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5 NO BOX Polycom SoundPoint IP 320 #Asterisk #Switchvox #VoIP

Thursday, May 28, 2009


Make an Offer - sales@e4strategies.com
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posted by .e4 Technologies, 9:24 AM | link | 0 comments |

Aastra 9133i On Sale

Wednesday, May 27, 2009

Popular amongst Asterisk and Switchvox users, the Aastra 9133i is an advanced, fully featured multi-line IP Telephone that takes full advantage of VoIP technology by offering a flexible, interoperable solution at an affordable price.




.e4 is pleased to announce a price drop on the Aastra 9133i Phone. For a limited time the 9133i will be offered at $114.00 - Contact Support for VOLUME Discounts.
posted by .e4 Technologies, 7:27 PM | link | 0 comments |

New Polycom SIP software releases

Thursday, May 21, 2009

New SIP software releases are now available and affect the Polycom SoundPoint IP, SoundStation IP, and the VVX family of products.

Starting with SIP application release 3.1.2, software is being distributed in both a ‘combined’ and a ‘split’ format. Use of the ‘split’ format will result in a faster phone upgrade time. To take advantage of this improved upgrade time, all phones must be running BootROM 4.0.0 or later. In the event that a deployment has some phones running earlier BootROMs, it is recommended that all phones are upgraded to a newer BootROM. If this is not possible, downloading and unzipping both the ‘split’ and ‘combined’ distributions will give the benefits of the faster upgrade to newer phones and still allow older phones to upgrade.

SIP 3.1.3RevB

An updated software release SIP 3.1.3RevB is available from the Polycom Support web site:

  • A combined download for use where phones are running pre-4.0 BootROM
  • A split download for use where all phones are running BootROM 4.0 or later. Use of this distribution will result in faster upgrade times.
posted by .e4 Technologies, 7:25 AM | link | 0 comments |

Polycom Kirk 2010 vs. Polycom Kirk 5020 vs. Gigaset S657IP vs SL785

Wednesday, May 20, 2009

Here is a quick pic for all of the mobility enthusiasts out there... I personally think that this photo represents the best of breed devices in this space.

Pictured from Left to right...

Polycom Kirk 2010, Polycom Kirk 5020, Gigaset S657IP, Gigaset SL785.



CLICK PHOTO TO ENLARGE
posted by .e4 Technologies, 11:22 AM | link | 0 comments |

iPhone App for Switchvox

Tuesday, May 19, 2009

Here is a sneak peek at the iPhone app for Switchvox 4.0

Coming Soon...


posted by .e4 Technologies, 8:52 AM | link | 1 comments |

Digium Launches Switchvox Developer Central

Code, documentation, tutorials and discussion for developers creating unified communications applications.

http://developers.digium.com/switchvox/

05.19.2009 – HUNTSVILLE, Ala.--Digium, Inc., the Asterisk Company, today unveiled Switchvox Developer Central, an online community for developers who are integrating voice and web applications using the Switchvox unified communications solution. Switchvox is Digium's family of voice over IP (VoIP) phone systems for small and mid-sized businesses (SMBs). Switchvox systems, which are based on the open source Asterisk telephony platform, are cost-effective, easy to use and full of features that are typically found only in expensive PBXs.

Available since April, Switchvox SMB 4.0 includes the new Switchvox Extend API. This new toolset lets developers integrate Switchvox with their business applications using an XML API, IVR management tools and event notifications. Utilities such as Fire Dialer, the click-to-call extension for Firefox, or the Switchvox Outlook Plugin are examples of the applications that can be created using the Switchvox Extend API. The newly released API is currently in beta.

Switchvox Developer Central is a website for developers to connect with one another to share ideas and solve problems. It includes a wiki containing all documentation for the Switchvox Extend API, a forum for ongoing discussion, a blog for the Digium engineering team to post news to the community, and tools to simplify development and testing. Digium’s new developer crossroads, http://developers.digium.com, lets users choose their development platform or path—Asterisk.org if they want to contribute directly to the open source software or Switchvox Developer Central if they want to integrate with Switchvox using the Switchvox Extend API.

"The Extend API was one of the most important new capabilities released in Switchvox SMB 4.0 and we want to provide documentation for it in a living format," said Joshua Stephens, general manager of Digium's San Diego operations, where Switchvox is developed. "An administrator or reseller of a Switchvox system can integrate their phone system with a custom web application that's completely tailored to their business or an employee's job function. If they have the skills to create the web application, integrating with Switchvox will be easy because they can use whatever programming language they're comfortable with, so there's virtually no learning curve or specialized knowledge required. If they've worked with any web-based API before, this is going to look really familiar, so they should be able to ramp up quickly."

posted by .e4 Technologies, 8:49 AM | link | 0 comments |

Free VoIP for your Switchvox PBX

Friday, May 15, 2009

In a world where VoIP providers are in a race to the bottom in terms of pricing http://www.ipcomms.net/ has officially one up'd the competition by offering prospective clients unlimited free inbound VoIP service.

Here's what you get...

-Unlimited free incoming minutes
-No commitment, No Purchase Necessary
-NO Credit Card required
-Receive one free number+ with two ports
-SIP Delivery

To claim your free DID CLICK HERE

The IPCOMMS free DID works with all Asterisk based PBX systems. Today we'll focus on Digium's Switchvox solution and outline the complete set up with 4.0 and Free Edition.

Once you've received your welcome letter navigate to your Switchvox admin screen.

located at https://IPADDRESSOFSWITCHVOX/admin

Once you've logged in select System Setup --> VoIP Providers


Once on the VoIP provider set up screen select SIP Provider from the pull down menu and click go.



Once you've selected the SIP provider enter the following information found within your IP communications provisioning letter.

Enter the following information...

  • SIP Provider Name = IPC-Inbound
  • Your Account ID: 1
  • Your Password: 1
  • Hostname/IP Address: Origination IP found in the IPC provisioning letter
  • Callback Extension: Default ext. to ring when receiving a call from this trunk


Once finished, click Add SIP Provider.

posted by .e4 Technologies, 7:10 PM | link | 0 comments |

Product Spotlight: Edgewater Networks - EdgeMarc Appliances

Wednesday, May 13, 2009


EdgeMarc Appliances:

The EdgeMarc suite of Network Services Gateways offer small to large enterprises reduced operational and capital expenses of up to 90% while improving call quality, security, and simplifying the user experience. They are an all-in-one solution that contain all of the features needed to deliver business class VoIP services to the enterprise. All EdgeMarc products are Plug & Dial ready and feature an Integrated VoIP Test Agent allowing service providers and enterprises to install hosted VoIP service in minutes, not hours.


EdgeMarc Product Family


200 Series - The EdgeMarc 200 Series combines multiple voice and data features into a single, easy to use network services gateway. It includes models that integrate a router with an ethernet wide area network (WAN) or ADSL uplink, ethernet LAN switch, integrated access device (IAD) analog ports, PSTN gateway, call quality probe and optional 802.11 wireless access point for data. Designed for Small Office/Home Office (SOHO) environments it supports up to 10 concurrent WAN VoIP calls.

4500 Series - The EdgeMarc 4500 Series combines multiple voice and data features into a single, easy to use network services gateway. It includes models that have up to 4 T1 WAN interfaces or a single ethernet WAN, a 4 port managed VLAN switch, call quality probe optional 802.11 Wireless Access Point and optional integrated analog phone and line ports. Designed for SOHO and small to medium enterprise deployment the 4500 Series contains models that support 2, 5, 10 or 30 concurrent WAN VoIP calls.


5300 Series - Ideal for medium to large enterprise locations the 5300 Converged Network Appliance includes 2 x 10/100/1000 Ethernet interfaces, a 1 x 10/100 Mbps Ethernet out of band management interface and support for up to 300 concurrent WAN VoIP calls.

6400 Series - Designed for demanding environments the 6400 Network Services Gateway offers 2 x 10/100/1000 Mbps Ethernet interfaces and dual redundant, hot swap AC or DC power supplies and support for up to 1000 concurrent WAN VoIP calls.


Contact .e4 Today- Our trained team of VoIP Specialists are standing by to help you make an educated decision.


By Phone: 877-434-8647

By Email: info@e4strategies.com

LIVE CHAT: CLICK HERE!
posted by .e4 Technologies, 8:45 PM | link | 0 comments |

The VoIP Users Conference


If you've been hanging around twitter on a Friday you've probably seen the Asterisk/VoIP tweeters buzzing about the VoIP Users Conference. The VUC as it is often referred to by regulars is a unique way for Fans of Asterisk and other VoIP/IP Telephony technologies to get their geek on.

About the VUC:

The VoIP Users Conference is a weekly live discussion about VoIP, SIP, Asterisk and all kinds of telephony-related topics. The conference has been running for over two years.


Hosts:
Randy Resnick aka Randulo/Zeeek @randulo
Michael Graves
- @mjgraves


On the Web:

http://www.voipusersconference.org/


When:

Every Friday at 12 Noon Eastern Time (9AM Pacific, 5PM GMT, 18h France)


How:

The VUC is available through various listening methods. My personal favorite being the ZipDX Wideband Bridge. For those lucky enough to have a G722 "HD VoIP" Compliant Device the Conference can be reached via SIP URI at 200901@login.zipdx.com.


The VUC is also available on Talkshoe- Instructions for joining can be found HERE


On irc.Freenode.net #voip-users-conference or http://tr.im/vucirc


Episodes Archive:

http://www.voipusersconference.org/listen-now/


Sponsors:

http://www.digium.com/

http://zipdx.com/

http://www.onsip.com/

http://www.e4strategies.com/


posted by .e4 Technologies, 7:55 PM | link | 0 comments |

Digium and snom Simplify Phone Deployments for Small and Medium Businesses. #Switchvox #Asterisk #VoIP

Tuesday, May 12, 2009

HUNTSVILLE, Ala. - (via Business Wire) With the release of Switchvox SMB 4.0, Digium®, Inc., the Asterisk® Company, now makes it easier than ever to deploy snom VoIP phones. The partnership between Digium and snom technology AG, a leading developer of VoIP phones, allows Switchvox®, the web-aware IP PBX designed for small-to-mid-sized businesses, to automatically detect and provision snom 3 series phones, as well as the company’s new 820 phone. This capability reduces setup time and allows businesses to easily and inexpensively deploy phones to the desktop.


snom’s devices are developed specifically to meet the needs of the SMB by offering desktop and wireless devices that combine the latest technology with innovative designs. The company’s desktop 3 series handsets have long had a place in offices around the world while its new m3 wireless mobile device takes advantage of the latest technology to offer clear voice and strong data transmissions.

“Digium and snom have a strong history of partnership and innovation that makes it easy for the business community to use Asterisk and snom together,” said Digium’s Bill Miller, vice president of product management. “Now this extends through to the latest version of Switchvox SMB, and is especially important in the European market, where snom holds a particularly strong position.”

Switchvox SMB is designed for businesses that want a full-featured Voice over IP PBX for hundreds of employees per server at a fraction of the cost of traditional PBXs. With Switchvox SMB, Digium offers the power and functionality of Asterisk—the most popular open source telephony software in the world—combined with advanced yet easy-to-use administrative features and close integration with several communications methods and the web.

“snom and Digium have a long and strong relationship in ensuring that our products work well together,” said Dr. Michael Knieling, sales and marketing director for snom. “Our goal is not only to make the products interoperate, but to take full advantage of all that Switchvox has to offer.”
About snom

snom technology AG develops and manufactures Voice-over-IP (VoIP) telephones and related equipment based on the IETF open standard, SIP (Session Initiation Protocol). Recognized for its high quality, customizable and cost-effective business solutions, snom is also differentiated by the company’s history in the VoIP industry, and its dedication to high security standards. All of snom’s software exists in the firmware on the phones – making it easier for users to download updates and new features. snom customers benefit from the interoperability and flexibility that the snom telephones offer, including plug and play integration and universal compatibility with any SIP-based telephony platform. Founded in 1996 and headquartered in Berlin, Germany, snom technology AG has offices in North Andover, MA, Wuxi, China, Milan, Italy and Paris, France. The company distributes its third generation SIP phones through its network of authorized reseller partners in Europe, South America, Asia-Pac, Africa, and Australia. For more information on snom, please visit our website at http://www.snom.com/.

About Digium

Digium®, Inc., the Asterisk® Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX Software to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of hardware to enable resellers and customers to implement turnkey solutions or to design their own voice over IP (VoIP) systems. More information is available at http://www.e4strategies.com/.

The Digium logo, Digium, Asterisk, Switchvox and the Asterisk logo are trademarks of Digium, Inc. All other trademarks are property of their respective owners.
posted by .e4 Technologies, 8:00 AM | link | 0 comments |

Eliminate the ability for your Switchvox users to call paid 411 - Goog 411 Re-Direct

Monday, May 11, 2009

One of the nice things about Switchvox is its ability to send calls out of specific trunks when specific dial patterns are input in to the user's phone. A good example of what can be done with this would be changing the outbound number dialed when users hit 411 to dial local directory assistance. As all of you know this cost the company money and should be avoided - We have in the past been asked to block this number in many systems however, I find it better to give them a free alternative.

First things first..

As you probably noticed, Switchvox makes the default names directory in the system 411 - Assuming that you are dialing 411 for local directory assistance this would get you to the internal directory eliminating the problem listed above. I generally change this ext number to something else since it can be accessed from an IVR menu using a single digit and folks internally rearely need to call the directory- (Switchvox 4.0 has one built right in to their Switchboard)
The Solution:

Navigate to the /admin screen of your Switchvox PBX. Once you've logged in goto:

System Setup>>Outgoing calls.

Press the Blue Add Outgoing Rule Button

Create an outgoing rule that looks like this....


CLICK TO ENLARGE!




Congrats! When users dial 411 they will be routed to 1-800-GOOG411

NOTE: you can also use this same methodology for 15551212 as well as creating a dial 0 rule for internal callers to reach reception. Just be sure that your Polycom's digit map settings are correct and support the dial patterns for the calls you are trying to re-direct.

Thanks for checking this out!
MW .e4
posted by .e4 Technologies, 7:43 AM | link | 0 comments |

Setting up a Polycom with Asterisk or Switchvox using the phone's WUI

Saturday, May 9, 2009

I was browsing the twitter stream for interesting readables and came across a Switchvox user in need. Here is what I came up with...

First Find the IP address of your Polycom Phone. This is easiest when using the menu on the phone. go to “menu -> status -> network -> tcp-ip parameters”

Once you have this you'll need to open a web browser window on a computer that is on the same network as the phone. Now you can access the phone’s web configuration page by going to http://x.x.x.x where x.x.x.x is the IP address of the phone.

The factory default username and password for accessing web configuration on Polycom phones is “Polycom” and “456” (make sure Polycom starts with a capital P).

On the main Polycom page (image above) click on the "Lines" link in the top nav. On the lines page, under the "Identification" section, enter the extension you have selected for the phone in every field except password (IE: Display Name, Address, Auth User ID, Label, Address, Third Party Name). Enter in the password that you have selected for this extension in the password field. Under Server 1, enter in the IP address of your Switchvox PBX along with the following info:

Port: 5060
DNS Lookup: DNSnaptr
Expires: 120
Register: 1

Click the first submit button and your phone should reboot. Be prepared for a long wait before you can reconnect to the web interface of the phone. Even when the phone is done rebooting, the web interface won't be available for some time.

When it reboots, go back to the Polycom admin and click on SIP. Scroll to the bottom and enter this for the Digitmap string:

FREE ED and SOHO: [0-8]xx911941196119011xxx.T91xxxxxxxxxx9[2-9]xxxxxx

SMB:
[0-8]xx911941196119011xxx.T91xxxxxxxxxx9[2-9]xxxxxx*xx.T

DIGITMAP EXPLAINED:
Each piece of the dial plans listed above will work with switcvox to match specific dial patterns and make the Polycom automatically dial when a rule is matched. Each dial plan is seperated by a pipe or a bar.

What are the rules? By default Switchvox has a set of default outgoing rules. The following Dialplans are designed to work with these rules.

[0-8]xx - Extension rule.. Add more x's for longer ext numbers - [0-8]xxx is a 4 Digit EXT Rule 911 - Emergency Dialing
9411 - Information - I generally change the outgoing rule to make this dial goog 411
9611 - POTS repair
9011xxx.T - International Dialing
91xxxxxxxxxx - Long Distance 1+
9[2-9]xxxxxx - Local Calls
*xx.T - Switchvox Feature Codes

Note that this string may be different if you are not using 9 as an outgoing prefix or if you do not have 3 digit extensions.

Click Submit. Your phone will reboot.

Time Settings:
Go to the General option and edit the SNTP Server and GMT Offset settings. If you don't have a handy NTP server, you can enter: pool.ntp.org. GMT Offset is the offset of your timezone from GMT (IE: New York is -5, California is -8). Click Submit to reboot once more.

Return to the phone's web configuration. In the Lines section, under 'Line 1' at the bottom of the table (not the page) there is a Message Center section. Here's what you need to change:
Leave Subscriber blank. Set 'Callback Mode' to be 'Contact', and then type in the extension you have set for Voicemail Access into the Callback Contact field (DEFAULT 899).

Click submit and a long reboot ensues...

Congratulations your Polycom phone should now be set up and ready to use with your Switchvox PBX.
posted by .e4 Technologies, 5:18 PM | link | 0 comments |

Power Over Ethernet Myths- The Facts about PoE

Thursday, May 7, 2009

Power-over-Ethernet (PoE) technology integrates power and data across standard Cat5/5e/6 network cabling and provides more flexibility in today’s workplace. PoE enables power to be supplied to network devices, such as IP phones, network cameras, and wireless access points through a single, most often existing, network cable. When combined with an uninterruptable power supply (UPS) a PoE network delivers continuous operation and minimizes business downtime by eliminating most power interruptions. With the ability to install endpoints in any location PoE technology provides a scalable and flexible networking infrastructure geared for growth and efficiency.

Myth 1: PoE Switches can provide all the power I need or will need.

Today most switches are merely PoE-enabled. This means the majority rely on power management to share available power across the switch ports. The switches are designed with a smaller power supply that is typically capable of powering the switch itself and providing the required 15.4 watts of power over a limited number of ports.

For example: A 24-port PoE Switch with power management typically has a 195-watt power supply. After the 40 watts needed to power the switch, you have approximately 155 watts remaining. If 12 of the 24 ports are used to connect end devices using 11.5 watts each, you would only have 17 watts remaining to provide power on the last 12 ports. The math doesn’t match the ports: 195W – 40W (switch) – 138 (12 devices @ 11.5W/ea) = 17W left for power on 12 ports

Myth Busted: A PoE Switch is often not the best and most cost effective solution.

Myth 2: A midspan and a PoE switch are the same.

A PoE Midspan is not a switch. A Midspan is an additional PoE power source that can be used to offer full power to all endpoint devices. PoE Midspans (Power Hub or Power Injector) pass data from a switch and ‘inject’ safe power acting as a patch panel of sorts. Midspans are commonly used with either a non-PoE switch, an existing PoE switch, or a new PoE switch in a network. In addition to offering full power across all available ports, midspans costs substantially less per port and overall than a new PoE enabled switch.

Myth Busted: Midspans do not switch – they make use of existing best-in-class switches. They inject safe power across all ports and cost less than PoE switches.

Myth 3: Only a switch that has PoE built in should be used to power devices like IP Phones, Access Points, and IP Security Cameras.

Switches were designed to, well, switch. PoE Switches are designed with power management and have to distribute different power as required to ports but there is often not enough power for all devices plus the power required to complete the primary task - switching. Networks that have multiple devices like IP phones, IP cameras, wireless access points quickly go beyond the limited capacity of managed power PoE switches. As more PoE devices continue to grow in capabilities and market share this managed power limitation will become more and more evident. Midspans, in contrast to switches, were designed to provide full power on every port and deliver safe and reliable power based on the industry standards (IEEE802.3af/at).

Myth Busted: Rather than relying on power management in a switch use a midspan that can deliver full power (15.4W) to every port for all PoE-enabled devices now and in the future.

Myth 4: Ethernet devices not PoE-enabled (non 802.3af/at compliant) cannot be powered using PoE technology.

Many devices do not directly accept Power-over-Ethernet but can still use PoE technology. If the device uses less than 12.5 watts (802.3af) or less than 50 watts (802.3at+) and connects to an IP Ethernet network you can use a PoE splitter. PoE splitters enable you to accept PoE power from any IEEE 802.3af/at compliant switch or midspan then separates the data and power on to two seprate cables. The data is connected to the end device through a standard RJ45 plug while the power is connected using a standard 5.5 x 2.1 x 12mm Adapter Plug. Splitters can also convert the input voltage to the required voltage for a non-PoE device. Splitters are traditionally used with older network products which only accept power through their (DC) jack and data through their RJ-45 jack.

Myth Busted: PoE splitters can be used in conjunction with PoE midspans and switches to provide both the data connectivity and power required by most endpoint devices.

Myth 5: I need/will need additional PoE switch ports to power my IP cameras and high-power pan, tilt, and zoom (PTZ) cameras.

Today, many devices have evolved into more advanced solutions with higher power requirements. The traditional approach was to endure a “forklift upgrade”. This meant buying new PoE switches at considerable cost and physically swapping out the existing switches to meet higher power requirements or add more powered ports. There is an easy and more cost-effective way – separate the data and power in the wiring closet (IBF). It is more efficient and costs less to separate your data and power allowing you to keep your best-in-class business switch for your IP needs and supplement it where required with best-in-class midspan technology to power the endpoints.

Myth Busted: A PoE Switch is often not the best and most cost effective solution.

Myth 6: All midspans are created equal . . . they are all the same.

Always select a best-in-class midspan. If you wanted to enhance your switched network wouldn’t use a best-in-class network switch? Of course you would. A midspan designed and manufactured by a leading power supply company that understands power, power requirements, and one that delivers enterprise-level solutions.

Select a midspan manufacturer that has multiple members on the IEEE (PoE) committee helping to define safe, new PoE standards. This ensures that every midspan is designed to meet current and future IEEE specifications for Power-over-Ethernet.

Select a midspan manufacturer that designs, manufactures, and tests its own product rather than outsourcing these tasks across the globe to cut costs.

Select a midspan that has a high-speed, common interface to access the management console. A USB port is not as cheap as a serial port (RS-232) but is faster, more user-friendly, and more common on high quality midspans.

Myth Busted: Although there are many midspan manufacturers out there, few have the power supply experience, quality controls, and manufacturing capability to produce best-in-class midspans. All midspans are NOT created equal.

Contact .e4 Today- Our trained team of PoE Specialists are standing by to help you make an educated decision:

By Phone: 877-434-8647
By Email:
info@e4strategies.com
LIVE CHAT:
CLICK HERE!
posted by .e4 Technologies, 10:43 AM | link | 0 comments |