Tag Archives: Digium

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.

 

New Line of Digium Phones (D40, D50, D70) are Now Available to Ship!

These new phones from Digium designed exclusively for Asterisk are in high-demand, so get yours before they’re sold out!

Digium Phones

Highlights: 

  •              Easy to Install, Integrate, and Provision
  •              Fully Compatiable with Asterisk & Switchvox
  •              Built-In & Custom Applications
  •              Crystal Clear HDVoice
  •              And Great Value!

Digium D40 – Entry-level HD IP Phone with 2-line keys – Digium’s best value phone designed for any employee.
Digium D50 – Mid-level HD IP phone with 4-line keys and 10 rapid dial buttons with busy lamp field indicators for most important contacts. Perfect for heavy users.
Digium D70 – Executive-level HD IP PHone with 6-line keys, 10 digital rapid dial buttons with real-time status information and busy lamp field indicators for 100 contacts. Top-of-line features ideal for administrators and/or executives.