Quantcast
Channel: VOIP-info.org Wiki Changes
Viewing all 5767 articles
Browse latest View live

Cisco SPA525G2 Business IP Phone

$
0
0

Cisco SPA525G2 5-Line Business IP Phone with Enhanced Connectivity and Media for a New Level of User Experience



MKI04001.jpg


Highlights


  • Full-featured and stylish business IP phone supporting up to two Cisco® SPA500S Expansion Modules (32 button attendant consoles)
  • Cisco Mobile Link: Bluetooth enhanced integration with mobile phones to make and receive calls, import your personal contacts, and charge your mobile phone
  • Enhanced network connectivity with Power over Ethernet (PoE), 802.11g Wi-Fi client with Wi-Fi Protected Setup (WPS), and Bluetooth headset support
  • Graphics-rich, high-resolution 3.2-inch QVGA 320 x 240 color screen
  • Cisco AnyConnect VPN Client: Highly secure Internet phone connection for remote users that is simple and easy to set up
  • MonitorView for monitoring up to four video surveillance cameras from your phone
  • Cisco XML services framework: Support for productivity applications directly on your phone
  • Support for multimedia functions, such as playing MP3s, displaying digital photos, and viewing RSS feeds
  • Wideband audio for unsurpassed voice clarity and enhanced speaker quality
  • Support for both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco Unified Communications 500 Series for Small Business



Overview


The Cisco SPA525G2 5-Line IP Phone with Color Display is an excellent choice for businesses that require an enhanced user experience with a hosted IP telephony service, an IP private branch exchange (PBX), or a large-scale IP Centrex deployment. Part of the Cisco Small Business Series, the SPA525G2 uses industry-leading SPA voice over IP (VoIP) technology from Cisco, with high-quality hardware providing additional connectivity via Bluetooth, PoE (802.3af), or a Wireless-G client (802.11g).

Standard Cisco SPA525G2 features include five active lines, VLAN-capable dual switched Ethernet ports, 802.3af PoE support, a 3.2-inch QVGA color display, a full-duplex, high-quality speakerphone, a Bluetooth interface, a Wireless-G (802.11g) client, a 2.5-mm stereo headset port, and a USB 2.0 host port. Each line can be configured independently to use a unique phone number (or extension) or can use a shared number that is assigned to multiple phones. The power supply for the SPA525G2 is sold separately. The optional SPA500S 32-button attendant console adds up to 64 buttons for receptionist positions.

The Cisco SPA525G2 IP Phone further improves the user experience with VPN and video surveillance applications. It includes an embedded AnyConnect Secure Sockets Layer (SSL) VPN client that allows remote users to securely connect to their phone system and make calls over the Internet, without the need for additional hardware. The SPA525G2 also provides users the ability to view video feeds from Cisco WVC2300 and PVC2300 Business Internet Video Cameras, allowing users to quickly see different locations around the business in order to improve physical security. The SPA525G2 is part of the Cisco SPA500 Series IP Phones, a robust portfolio of small business phones providing a rich user experience that includes HD voice, on-phone applications, and intuitive menu options.

The Cisco SPA525G2 provides encrypted signaling, media, and provisioning information, using state-of-the-art technologies such as Session Initiation Protocol (SIP) over Transport Layer Security (TLS), Secure Real Time Protocol (SRTP), and HTTPS to secure communications between the phone and service provider. Cisco SPA Secure Remote Provisioning provides a highly secure mechanism for the service provider to remotely manage the phone/user configuration and software upgrades.

The Cisco SPA525G2 IP Phone can also be used with productivity-enhancing features such as VoiceView Express and Cisco XML applications when connected to the Cisco Unified Communication 500 Series in SPCP mode. ...

SIP Speaker——iSpeaker C20

$
0
0
iSpeaker-c201.jpg


iSpeaker-c202.jpg


iSpeaker C20

The SIP based audio system iSpeaker C20 utilizes the built-in intercom and paging capability already inherent in most modern IP PBX systems and enhances this to improve end user experience by providing a dedicated high performance digital amplifier on which to broadcast announcements or play background music.

Basic Features
Support G.711,G.722,G.723,G.726,G.729 audio codecs;
Support SIP2.0 (3621)and related RFC protocol;
Support Power over Ethernet(PoE)
The DSP integrated echo cancellation and noise suppression. If special or higher requirement for voice quality is required then ZYCOO hardware echo cancellation module is an excellent choice.
Easy management and configuration from Web interface.
Support encrypted communication;
Online update of software.



Contact us:



ZYCOO China
Web: www.zycoo.com
Tel: +86 (28) 85337096
Address: 7F, B7, Tianfu Software Park, Chengdu, China.

ZYCOO UAE
Web: www.zycoo.ae
Tel: +971 (4) 3798839
Address: #305, Al Riffa Plaza Bldg., Khalid Bin Alwaleed RD, DUBAI, UAE

ZYCOO UK
LS21, Armstrong House, First Avenue, Robin Hood Airport, Doncaster, UK (DN93GA)


ALL THE RIGHTS ARE RESERVED BY ZYCOO Co.,Ltd

How to start a VOIP Business

$
0
0

The first thing to do is decide what part of VOIP marketplace you want to serve. Here are some possibilities:

  • VOIP Provider services
  • VOIP consulting
  • Independent Sales/Service Agent for existing VOIP service providers
  • Value Added services with VoIP
  • etc.

Some general suggestions:

  • Pick an area that plays to your strengths. For example, if your strength is sales and marketing, pick an area where you can leverage those abilities
  • Learn all you can about the maketplace
  • Attend industry tradeshows
  • Read industry magazines, blogs, forums, etc
  • Read books
  • Do market research - talk to your potential customers
  • Ask questions
  • Test the waters — to the extent possible try before you buy, test the waters before making large commitments of time or money

Value Added services

If you have experience with VoIP or already in VoIP business, you can get benifit / new customers by introducesing some value added services on VoIP. Few value added services are mentioned in following, Within each service there are many choices.
  • PBX sales and service
    • Hosted PBX
    • Virtual Numbers
    • Hosted IVR / Auto attendents
    • etc

  • Message broadcasting / Call Center Solutions

  • Prepaid Cards
    • Retail prepaid cards from existing wholesale providers
    • Start your own brand of prepaid cards using services from existing wholesale providers
    • Start a new prepaid card provider company
    • Create new software package for prepaid card services
    • Create a Free Phone Booth
    • etc.

VoIP Business - Startup No Money Down:

Start a VOIP Business with no money down using ITSPtec Hosted Systems to do HostedPBX, Carrier/Termination, Residential and Callingcard Services.
See the Hosted Model

Become a VoIP provider with no costs. Speedflow offers hosted Class 5 Softswitch for free. Use it with our high-quality routes and pay only for traffic. No setup fee, no hidden payments.


VoIP Business: Buy, Build or Resell?

Starting a VoIP business can be tricky Its a lucrative business that makes it a very Competitive business too. ...

ICTFAX

$
0
0

Release Note


Released new version of ICTFax ICTFAX Version 2.2.0 on Feb 13, 2014 , Fax over IP software implementation based on T.38 protocol also support G.711 pass through faxing and PSTN faxing

Released new version of ICTFAX Ver 3.0 on Nov 28, 2014 , completly rebased on ICTCore after dropping plivo , Old version of ICTFAX was based on Plivo and has several issue during installation.

ICTFAX


ICT FAX is an open source (GPL v 3) based buisness solution especially for faxing along with support of SMS and Voip with advance web based billing capabilities featuring TIME, Per PAGE and Per SMS based Billing , It supports G.711 , T.38 and PSTN faxing .ICTFAX is complete faxing solution and does not need to be integerated with other open source projects to function properly that makes ICTFAX a unique and innovative faxing solution.

ICTFAX, a Faxing solution


ICTFAX can be used in following faxing scenarios

  • Email to fax
  • Web to fax
  • Fax to email

ICTFAX, a SMS solution


ICTFAX can be used in following SMS sending scenarios

  • Email to SMS
  • Web to SMS

Screenshots


http://sourceforge.net/projects/ictfax/

Download


Download open source Online FAX solution

Documentation


for further help please visit ICTFAX Forum

ICTFAX is developed by ICT Innovations

T.38

$
0
0
T.38 is an ITU standard for sending FAX accross IP networks in a real-time mode.
FAX messages are sent as UDP or TCP/IP packets.

  • The IETFRFCRFC 3362 implements a media type called image/t38 for T.38 faxes.

From RFC 3362:

ITU-T Recommendation T.38 T.38 describes the technical features necessary to transfer facsimile documents in real-time between two standard Group 3 facsimile terminals over the Internet or other networks using IP protocols. The Recommendation allows the use of either TCP or UDP depending on the service environment.

ITU-T Recommendation T.38 T.38 Annex D describes system level requirements and procedures for internet aware facsimile implementations and internet aware facsimile gateways conforming to ITU-T T.38 to establish calls with other ITU-T T.38 implementations using the procedures defined in IETF RFC 2543 SIP-99 and IETF RFC 2327 SDP.

Note that ITU-T T.38 Recommendation T.38 (04/02) T.38 is an aggregation of the original ITU-T Recommendation T.38 (06/98)T.38-98 and all of the subsequent Amendments and Corrigendum including T.38D-00. While T.38 and T.38D-00 describe SIP procedures per SIP-99, the procedures can also be applied to the revised Session Initiation Protocol specification SIP.


  • Commetrex has completed significant interoperability testing in the T.38 Interoperability Test Lab. http://www.commetrex.com
  • Brooktrout White paper: http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf (link no longer works as Brooktrout became part of Cantata Technology, which is now itself part of Dialogic). The document can be found at www.viatechnology.es/Documents/Productos/fax_over_ip.pdf


T.38 with Fax Voip T38 Fax & Voice

FaxVoip Software develops solutions for the transmission of a fax via the Internet Telephony (FOIP). The main emphasis has been placed on the transfer T.38 and audio via SIP, H.323 and ISDN CAPI 2.0.
Fax Voip application operates with T.38 faxes via standart COM port interface.
What is Voip? For your fax or voice application, it's a Voice Fax Modem. You can setup your Fax & Voice program to operate with Fax Voip Virtual COM ports or virtual "FaxVoip Modems 14400 bps Voice-Fax" modems. From the perspective of your VOIP internet network, it’s a SIP/H.323 client with T.38 and G.711 Fax support.From the perspective of your ISDN line, it’s CAPI 2.0 client with audio fax support. You can send/receive T.38 and audio (color/black-and-white) faxes and voice messages without any hardware, using your favourite Fax & Voice program. Fax Voip is the ideal solution for the implementation of Fax and Voice Mailbox into SIP/H.323/ISDN network. You can use Fax Voip with your VOIP or ISDN-based PBX or with your SIP/H.323/ISDN Provider. Up to 100 virtual modems can be used simultaneously.
Multiple SIP Registrations and Call Routing functions make your system the most flexible as well as allowing you to work with different SIP and H.323 providers simultaneously.
Fax Voip supports Caller ID and Dial a Phone Number Extension features.
You can send T.38, audio and CAPI faxes via Fax Voip Virtual Printer and receive faxes directly in TIFF, PDF or SFF files, and without limiting the number of incoming fax sessions. You can manage faxes with Fax Voip Console.
Fax Voip is a fully-functional system for sending faxes via e-mail (Mail to Fax) and for receiving faxes to e-mail (Fax to Mail).
Incoming Routing Methods (Route through e-mail, Store in a folder, Print) allow you to route incoming faxes to recipients on the network.
Fax on Demand function allows callers to retrieve information via fax on the same call (from a fax machine). ...

Open Source VOIP Software

$
0
0

Open Source VOIP applications, both clients and servers.

Open source means all source code is available!! Do not post any "free but not open" software here!

SIP Proxies


  • JAIN-SIP Proxy
  • Mini-SIP-Proxy A very tiny perl POE based SIP proxy
  • MjServer cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
  • MySIPSwitch SIP Proxy server which allows using multiple SIP accounts with a single SIP login
  • NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
  • Net-SIP A Perl SIP framework that includes a stateless proxy
  • OpenJSIP Opensource distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. ...

VOIP Service Providers

$
0
0
For a list of VOIP to PSTN service providers, indexed by country, please see:


PLEASE DO NOT ADD NEW CATEGORIES HERE.

VOIP provider services, exchanges and other business deals belong under VOIP Service Providers B2B

Please keep your entry in ALPHABETICAL ORDER in relation to the other entries in your section.
If you add a new entry, including an 'added on dd/mmm/yy' would make it easier to notice.

Miscellaneous VOIP related services, including peer-to-peer services, are listed below.

Peer to Peer Service


  • ALTOTELECOM VoIP provider for business and Call Centers - AltoTelecom is a USA based VoIP company provides VoIP services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute (0.008) to USA, Canada and UK www.altotelecom.com

  • 2 - Tel2 - UK based Feature-rich VoIP provider with FREE signup and UK and European DIDs. Asterisk friendly ITSP with support for both the SIP and IAX2 protocols. Offer a host of services including call recording, web and video conferencing and collaboration services, faxmail (+ T.38 passthru), locate me, Smartphone and Desktop Apps and more. We offer all UK landline and tollfree numbers. Suitable for all users from residential, business through to large call centres. ...

ICT Innovations

$
0
0
images.jpg


ICT Innovations is a software development company having experienced and dedicated professionals with expertise in LAMP Stack and computer telephony integration (CTI). ICT Innovations has strong knowledge of Open Source communications technologies and applications such as Asterisk, Freeswitch, Drupal, Plivo,voip, Elastix and OpenSIPS/ Kamailio.

Products

ICTBroadcast

ICTBroadcast is a multi-tenant unified communications telemarketing software solution that supports Voice, SMS, Email & Fax broadcasting. It is suitable for small business owners, enterprises and Internet telephony service providers. ICTBroadcast is a smart auto dialer software with advanced autodialing multilingual supported features.

ICTDialer

ICTDialer is a Open Source unified communications marketing software. The ICTDialer is multi-tenant and allows voice, SMS & fax broadcasting. These capabilities have been developed using the Open Source content management system Drupal and the powerful Freeswitch and ICTCore Communication Framework. ICTDialer has the operating capacity to make thousands of simultaneous calls using VoIP, FoIP or PSTN.

ICTFAX

ICTFAX is an Open Source (GPL v 3) multi-user, web based software solution for service providers based on Open Source Spandsp, Drupal and ICTCore Framework. ICTFAX is an email to fax , web to fax gateway, it supports G.711 faxing, PSTN faxing and T.38 origination and termination.

ICTInvoice

ICTInvoice is an Open Source Elastix PBX module for invoice management. It converts Elastix PBX into a multi-tenant hosted PBX platform suitable for offering hosted PBX services to small business owners and enterprises. It is Open Source GPL v 3 software and is developed and maintained by ICT Innovations. ICTInvoice enhances the capabilities of Elastix billing and empowers Elastix administrators. It allows email invoices to be automatically generated and emailed to users on monthly basis.

Services

Consultancy services

ICTInnovations offer consultancy services to its clients and work with them to fully understand their exact business requirements. ICTInnovations customize, develop, migrate, integrate and provide network architecture and deployment plans to our international client base.

Support services

ICTInnovations provide support services for Open Source communications technologies such as Asterisk, Freeswitch, OpenSIPS, Plivo and Drupal.

Integration and Development

ICTInnovations offer services to allow the integration of Open Source telephony projects and components into any existing network. ICT Innovations can provide a complete business solution per each individual client's requirement. We will develop tailored API's on request to integrate Open Source VoIP projects with your existing communications infrastructure.

Monitoring and Support services

ICTInnovations provide 24/7 support services to monitor VoIP/Linux Servers. This ensures that any outage causes minimal service disruption. We provide immediate support for any issue arising and our professional support team is always available during our client’s business hours. Our monitoring support services include
  • Monitoring Linux servers during client’s business hours. ...

Predictive dialer

$
0
0

What Is A Predictive Dialer?


  • "A predictive dialer is a computerized system that automatically dials batches of telephone numbers for connection to agents assigned to sales or other campaigns. Predictive dialers are widely used in call centers." - Wikipedia

"Definitions of Predictive dialer on the Web:

  • A predictive dialer is an outbound call processing system designed to maintain a high level of utilization and cost efficiency in the contact center. The dialer automatically calls a list of telephone numbers, screens the unnecessary calls such as answering machines and busy signals, and then connects a waiting representative with the customer.

note: above text may be copyrighted by it's respective owners)

A VOIP Predictive Dialer, a.k.a. soft predictive dialer, is a software product capable of predictive dialing using VOIP service directly. Besides computer and internet connection, there is no equipment needed in order to use VOIP predictive dialer.


Software Only Predictive Dialer

New predictive dialing technology, together with faster computers and bigger broadband bandwidth, enables software only predictive dialers to work as good as or even better than hardware based dialers. Software based solution avoids expensive telephony board and associated hardware maintenance cost. It is easy to install and configure. For example, it is very easy to setup remote agent (at home agent).

SOFTWARE ONLY PREDICTIVE DIALER

  • Unique Call center Suite Predictive/ Progressive/Auto Dialer and Web CRM Solution. For more details contact: avil@uniqueperform.com

  • Texo.cc auto dialer is an Agile Customer Interaction hub on cloud, enabling you to build Personalized engagements across the customer journey. Texo.cc empowers users to quickly innovate with minimal technical Intervention, thereby challenging the status quo of traditional inbound, outbound, or blended contact centers. Contact Texo.cc for the best voip predictive dialer. The Texo.cc auto dialer has been trusted by leading outbound sales teams worldwide, checkout Texo.cc predictive dialer reviews for more.


Aavyukta Intel e Call Dialer and VoIP Solutions to Call Centers: Predictive Dialer (Unlimited Seats) + VoIP (US/UKLL/Canada) + Hosted/Cloud Server @ 1 US cent/Min, Reach us on www.dialerphilippines.com or catch us on skype on id avyukaindia +91-9549999916

  • 3CLogic Cloud-Based Contact Center Solutions - No hardware is required. Agents can be working from home or multiple locations. Automatically initiate contact with the next prospect before a rep finalizes a call, reducing call center queue times and operational costs. With 3CLogic, you will have complete visibility into call center operations including advanced scripting and reporting.
  • Astral Predective Dialer Solutions from Astrizon Technology Solutions, Predictive Dialer and Web CRM Solution, Provider With Operations in India, Europe, United State, United KIngdom, Singapore, Middle East etc Tel: +91 9447995111
  • BrightPattern - Next generation cloud-based contact center solution, ServicePattern. Includes a predictive dialer, and other dialing modes, in the outbound contact center solution. ...

Autodialer

$
0
0

What Is An Autodialer?

The following definition for Autodialer appears on Wikipedia.

  • '"An autodialer is an electronic device that can automatically dial telephone numbers to communicate between any two points in the telephone, mobile phone and pager networks. Once the call has been established (through the telephone exchange) the autodialer will announce verbal messages or transmit digital data (like SMS messages) to the called party."'

Autodialing Techniques

Autodialing can take on many forms including connecting calls to agents versus simply playing a recorded message. The following Auto Dialer Types are available from companies such as 3CLogic and Database Systems Corp.:

  • Voice Broadcasting delivers a pre-recorded message to live answers and answering machines. If another call status is detected (busy, etc.), the phone systems can reschedule the call for a later time. Simple messages can be delivered or the call recipient can be presented with an IVR script that accepts touch phone responses.

  • Preview Dialer allows a phone agents to view the call information prior to the call being placed. The agent can decide not to initiate the call.

  • Progressive Dialer passes the call information to the agent at the same time the number is being dialed by the phone dialer. The agent usually has a few seconds to view the call information, but cannot stop the call process. Often referred to as Forced Preview Dialing.

  • Predictive Dialer is more sophisticated because the phone dialer automatically calls several numbers and only passes a call to an agent when a person has been contacted. This eliminates busy signals, answering machines, etc.

  • Smart Predictive Dialer places calls, plays recorded messages and prompts, and passes the calls to agents only when the called individual requests a contact. "

Autodialing Equipment

Autodialers are usually desktop computers equiped with either special telephony voice boards or off-the-shelf voice modems. Traditionally, voice modems lack many of the features telephony boards offer. For example, call progress detection, call transfer, etc. However, with the ever improvement in today's computer speed, many of the telephony board features are implemented in software, which drastically reduces the cost and improves flexibility. Examples of such equipment are 3CLogic's Autodialer and Voicent BroadcastByPhone Autodialers, which offer all features offered by telephony board based autodialer systems and many more.

Another advantage of using a voice modem based system is that it can turn a laptop computer to an autodialing system. There are good quality USB based voice modems, such as those from Creative Lab and Zoom.


See Also

VoIP Softphones

$
0
0

What is a VoIP Softphone?

A VoIP softphone is a program that installs to and runs from your computer. A VoIP softphone enables you to make calls with just your computer using a VoIP service. Skype, iChat, and GoogleTalk are some of the more popular services, but there are many different ones available for you to choose from.

A VoIP softphone is accessible wherever you have a computer with you and there are both paid and free VoIP softphones. Companies often offer their own proprietary softphones that are configured to work with their service like Cisco and Counterpath.


The VoIP softphones are designed to be intuitive to use and most resemble an actual phone handset. Or you can choose to have the layout show your contacts if that makes calls easier. You can either click the buttons on the interface to dial or use the number pad on your keyboard.

VoIP softphones are only a program on the computer, so a headset with a microphone or an internal microphone and speakers are also needed to make the calls. Headsets prices start at a very affordable $5-$10 for a standard configuration and can get up in the $100-$200 area for a high-quality wireless headset with a long battery life and interchangeable ear pieces.

Who Can Benefit From VoIP Softphones?

Softphones can be used by anybody with a computer. There are a few types of users who can really benefit from the features of a VoIP softphone:

  • VoIP beginners
  • Heavy travelers
  • Telecommuters
  • Call center employees
  • Small businesses
  • Frequent long-distance callers


VoIP beginners can quickly and cheaply explore how the service works by downloading a VoIP softphone to make free computer-to-computer or computer-to-phone calls. Heavy travelers can avoid racking up large bills on their mobile phones or at hotels by using a low-cost VoIP service with a VoIP softphone. Telecommuters can register a VoIP softphone with their office PBX system to enjoy the same call features available to them at the office while they are on the move. Call center employees and small businesses can save on costs by pairing a VoIP service with a softphone to avoid purchasing and maintaining desk phones. International and long-distance rates are much lower when using a VoIP softphone, so those making regular or frequent calls out-of-state or country can cut some major costs.

VoIP Softphone Features

VoIP softphones offer the same features that traditional phones offer and more:

  • Call forwarding
  • Call conferencing
  • Hold capabilities
  • Call transferring
  • Voicemail
  • Greeting capabilities
  • Text, IM, and video capabilities
  • Echo cancellation to improve sound quality
  • Contact list/address book


Softphones also use less energy than phone and phone system hardware, which saves on costs and is useful for green companies.

VoIP Softphone Protocols

VoIP service uses different protocols to determine how the data is processed and transferred over the network. Your VoIP softphone needs to support the same protocol your VoIP service uses. ...

VOIP Phones

$
0
0
This page is for listing brief details of VoIP Phones including details, where to buy, specifications, and any other relevant VoIP Phone information. Please read the Posting Guidelines for Promoting Products and Services before adding to it.

Hard Phones

Standalone Ethernet Hard Phones (voice only)
An Ethernet hard phone is a self contained IP telephone that looks just like a conventional phone but instead of a conventional phone jack, it has an Ethernet port through which it communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone it does not require any personal computer nor any software running on a personal computer to make or receive VoIP phone calls. It can be used independently, all that is required is an internet connection. While PC based software solutions are cheaper, a hard phone is the best solution for IP telephony.

General


ageet Corporation

$
0
0

About ageet Corporation



ageet Corporationspecializes in innovative IP telephony software solutions for mobile, desktop and web systems. We are an experienced international team of VoIP software development specialists located mostly in Japan, but also in North America and Europe. For more than 10 years, ageet has been developing VoIP software across a variety of platforms from embedded systems over smartphones and tablets to desktop computers and servers. Apart from our softphone brand “AGEphone” "AGEphone for iOS" , "AGEphone for Android" , "AGEphone for Windows" and its "OEM versions”, we also provide custom tailored software solutions to our customers.


News Releases:


Kyoto, June 10th, 2008 ageet publishes AGEphone Mobile 2.5 Speakerphone Edition(HTML, external page)
Kyoto, February 28tht, 2008 "AGEphone Mobile 2" released(HTML)
Kyoto, October 29th, 2007 "AGEphone Google Gadget" for Google Sidebar announced(PDF)
Kyoto, August 29th, 2007 ageet announces its new and innovative softphone BIZTEL P-Free
Kyoto, June 25th, 2007 CallPicker: The whole Desktop as an Address Book - Quick Calls with AGEphone or Skype (PDF)
Kyoto, February 28th, 2007 ageet Releases IPv6 Supporting Softphone and VoIP SDK (HTML)
Kyoto, January 10, 2007 VoIP Gadget AGEphone Gadget for Windows Vista's Sidebar Announced (HTML)
Kyoto, November 15, 2006 VoIP on Your Mobile - AGEphone for Windows Mobile 5.0 announced (HTML)
Kyoto, September 29, 2006 New AGEphone Allows Call Forwarding and Provider Crossover (HTML, External Page)
Kyoto, September 20, 2006 ageet Corporation Launches Distribution and Sale of ActiveX VoIP Plug-In Series miroSIP Web Object (HTML, external page)

Address:


ageet Corporation
URL: http://www.ageet.com/us/
Head Office Address: 13-98 Shurishiki Terado-cho Mukou City Kyoto, 617-0002 Japan
Representative Director: Masato Okazaki
Established: July 2005 Capital: 21. ...

Voicepulse

$
0
0

VoicePulse FIVE


To learn more about VoicePulse FIVE and chat with a Live Representative click here.



VoicePulse FIVE is our fifth generation VoIP platform serving the residential, SMB and Wholesale segments. FIVE is a completely rewritten platform designed from the ground up to provide Internet telephony in the most powerful, flexible and easy to use way possible. Sign up for a free, no obligation evaluation account and be up and running in just minutes.

Company News

2014-11-06 VoicePulse Introduces VoicePulse FIVE, the Next generation in VoIP Services

Pricing

Business Gateway Pricing click here.

Ideal for SMB and Enterprise

Use your existing SIP enabled on-premise softswitch or PBX to make and receive phone calls regardless of call volume.

Channels

Channels are included with your Business Gateway at no additional cost

Endpoints

Endpoints are included with your Business Gateway at no additional cost

Phone Numbers

Choose your number from anywhere in the US or move your existing DIDs to VoicePulse FIVE. There is no fee to port your phone number to VoicePulse FIVE.

Incoming Calls

Incoming calls to a U.S. phone number are $.01 per minute

Incoming Toll-Free Calls

Incoming calls to a U.S. toll-free phone number are $.029 per minute.

Outgoing US & North American Calls

Outgoing calls to the US48 are $.02 per minute

Outgoing International Calls

Competitive International rates.
Look up international termination rates here.


Account Center


  • View your current Statement Balance
  • Monitor real time costs for usage, Endpoints, Trunks, Gateways, Call Apps, Channels, and E911
  • Make instant payments by credit card
  • Add unlimited Channels or call paths to your Trunk
  • View your active phone numbers
  • See our inventory of numbers
  • Instantly activate new numbers
  • Manage E911


Customer Support



Chat with a Live Representative. M-F 9am to 5pm EST


Supported Protocols


  • Session Initiation Protocol (SIP)

Supported Codecs


  • G.729a
  • G.711ulaw
  • G.711alaw
  • GSM
  • ADPCM
  • ILBC

Supported User Agents




  • Asterisk - The Open Source PBX, AsteriskNOW, AA50, SwitchVox
  • Fonality PBXtra
  • trixbox CE, SE, EE, CCE
  • FreeSWITCH
  • Elastix
  • 3CX
  • FreePBX
  • PBX-in-a-Flash
  • OpenSER / Kamailio
  • OpenSIPS
  • Cisco/Linksys SIP devices
  • SIPfoundry
  • Yate
  • Aastra, Grandstream, Snom SIP devices
  • IPitomy
  • OBIHAI technology
  • Kerio Technologies
  • Softphones
  • And more

Cisco

$
0
0

Cisco Systems, Inc

Cisco's IP telephony products offer great flexibility, a large feature set, and stand up well to every day corporate use. That being said, they're also not the cheapest in the field.

Products


Most likely you'll need access to firmware packages for upgrades or version changes, i.e. converting a phone from SCCP to SIP. The files are available on Cisco's website along with a good amount of documentation but you'll need a service contract. The service contract for a Cisco phone is around $8/year. The service contract price will be based on the value of the item you want covered. Cisco has several support options. You just need the most basic support in order to be able to get updated software.



GKTMP

GKTMP is Cisco's open, but proprietary, protocol used for communication between a call control device such as a Cisco Gatekeeper or SIP Proxy and an external application such as a route server. GKTMP, which stands for GateKeeper Transaction Message Protocol, was originally developed for H.323 gatekeepers but since it is a rich and lightweight Operations and Billing Support System (OSS/BSS) protocol, it is equally useful as an API to external applications for SIP proxies or B2BUAs. A major benefit of GKTMP is the ability to offload complex routing algorithms, such as least cost routing tables with hundreds of thousands of routes, to an external server.

An open source module which provides a GKTMP interface to open source OSP servers, such as OpenOSP and RAMS on www.sipfoundry.org/OSP, is available at sourceforge.net/projects/gktmp-to-osp.

Service Contract


Obtaining a service contract directly from cisco can be a lot of fun. When last checked, there's no online registration available, only a page with an email address that's no longer valid. ...

Asterisk Paging and Intercom

$
0
0

Paging and Intercom

On legacy phone systems you can find the following kinds of paging:

  • Dial a code to connect to a separate overhead paging and announcement system (like in an airport)
  • Dial a code and connect directly to a built-in one-way announcement speaker on one or more phones
  • Dial a code and connect directly to a built-in two-way announcement and talkback function on one or more phones

Some overhead paging systems also provide a talkback system so that the person being paged can just speak to respond. Background noise issues limit where this feature can be used. The talkback function is usually setup to be hands free. That means that the person responding to the page does not need to take any action other then speaking.

New in Asterisk 1.2: The new dialplan command Page utilizes MeetMe to page one or more phones.

New in Asterisk 1.8: A new RTP engine and channel driver have been added which supports Multicast RTP.
The channel driver can be used with the Page application to perform multicast RTP paging. The dial string format is:

MulticastRTP/<type>/<destination>/<control address>

Type can be either basic or linksys. Destination is the IP address and port for the RTP packets. Control address is specific to the linksys type and is used for sending the control packets unique to them.

Advanced Paging / Intercom

There is also another system available since many years, the best one, combining paging and intercom. Here the talback system is limited to only one phone. The paging is done in one way mode through a group of phones, and the person being paged can respond pressing a digit to switch the nearer phone to two-way mode, simultaneously hanging-up all other phones speakers.
This mode combine the best of the two world, eliminate the noise problems, and keep the communication private as soon as the paged person pressed the right digit on a phone.
It should be possible to implement this mode on Asterisk with a managed conference and a feature map application.

Multicasting begin to be supported at all major phones manufacturers, Aastra firmware v2.2, Snom v7, Linksys,... allow the setting of a multicast listening address. This will permit to reduce the generated trafic for an extensive paging.

If a phone is in use when a page arrives, some systems can do a "whisper page" so that only the person being paged can hear the page.

SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. The phones most often mentioned supporting this are:

VOIP Service Providers Business

$
0
0
This is a list of Business VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


VOIP Service providers divided by country/continent:

Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.


List of Service Providers:


1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 50 countries. We have regional network facilities spread across the globe.
  • Actual US CLEC
  • Branded customer portals
  • Multiple geographic locations on one Hosted PBX
  • Coverage in over 50 countries
  • Unlimited inbound on each channel
  • Great for inbound call centers
  • No fee's per user or extension



Amivox Amivox is a VOIP service provider supporting SIP and IAX connectivity. Amivox also provides APPs for business calling and messaging. ...

SIP Trunk Providers UK

$
0
0
This page is a list of SIP trunk providers in the UK (United Kingdom) including England and Scotland. Please keep this list in alphabetical order. UK SIP providers looking to add their services can do so in the list below.
__

ALTOTELECOMVoIP provider for business and Call Centers - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK www.altotelecom.com

1comms VoIP provider for UK Businesses. Sip trunk services, UK DDI's and Asterisk Support for business users and resellers. White label & fully itemised per second billing.

1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 50 countries. We have regional network facilities spread across the globe.
  • Actual US CLEC
  • Branded customer portals
  • Multiple geographic locations on one Hosted PBX
  • Coverage in over 50 countries
  • Unlimited inbound on each channel
  • Great for inbound call centers
  • No fee's per user or extension

2 - Tel2 - Feature-rich VoIP provider with FREE signup and UK DIDs. Offer a host of services including call recording, web and video conferencing and collaboration services, faxmail (+ T.38 passthru), locate me, Smartphone and Desktop Apps and more. We offer all UK landline and tollfree numbers. Suitable for all users from residential, business through to large call centres. Wholesale and reseller programs and a white label 'Telco in the Cloud' product available. Lowest Rates. Save with calling bundle rates of 0.6p for landlines, 2p for Mobiles. 40+ countries at 1p/min. Build your own Telco in the cloud under your own branding and set your own rates and create your own calling plans and bundles using our fully automated web portals.

Aloha Connect Part of the Aloha Telecommunications Group, a UK National operator. Aloha Connect provides a simple platform to provision a prepaid Free SIP trunk with options to purchase DIDs (numbers) from over 50 countries. Aloha focuses purely on the quality side of the market (especially in regards to international calling). UK and Mobile calls are some of the cheapest rates on the market.

  • AVOXI is a Sip Trunk Provider. AVOXI virtual call center solutions provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers.

  • CallForwarding - Be present anywhere in the world with toll free forwarding services from CallForwarding.com.
- Interactive Voice Response (IVR), Call Forwarding Services, Automatic Call Distributor



VOIP Service Providers Business North America

$
0
0
This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

Please describe services in neutral language and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!

Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

If you like this page, please link to it, so Google and other search engines will consider it more important.

Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page.

Providers in other countries/continents can be found here:


Canada


VoIP Gateways

$
0
0
Information about VoIP gateways, including VoIP media gateways, FXO gateways, and other VoIP gateways can be found on this page. If you want to add your company's products, please read the Posting Guidelines for Promoting Products and Services

Media Gateways

Media gateways, also commonly referred to as VoIP gateways are devices which bridge conventional telephone networks and equipment to VoIP telephone networks. A typical media gateway has at least one conventional telephone port and at least one ethernet port.

Analog FXO gateways


Viewing all 5767 articles
Browse latest View live




Latest Images