.e4

April 28th, 2009

New snom leasing program gives Resellers the ability to offer small and medium size businesses snom phones on “Capex friendly” terms - small monthly payments with no money down and no payments for first 60 days.

Berlin, Germany and North Andover, Mass. (April 28, 2009) – snom Technology AG (http://www.snom.com/), a leading developer and manufacturer of advanced Voice over IP phones for enterprise, SMB and residential markets, today introduced a bold new VoIP stimulus plan - a new leasing program with Integrity Leasing and Financing for snom’s portfolio of SIP phones making it easier and more affordable for SMBs to transition to IP telephony for their business. The program, designed for snom’s growing network of resellers that include telecom and interconnect VAR’s and solutions providers and Internet Telephony Service Providers offering Hosted VoIP solutions, significantly reduces the upfront investment and total cost of ownership for snom VoIP phones.“In today’s tough economic climate, companies are looking for new ways to stretch their telecom budgets and manage their cash flow,” said Michael Knieling, executive vice president of marketing and sales for snom. “The snom Leasing Program with Integrity Leasing and Financing addresses this need enabling companies to invest in the latest productivity enhancing VoIP technology that can help drive their top line while continuing to aggressively manage their bottom line.”

“In today’s tough economic climate, companies are looking for new ways to stretch their telecom budgets and manage their cash flow,” said Michael Knieling, executive vice president of marketing and sales for snom. “The snom Leasing Program with Integrity Leasing and Financing addresses this need enabling companies to invest in the latest productivity enhancing VoIP technology that can help drive their top line while continuing to aggressively manage their bottom line.”

The snom Leasing Program The snom Leasing Program with Integrity Leasing and Financing provides its reseller partners who are selling IP PBX and hosted VoIP solutions to SMBs with the option to lease a “bundled package” of phones and VoIP services that can be offered to their end customers as part of a monthly payment plan with no money down and no payments for 60 days. All snom products are included in the leasing package that also incorporates standards based SIP VoIP IP PBXs snom partners and related VoIP equipment presenting VARS with the most complete end-to-end VoIP solution.

VIEW .e4's COMPLETE LINE OF SNOM VOIP PHONES

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

By Phone: 877-434-8647

By Email: info@e4strategies.com

LIVE CHAT: CLICK HERE!

Released on 2009-04-23

Fixed an issue where Caller-ID was not always preserved with RingAll and external numbers.

Caller-ID for voicemail messages is now retrieved later in the recording process to accommodate assisted transfers.

Fixed various issues for the IVR action Send an Email.

Fixed an issue with special characters in external number entries in the Switchboard phonebook.

The SMS email template now sends text-only email when the subject is blank, and the wav is not attached.

Fixed various issues with peering multiple systems over XMPP

Voicemail audio is again streamed over HTTP, which fixes an issue with IE7, Windows Media Player, and HTTPS.

Fixed an issue where simultaneous IAX calls could deadlock VoIP traffic, resulting in phones unregistering.

Fixed an issue that prevented user-initiated call recordings from being offloaded to an external FTP server.

Added the ability to use the + character in outgoing call rules.

Fixed an issue that prevented Aastra phones with newer firmware from registering with the PBX.

Added the ability to disable fax detection for FXS channel groups.

Resolve an issue that prevented single-digit outgoing rules from being used.

Fixed an issue when prepending a message before forwarding in voicemail.

We accidentially ordered the incorrect SKU for a customer so we are unloading these below our cost. Our loss is your gain.







PRODUCT INFO:

The CHAT 50 personal speaker phone is a mobile audio peripheral that connects to a wide variety of devices and provides crystal-clear, hands-free audio communications.

It provides unmatched full-duplex capability, which allows users to simultaneously speak and listen without audio cutting in and out. It also provides high-quality audio playback for music, gaming and other sound files.

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

By Phone: 877-434-8647
by Email: info@e4strategies.com
LIVE CHAT: CLICK HERE!

PRODUCT CHAT 50 USB

Intro:
The continuing advance of networking technology is enabling new and better forms of communication, but also adding complexity to the process. Companies must maintain flexibility in a changing market, while also providing opportunities for growth. Older technologies need to be maintained, while new and improved capabilities are implemented. And of course, costs must be controlled, as employees come up to speed on the latest communication options.

SIP Trunking Defined:
Protocols are conceptual models that let applications exchange data through a communications network. SIP, or Session Initiation Protocol, is an application-layer standard for creating, modifying and terminating Internet Protocol (IP) sessions with one or more participants. SIP has its roots as a protocol for bridging traditional analog telephone networks and IP networks, but it can also support a number of emerging technologies, such as VoIP, presence management and Instant Messaging. These technologies allow information to be converted into a common format, so many kinds of devices can exchange information. Essentially, SIP lets different kinds of IP traffic share the same network connections, which opens new possibilities for integrating voice with other communication options.

Trunking is a communications concept that lets multiple users share network assets by defining access rules for lines, frequencies or bandwidth. A SIP Trunk can support multiple users with voice calls, conference calls, multimedia distribution and other features. SIP Trunking actually offers a number of inherent advantages over traditional telephony. SIP connections can support very high audio quality, and with compression algorithms, can fit more calls within a given amount of bandwidth. Transmitting call-related information (such as caller ID) is easy, and calls can even be enhanced with new features, such as pictures of the caller. With a SIP connection, a telephone number isn’t limited to ten digits, so Direct Inward Dialing could be implemented for every extension.

Why SIP Trunking Matters:
The trend toward converged networks means that SIP-supported communications are the way of the future. Companies can use their IP-enabled networks for both voice and data, making the most of their high-speed Internet connections. They can converge local, long distance, toll-free, private voice and data traffic onto an MPLS IP-based network. Years of investment in data networks can therefore deliver an additional return, when voice and data traffic are moved to a single network platform.

SIP also allows for dynamic bandwidth allocation, ensuring that voice traffic is given the highest priority – calls always go through. There’s no need to pre-allocate space to fixed voice channels, as long as enough total bandwidth is available. That means SIP supports highly efficient use of network resources, allocating bandwidth based on the needs of the moment. Companies can also increase the performance of their call centers by using SIP to enable new routing, tracking and call transfer functionality. SIP supports a number of refer-and-redirect capabilities, which can be used to define communications-enabled business processes. Intelligent routing options can improve process efficiency and increase customer satisfaction. SIP-enabled VoIP systems can also support “Next Generation” contact centers, in which voice communication is integrated with digital communication, supporting features like Web-based “click to call” options. SIP is based on an open standard, and can be the foundation for building new and innovative services. SIP standards add value and functionality, with extensions that cover data compression, messaging, security and call control. SIP allows for the potential to define new services suggested by end users and companies alike, and simplifies interoperability in a multi-vendor environment. SIP can support a number of advanced communications capabilities. For example, hotels could deliver ads for local businesses or restaurants to the screens of SIP-enabled telephones, and also track room availability. Lawyers could track time automatically, with call-logging systems feeding information directly to billing and account management.

A key advantage is that SIP Trunking lets companies implement VoIP technology at their own pace. SIP Trunks can connect existing key systems and TDM PBXs to VoIP networks, allowing for a phased migration. The SIP standard also makes for easier interoperability. Companies can manage their infrastructure investments while migrating to business class VoIP.

Finally, SIP trunking can reduce the Total Cost of Ownership (TCO). By converging voice and data networks, companies can reduce access costs, improve bandwidth utilization and bring down operational expenses. Routine tasks (such as Moves, Adds and Changes) are quicker and easier in a VoIP environment. VoIP networks can be easier to manage than legacy systems, and can often be controlled remotely, using Web-based tools.

Business Implications:
Implementing a SIP Trunk as a road to VoIP offers definite business value, but the specific value delivered depends on the size of the enterprise. Large companies can save a significant amount of money by converging their voice and data networks, particularly if they’re covering multiple locations. Smaller companies can chart a phased migration path toward next-generation technologies without a large up-front investment. Companies of any size will benefit from features like dynamic bandwidth allocation.

The VoIP marketplace today is divided between equipment vendors and service providers, and many companies feel pressured to act sooner rather than later. SIP Trunking services are a good way for companies to experiment with VoIP capabilities while using their existing equipment, eliminating the need for significant capital investment and extensive user training programs. This ability to implement the change gradually is a way to reduce TCO, something that’s often overlooked in purchasing decisions.

Looking Forward:
In the future, SIP will dominate telecommunications, as carriers establish peer relationships to expand on-net calling. SIP applications will become more intelligent, learning how to prioritize messages, adapt to changing user preferences and ensure privacy.

SIP Trunking delivers tangible benefits in cost reduction, productivity and stronger business interactions, in both a B2B and a B2C context. Although still a new concept for many companies it already offers a broad range of features and significant business benefits.

Asterisk SBC: Intertex' new IX78 E-SBC is an ADSL Modem/Router/Firewall with built in Enterprise Session Border Controller (E-SBC) functionality for SIP Trunking.

SIP Trunking is connecting a company PBX to a carrier's IP SIP telephony service instead of via TDM lines, such as PRI, BRI or multiple analog lines. The IX78 E-SBC now allows service providers to mass deploy high quality SIP Trunks cost effectively over ADSL or Ethernet access, solving the SIP NAT Traversal problem, PBX interoperability with the SIP Service,Security and QoS (Quality of Service).

With the IX78 E-SBC, a service provider can roll out a SIP Trunking service over ADSL, where VoIP is delivered over a quality assured PVC and the unused bandwidth is fully used for non-prioritized Internet traffic over another PVC on the same line. The IX78 E-SBC can alternatively be used with Ethernet access -- VLAN tagged or not -- and also prioritize and traffic shape over a single Internet pipe.

The general SIP functionally and handling of multiple high capacity WAN connections with different priority, is inherited from the triple play,any port, any service IX78 broadband product. In the SIP Trunking case,the IX78 E-SBC creates a unified VoIP and data LAN, where the company gets global SIP connectivity both for the PBX and its phones and for other SIP services, e.g. remote SIP users (home workers and road warriors) connected to the PBX over the Internet.

The capacity is high -- over an ADSL2+ Annex M line, more than 30simultaneous calls can be supported, replacing a full TDM PRI line. For larger installations, Intertex' sister company Ingate Systems, with which it shares an unequalled interoperability listing between PBXs and SIP Trunking service providers, can provide higher capacity E-SBC products.

VIEW THE FULL LINE OF INTERTEX IX78 SIP AWARE FIREWALLS

Intuitive, visual control and management of a four-party conference call right from the phone's display

The Visual Conference Management application is part of the Polycom Productivity Suite, a group of applications that increases the value of Polycom SoundPoint IP phones to help companies communicate and collaborate more efficiently.

This application provides an intuitive visual interface to set up and control a four-party conference call from the phone’s display, including the ability to easily add, drop, split, mute, and put callers on and off hold. Thus, the Visual Conference Management mitigates the complexity of using cumbersome “codes” to manage conferencing features.

Supported Phones Visual Conference Management:
SoundPoint IP 450, IP 550, IP 560, IP 650, IP 670, SoundStation IP 6000, IP 7000

Learn More
To learn more about Polycom’s Productivity Suite for SoundPoint IP Phones, please call 877.434.8647, or visit www.e4strategies.com

Increase productivity while reducing operating costs by recycling your SpectraLink Wireless Telephones today!

Program Summary:
Polycom’s Wireless Trade-in Program rewards end users when they adopt the newest generation SpectraLink Wireless Systems. Purchase new Polycom SpectraLink 6000 or 8000 Wireless Telephones or infrastructure from a .e4 Technologies and receive a manufacturer rebate of up to $600 per device!

Why Upgrade?
Check out the Top 10 Reasons customers are choosing to upgrade to the newest generation of SpectraLink Wireless Telephones:

10. Easier to carry Smaller form factor and 35% lighter weight

9. Extend productivity 2x improvement in battery life and talk-time

8. Increased durability Certified for liquid, dust and shock resistance

7. Improved collaboration features Office quality speakerphone

6. Easier to convey information 50% larger display with higher resolution

5. Easier to use One-button speed dialing, side volume adjustment keys, backlighting, larger display and programmable softkeys

4. Environmentally friendly Recycle discontinued handsets and replace with RoHS compliant handsets

3. Renewed warranty Includes one-year wear and tear handset warranty coverage

2. Investment protection Easily integrate into existing wireless infrastructure

1. Cost savings! Limited-time financial upgrade

How to Participate

STEP 1 Purchase new qualifying SpectraLink 6000 or 8000 products from a Polycom Reseller or Channel Partner between January 1, 2008 to June 30, 2009. (See table of qualifying products)

STEP 2 Complete the Wireless Trade-in Claim Form and Letter of Return.

STEP 3 Submit completed Wireless Trade-in Claim Form, Letter of Return, Proof-of-Purchase (.e4 invoice) and trade-in equipment within 90 days of purchase.

STEP 4 Polycom will issue each rebate check directly to the End User customer within 6 weeks of receiving qualified and verified claims.*

*The rebate check will be issued to the End User customer unless item #5 under termsand conditions is properly completed.

For more information on taking advantage of this exciting Polycom Promotion call .e4 at

877-434-8647 or Click LIVE SUPPORT

The NetAlly Network Assessment and Troubleshooting package quickly and automatically determines the network readiness for VoIP and provides tools to identify and resolve service and readiness problems across the entire network.

As a network engineer responsible for VoIP deployment projects and network troubleshooting, NetAlly quickly identifies most of your network issues so that when VoIP deployment is started you are confident that you will not be faced with dissatisfied users complaining of poor call quality.

As a system integrator or VoIP equipment reseller responsible for evaluating whether the network is ready for the VoIP application, use NetAlly to validate the network to avoid delayed deployments, cost overruns and dissatisfied customers.

Pre-deployment Assessment

The network pre-deployment assessment is specifically designed to assess the readiness of the network, and certify the performance for supporting VoIP applications prior to deploying VoIP traffic. The assessment process is divided into three different phases, each of which has a definite objective in the overall assessment scheme.

Don’t take the risk of starting a live VoIP deployment without first conducting a network assessment.

The Network Configuration phase checks aspects of the network and consists of three different tests

• Connectivity
• VoIP traffic precedence (QoS policy)
• Route quality and utilization measurements

The readiness assessment provides the most accurate picture of VoIP readiness for the network that is being tested by:

• Identifying the maximum call capacity
• Identifying major contributors to call quality degradation
• Validate existing capacity planning assumptions

End-to-end call quality is measured using MOS scoring method and synthetic calls can be mixed with real data traffic during normal production hours to get a real world assessment of service quality or scheduled for off-hours operation when criti­cal business applications are not in use.

The performance certification test phase verifies the accuracy of the Readiness Assessment test data by emulating a fixed number of calls over an extensive period of time. This test makes it possible to assess VoIP performance according to time of day and day of week, under varying network traffic conditions.

Recommended Solution for Asterisk and Switchvox Users.

Polycom Experience HD Voice Rebate Program

For a limited time, End User customers can receive a rebate of up to $150 for a SINGLE purchase of TWO new HD Voice equipped SoundPoint IP telephones through .e4.

This offer is valid on new qualified SoundPoint IP products purchased from April 13, 2009 through September 30, 2009. Customers are limited to one rebate claim during the rebate period.

How to Participate

STEP 1 Purchase a minimum of TWO new qualified SoundPoint IP phones with Polycom HD Voice Technology between April 13, 2009 and September 30, 2009. (See table of qualifying products)

STEP 2 Submit one completed Experience HD Voice Phone Rebate Claim Form, proof of purchase, and customer invoice or receipt via mail, fax or email as directed on the claim form. All claim forms must be submitted within 30 days of purchase.

STEP 3 Polycom will issue a rebate check directly to the End User customer within 6 weeks of receiving qualified and verified claims.

SoundPoint IP 450 - 2200-12450-025
SoundPoint IP 450 w/PSU - 2200-12450-001
SoundPoint IP 550 - 2200-12550-001
SoundPoint IP 560 - 2200-12560-025
SoundPoint IP 560 w/PSU - 2200-12560-001
SoundPoint IP 650 - 2200-12651-001
SoundPoint IP 670 - 2200-12670-025
SoundPoint IP 670 w/PSU - 2200-12670-001

Any two SoundPoint IP 450s and/or 550s $100
Any two SoundPoint IP 560s, 650s and/or 670s $150

Labels: polycom

How can VoIP save your business money?

You want to or already have deployed a VoIP (Voice over Internet Protocol) capable phone system for your business, but where are the monthly cost savings, VoIP? You’ve seen some savings by reusing your existing company infrastructure, like network wiring, and you’ve seen a boost in productivity because of all the features that can come with VoIP, and specifically an IP PBX, but do you really need to entrust your voice to the wild west of the Internet to see any real impact on your monthly bill? We’ll explore ways to get the most out of an IP PBX (Internet Protocol Private Branch eXchange) deployment so that your calls are as cheap and as reliable as you are willing to make them. And we’ll look at ways to help you decide how much risk your company can tolerate in the name of slashing phone bills.

Wait, but isn’t VoIP free?

Not exactly, no. If you make a call using VoIP to another user of the same VoIP network, then yes, this call could potentially be free. This is really dependent on what the owner of that network has decided for their policy. If the owner of the network is you, as in the case of multiple IP PBX systems joined together, then yes, those calls are free.

So what are you paying for then?

If you’re not calling another VoIP user, like in the case where a VoIP call is made to a cell phone, somewhere, somehow, that call needs to jump out of the VoIP network and “terminate” into the PSTN (Public Switched Telephone Network). That’s the service you’re paying for when you’re paying for VoIP service.

The main reason that your phone calls are less expensive when using a VoIP provider is because they’re sending your call as far as they can with VoIP, and only sending it as short a distance as they can out on the PSTN. In other words, they’re saving by not sending the call long distance either.

An ITSP (Internet Telephony Service Provider) with many termination points all around the world can have rates well below a traditional carrier for this reason. Take, for example, a call you want to make from Los Angeles to someone’s regular home phone in Paris. If the VoIP carrier you’re using has a termination point in Paris, you’re in luck and the call will travel across the distance just like any other internet traffic (like if you sent an e-mail to someone in Paris), and then when it needs to go from that termination point in a data center out to the PSTN network in Paris, its just a local call, and therefore, cheap!

But all this goes out the window when you consider that most ITSPs are actually just reselling a larger, wholesale carrier’s minutes. So shopping for an ITSP can just come down to shopping for the lowest rates. But buyer beware! Just like anything else, you tend to get what you pay for. There are definitely bargains to be had, but it’s important to know if the carrier you’re researching is reselling someone else’s minutes or if they actually have their own network. It may be better if they’re reselling a larger carrier’s minutes because that large company has a lot of infrastructure, presence worldwide, and support staff. On the other hand, you will get some frustrating answers from ITSPs that don’t own their own network if they’re experiencing an outage. Basically, there’s not much they can do about it. So if you are going to choose to go with a provider that resells rather than owns their own network, the best bet is to choose a carrier that resells several larger carriers’ minutes, instead of depending on just one.

Big Impact: Routing Calls Wisely

The whole goal here is to explore how routing your business calls through the right channels can impact your bottom line, without forcing you to jump into VoIP “head first” at the outset. VoIP may be cheap, but it’s typically no more reliable than the internet, so balancing with PSTN calls would be the wise deployment, due to the government regulations placed on our old telephone network.

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

By Phone: 877-434-8647
by Email: info@e4strategies.com
LIVE CHAT: CLICK HERE!

For a limited time .e4 is offering a FREE Digium Laptop Backpack to all customers who order and FXO/FXS card with a port density of four ports and up. .e4 will also extend this offer to all customers ordering a T1 card of any port Density.

Announces call for speakers, sets October date in Phoenix and invites developers, systems integrators and entrepreneurs to discuss all things Asterisk.

HUNTSVILLE, Ala.-- via (BUSINESS WIRE)--Digium®, Inc., the Asterisk® Company, today released details about the sixth annual AstriCon Open Source Telephony Conference and Exhibition. The event brings together open source and telephony developers, systems integrators, entrepreneurs and Digium partners to discuss Asterisk, the most widely used open source telephony platform for creating custom communication solutions. Digium invites those who would like to speak at AstriCon to submit information for consideration by June 1, 2009, at http://www.astricon.net/.

The event will take place from October 13-15, 2009, at the Renaissance Glendale Hotel and Spa near Phoenix, Arizona. Registration is now open and early bird rates are available until July 1, 2009.

AstriCon 2008 proved to be the open source telephony event of the year, attracting over 600 Asterisk enthusiasts, a record number of attendees, for three days of in-depth discussions. This year’s convergence of users, developers, resellers, entrepreneurs and other fans of open source technology will continue the celebration of one of the most influential open source projects with educational sessions devoted to the developing Asterisk ecosystem, trends in Asterisk use, the latest applications and a broad range of technical topics.

“Attendees will learn from their peers during the numerous technical and business-oriented sessions and just as valuable are the opportunities for networking with other members of the Asterisk community,” said Mark Spencer, creator of Asterisk and Digium’s chief technology officer. “Often the most interesting data is exchanged over a lunch or coffee, and business relationships are formed between people sitting next to each other in a session. The people who attend are as much a part of the conference as the information provided in the talks, and Digium encourages new attendees and welcomes back those that attend each year—both groups contribute enormously to the success of the project and of the conference.”

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.digium.com/.

The Digium logo, Digium, Asterisk, Asterisk Business Edition, AsteriskNOW, Asterisk Appliance, Genuine Asterisk, Switchvox and the Asterisk logo are trademarks of Digium, Inc. All other trademarks are property of their respective owners.

Popular amongst Asterisk and Switchvox users, the Linkys SPA942 is Stylish and functional in design, the SPA942 IP Phone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment. The SPA942 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP phone that is unparalleled in features, value, and support.


.e4 is pleased to announce a price drop on the Linksys SPA942 Phone. For a limited time the 942 will be offered at $106.95 - Contact Support for VOLUME Discounts.

Popular amongst Asterisk and Switchvox users, the Aastra 6755i offers powerful features and flexibility in a standards based, carrier-grade advanced level expandable IP telephone. With a sleek and elegant design, 144 x 75 pixel backlit LCD display and 6 dynamic context-sensitive softkeys, the 6755i is fully interoperable with leading IP Telephony platforms, offering advanced XML capability to access custom applications and support for up to 9 calls simultaneously.

.e4 is pleased to announce a price drop on the Aastra 6757i Phone. For a limited time the 6757i will be offered at $169.95 - Contact Support for VOLUME Discounts

Ideal for Small offices that don't have a computer rack and need the space-savings of a small platform.

Form factor

AA60 Desktop/Wallmount

Users/Calls

• Supports 1 to 30 users


• Up to 12 concurrent calls
  

Recording/Conferencing

• Up to 5 concurrent recorded calls


• Up to 5 simultaneous conference users

Redundancy/Failover

• Optional cold spare failover

Subscription Options

• Silver Subscription Plan


• Gold Subscription Plan


• Platinum Subscription Plan

Warranty Options

• Standard 1 Year Warranty
  
• 3 Year Extended Warranty

.e4 is pleased to announce a price drop on the Polycom VVX 1500 Video Phones. For a limited time the 2200-18061-025 will be offered at $799.99

Labels: 2200-18061-025

Tomorrow's opportunities will be seized by organizations of all sizes that can effectively meet, communicate, and collaborate. World-class collaboration requires easy-to-use technology that replicates the natural clarity and efficiency of face-to-face meetings. Polycom—the world leader in voice, video conferencing, and content sharing brings these technologies together in a next generation collaboration architecture called Polycom® UltimateHD™.

Polycom UltimateHD architecture enables the world's most life-like and engaging experience that makes possible an entirely new class of conferencing and collaboration applications. Polycom UltimateHD solutions, such as Polycom HD Voice™ technology, HD Video solutions, HD infrastructure products, and HD Global Services, give users the greatest visual, audio, and content detail in a multimedia collaborative meeting. This pure clarity, rich detail, and fidelity can improve productivity and efficiency on a scale never before available in either on-demand or scheduled collaborative meetings.

·Enhances meeting productivity, effectiveness, and efficiency – An engaging, life-like experience allows you to manage globally dispersed teams, speed time-to-market, and build loyal relationships.

·Facilitates a new class of unified collaboration applications ­– Displaying details previously requiring face-to-face meetings are now possible and supported by Polycom UltimateHD applications.

·Ensures a life-like user experience everywhere – Consistent premium experience is available from mobile, desktop, and conference room solutions.

·Enables the next generation of unified collaboration –Enjoy conferencing, broadcasting, streaming and archiving.

·Becomes the collaboration core of any unified communications strategy – Leverage and enhance your existing infrastructure telephony and presence-based systems.

·Provides unmatched flexibility – Supports simultaneous on-demand or scheduled HD collaborative meetings.

·Delivers quality without compromise – Only Polycom can provide all of the essential elements with best-in-class Polycom HD Voice, HD Video, HD Content, HD Infrastructure, and HD Services.

Polycom UltimateHD architecture provides a structure for solutions to make meetings as timely and engaging as face-to-face meetings. As a result, they are dramatically more productive than traditional remote meetings.

Polycom delivers the best collaborative communications experience for every application bandwidth and price point.

G.722.1 is a state-of-the-art international standard wideband audio compression algorithm. It is based on the third-generation of Polycom Siren compression technology and is derived from Polycom's field-proven PT716 plus algorithm. Polycom developed this technology to meet the demanding audio needs of the multimedia community. It provides high-quality audio at low bit rates with low delay and very low complexity. It works for all kinds of audio signals including speech, music, and singing, for example.

G.722.1 provides wideband audio, closer in quality to FM radio than to ordinary telephone.

For example, in a telephone conversation, have you ever misheard the words "see" and "fee"? The "f" and "s" sounds are easily confused because their intelligibility is lost with inadequate rendering of the high frequencies. Such confusion never occurs with wideband coding, because all the frequencies required for speech are now fully represented. The whole audio experience when using wideband is far more natural and relaxing to the ears.

Because of its low complexity (~ 14 fixed-point MIPS, ~ 2 to 9 floating-point MIPS), G.722.1 may be used on a variety of processing platforms from digital signal processors (DSPs) to host CPUs, and still leave ample cycles for other process-hungry tasks.

The G.722.1 standard specifies operation at 24 kbps and 32 kbps for wideband (50 Hz - 7 kHz) and is currently defined in the form of a fixed-point arithmetic implementation. In addition, all licensees also get Polycom's 16 kbps extension of the standard.

Until now wideband coding was only possible using ITU-T Recommendation G.722 (at double the bit rate of G.722.1) or a proprietary algorithm. This is a big bandwidth saving in any wideband application, and it opens a whole new world of applications.

Examples of G.722.1 and Polycom Siren at 16 kbps applications:

-Wideband IP telephony
-Streaming audio (including music!) over the Internet
-Video conferencing
-Audio conferencing
-Audio storage playback (recorders)
-Store-and-forward messaging (voice mail)
-Audio-enabling your Web site

L256: The simplest of all wideband codecs, the 7 kHz L256 directly sends all the bits of digital audio sampled into 16-bit words at 16 kilo samples per second (ksps), using no compression whatever, hence the name ("Linear 256” ksps). L256 is a basic requirement in all VoIP phones, but is seldom used because of its high bit rate.

G.719: Perhaps the best match among requirements for communication systems at 20 kHz, G.719 is a recent ITU-approved arrival that combines excellent quality for music and voice with low latency, modest processor load, and network-friendly bit rates.

G.722: This is the grandfather of 7 kHz wideband VoIP codecs, and the most widely deployed so far. G.722 applies adaptive differential pulse code modulation (ADPCM) to high and low frequencies separately, yielding an algorithm that works equally well with music or voice.

G.722.1: Also known as "Siren 7," this modern 7 kHz audio codec is in almost every videoconferencing system today and is gaining traction in VoIP because of its higher efficiency and lower bit rate. G.722.1 is a "transform" (as in "Fourier transform") codec and works by removing frequency redundancies in any kind of audio.

G.722.2: This codec, "AMR-WB," is a 7 kHz wideband extension of the popular adaptive multi-rate (AMR) cellphone algorithm, and excels in delivering wideband high-quality voice at the lowest bit rates. G.722.2's algebraic code excited linear prediction (ACELP) algorithm is optimized for speech, and works by sending constant descriptions of how to shape and stimulate a human speech tract to reproduce the sound you feed into it.

G.722.1: Annex C. Also known as "Siren14," this is a 14 kHz extension of G.722.1 and is popular because of its wider bandwidth, its efficiency, and its availability (under license) for zero royalty.

Speex: Speex is an open-source CELP codec. MPEG. There are more than 25 versions of the moving pictures expert group (MPEG) transform codecs, each delivering a set of performance levels optimized for various parameters. The variant best suited to telecommunications is MPEG4 AAC-LD, a lower-delay version of the intended MP3 successor, MPEG4 AAC.

MP3: The popular MP3 format uses a form of transform coding, and is optimized for media distribution.

FLAC: The Free Lossless Audio Codec (FLAC) produces much higher bit rates than most other codecs, but compensates by preserving complete audio quality.

A VoIP codec ("coder - decoder") is an algorithm that squeezes (the "coder") digitized audio so it fits more easily into a VoIP data channel, and then re-expands it (the "decoder") so you can hear the audio once again. "Wideband" means that the audio has much higher fidelity than old-fashioned analog phones and 20 kHz is wideband audio, extended to the very top edge of human hearing.

Polycom has once again redefined the standard for crystal-clear, natural voice communications with Polycom HD Voice technology. Leveraging over 15 years of leadership in communications technology,Polycom HD Voice is revolutionizing voice communication and collaboration by bringing astoundinghigh-fidelity to the telephone, creating a richer and more natural experience.

Polycom HD Voice delivers over twice the clarity of ordinary phone calls for life-like, vibrant conversations.It’s like switching from AM radio to CD-quality audio. The difference is so astounding, you will never wantto go back to regular phone calls.

This increased clarity enables much more natural conversations, which significantly boosts recognition andenhances productivity. It’s like being in the same room with the other participants on the call. You can hearevery word without repeating, which saves time and cuts down on misunderstandings. User satisfaction isalso increased by reducing frustration during phone conversations.

As global business and remote collaboration continue to increase, the need for clear communications hasbecome more critical than ever. Accents and cultural differences add to the challenges. The clarity delivered by Polycom HD Voice makes remote collaboration easier than ever, enabling geographically dispersed teamsto communicate as effectively over the phone as they can in person.

.e4 is pleased to announce a price drop on the Polycom SoundPoint IP 670. For a limited time the IP 670 PoE will be offered at $367.99

.e4 is pleased to announce a price drop on the Polycom SoundPoint IP 670. For a limited time the IP 670 with 48v Power Supply will be offered at $367.99

Labels: ip 670

Somewhere between the Snom M3 Dect System and the Aastra SIP DECT solution lies the new Polycom Kirk 2010 - Research leads me to believe that this sexy little device is just around the corner. A few google search results yield that the 2010 is already available outside of the USA.

An updated software release SIP 3.1.2 is available from the Polycom Support web site:

A combined download where phones are running pre-4.0 BootROM
A split download where phones are running BootROM 4.0 or later

This is a maintenance patch release. The Release Notes are available from the Polycom Support web site.

Note: The Release Notes should be carefully reviewed before deploying these releases.
See the Software Release Matrix page for the latest release information.


Technical Bulletins
The following informational technical bulletins have been recently posted to the Polycom Knowledgebase at the Polycom Support web

Digium Releases Switchvox 4.0 rather silently today under the guise of release 17041

Enhancements:
Unified Communications
Fax - Send and receive faxes using Switchvox. You can use your fax machine, or fax files from your desktop computer.

Video calling - You can now use video phones with Switchvox. Video phones do not require any special set up, but for video quality we recommend that you use the same kind of phone on both ends of the call.

Chat - The new Switchvox private chat server uses the XMPP protocol. The Switchboard offers a Chat Panel, or you can use your favorite XMPP-based client. Also, you can set the chat(Jabber) hostname for Switchvox in Network Settings, and the chat (Jabber) hostname for any VOIP Providers that run a Jabber-based chat server that you will use (including a peered Switchvox).

IMAP Mailbox - Each extension now has a one-stop IMAP Mailbox for both voicemail and faxes.
Admin Features :

Phone Setup for snom technology, Inc. phones - Phone Setup is available for VOIP phones manufactured by snom technology, Inc.

(Phone Setup for Polycom, Inc. phones has been available since v9337).

Disk-space Quotas - You can set a disk-space quota for voicemail and fax usage for each extension and extension group. Also, you can see the voicemail and fax disk-space usage, in total and for each extension.

Call queue improvements - Queue members can log into, out of, and pause their status on each queue (rather than a set of queues). In the Switchboard, queue members can pause their status (when they are logged in), and add a comment about why they are unavailable or when they might be back. Also, queue status changes are now available as URL Manager events.

Notification Templates - Admins can customize multiple templates for voicemail notifications. These templates are available to users as 'default' templates. When Users sets up their notifications, they can select from these default templates or (if they have permission) customize their own templates.

Centralized Voicemail on Peered Switchvoxes - You can set up a peered Switchvox as the SIP Provider for External Voicemail.

Bulk Import Extensions tool - You can use a CSV file to create and modify extensions.

IVR Menu Actions:

Send an email (with or without a sound)

Set global (system-wide) variables

Get global (system-wide) variables

Get extension status

Get extension type

Check a user password

Perform math

Concatenate variables

Send a recorded sound to a voicemail box

Store a recorded sound in the Sound Manager

Upload a sound to a Web application

Play a sound from a web application

Merge sounds

Access Control - You can set up multiple rules to limit network access to Switchvox services.

BRI: B410P Support - This card is configured in the Channel Admin.

HD Voice G722 Support - High quality wideband audio.

6-digit extensions - Extensions can be 3, 4, 5, or 6 digits.

Scheduled Reports - You can set up regularly scheduled reports to be emailed as HTML, XML, or a chart.

Advanced reports on inbound DIDs - Call Reports can now be created based on the Incoming DID.

Network and VOIP Provider Settings - Network Settings: IP ToS settings (audio and video)

VOIP Providers: RTP Port Range SIP provider:

Caller-ID method

Include user=phone in SIP

Use Local Address in From Header

Enable Jitterbuffer

Allow Reinvite Channel Admin:

Ring Debounce (for analog lines with FXO signaling)

Network Specific Facility Code (for PRI T1/E1 Bearer channels)

PRI Reset Interval (for PRI T1/E1 Bearer channels)

Password-strength indicator - When an Admin or a User is creating a new password, a password strength-indicator lets him know how strong his new password is.

Subscription-Expiration Alerts - New alerts let you know if your Subscription is expiring soon (or if it has already expired), and Renewal links help you easily get your Subscription renewed.

XML API: Switchvox Extend - An XML-based API lets administrators access call logs, call reports, and extension lists.

Digium Addon Products - Admins can easily register Digium products for Switchvox. In particular, you can register your fax license (in the Admin Suite, select Machine Admin -> Digium Addon Products).

Advanced Debugging - You won't need to use these tools unless requested by a Switchvox Service Representative. It includes SIP and PRI debug, and Asterisk console viewer.

User Features:

Voicemail Greetings - You can customize multiple voicemail greetings, specify which greeting should be active, and assign different greetings to play when you are unavailable versus on the phone.

Organized Phonebooks - You can create and manage multiple phonebooks, to keep your contacts organized. You can also see each contact's `Additional Numbers' that they publish, and create and modify your own additional numbers for them.

Additional Numbers - You can set up your own `Additional Numbers' so that co-workers can easily call you at the numbers you publish (your mobile phone, or an extension on the manufacturing floor).

Call queue improvements - You can log into, out of, and pause your status on each queue. When you pause your status, you can add a comment to let people know when you might be back.
Conference Announcement - A caller can record his name before entering a conference room. Then, when he enters and exits the conference room, the recording is played.

Switchboard:

Chat Panel - The chat panel includes the company directory for easy communications.

Company Directory Panel - The Company Directory panel offers type-to-find, to help you quickly find your co-workers' extensions.

Multiple Phonebook Panels - Each of the your Phonebooks is available as a Switchboard panel.

Centralized presence - Across peered Switchvox PBXs, Phonebook entries show consistent Presence and call details.

Presence Settings - Presence settings include:

Available

Away (includes a comment)

Chatty

Extended Away (includes a comment)

Do Not Disturb

One-click options - These features are now available with quick one-click access in your

Switchboard panels:

Phonebook entries let you dial additional phone numbers.

Extended entries let you Barge and Whisper.

In/out/pause queue-status icons let you set your status for each queue (including a comment for a paused status)

Changes:

The extension type `Call Queue Agent' has been removed. If you have extensions of this type, you can still maintain them. However, we recommend using a `Virtual' extension instead, which offers the same features as the `Call Queue Agent,' but without the limitations.