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VOIP Service Providers B2B

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Here is a list of VOIP Service Providers focusing on Business-To-Business services. This includes VoIP origination and VoIP termination, plans aimed at call centers, IVR providers and generic Asterisk users. See also:

Services which require the use of locked ATA devices should not be listed on this page. Nor should services which do not permit simultaneous calls — most services here support at least 4 simultaneous incoming calls. Please list only services which support Asterisk connections, via SIP or IAX2, to the PSTN.


Grandstream GXP1405

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Grandstream GXP1405 Small to Medium Business HD PoE IP Phone



Grandstream-VoIP-phones-gxp1400-gxp1405.jpg



The Grandstream GXP1405 IP Phone is a next generation small-to-medium business IP Telephone that features 2 lines with 1 SIP account, a 128x40 graphical LCD, 3 XML programmable context sensitive soft keys, dual network ports with integrated PoE and 3-way conferencing.

The Grandstream GXP1405 IP Phone delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.

The Grandstream GXP1405 IP Phone is a perfect choice for small-to-medium businesses looking for a high quality feature rich IP phone at an affordable cost.


Features


  • 128x40 pixel graphical LCD display
  • 2 line keys with dual color LED 1 SIP account and up to 2 call appearances 3 XML programmable context sensitive soft keys 3-way conference
  • HD wideband handset hands free speakerphone with advanced acoustic echo cancellation
  • Phonebook with up to 500 contacts and call history with up to 200 records
  • Automated personal information service (e.g. local weather) personalized music ring tone/ring back tone
  • Dual switched auto sensing 10/100 Mbps network ports with integrated PoE
  • Automated provisioning using TR 069 or AES encrypted XML configuration file, SRTP and TLS for advanced security protection, 8021x for media access control


Protocols and Standards


  • SIP RFC3261
  • TCP/IP/UDP
  • RTP/RTCP
  • HTTP/HTTPS
  • ARP/RARP
  • ICMP
  • DNS (A record, SRV, NAPTR)
  • DHCP
  • PPPoE
  • TELNET
  • TFTP
  • NTP
  • STUN
  • SIMPLE
  • TR-069
  • 802.1x


Feature Keys


  • 2 line keys with dual-color LED and 1 SIP account
  • 3 XML programmable context sensitive soft keys
  • 5 navigation/menu/volume keys
  • 8 dedicated function keys for:
    • Hold
    • Transfer
    • Conference
    • Volume
    • Headset
    • Mute
    • Speakerphone
    • Send/Redial


Voice Codec


  • Support for:
    • G.723.1
    • G.729A/B
    • G.711µ/a
    • G.726-32
    • G.722 (wide-band)
    • iLBC
    • in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)


GXP1400.jpg

Grandstream GXP 2100

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Grandstream GXP2100 HD Enterprise 4-line PoE IP Phone



GRGXP2100.jpg



The Grandstream GXP2100 IP Phone is a next generation, enterprise grade IP phone that features 4 lines a 180 x 90 backlit graphical LCD, 3 XML programmable context-sensitive soft keys,
7 XML programmable BLF extension keys, dual network ports with integrated PoE and 5 way conferencing.

The Grandstream GXP2100 IP Phone delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service,
automated provisioning for easy deployment, advanced security protection for privacy and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.

It is a perfect choice for enterprise users looking for a high quality, feature rich multi-line IP phone with the best values.

Feature Highlights


  • 180x90 pixel backlit graphical LCD display with up to 8 level grayscale
  • Dual switched auto-sensing 10/100Mbps network ports with integrated PoE
  • Large phonebook (up to 2,000 contacts) and call history (up to 2,000 records)
  • 4 dual-color line keys (with 4 SIP accounts), up to 5-way conference (max 4 concurrent calls), and up to 11 call appearances with 7 dual-color BLF (Busy Lamp Field) extension keys
  • 3 XML programmable context-sensitive soft keys, a desktop stand to allow 2-angle positions
  • HD wideband audio, superb full-duplex hands-free speakerphone with advanced acoustic echo cancellation and excellent double-talk performance
  • Automated personal information service (e.g. local weather, stock, currencies, RSS news, etc), personalized music ring tone/music ring back tone/music-on-hold using Internet music streaming or local music files, flexible customizable screen content & format using Web and enterprise applications integration (pending)
  • RJ9 and 2.5mm jacks for professional and casual use
  • Can be used as a desktop phone or mounted on a wall


Protocols/Standards


  • SIP RFC3261
  • TCP/IP/UDP
  • RTP/RTCP
  • HTTP/HTTPS
  • ARP/RARP
  • ICMP
  • DNS (A record, SRV, NAPTR)
  • DHCP
  • PPPoE
  • TELNET
  • TFTP
  • NTP
  • STUN
  • SIMPLE
  • TR-069
  • 802.1x
  • IPV6 (pending)

Network Interfaces


  • Dual switched 10/100Mbps ports with integrated PoE

Feature Keys


  • 4 line keys with dual-color LED and 4 independent SIP accounts
  • 7 programmable BLF extension keys with dual-colour LED
  • 3 XML programmable context sensitive soft keys
  • 5 navigation/menu/volume keys
  • 11 dedicated function keys for:
    • Hold
    • Speakerphone
    • Send/Redial
    • Transfer
    • Conference
    • Mute
    • Headset
    • Intercom/Paging
    • Volume
    • Phonebook/Contacts
    • Message (with LED indicator)

Voice Codec


  • Support for
    • G.723.1
    • G.729A/B
    • G.711u/a-law
    • G.726
    • G. ...

Grandstream GXP1450

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Grandstream GXP1450 Enterprise 2-Line POE IP Phone



GRGXP1450.jpg



The Grandstream GXP1450 IP Phone is a next generation, enterprise grade IP phone that features 2 lines with 2 SIP accounts, a 180x60 backlit graphical LCD, 3 XML programmable context sensitive soft keys, dual network ports with integrated PoE and 3-way conferencing.

The Grandstream GXP1450 IP Phone delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service automated provisioning for easy deployment, advanced security protection for privacy and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.

It is a perfect choice for enterprise users looking for a high quality feature rich IP phone with affordable cost.


Features


  • 180x60 pixel backlit graphical LCD display with up to 4 level grayscale
  • 2 dual color line keys (with 2 SIP accounts and up to 2 call appearances)
  • 3 XML programmable context sensitive soft keys
  • 3-way conference
  • HD wideband audio, superb full duplex hands free speakerphone with advanced acoustic echo cancellation and excellent double-talk performance
  • Large phonebook (up to 2,000 contacts) and call history (up to 2,000 records)
  • Automated personal information service (e.g. local weather, etc.)
  • Personalized music ring tone/ring back tone, flexible customizable screen content & format using XML and advanced Web and enterprise applications integration
  • Dual switched auto sensing 10/100Mbps network ports with integrated PoE
  • Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
  • RJ9 and 2.5mm headset jack
  • Allow 2 angle positions for the base stand and can be mounted on a wall


Feature Keys


  • 2 line keys with dual-color LED and 2 independent SIP accounts
  • 3 XML programmable context sensitive soft keys
  • 5 navigation/menu/volume keys
  • 10 dedicated function keys for:
    • Hold
    • Speakerphone
    • Send/Redial
    • Transfer
    • Conference
    • Mute
    • Headset
    • Volume
    • Phonebook/Contacts
    • Message (with LED indicator)


gxp1450__1.jpg

Asterisk Consultants Austria

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Asterisk consultants: Austria


Add your entry here (Alphabetical order by country and company):
This page is growing large. Please don't post logos!!

ACCM Gmbh, Wien

ACCM is an austrian telephone-system-provider, combining the benefit of traditional telephone systems with the possibilities of Internet. ACCM configures the best solution for your companies needs, regarding least cost, efficiency and scalability.

ACK EDV Dienstleistungs-GmbH

ACK provides consulting, development and deployment for providers and large companies. Several enhancements and tools for integration with other systems are available. (e.g. billing module, webinterfaces, intelligent GSM gateways etc.)

Alexander Topolanek EDV-Beratung und Support

Company with a ten years expierience in the field of telecommunications and call center solutions. We're providing tailored solutions for your business, based on the open source telephony system Asterisk.

Armstrong Consulting GmbH

Armstrong Consulting helps international companies to carry out VoIP deployments and call-center legacy-system integrations.

Asterisk Consultant.com

Brings Asterisk, VOIP, Linux and Network consultants from around the world including Austria under one roof. You simply post your project requirements, maximum budget and time constraints. Consultants will bid on your project in the reverse style auction, giving you the best price possible and luxury of choosing which consultant/company to use.

Post your Project Requirements today at
Web: www.asterisk-consultant.com


Avalaris, Wien

Asterisk and legacy PBX integration, migration
Integration services with SIP, H.323 networks and legacy systems.
Development, tuning and integration of Open Source VoIP networks. ...

Zycoo

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zycoo_logo2.gif


Zycoo is an original manufacture of China in VoIP communication field. It is located in Software Incubation Park of Chengdu, China. We engaged in R&D of Asteirsk hardware,such as 4 and 8 ports Embedded ippbx, 4/8/16 ports analog PCI cards,etc.
Website: http://www.zycoo.com/

News: Zycoo IP PBX with Plastic case---ZPXP is released!

PBX front---SMALL.JPG

PBX back small.JPG


Products

  • Asterisk Analog Cards
    • ZA4P 4 ports Asterisk Analog PCI Card, like Digium’s TDM400P
    • Zycoo ZA4E 4 ports Asterisk Analog PCI-E Card
    • ZA8P 8 ports Asterisk Analog PCI Card, up to 8 total FXO/FXS connections
    • Zycoo ZA8E 8 ports Asterisk Analog PCI-E Card
    • ZA16P 16 ports Asterisk Analog PCI Card, up to 16 total FXO/FXS connections
    • Zycoo ZA16E 16 ports Asterisk Analog PCI-E Card
    • ZD1P single E1 ISDN PCI card up to 30 channels

  • Embedded IP PBX
  • IP Phones
    • ZP302 IP Phone
    • Zycoo ZP306 IP Phone
    • Zycoo ZP502 IP Phone

  • Customized case
    • We provide customized service for our clients, and till now, we have finished customized cases excellently for clients in US, Poland,etc.
To look for more info of these products, please visit our website and download directly from: http://www.zycoo.com/download.php








Now, we are looking for partners in the whole world for our products in competitive price and good after-sale service.

PromotionActivities
ZPXPisinpromoting. ...

MapleLeaf Technologies

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Looking for VoIP hardware?

looking4voip.jpg

You're in the right place!


Benefits
  • get reseller discounts
  • buy vat-free
  • enjoy subsidized shipping
  • earn bonus discounts

VoIP Technologies & Solutions
  • Appliances
  • ATAs
  • Channel Banks
  • Gateways
  • IP Phones
  • PRI Bridges
  • BRI Telephony Cards
  • FXO/FXS Telephony Cards
  • Gateway Cards
  • GSM Telephony Cards
  • Hybrid Telephony Cards
  • PRI Telephony Cards
  • Transcoder Telephony Cards

Manufacturers
  • ALLODOTCOM, Inc.
  • ATCOM Technology Co. Ltd.
  • beroNet GmbH
  • Digium, Inc.
  • OpenVox Communication Co. Ltd.
  • Redfone Communications, LLC
  • Rhino Equipment Corp.
  • Sangoma Technologies Corp.
  • Xorcom Ltd.
  • ZYCOO Co. Ltd.

Why select MapleLeaf Technologies as your supply partner

Product Portfolio
  • Best-in-Class products from the market's premier Manufacturers!
  • Outstanding market knowledge, experience, and expertise!
  • Highly competitive "fair-price" Pricing Policy!

Customer Service
  • Online and available for you 24/7 - every single day of the year!
  • Orders processed and acknowledged the same day!
  • Shipment notification and package tracking data by email!
  • All shipments pre-insured to ensure maximum integrity of supply!

Customer Support
  • Manufacturer's Product Warranty Terms & Conditions Apply!
  • Pre-Exchange Warranty Replacement Unit Program!
  • Conditional 30-day "try 'n buy" Return Program!

Supply Logistics
  • Product Inventory sited in Europe for rapid deployment!
  • Subsidized Europe-wide shipping costs!

MapleLeaf Technologies
Hauptstrasse 9
83052 Bruckmühl, Germany
Tel: (+49) 8062-7269930
Email: Sales@mapleleaf-technologies.com
Internet: www.mapleleaf-technologies.eu


VoIP Gateways

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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



  • Allywll-WIA2008 GSM/CDMA VoIP gateway
  • 1Telecom Ltd - FXO/FXS VoIP GSM Gateways
  • 2N Telekomunikace - FXO GSM Gateways
  • Aastra Aastra Venture FXO Gateway
  • Abilis Abilis the all-in-one VoIP gateway with ISDN backup
  • AirTouch - FXO/FXS Skype VoIP Gateways
  • ALLYWLL - FXO/FXS/SIP VoIP Gateways
  • Allwin Tech SIP/H.323 dual protocols, 2/4/8 FXO/FXS ports ,NAT, Router, register up to 4 servers simultaneously
  • Anketechnology - FXO gateway--VoicePixie-211 www.anketechnology. ...

SIP Trunking

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From the SIP RFC 4904:


A Session Initiation Protocol (SIP) to PSTN gateway may have trunks that are connected to different carriers. It is entirely reasonable for a SIP proxy to choose — based on factors not enumerated in this document — which carrier a call is sent to when it proxies a session setup request to the gateway. Since multiple carriers can transport a call to a particular phone number, the phone number itself is not sufficient to identify the carrier at the gateway. An additional piece of information in the form of a trunk group can be used to further pare down the choices at the gateway. As used in this document, trunks are necessarily tied to gateways, and a proxy that uses trunk groups during routing of the request to a particular gateway knows and controls which gateway the call will be routed to, and knows what trunking resources are present on that gateway.


In an architecture where calls can be terminated on multiple gateways it is wise to consider routing the call to a destination based on some significant criteria such as cost, quality or proximity. Where a proxy has the ability to evaluate a call based on one or more of these criteria, as well as knowledge of the TDM trunk resources available, the proxy can "tag" the call using the tgrp and trunk-context values in the SIP Contact field of the INVITE. It is important to note that the tgrp and trunk-context values can only be used with a TEL URI, not with a SIP URI.

Unlike in traditional telephony, where bundles of physical wires were once delivered from the service provider to a business, a SIP trunk allows a company to replace these traditional fixed Public Switched Telephony Network (PSTN) lines with PSTN connectivity via a SIP trunking service.

To be added...

  • PSTN-to-PSTN calls traversing SIP networks

See Also:

Hosted IVR Services

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Hosted IVR Services over VoIP and traditional landlines are not very prevalent in India. This is primarily due to lack of complete automation in SMEs, high access deficit charges (ADCs) and non-availability of such services in the market.

Commercial Vendors


  • Database Systems Corp. provides a wide range of IVR outsourcing and IVR hosting services. Hosted IVR is a popular program with many organizations that do not have the telecommunications equipment and infrastructure to maintain their own calling applications. DSC can develop and manage virtually any sized IVR campaign.
  • EasyIVR provides a complete line of IVR solutions including IVR services, IVR systems, and hosted services and development services.
  • Hosted IVR solutions from Custom IVR
  • IVR Service including IVR development services as well as hosted IVR services.
  • Phonon is one such company that has just started offering hosted IVR services in this market.
  • Vconnect Hosted IVR service provider in India, pay as you use model. Plans starting as low as Rs.2000/month. KooKoo Cloud IVR based on Vconnect. Interested customers can contact Atul, 9989327222.

DID Service Providers

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A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

Cheapest DID Providers


Algeria

  • Algerian DID numbers Currently only national Algerian numbers.. Also 68 other countries available. Free forwarding to Voip/SIP, IAX,H323 Google Talk, etc many providers preconfigured. Forwarding to PSTN (landlines and mobiles) from 2cent/min, including mobiles. Mobiles Europe 6cent/min.

Argentina

  • Phone2call Virtual Numbers DID are available in 60+ countries including Argentina. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership.
  • Argentina DID Numbers€ 4.99/month| DomesticNumbers offers Argentina virtual phone numbers from 19 different cities in Argentina, including Buenos Aires. Website also in Spanish.

Australia

  • Phone2call Virtual Numbers DID are available in 60+ countries including Australia. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership.
  • Comfax Australia Free Fax Free Fax to Email Corporate Services

Austria

  • Sipgate.at Free personal 0720- national number in Austria.
  • TeleCallMart Unlimited Incoming Phone Numbers, Voip calls, SIP Phone, Auto Attendant, DTMF. No monthly fees. Low prices!


Bahrain

  • Phone2call Virtual Numbers DID are available in 60+ countries including Bahrain. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership. ...

voip-info.org

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Welcome to the VOIP Wiki - a reference guide to all things VOIP.


This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.

Update: We have added theFacebook Like and Google +1 button to the top right corner of all pages on Voip-Info.org. Please help recommend the wiki by clicking on them. We also now have Google+ page here and a Facebook page here. Visit them and add us. Thanks!


NEWS

HyperMedia Systems

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Hypermedia Systems


Hypermedia Systems develops and manufactures scalable, professional GSM Gateways and VoIP GSM Gateways for low to high volumes of traffic, starting from 8 ports, with up to 72 cellular channels/ 288 SIMs. Available interfaces: VoIP, E1/T1-PRI and Analog interfaces, as well as GSM, 3G UMTS, CDMA.

These carrier grade Voice Gateways provide opportunities for Alternative carriers, Call termination vendors, and Service providers, as well as for SMB and corporate enterprises.

Read our Testimonials page.


Visit Hypermedia's NewWebsite

We encourage you to visit and get to know it. Find many interesting new areas, like a new downloads page where you can find White Papers, Data Sheets, and Application Notes and more. See our latest addition: Clips & Presentaions.

GSM Gateways and VoIP Gateways for Call Termination

Compact bi-directional gateways for providing cost-effective termination services to or from GSM, UMTS/3G, CDMA, VoIP, E1/T1-PRI or Analog.

  • Fast Start Series for Call Termination - Plug 'n Sell VoIP GSM Gateway. An ALL-IN-ONE solution for the termination business with a minimum investment.
  • Professional VoIP GSM Gateway. HG-4000 Series. From 12 to 72 cellular ports.
  • Call Termination Systems for VoIP/3G, GSM & CDMA.
  • Combination UMTS / GSM Gateway - HG-5000 Series. The right opportunity for anyone who plans to provide termination services by using both GSM and UMTS cellular networks. You can have several combinations of technologies in the same unit, and receive the traffic by using Embedded VoIP Gateway or by E1/T1-PRI interfaces.


HG4000_side.jpg



GSM Gateways for Cutting Telecom costs

Hypermedia provides a wide range of Call Cost-Saving solutions ideal for cutting costs on daily communication expenses and overall corporate telephony expenditure, while leveraging existing SMB & Corporate business telephony infrastructure.

You can save on:
  • Office-to-Cellular and Cellular-to-Office call costs,
  • on GSM Roaming charges with Callback solutions,
  • and on national and local calls -

BULK SMS

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CDTECH a rapidly growing software based company in Chennai,India developing various popular software products to provide effective and advanced communication services .

With more than 10,000 customers across various states in India,CDTECH has achieved heights in Technology with various research centers in India.

Apart from developing product based softwares, CDTECH also offers software customization and support for Small Network Implementations using simple routers and switches.

Some of the products developed by CDTECH are...
1. Auto Call 2. SMS Broadcaster 3. SMS All 4. SMS Interactor 5. E-mail All 6. Video streaming and surveillance 7. Home Automation with PC 8. College management software 9. School management software and many more...

Visit our website at www.cdtech.in to see how you can be benefitted through these technological products or customized solutions.


CDTECH takes Vo-Ip to the normal phone user. AutoCaller was developed for this purpose. It can help you automate phone calls with recorded voice messsages and act as an IVR.
Check out http://autocaller.in for info about it

Bulk SMS Gateways


Hypermedia Systems develops and manufactures Customer Premise (CPE) SMS server gateways that give you independency and full control over the information flow so you can create, manage and deploy your own SMS services and campaigns anytime and anywhere!

Watch our Bulk SMS Gateway video!

Hypermedia Bulk SMS Gateway Pro is a SIM-based platform designed for 2-way SMS messaging directly via the cellular networks to and from any authorized SMS client. Using SIM cards with an unlimited text plan on an SMS gateway device can drive bulk SMS costs down in orders of magnitude!
Hypermedia SMS Gateway is independent equipment that takes into consideration SMS cloud services security and database integrity issues, and overcomes the risk using direct SMSC gateway.

Diafaan SMS Server for Hypermedia's HG-7000: Diafaan releases new plugin for Diafaan SMS Server for Hypermedia's HG-7000

One SMS box - A thousand possibilities - Take your text to the next step!


Stay updated on SMS Hardware: Read about the advantages of Autonomous SMS Gateway hardware device, or about SMSing where Other's Can't Deliver

Related links:
SMS Gateway Hardware
SMS Campaigner
Multifactor Authentication Hardware Systems

How to start a VOIP Business

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

  • Prepaid Cards
  • VOIP consulting
  • PBX sales and service
  • Call Center Solutions
  • Independent Sales/Service Agent for existing VOIP service providers
  • VOIP Provider services
  • etc.

Within each category there are many choices. For example, for 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.

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

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.

White Paper on Starting an Internet Telephony Service Provider:

If you are considering starting an ITSP, then this paper is a must-read. It draws from years of experience in setting up ITSPs around the world.
"How to Start an ITSP"


Whitepaper on "How to Become and ITSP":

So how does one get started in offering such a service? It is easier than you might think!!!
Whitepaper"How to Become an ITSP"


Whitepaper on "How to Sell Wholesale VoIP Services:

Whitepaper

Asterisk system vendors INDIA

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This is a list of Asterisk system vendors in India.

AAAB - SOFTTEK SOLUTIONS PVT. LTD.- Asterisk Consultant India

Final Logo.jpg

Services Provided:

  • Predictive Dialer
  • Voice Broadcasting & Fax Blasting
  • Medium to Large Multi-Office Business Telephony Systems
  • Call Centers Solutions - Local and International
  • Asterisk Dialers and Bulk Calling Systems
  • International Office Telephony Systems
  • Specialty Calling Systems (Entertainment and Personals)
  • Healthcare Application Integration
  • Customer Relationship Management, CRM Application Integration
  • Distributed Server Architecture and Asterisk Load Balancing
  • SIP Express Router, SER Load Balancing
  • Hospitality Telephony Systems (Hotel PBX Integration)
  • Complete IVR Development
  • Local or Datacenter PBX Customization
  • Wireless Telephony Installations
  • Database Integration and Customization with Applications
  • Custom Application Development on Asterisk ( Voice and Data )
  • vtiger vicidial integration
  • End to End VOIP Set-up
  • VOIP Minutes
  • Asterisk - IP PBX
  • Elastix
  • FreePBX
  • AsteriskNow
  • sipXces
  • FreeSwitch
  • Kannel
  • IVRS
  • Gtalk Integration with asterisk ( Request/Response )
  • Custom applications : Hosted IVR, on demand IVR, IVR on cloud, OBD, Click2Call.
  • Enterprise VAS

IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices.

Support for digium / openvox / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards /grandstream

Asterisk IPPBX India (Mumbai / Pune / Delhi / Hyderabad / Bangalore / All India)

Asterisk Consultant : Kokkula Chandhramohan

Asterisk Ph : +91 9821046774 / +91 9223333464

mailto:softtek.mumbai@gmail.com

mailto:chandhramohan.kokkula@softteksolutions.in

http://www.softteksolutions. ...

Broadview Networks, Inc.

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Broadview Networks, Inc. is a communications and information technology provider for business a crossed the States. Founded in 1996 as a regional telecom company; they grew into national provider of Cloud Phone System and other cloud solutions.

VoIP Providers USA

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This is a list of VoIP providers in the USA. These companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP. Please add VoIP Providers USA to the list below.

  • 011now.com Cheap International long distance phone to phone calling by 011Now offers you great rates from any phone in US and Canada. Save 90% on international long distance calls. voip-info Page for more information.
  • 1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
  • 45meeting Simple online conference solution for small business.
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FoIP

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Fax over IP - FoIP

Fax over IP can be done in many ways, either realtime or store-and-forward

Realtime

  • Transmission of a fax audio stream over a VoIP protocol, like SIP, IAX, H.323. This requires an uncompressed 64 kbit codec to have any chance of working, resulting in the need of around 80 kbit bandwidth. For a number of reasons it may or may not work even then.
  • Fax over IP with a special protocol designed for this application. The key standardised protocol for this is T.38.

Store and forward

  • A FAX, from a FAX machine, is decoded and stored by a FoIP gateway. It is then sent, using SMTP or a similar transport, to another FoIP gateway where it may be sent out to a destination FAX machine. The key standardised protocol for this is T.37.


FoIP Resources

All about FoIP As IP networks and VoIP adoption expands, and traditional Public-Switched Telephone Network (PSTN) phone services change, organizations with paper-intensive communication processes increasingly look to Fax over IP (FoIP) as part of their solution. AllAboutFoIP.com is a bold initiative which provides VoIP providers, integration specialists and their clients with the most instrumental offerings of FoIP components and solutions, including Audio Codes, Alcatel, Avaya, Biscom, Cisco, Commetrex, Dialogic (Brooktrout), EtherFax, Faxback, Faxcore, GFi, Open Text (RightFax), SagemCom (XmediusFax), ShoreTel, and TargetFax.

FoIP Software

T.38 with Fax Voip T38 Fax and 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 fax using SIP and H.323.
Fax Voip application operates with T.38 faxes via standart COM port interface.
Fax Voip T38 Fax & Voice - Fax and Answering Machine for your SIP/H.323/ISDN CAPI 2.0 line. Multiple SIP Registrations. Call Routing. Voice Fax Modem for your Fax & Voice software. Color faxes over VOIP and ISDN. Incoming Routing Methods. Fax-On-Demand.
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 allow you to work with different SIP and H.323 providers simultaneously.
Caller ID and Dial a Phone Number Extension features are supported.
You can send T.38, audio and CAPI faxes via Fax Voip Virtual Printer and receive faxes directly in TIFF, PDF or SFF files. You can manage faxes with Fax Voip Console.
Fax Voip allow to send faxes via e-mail (Mail to Fax) and receive 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 allow callers to retrieve information via fax on the same call.
Fax Voip T38 Fax & Voice has been successfully tested with Venta Fax & Voice (include color faxes), Microsoft Fax,

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