Friday, March 31, 2006
Wednesday, March 29, 2006
Introducing the D-Link® DPH-540 Wireless G Flip-Style Wi-Fi Mobile Phone!
• Call Anywhere in the World Wherever you Have a Wireless Connection*
• 802.11g Wireless Connectivity
• Easy to Read Colorful LCM Display with Backlight
• WPA-PSK for Enhanced Wireless Security
Product Description:D-Link®, introduces the DPH-540 Wireless G Flip-Style Wi-Fi Mobile Phone, giving you the freedom of wireless connectivity and the benefits of Voice over IP service. Utilizing the DPH-540 and an Internet (VoIP) phone service plan can reduce telephone charges compared to standard telephone services.
When used with an Internet (VoIP) phone service plan, the Wireless VoIP Telephone works like a cellular phone — no PC is required! The DPH-540 connects to your wireless network using 802.11b/g, giving you greater freedom to roam throughout your network while making phone calls. In fact, you can make calls wherever you have Internet access.
The DPH-540 includes a large colorful LCM to display call information such as the numbers you dial, address book entries and caller ID numbers. The DPH-540 also supports several advanced calling features, including last number redial, mute, hold and text messaging.
The DPH-540 uses Session Initiation Protocol (SIP), and is ready to use with an Internet (VoIP) phone service plan. With echo cancellation, packet delay compensation and lost packet recovery, a VoIP call over a wireless network will sound similar to regular phone service.
Using the DPH-540 Wireless G Flip-Style Wi-Fi Mobile Phone gives you the advantages of VoIP, while giving you the ability to use the freedom of wireless networking. When used with an Internet (VoIP) phone service plan, you will be able to enjoy all of the advanced features that VoIP offers.
If you're interested in purchasing this phone, or want more information, please click here.
Saturday, March 25, 2006
Introducing the snom 300
For more information Please Visit the IP Phones Section of our store@
As the basic model of the snom business telephone family, the snom 300 fulfils the most important requirements of VoIP telephony and additionally offers numerous functions that are indispensable in the business world. For effective everyday work, the snom 300 provides all relevant office functions such as choice of trunk line, status display, group lines, the engaged option or picking up calls.
When it comes to user friendliness, the snom 300 sets new standards: A two-line graphical LCD display enables the display of call information, and the menu-driven user interface provides the simplest of feature management. Via the navigation key, the user is guided intuitively through the telephone menu. More complex telephone functions, call details and configuration possibilities are accessible via the browser over the connected PC.
Six free user or administrator-configurable (or carrier-preconfigurable) function keys can be easily allocated to security-related menu functions, or assigned to multiple lines.
The snom 300 comes factory-equipped to enable two of its six programmable keys to be configured as line appearances, and snom provides upgrades that let you configure (up to) all six function keys in this way - flexible enough to suit the needs of every user. This option enables an individual adaptation of the device to specific areas of application and the personal user behavior - a functionality that is becoming increasingly popular, particularly in call centers and for sales agents.
The snom 300 is designed for different environments: for small offices, call centers, lobbies, recreation rooms, or in the home. It fits into its environment without any troubles. Through numerous telephone functions, the demands of everyday office communication can be easily managed. The snom 300 has a headset connection and can be used as a freestanding or wall-mounted model.
As the snom 300 supports all of the common compression codecs such as G.729a and G.723.1, it is compatible with numerous components of other manufacturers and can be used in low-bandwidth environments. An integrated 2 Ethernet port switch enables connection to the network over an RJ-45 interface simultaneously with the PC connection!
Thursday, March 23, 2006
.e4 VoIP Fulfillment Services
The .e4 Advantage!
• Real time XML stock availability.
• Online ordering
• Branded dispatch note (your logo, not .e4)
• Configuration server options
o .e4 redirect endpoint to service providers own configuration server
Typical Scenario From End users perspective
1. End User visits service provider's website and orders a VoIP service
2. Website offers options of hardware for the customer to select
3. End user adds hardware to shopping basket, and pays for service and hardware together
4. VoIP equipment arrives next day, preconfigured and ready to go
Behind the scenes
1. Service provider's website reads XML information from .e4 to display stock availability
2. Service provider electronically transfers the order information (Name, Address, product, SIP username, SIP password) to .e4.
3. .e4 dispatch ordered hardware directly to the end user, including a dispatch note with the service providers branding
4. The SIP details are loaded onto .e4's configuration server
5. Service provider receives a combined weekly invoice for all orders
• No need to Stock or pay for hardware up front. Also no inward shipping fees to your warehouse.
• No need to get involved in packing
• Benefit from reduced shipping rates, because of the shipping volume of .e4
• Ordering can be completely electronic.
• Online monitoring of order status
This service is generally available to service providers who will buy at least 50 VoIP Endpoint units per month, subject to .e4's approval. Please contact .e4 to discuss your exact requirements.
• Agree pricing, invoicing terms and sales forecasts
• We need a PDF of your company letterhead
• Financial setup - agree payment terms
• Agree default SIP configuration for devices.
o Firmware version
o SIP servers
o Which settings customers can change
o NTP Server
Tuesday, March 21, 2006
Introducing Ranch Networks
Ranch Networks was established in 2000 by an award-winning group of data networking professionals who were veterans of successful data networking startups such as Scorpio Communications (acquired by US Robotics) and Lannet (acquired by Lucent) or were from advanced data networking projects in Bell Labs. www.ranchnetworks.com
For more information call (231) 946.4162
Friday, March 17, 2006
Introducing the Linksys WBP54G Wireless-G Bridge
Wireless-G Bridge for Phone AdaptersConvert your IP Phone to use Wireless-G networking!
Or visit http://www.digiumcards.com
- Put your IP Phone wherever you want, with no cabling hassle
- Connects your IP Phone to your Wireless-G network
- Shares power with the IP Phone -- only one AC Adapter necessary
- Wireless connection protected by WEP, WPA or WPA2 encryption
Now you can put your IP Phone almost anywhere in the building, without the cost and hassle of running network cables. The Wireless-G Bridge for Phone Adapters was specially designed to convert your IP Phone into a wireless device, so it can connect to your network without an Ethernet cable. This lets you put your IP Phone where it's most convenient and frees you from the contrains of plugging into the nearest network port.
To make installation even more convenient, the Wireless-G Bridge shares electrical power with the IP Phone, so only one AC Adapter is needed. To get connected, just plug your existing IP Phone's power jack into the Wireless-G Bridge, and the Bridge's power and data cables to the IP Phone. The included Setup Wizard makes it easy to configure the Bridge to your wireless network's settings. To protect your privacy, all wireless voice transmissions can be encrypted with WEP or industrial-strength Wi-Fi Protected Access (WPA/WPA2) security.
So don't hassle with running cables around the room to your IP Phone — get connected the easy way with the Linksys Wireless-G Bridge for Phone Adapters.
Wednesday, March 15, 2006
Take the RISK out of Asterisk!
For the best price available please contact
Or order on line @
Asterisk Business Edition™ is an enterprise-grade version of its acclaimed open source PBX for the Linux operating system.
Asterisk Business Edition provides tested reliability of critical functions and features, tailored for small- and medium-sized business applications. An Asterisk technical manual and quick-start documentation supplements the package, making Asterisk even easier to install, configure, and use.
Asterisk Business Edition Includes:
-Asterisk Business Edition binaries, drivers, installer, and scripts on CD-ROM
-An all-new Asterisk Technical Manual
-Authentication key for activation and support
-Special offers and discounts from Digium and its Partner companies
Asterisk Business Edition supports up to 120 simultaneous calls. Upgrades available to allow up to 240 calls.
Asterisk Business Edition™ is backed by Digium's professional support team for one year. This provides enterprise environments with a PBX and telephony platform suitable for critical business applications.
Digium's comprehensive test program ensures Asterisk Business Edition's reliability, performance, and interoperability with key hardware, software, and protocols. Digium hardware cards are tested for full compatibility with Asterisk Business Edition, as are several select models of servers, VoIP, and TDM devices. All major software features in Asterisk Business Edition are thoroughly tested for functionality and reliability. Test bed systems are also subjected to extreme stress conditions using Empirix™ test equipment to simulate hundreds of thousands of calls in various real-world combinations and configurations.
As a result, customers can rely on their combination of proven Asterisk software and Digium hardware to work together to provide a feature-rich PBX or VoIP system.
Please note that the Business Edition is not a downloadable product and is not eligible for refunds once installed and activated.
http://www.digiumcards.com announces reseller program
March 15, 2006
The new Partner Program offers technology partners distribution services, IP Telephony training, access to the latest in IP technology and products and services, pre and post sales engineering, technical support. A main feature of the program is the .e4 University, a training and program for resellers focusing on SIP and VoIP technology.
"Our reseller program is designed to help newer companies quickly generate revenue in the booming VoIP market," says Michael White, President of .e4. "In terms of one-stop shopping and support our resellers can achieve a low-cost entry path to deploying VoIP services that adhere to open standards are vendor independent and interoperable. Our solutions are SIP (Session Initiation Protocol) based, a new protocol for VoIP communications and Telephony which offers partners unprecedented power and flexibility to intermesh communications with application based intelligence and build very distinguishable solutions with unique combinations of features and services."
http://www.digiumcards.com specializes in VOIP products; .e4 LLC is based in Traverse City, Michigan and has a sales organization that covers the US, Canada, Mexico and Latin America. Key products distributed by http://www.digiumcards.com are Digium, Linksys SPA, snom technology IP Phones, Aastra, Sangoma, Mediatrix, and AudioCodes gateways.
Additional information about the program can be obtained by contacting:
Tuesday, March 14, 2006
Introducing the Linksys SPA922, SPA-922
- Full featured one-line business class IP Phone supporting Power over Ethernet 802.3af
- Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
- Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more
- Easy installation and secure remote provisioning. Menu based and web based configuration.
Stylish and functional in design, the SPA922 VoIP 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 SPA922 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP Phone that is unparalleled in features, value, and support.
Standard features on the SPA922 include dual switched Ethernet ports, 802.3af PoE, a high resolution graphical display, speakerphone and a 2.5 mm head-set port. The SPA922 supports one line with two call appearances and provides support for three way conferencing, attended call transfer, and placing a call on hold to answer an incoming call. The line can be configured as a unique phone number (or extension), or can be configured to share a number that is assigned to multiple phones.
Comprehensive Interoperability and SIP Based Feature Set
Based on the SIP standard, the SPA922 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable sevice parameters, the SPA922 addresses the requirements of traditional business users while leveraging the advantages of IP telephony. Features such as easy station moves, presence, and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA922.
Carrier-Grade Security, Provisioning, and Management
The SPA922 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, pre-loading, and re-configuring customer premise equipment (CPE).
Music on Hold for Asterisk
The average business receives one phone call every five minutes; approximately 100 calls per day.
70% of callers are placed on hold for an average hold time of 45 seconds.
60% of business callers put on hold hang up.
30% of these callers never call back.
The average executive spends 15 minutes per day or 60 hours a year on hold.
Playing music, a radio or CD on your hold line is illegal and you may be responsible for copyright fees.
When you subscribe to eMagin On hold we take care of the hassle! We use only licensed music to create your productions and any fees are included in your service agreement.
Think you won’t put callers on hold?
What happens to callers while you finish up another call, look up the status of their order, or the answer to a question? Just a few seconds can seem forever to someone waiting impatiently on hold. You have a captive audience! Don’t subject them to that annoying beep, a radio selling other companies’ products, or endless silence. Inform, entertain and enlighten them with eMagin On hold's virtual magazine. Interesting articles about health, fitness, lifestyles, fashion and many other topics.
Monday, March 13, 2006
Linksys Voice System LVS9000 SPA9000
in music of your choice. Park calls and use the intercom for paging. There are over one hundred telephony features built-in!
The LVS9000 System also features a wide range of inexpensive, Linksys IP Phones to meet your needs and your budget. Add up to 16 lines as your business grows. Choose an IP Phone with one, two or four lines with a high resolution graphical display or the basic no-display model.
optional Wireless Adapter, you can even install IP Phones in hard to reach places without running cables.
Introducing The Linksys Voice System 9000
Inexpensive and easy to install, this IP PBX telephony system offers big business features like auto-attendant, music or message on hold and much more, all on a small business budget. The LVS 9000 provides you the opportunity to address the special telephony needs of your SOHO and small business clients while generating bounty payments or recurring revenue streams through partnerships with participating small business service providers.
Wednesday, March 08, 2006
Please visit our store located @ http://www.digiumcards.com
Not the price you were looking for? Contact firstname.lastname@example.org or call
Digium FXO Boards:
Digium TDM01B TDM Card with 1 FXO
Digium TDM02B TDM Card with 2 FXO
Digium TDM03B TDM Card with 3 FXO
Digium TDM04B TDM Card with 4 FXO
Digium FXS Boards:
Digium TDM10B 1 FXS port PCI Card
Digium TDM20B 2 FXS port PCI Card
Digium TDM30B 3 FXS port PCI Card
Digium TDM40B 4 FXS port PCI Card
Digium TDM11B TDM Card with 1 FXS, 1 FXO
Digium TDM12B TDM Card with 1 FXS, 2 FXO
Digium TDM13B TDM Card with 1 FXS, 3 FXO
Digium TDM21B TDM Card with 2 FXS, 1 FXO
Digium TDM22B TDM Card with 2 FXS, 2 FXO
Digium TDM31B TDM Card with 3 FXS, 1 FXO
Digium TDM FXO Module FXO Module X100m
Digium TDM FXS Module FXS Module s110
Digium T/E 110P Single Span T1/E1 Card
Digium T/E 205P Dual Span T1/E1 Card (5v)
Digium T/E 210P Dual Span T1/E1 Card (3.3v)
Digium T/E 405P Quad Span T1/E1 Card (5v)
Digium T/E 410P Quad Span T1/E1 Card (3.3v)
Digium T/E 406P Quad Span T1/E1 Card (5v) with echo Cancellation module
Digium T/E 411P Quad span T1/E1 Card (3.3v) with echo Cancellation module
Digium IAXy Device
Digium Asterisk Software
Digium Asterisk Business Edition
Digium TDM24XXP Series FXO:
Digium TDM2401B TDM Card with 0 Quad FXS, 1 Quad FXO
Digium TDM2402B TDM Card with 0 Quad FXS, 2 Quad FXO
Digium TDM2403B TDM Card with 0 Quad FXS, 3 Quad FXO
Digium TDM2404B TDM Card with 0 Quad FXS, 4 Quad FXO
Digium TDM2405B TDM Card with 0 Quad FXS, 5 Quad FXO
Digium TDM2406B TDM Card with 0 Quad FXS, 6 Quad FXO
Digium TDM24XXP Series FXO with Echo Cancellation Module:
Digium TDM2401E TDM Card with 0 Quad FXS, 1 Quad FXO w/ Echo Cancellation
Digium TDM2402E TDM Card with 0 Quad FXS, 2 Quad FXO w/ Echo Cancellation
Digium TDM2403E TDM Card with 0 Quad FXS, 3 Quad FXO w/ Echo Cancellation
Digium TDM2404E TDM Card with 0 Quad FXS, 4 Quad FXO w/ Echo Cancellation
Digium TDM2405E TDM Card with 0 Quad FXS, 5 Quad FXO w/ Echo Cancellation
Digium TDM2406E TDM Card with 0 Quad FXS, 6 Quad FXO w/ Echo Cancellation
Digium TDM24XXP Series FXS:
Digium TDM2410B TDM Card with 1 Quad FXS, 0 Quad FXO
Digium TDM2420B TDM Card with 2 Quad FXS, 0 Quad FXO
Digium TDM2430B TDM Card with 3 Quad FXS, 0 Quad FXO
Digium TDM2440B TDM Card with 4 Quad FXS, 0 Quad FXO
Digium TDM2450B TDM Card with 5 Quad FXS, 0 Quad FXO
Digium TDM2460B TDM Card with 6 Quad FXS, 0 Quad FXO
Digium TDM24XXP Series FXS with Echo Cancellation Module
Digium TDM2410E TDM Card with 1 Quad FXS, 0 Quad FXO w/ Echo Cancellation
Digium TDM2420E TDM Card with 2 Quad FXS, 0 Quad FXO w/ Echo Cancellation
Digium TDM2430E TDM Card with 3 Quad FXS, 0 Quad FXO w/ Echo Cancellation
Digium TDM2440E TDM Card with 4 Quad FXS, 0 Quad FXO w/ Echo Cancellation
Digium TDM2450E TDM Card with 5 Quad FXS, 0 Quad FXO w/ Echo Cancellation
Digium TDM2460E TDM Card with 6 Quad FXS, 0 Quad FXO w/ Echo Cancellation
Digium TDM24XXP Series FXS/FXO:
Digium TDM2411B TDM Card with 1 Quad FXS, 1 Quad FXO
Digium TDM2412B TDM Card with 1 Quad FXS, 2 Quad FXO
Digium TDM2413B TDM Card with 1 Quad FXS, 3 Quad FXO
Digium TDM2414B TDM Card with 1 Quad FXS, 4 Quad FXO
Digium TDM2415B TDM Card with 1 Quad FXS, 5 Quad FXO
Digium TDM2421B TDM Card with 2 Quad FXS, 1 Quad FXO
Digium TDM2422B TDM Card with 2 Quad FXS, 2 Quad FXO
Digium TDM2423B TDM Card with 2 Quad FXS, 3 Quad FXO
Digium TDM2424B TDM Card with 2 Quad FXS, 4 Quad FXO
Digium TDM2431B TDM Card with 3 Quad FXS, 1 Quad FXO
Digium TDM2432B TDM Card with 3 Quad FXS, 2 Quad FXO
Digium TDM2433B TDM Card with 3 Quad FXS, 3 Quad FXO
Digium TDM2441B TDM Card with 4 Quad FXS, 1 Quad FXO
Digium TDM2442B TDM Card with 4 Quad FXS, 2 Quad FXO
Digium TDM2451B TDM Card with 5 Quad FXS, 1 Quad FXO
Digium TDM24XXP Series FXS/FXO with Echo Cancellation Module:
Digium TDM2411E TDM Card with 1 Quad FXS, 1 Quad FXO w/ Echo Cancellation
Digium TDM2412E TDM Card with 1 Quad FXS, 2 Quad FXO w/ Echo Cancellation
Digium TDM2413E TDM Card with 1 Quad FXS, 3 Quad FXO w/ Echo Cancellation
Digium TDM2414E TDM Card with 1 Quad FXS, 4 Quad FXO w/ Echo Cancellation
Digium TDM2415E TDM Card with 1 Quad FXS, 5 Quad FXO w/ Echo Cancellation
Digium TDM2421E TDM Card with 2 Quad FXS, 1 Quad FXO w/ Echo Cancellation
Digium TDM2422E TDM Card with 2 Quad FXS, 2 Quad FXO w/ Echo Cancellation
Digium TDM2423E TDM Card with 2 Quad FXS, 3 Quad FXO w/ Echo Cancellation
Digium TDM2424E TDM Card with 2 Quad FXS, 4 Quad FXO w/ Echo Cancellation
Digium TDM2431E TDM Card with 3 Quad FXS, 1 Quad FXO w/ Echo Cancellation
Digium TDM2432E TDM Card with 3 Quad FXS, 2 Quad FXO w/ Echo Cancellation
Digium TDM2433E TDM Card with 3 Quad FXS, 3 Quad FXO w/ Echo Cancellation
Digium TDM2441E TDM Card with 4 Quad FXS, 1 Quad FXO w/ Echo Cancellation
Digium TDM2442E TDM Card with 4 Quad FXS, 2 Quad FXO w/ Echo Cancellation
Digium TDM2451E TDM Card with 5 Quad FXS, 1 Quad FXO w/ Echo Cancellation
Digium Quad FXS Module
Digium Quad FXS Module
Echo Cancellation Module
Echo Cancellation Module for TDM24XXP
Introducing the ZyXEL Prestige 2000W - VoIP WiFi Phone
VoIP WiFi Phone
Enjoy the Benefits of VoIP, Wirelessly
- Mobility with IEEE 802.11b wireless standard compliance
- High voice quality with low communication cost
- Support Auto Provisioning for ease of deployment
Mobility and Availability
The call control protocol of the P-2000W is based on SIP v2 (Session Initiation Protocol version 2, RFC 3261) open standard, which is interoperable with major SIP-based call servers, IP-PBXs, and other standard SIP-based client devices. The P-2000W_v2 is compliant with the IEEE 802.11b standard and interoperates with any existing 802.11b or 802.11g wireless AP and gateway. It may be used as a cordless handset for residential users or for business users in an office environment. The small form factor of the handset is easy to transport and allows users to place VoIP phone calls in public 802.11-based environment.
High Voice Quality with Low Communication Costs
The P-2000W_v2 is capable of tagging features that support a service provider’s QoS (Quality of Service) planning, such as ToS (Type of Service). It allows gateways or central side equipment to identify and prioritize voice and data traffic. By supporting G.711 and G.729 voice compression technology, the P-2000W_v2 effectively reduces bandwidth consumption caused by voice traffic.
Travel Charger Design for Mobile Users
The P-2000W_v2's travel charger design makes it easier for users to carry and use the phone in different site. The mobile phone style design also let user to have similar and better experience to use VoIP service.
Direct IP-to-IP Call and Ad-hoc Intercom Mode Support
By configuring a remote IP address in the built-in phone book, the P-2000W_v2 provides a direct IP-to-IP call feature when there is no intermediate SIP proxy server available in the network. The P-2000W_v2 an also establish an 802.11 ad-hoc network (computer-to-computer network without Access Point), which allows users to use the handsets as wireless intercoms.
To order: http://www.digiumcards.com/p2000w_v2.html
Introducing the ClearOne MAX IP SIP VoIP Conference Phone
The MAXAttach IP and MAX IP offer a unique ability, unmatched by competitive products, that allows customers to daisy-chain multiple phones together, up to a total of four. This provides multiple speakers, multiple microphones, and multiple dial pads distributed throughout the room for unrivaled coverage.
In addition, these VoIP conference phones offer the advanced audio signal processing technologies that were originally developed for ClearOne's market-leading professional audio conferencing products. These technologies deliver crystal-clear audio to participants on both ends of the call, and include:
Distributed echo cancellation
First microphone priority
Automatic gain and level controls
These new VoIP products also offer a suite of SIP features, including:
3-way calling - allows for ad-hoc conferences without need for a conference bridge
VLAN tagging - allows users to manage bandwidth usage on the network
TLS & SRTP encryption-ready (with future release of firmware upgrade) - secures voice communications over the network
Field upgradeability - allows users to easily download firmware upgrades from ClearOne website and load directly into the conference phone
MAXAttach IP (comes with two VoIP conference phones):
Medium to large conference rooms
Unique room configurations, such as U-shaped table layout
Can expand up to four linked phones
Phones can be used in separate rooms with extra base unit
MAX IP (comes with one VoIP conference phone):
Small conference rooms — up to 8 people
Can also expand up to four linked phones
Monday, March 06, 2006
To order visit
The Digium™ S101I, affectionately known as the IAXy™, takes Asterisk™ from the PC to the CPE. The IAXy provides a single, fully featured FXS interface with an Ethernet back-end, speaking the Asterisk-native IAX protocol, at a highly competitive price. The IAXy is aimed at Voice Over Broadband and Internet Telephone Service Providers. The IAX protocol provides complete NAT transparency, enabling full operation behind NAT and PAT firewalls. This includes the ability to robustly transfer calls between endpoints, allowing on-net calls to be moved off of a service provider's network for better quality and lower cost.
Saturday, March 04, 2006
Introducing the Epygi 4x-4l
Voice Coding G.711, G.723 (5.3, 6.3 kbit/s), G.726 (16, 24, 32, 40 Kbps), G.729, iLBC (13,33 kbit/s , 15,2 kbit/s); (RFC 3951, ITU-T: G711, G.723.1 Annex A, G.726, G729 Annex A; IETF; ITU-T Q.23, Q.24, Bellcore GR.506, GR.181; ITU-T G.168-2000, 2002; ETS_300659_1,2,3)
NAT traversal (both manually and STUN)
VAD, CNG, G.168 echo cancellation
Lifeline POTS (single line)
Per call WAN bandwidth requirements for the following codecs (non-encrypted/VPN encrypted):
G.711a/G.711u 20 msec 84 kbps/105 kbps
G.726-16 20 msec 37 kbps/59 kbps
G.726-24 20 msec 45 kbps/65 kbps
G.726-32 20 msec 52 kbps/74 kbps
G.726-40 20 msec 60 kbps/80 kbps
G.729a 20 msec 29 kbps/49 kbps
G.723 30 msec 21 kbps/34 kbps
iLBC 30 msec 27 kbps/41 kbps
Call block, forward, hold, ID, park, relay, transfer, wait
Multilevel auto attendant with Interactive Voice Response (IVR)
Distinctive ring, hold music, speed dialing
Many extensions ring
Do not disturb service
T.38 fax, fax relay and clear channel fax
Unified Fax Messaging
Busy auto redial
Multiple user extensions (up to 70 physical/virtual extensions)
SIP on the WAN and LAN side (RFCs: 3261, 3263, 3265, 3311, 3428, 3515, 3842, 3856, 3891, 3892, 3581, draft-ietf-sip-session-timer-15, draft-ietf-sipping-dialog-package-05; Presence: RFCs: 3842, 3856, 3863, draft-ietf-sipping-dialog-package-05)
SDP (RFC 2327)
RTP (RFCs: 1889, 1890, 2833, 3389, 3550, 3551, 3555, draft-ietf-avt-rfc2833bis-05, draft-ietf-avt-rtp-ilbc-o5), in band and out of band signaling support
MGCP with support of the full MGCP business extensions and full screen and soft buttons control of the MGCP phones on the LAN side (RFCs: 3149, 3435, 3660, draf t-foster-mgcp-returncodes-01, PKT-SP-EC-MGCP-I10-040402)
Fax over IP (ITU_T: T4, T30, T38, V17, V21, V27 ter, V29)
FSK and DTMF Caller ID support
NAT address translation
STUN/NAT traversal (RFC 3489)
IPSec VPN with DES, 3DES and AES encryption in tunnel mode (RFCs: 2402, 2406, 2409)
Manual and automatic IKE key support
Firewall security via:
Intrusion Detection System
NAT (Network Address Translation)
Policy and service-based filtering
Stateful inspection firewall
DHCP server on the LAN side
DHCP client on the WAN side
DNS server with forwarding functionality
SNTP (Simple Network Time Protocol) server/client for computer clock synchronization
PPPoE connection to the ISP with PAP/(MS)CHAP authentication
IP DIFFSERV for QoS
Mail client to send voice messages as e-mail attachments (.wav) and system notifications
DNS (DYNDNS) support with third party
Friday, March 03, 2006
IP PBX Grand Rapids, Michigan
on the web @ http://www.e4strategies.com