set.
From now until June 30th, for every 2 additional handsets you purchase, you get 1 absolutely free! Use the promo code “DECT” during checkout at the Chromis.com store or contact us at 602.357.8070 for more details.
Panasonic KX-TGP500
KX-TGP500 – The system features a wall-mountable base unit and one cordless handset. It is expandable up to 6 DECT 6.0 cordless handsets and supports up to 8 phone numbers and 3 simultaneous calls. It boasts Wide Band Audio (G.722) and 5 hours Talk Time, 5 days Standby. Its elegant design features a white backlit large LCD on the handset and a handset Call Button on the base unit. It also has a handset speakerphone, 2.5mm headset jack and belt clip.
Panasonic KX-TGP550
Also available is the KX-TGP550 – This unit has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD and 5 hours Talk Time, 5 days Standby, plus a Hands-Free Speaker phone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication. MSRP $329.95
Upgrade your Asterisk to Switchvox and $ave!
Tuesday, June 1st, 2010
Does your company have an Asterisk telephone system? Ask yourself this, how valuable would professional support, improved performance, and enhanced features be to your company? Would you like to make a move without having to switch out all your phones or Digium TDM cards? Chromis Technology, a leading Digium Select reseller, can help migrate you to one of Digium’s Asterisk-based, turn key telephone solutions, Switchvox SOHO or SMB.
Asterisk, the open source telephony toolkit, is an industry-changing project that many organizations have used to replace old telephone systems. Asterisk solutions are known for technical advancement and cost savings compared to traditional PBX solutions. Unfortunately the technology world can be complicated and often requires customization to each organization. Asterisk is no different. Chromis has helped several of our customers migrate their Asterisk systems to the powerful, friendly, and commercially supported Digium Switchvox solutions.
Switchvox is created by Digium, the Asterisk company. Switchvox offers easy to use features like administrator and user web tools that reliably allow you to configure your system, control your telephone, run call reports, and interact with other software (like Customer Relationship database “CRM” or Microsoft Outlook). And when you buy a Switchvox SMB solution from Chromis Technology we provide one year of technical support and software updates.
Through August 31, 2010, Chromis Technology is offering our “Asterisk Upgrade Promotion”. Chromis will discount the cost of a Digium Switchvox solution by up to to $1,000! You can save even more by keeping your existing Digium telephony cards and standards based telephones like Polycom, Linksys, Cisco, Aastra, or snom. Contact our sales department for more information via email or telephone at 602.357.8070 if you would like to upgrade your Asterisk and win the benefits of Switchvox.
Chromis Technology Earns Digium Integration Partner Certifications
Wednesday, May 5th, 2010-
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Chromis Technology is pleased to be one of the first Digium Resellers to be both a Certified Switchvox Partner and also a Certified Asterisk Integration Partner under Digium’s new certification requirements.
Digium recently launched these programs to enable customers to to know that they are dealing with serious Asterisk professionals that have met Digium’s stringent requirements.
The certification program tests channel partners and their confidence to design, implement, and support telephone solutions such as a custom Asterisk solution or the turn-key Switchvox IP-PBX. To learn how Chromis Technology can implement your Asterisk or Switchvox solution, please click here to contact us.
Digium Announces AstriCon 2010 Details
Thursday, March 4th, 2010
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.
Enter to Win an Extreme Phone Makeover from Digium, Polycom, and Chromis Technology
Friday, 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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Switchvox VoIP Security
Monday, 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
Wednesday, 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.
Voice Over the Grand Canyon: A Switchvox Case Study
Tuesday, 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.)
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.
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.
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@chromis.com.