Tag Archives: Switchvox

Digium TE133/TE134 Single-Span Digital PBX Cards 2013 Promotion

Looking for single span digital telephony cards?  Digium offers their new TE133 and TE134 cards with built-in echo cancellation at an extremely competitive price point all while maintaining Digium quality and ease of implementation.  From the people that brought you the Asterisk platform paired with their rock-solid 5-year warranty, it’s hard to beat.

Digium TE133 Digium TE134

The TE133/TE134 cards are designed as replacements to the TE121BF and TE122BF cards and are fully compatible with existing software applications as well as integrate fully with Asterisk platform.  Both line-side and trunk-side interfaces are supported as well as many advanced call features.  The cards support industry standard protocols and are capable of running in E1, T1 or J1 modes.  The Octasic based hardware echo cancellation is built into the cards to eliminate the task of echo cancellation from systems CPC which increases overall system performance and call quality.

Specs:

  • Up to 24 (T1/J1) or 30 (E1) simultaneous calls
  • Selectable T1, E1 or J1 Mode / Half-Length, Half-Height, Digital Card
  • PCI-Express (TE133) or PCI (TE134)
  • 1x RJ48 Interface Port
  • Supported protocols include: ISDN PRI, Robbed-Bit, CAS, MFC/R2
  • Built-in 128ms Octasic DSP hardware echo cancellation
  • Supported with both Switchvox and Asterisk
  • 5 year manufacturer warranty

Both the Digium TE133 and TE134 digital telephony cards are in stock and ready to ship at The Telecom Spot.  The current promotional pricing at $495 MSRP will expire end of year and increase by $235 more in 2014, so now is the right time to buy!

Additional resources:

Free User Training with Digium Switchvox Purchase!

From now thru Dec 31, 2013 March 31, 2014 get free Switchvox User Training ($995 value) with purchase of any eligible Digium Switchvox system or software.  The Digium Switchvox training, Introduction to Switchvox:  User Interface On-Line Training – English, is ideal for new users that want to learn how to get started using their Switchvox business phone systems.

Eligible Switchvox models include:

Please Note:  The free user training offer is only available at the time of a qualified purchase, and will not be available separately post-purchase.  Contact us for details.

 

New Digium Switchvox 80 and 300 Series Appliances – Now Smaller and More Powerful

Digium recently released new Switchvox 80 and 300 series appliances for the SMB market, replacing the older Switchvox 65, 305 and 355 appliances.

The new Switchvox 80 and 300 series have updated technology, smaller form factor, green design and are competitively priced.  Switchvox 80 replaces 65, 310 replaces 305, and 360 replaces 355.  The older Switchvox appliances (65, 305, 355) are discontinued as of July 30th (limited stock remains, check with us for availability).

If you’re a current user of the older Switchvox appliances, you can upgrade by purchasing a cold spare of the appliance that you choose instead of purchasing Switchvox with the software license.  Then you can upload the last backup to the new appliance and be back up and running!  Just make sure the Switchvox software version is 5.7.1 or newer. The old appliance can be put to good use as a failover cold spare for your disaster recovery plan.

See which one of the Digium Switchvox appliances is the best fit for you via the convenient comparison chart below.  You can build your custom bundle via our easy system configurator or simply contact us today for quote!

Digium Switchvox Comparison

Click Image to Enlarge

*About Digium Switchvox phone systems:  Digium Switchvox offers a single powerful set of features you can afford.  You pay one low price for all its communication power – it’s an United Communication system that integrates all office communications including phone, fax, chat and web mash-ups.  Business that wants to do more than just talk can count on Switchvox to help them easily transition from simple telephony to a feature-rich UC solution.  Business users can improve productivity no matter where they are – on a mobile phone or in the office.

New Digium TE133 and TE134 Digital Telephony Cards – Now Available!

Digium just released their best PBX cards for any Digium installation with the TE133 and TE134 digital telephony cards.  They are designed as replacements to the TE121BF and TE122BF cards and offer an incredible introductory price of $495 (log into our site for even better pricing) and include echo cancellation and the single span card.

Digium TE133 Digium TE134

Specs:

  • Up to 24 (T1/J1) or 30 (E1) simultaneous calls
  • Selectable T1, E1 or J1 Mode / Half-Length, Half-Height, Digital Card
  • PCI-Express (TE133) or PCI (TE134)
  • One (1) RJ48 Interface Port
  • Protocol support includes: ISDN PRI, Robbed-Bit, CAS, MFC/R2
  • Built-in 128ms Octasic DSP hardware echo cancellation
  • 5 year manufacturer warranty

The TE133 and TE134 cards support industry standard telephone protocols including Primary Rate ISDN (North American and Euro standard) protocol families for voice.  Both line-side and trunk-side interfaces are supported as well as advanced call features.  Octasic based hardware echo cancellation is built into the cards to eliminate the task of echo cancellation from systems CPC which increases overall system performance and call quality.  They are supported with both Switchvox and Asterisk, plus a 5 year manufacturer warranty from Digium.

Both the Digium TE133 an TE134 digital telephony cards are now in stock and ready to ship at The Telecom Spot!  Initial stock is limited though, so don’t wait too long to get yours.

 

Free Digium Phone App Engine API Webinar – Register Now

On May 16th at 2pm CDT, Digium will offer a free webinar to show you ins and outs of the Digium Phone App Engine API.  The free phone API allows developers to customize the user experience using Asterisk (or Switchvox) with simple set JavaScript APIs.

The webinar will cover topics like:

  • What makes Digium’s Phone App Engine different
  • How Business Apps help in your niche
  • JavaScript API vs XML browser pages
  • The anatomy of a business phone app
  • Developer and non-developer resources
  • Live Q&A session

If you want to build custom apps for your Digium IP phone to take your business communications to a new level, this would be a great free webinar for you.

Register for the webinar here.

*The Digium App Engine for Digium IP Phones is a free tool that features an open API allowing developers to create custom apps. Developers can use JavaScript to easily create applications that extends the power of Digium phones with Asterisk or Switchvox.

Provisioning Polycom Phones with Switchvox Phone Setup Tool

Looking to provision your Polycom phones with Switchvox? If you purchased Phone Feature Packs for your Polycom phone and has one of the supported phones listed below, then you can take advantage of Digium Switchvox Phone setup Tool to help you configure and maintain the phones.

  • Fully Supported:  SoundPoint IP 320, 321, 330, 331, 335, 430, 450, 501, 550, 560, 650, 670, and SoundStation IP 6000, 7000
  • Partially Supported:  SoundPoint IP 301, 600, 601, and 4000
  • Not Supported:  SoundPoint IP 300, 500

Follow the steps below for Polycom IP phones running SIP firmware 1.6.7 or higher:

1)      To provision your Polycom phone with the Switchvox Phone Setup Tool, first make sure that:

  • Your Polycom phone is listed in the “Unknown Phones” section (Setup -> Phone Feature Packs -> Unknown Phones)
  • You are using the eth0 port (Server -> IP Configuration)
  • Confirm the phones are on the same local subnet as your Switchvox server
  • You have enough Phone Feature Packs to cover the number of phones you are setting up

2)      Once the top 3 criteria have been confirmed, reboot your phone to default (Menu -> Settings -> Advanced -> PW: 456 -> Admin Settings -> Reset to Defaults -> Reset local configuration).

3)       When the phone is completely back up, reset the ‘Device Settings’ (Menu -> Settings -> Advanced -> PW: 456 -> Admin Settings -> Reset to default -> Reset device settings) which will reboot the phone again.

4)      Check the radio box next to the phone MAC address you’re looking to configure in the “Unknown Phone” section of the “Phone Feature Packs”, then click the blue “confirm checked phone” button at the top of the page.  The phone should automatically reboot and begin the formatting process, but if it didn’t automatically reboot then you can manually reboot the phone either through its menu or by unplugging the phone.   Note:  This process could take several minutes and several reboots.  Be patient and let the phone do its work so it can pull all the necessary firmware.

5)      When step 4 is completed, the phone should display a static screen with the Digium Switchvox logo and the word “New” next to Line 1 on the phone.

6)      Next, locate the phone MAC address in the “Unconfigured Phones” section of the phone setup tool.  Once the phone’s MAC address has been located, click “Modify” and enter in the extension you wish to configure to this phone and save.   This should auto-populate the extension info such as first and last name.

7)      Check off the radio box again next to the MAC address and click the blue “confirm checked phones”.  This will reboot the phone again, but if not just reboot it manually by unplugging the phone power or go thru its menu options.  When the reboot completes, the phone MAC address should be listed under the “Configured Phones” section.  Step 7 is the final step, and will configure your phone with the proper extension, register the phone with Digium Switchvox, and populate the contact directory with the phonebook extensions for that particular extension.

8)      Congratulations!  Your Polycom IP phone should now be configured.


Note:  If on step 1 you couldn’t find your Polycom phone’s MAC address under “unknown phones”, or if the phone is on a different subnet / a remote phone, then try these steps to get your phone visible to Switchvox.

  1. Reboot your phone to default (Menu -> Settings -> Advanced -> PW: 456 -> Admin Settings -> Reset to Defaults -> Reset local configuration).
  2. When the phone is back up, reset the ‘device settings’ (Menu -> Settings -> Advanced -> PW: 456 -> Admin Settings -> Reset to default -> Reset device settings) which will reboot the phone again.
  3. After the phone is completely back up, go to Menu -> Settings -> Advanced (PW: 456) -> Admin Settings -> Network Configuration -> DHCP Menu -> Set “Boot Server” to “static” then go back one menu and select “Server Menu” field.
  4. On the next page, set “Server Type” to “HTTP” and the “Server Address” to “IP.of.the.PBX/pc”.  Note:  The server address field is case sensitive.
  5. If the phone remote connects via a WAN, the “server address” will be the WAN IP of the Switchvox network / pc.  (ie: 173.227.12.32/pc)
  6. Save your settings and reboot the phone again.
  7. Once the reboot completes and is back up, you should now be able to find the phone in the “unknown phones” or the “Unconfigured Section” of the Switchvox “Phone Feature Packs”.
  8. Go back up and follow the rest of instruction from either step 4 or 5 (depending on where you found the phone MAC address) to complete the phone configurations.

Any question?  Just leave a comment below or contact us for assistance.

Save with Digium Switchvox UC Solution

Are you a small or mid-sized business (SMB) looking to implement Unified Communication for your business?  With the right Unified Communication solution, your business could gain great efficiency, raise revenue and increase customer satisfaction.

Digium Switchvox UC

We realize it can be overwhelming as you begin your search for a new phone system not knowing where to start with so many options.  If you would like great features and flexibility all for a reasonable investment, we’d like to suggest Digium Switchvox as a starting point in your research.  Why?  To start with, Frost & Sullivan the current leading industry analyst have researched the Unified Communications market and did all the work for SMB consumers.  As a result of their findings, they’ve awarded Digium Switchvox their 2011 Customer Value Award winner as the best value solution for Unified Communication for SMBs.

Digium’s Switchvox phone system offers a solid UC solution including enhanced features and significant cost savings for SMBs compared to similar solutions offered by other vendors such as Avaya IP Office or ShoreTel ShoreGear VoiceSwitch.  And according to the graph below by Digium, a company with 100 employees could gain around 40,000 additional hours a year by utilizing the productivity tools offered by their Switchvox UC platform – that’s like adding another 18 employees to your staff!  Whether you’re trying to do more with less, or trying to be more efficient to fuel growth, the hours and dollars saved can be significant.

Switchvox Efficiency Gains Chart

Let’s take a closer look at Digium Switchvox and why they ranked best value for SMBs:

  1.  Competitive UC Features

  • True UC system – integrated PBX, unified messaging, conference bridge, fax server, instant messaging, presence, video calling, mobility, call queues, recording/monitoring, and call logging software functions onto a single rack mount or desktop appliance.
  • Customizable Switchboard UC client provide access to all features/applications in one place with personalized view.
  • Support SIP standards based phones (Digium also offers IP phones designed for Asterisk for best integration)
  • Excellent scalability, applications support and applications integration.

  2.  Value-added Capabilities at Competitive Price

  • A single activation license provides users access to all embedded applications and features including the Switchboard UC interface
  • Various applications and server hardware supports charged as extra add-on with alternate solutions are already included with Switchvox.
  • Software update subscription plans to receive new features and technical supports starting at only $10 a year per user.
  • No charge for Digium APIs.  MS Outlook and Firefox plug-ins, CRM interface panels, social media integration tools, and additional web mashups are included and built-in with Switchvox.
  • Develop CEBP integration by downloading software development kits or access Digium’s developer website and forums for free.

3.  Great Value

  • Cost-effective over system lifecycle including ongoing support/maintenance costs and upgrades
  • Single appliance reduces power, real estate and hardware requirements compared with multi-box alternatives with similar functionality.
  • Reduced training and time managing applications by using a simple, common Switchboard interface.
  • IP phones can be deployed with auto-provisioning features which reduce set-up time and frustration.
  • Save time and boost productivity with Switchvox UC.

For additional savings on your business investment, also check out the tax savings you could get via Section 179 for business equipments purchased during 2012.  Furthermore, as an additional bonus, we’re also offering a free iPOD Shuffle with each purchase of Digium Switchvox phone system or Switchvox system bundles now through end of 2012 (see full offer for details).

Sounds good?  Explore Digium Switchvox on our site or feel free to contact one of our helpful consultants for more information.  Due to manufacturer MAP restrictions, don’t forget to log in or contact us for our special pricing on your Digium Switchvox systems!

digium Switchvox Set to Raise Prices

We’ve just been notified by Digium that beginning September 1, 2012 there will be changes made with digium Switchvox systems both in pricing and software/subscription bundles.  The changes are:

  • Switchvox will no longer come bundled with 10 silver subscriptions and the 10 phone feature packs (this will affect all versions of Switchvox SMB/SOHO with or without appliance).
  • Switchvox appliance price will be increased reflecting manufacturing costs.

After flooding in Thailand last year, digium having been picking up the tabs in significate hard-drive prices expecting it to return to pre-flooding levels.  It was recently known that pricing will not be dropping back down, so it is now necessary for digium to make adjustments to the MSRP of Switchvox appliances.

Contact us today if you’ve been thinking about digium Switchvox as the current low pricing won’t be around for long!

How to Configure Panasonic IP Phone with Asterisk

The Panasonic SIP IP Phones are top-notch end points from a brand name you trust. Getting them setup and working usually isn’t very difficult if you know what info needs to go into what fields.  The following tutorial applies to the follow phones, however, some interfaces may change slightly from model to model: KX-UT113-B, KX-UT123-B, KX-UT133-B, KX-UT136-B, KX-UT248, KX-UT670, KX-TGP500, KX-TGP550.

Configure Panasonic

First you will need to setup the extension on your Asterisk System (the process can vary depending on the distribution). From this process, you will need the:

  • IP Address of the Asterisk Server: ie: 192.168.42.100
  • SIP Port of the Asterisk Server: ie: 5060
  • Extension SIP ID/Alias: ie: MAC Address of phone
  • Extension Number: ie: 7000
  • Extension Password: ie: 1234pass
  • Voicemail Access Number: ie: 8500

Next we will get started on configuring the phone itself.  Please note, this is tutorial on how to get your phone connected.  Advanced functionality is not covered:

  • Connect the phone to the network and power it on
  • Once connected to the network, you will need to turn on embedded web from the individual phone settings (turns off after a while). Press the Setup button > Select Embedded web > Select On
  • Open up a web browser of your choice and type in the IP Address of the Panasonic IP Phone into the address bar.  A login screen should appear.  The default login is L: admin P: adminpass.
  • Once logged in, you should make sure you have upgraded the firmware to the current version, the link to the admin guide and firmware files can be found here: http://www.panasonic.net/pcc/support/sipphone/download/us.html
  • Select the VoIP tab and click on Line 1 under SIP Settings.  Fill in these fields:
  • Phone Number: Extension # assigned in Asterisk. ie 7000
  • SIP URI: Extension SIP ID/Alias assigned in Asterisk. ie: MAC Address of phone
  • Registrar Server Address: IP Address of the Asterisk Server. ie: 192.168.42.100
  • Registrar Server Port: SIP Port of the Asterisk Server. ie: 5060
  • Proxy Server Address: IP Address of the Asterisk Server. ie: 192.168.42.100
  • Proxy Server Port: SIP Port of the Asterisk Server. ie: 5060
  • Presence Server Adress: IP Address of the Asterisk Server. ie: 192.168.42.100
  • Presence Server Port: SIP Port of the Asterisk Server. ie: 5060
  • Outbound Proxy Server Address: IP Address of the Asterisk Server: ie: 192.168.42.100
  • Outbound Proxy Server Port: SIP Port of the Asterisk Server: ie: 5060
  • SIP Authentication ID: Extension SIP ID/Alias on the Asterisk. ie: MAC Address of phone
  • SIP Authentication Password: Extension Password on the Asterisk. ie: 1234pass

Config Panasonic with Asterisk Graph1Click Save

Lastly, Select the Telephone tab.  Then Select Line 1 under Call Control.  Fill in these fields:  Voice Mail Access Number: Voicemail Access Number on the Asterisk: ie: 8500

Config Panasonic with Asterisk Graph2Click Save and reboot the phone.

Your Pansonic IP Phone should be configured to work on your Asterisk IP PBX.  Any question? Just leave a comment or contact us at The Telecom Spot for assistance.

How to Setup a T1/PRI on Switchvox

Configuring a T1/PRI circuit may seem like a daunting task, however, the Switchvox AA65, AA305 and AA355 platforms make it as easy as possible.  Here is an excerpt from Digium Switchvox Knowledgebase on how to configure a T1/PRI:

Obtain Configuration Information
Before you begin configuring your T1 PRI interface in Switchvox, it is very important that you obtain all of the configuration information from your T1 provider. Throughout this document we will refer back to this information.

Installing your T1 Card

First make sure you machine is completely shut down and the power cord is removed from the machine’s power supply.

  • Make sure the correct jumper setting is selected on your card (T1 or E1)
  • Install necessary card(s) and line(s) to your PBX machine
  • Start up your PBX

Configure the Card
To start configuring your T1 card, login to the Switchvox admin section (https://your.pbx.ip/admin) and go to System Setup –> Hardware Setup. Locate the T1 card you wish to configure and click Configure. Under Configure Span Options, enter in your T1 information exactly as it is specified by your provider. Click Save Settings and after the reload click Goto Channel Admin.

Configure the Channels
T1 lines consist of 23 Bearer Channels, and 1 Data Channel (channel 24).
E1 lines consist of 30 Bearer Channels and 1 Data Channel (channel 16).

The next step is to configure the channels of your T1 card. PRI T1s have two groups, the data and bearer channels. The data channel is a single channel, most often on channel 24 that provides signaling for the PRI line. Bearer channels are the channels that you use to place phone calls. For a full voice T1, channels 1-23 are all part of the bearer group. Partial T1 lines have some subset of this group.

Data Channel
To add the data channel, click Create New Channel Group. Select a name for your group, ‘data’ is a reasonable choice. For multi-span setups, you may want to specify the span in the group name (ex. Data Channels Span 1). Under Device Type select D (Data) Channel, Secondary Device Type should be set to No signaling (PRI D channel). You can leave the Callback Extension blank.

Scroll down and check the 24th channel on the card and click Save Channel Group Settings.

Bearer Channels
Next comes the bearer channel group. Click Create New Channel Group and fill in the Group Name (ex. T1 Voice Lines or T1 Voice Lines Span 1). Under Device Type, select B (Bearer) Channel. For Secondary Device type, select from the two PRI (Primary Rate Interface) options: CPE Side or Network Side. Your T1 provider should supply you with which setting to choose, but most providers use CPE Side. The Callback Extension is the default extension to ring for incoming calls on these lines. Ask your T1 provider about your settings for the PRI Switch Type, Overlap dial, and PRI Dialplan settings. (Note: For most North American PRI providers the default settings in Switchvox should work.) Next, fill in the caller id settings that you would like to appear on outbound calls through this t1.

Lastly, check the channels that apply to this T1. For full voice T1s, check channels 1 through 23. For partial T1s, check the appropriate subset of channels. Click Create Channel Group and your T1 should now be configured. Finally, check your System Status page to make sure your newly created channels are working properly when things are hooked up.

Multi-span users
For multi-span cards, you’ll see the full consecutive listing of channels. You’ll need to create multiple channel groups accordingly. Here is an example.

Group Name: T1 Voice Lines Span 1
-choose channels 1-23
Group Name: T1 Data Line Span 1
-choose channel 24
Group Name: T1 Voice Lines Span 2
-choose channels 25-47
Group Name: T1 Data Line Span 2
-choose channel 48
and so on…

Configure Outgoing Calls
You can now go to the System Setup –> Outgoing Calls section and configure outgoing call rules to use the bearer channel group you just created.

Incoming DIDs
You can also route incoming DIDs by adding rules to the Incoming Call Routing section under System Setup –> Incoming Calls. Please refer to the DID FAQ entry for more detail.