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DID Service Providers

May 24, 2013, 2:47 am
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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.
  • Bahraini DID numbers Currently only regional Manama numbers. ...
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MapleLeaf Technologies

May 25, 2013, 1:25 am
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maple_logo.jpg

Looking for VoIP hardware?

looking4voip.jpg

You're in the right place!


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


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Asterisk Consultants Belgium

May 26, 2013, 12:07 am
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AAA(Acube)


Professional support for office PBX systems based on asterisk.
Expertise in Trixbox, Asterisk, FreePBX, PBX-in-a-Flash
ViciDial
Voice Broadcasting Server
A2billing
Custom application development
IVR systems
Billing Systems
Remote Server administration & management
24/7 Helpdesk support
Asterisk Integration
CRM integration

25$ per hour

email: acube.asterisk@gmail.com


ACCRUE CONVERGENCE SOLUTION PVT. LTD.

  • Highly professional support for home and office PBX systems based on asterisk.
  • Expertise in ViciDial,Trixbox, Asterisk, FreePBX,Freeswitch, Hylafax, ATA, OpenSER, FreeRadius
  • Asterisk module development and source code customizations
  • Custom application development based on asterisk
  • IVR systems
  • Billing Systems
  • OpenSource Consulting
  • CRM Integration
  • Remote Server administration & management
mailto:info@accrueindia.com
http://www.accrueindia.com

ACOServices, Louvain-la-Neuve

Expert in professional Asterisk integration and development, ACOServices provides experienced and Dcap certified consultants.
ACOServices offers Asterisk solutions for CRM application integration, IP phones, IVR platform, fully integrated Call Centers with SLA, tailor-made PBX and specific development.

    • Contact: Danny Larbouillat / Benoît Carpent
    • Homepage:http://www.acoservices.be
    • Telephone : +32 10 84 33 12
    • Fax : +32 10 84 33 09
    • Email:info@acoservices.be

ADI ICT, Mechelen

ADI ICT offers services for small offices to large multi-site companies. Our focus is to deliver low costs solutions by using as much as possible Open-Source software.
We are specialized in Voice over IP (VoIP), Computer Telephony Integration (CTI), Contact center frameworks / CRM, Web design, Application design and Project Management.

    • Contact: Joop De Wel
    • Homepage:http://www.adiict.be
    • Telephone : +32 2 888 91 06
    • Fax : +32 15 55 91 84
    • Email:info@adiict.be

Alunys, Brussels

Alunys provides Asterisk turnkey IPBX and Voice Gateway solutions. (a) Our IPBX package comes with extensive web management interface and gives you access to business grade functionalities at the tip of the mouse. (b) Our Advanced Voice Gateway package connects your legacy PABX to the Internet, and extends its features. ...
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Asterisk Consultants Netherlands

May 26, 2013, 12:08 am
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Asterisk consultants: The Netherlands


Add your entry here (Alphabetical order by country and company):
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You might also be interested in VOIP Service Providers B2B in the Netherlands

aPeer

Het werkterrein van aPeer is uitsluitend Asterisk / Freepbx implementaties.
7 jaar ervaring met asterisk.
Geert Sillekens is DCAP Certified

Aproxius Internet

Aproxius Internet voor de implementatie van Asterisk VoIP. We verzorgen maatoplossingen voor
MKB en callcenter branche.
    • Home page:http://www.aproxius.net/
    • Email:info@aproxius.nl
    • Telefoon: +31 58 7501313
    • Postadres: Postbus 2553, 8901 AB LEEUWARDEN

Axeos Services B.V.

Axeos Services B.V. voor de implementatie van VoIP. VoIP oplossing via centraal hosted
platform of als stand-alone systeem. Axeos levert ook VoIP services aan ISP's en ASP's.

belcentrale.nl

belcentrale.nl is een VOIP provider die tevens Asterisk implementaties verzorgt op ASP basis. Veel ervaring in
trajecten binnen MKB omgeving. VOIP diensten, hardware en ASP oplossingen. Altijd zeven dagen per
week telefonische support. Bel ons 088 333 444 6


Chimit Communications

Chimit is een software ontwikkelingshuis met een speciale focus op datacommunicatie.
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Asterisk Consultants France

May 26, 2013, 12:08 am
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AB2L, Strasbourg

AB2L Intégrateur certifié dCAP en Alsace. Propose tout type d'infrastructure avec asterisk avec la voix sur IP.

Acropolis Telecom, Paris

Acropolis Telecom is an internet telephony service provider who interconnect with any equipements and install turn-key solution for small and medium companies

alloDom, Reunion Island

alloDom is an ITSP specialised in high-latency and low-bandwidth issues encountered in overseas territories. We deploy turn-key solutions in SMBs, such as gateways to ensure a seamless switch to IP.

A L S A C E C O M, France, Strasbourg

ALSACECOM VoIP Product Reseller - France and EU Distribution. dCAP certified.

Ambika, Paris

Ambika provides consulting and open source software based solutions for small and medium companies
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Asterisk Consultants Germany

May 26, 2013, 12:09 am
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Asterisk consultants: Germany


Add your entry here (Alphabetical order by country and company):
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Add your geo-position on Asterisk-users Counter.

ADDIX Internet Services GmbH, Kiel

ADDIX bietet auf Asterisk basierende Telefonanlagen, Datenbank gestuetztes Administrationstool fuer Admins und User, Echtzeitmanagement von Telefonkonferenzen, Vermittlungsarbeitsplätze, Mobile Applicationen für Smartphones. Anbindung an Junghanns BMS Callcenter Applikation, Master/Slave Systeme zur zentralen Verwaltung in groesseren VPN/MPLS Netzen mit x Systemen. Internet Services und eigene Data Center. Asterisk Hosting mit PSTN Gateway, VPN/MPLS (Filialvernetzung) und Programmierung.

ADES GmbH, Burscheid

ADES GmbH bietet von der Konfigurationsunterstuetzung bis zur kompletten ASTERISK-Anlage komplette Dienstleitungen an.
    • Home page:: http://www.ades.de
    • Telephone: PSTN +49.2174.64043
    • Email:asterisk@ades.de
    • Contact: Bent Weichert

Adimus GmbH, Bochum

Adimus bietet Beratung, Konzeption, Implementierung und Support von Asteriskbasierenden Telefonanlagen sowie VPN-Lösungen für klein- und mittelständische Unternehmen.


aixvox GmbH, Aachen

Die aixvox GmbH ist ein international tätiges Beratungs- und Dienstleistungsunternehmen. Unser Fokus ist es, Telekommunikationsinfrastruktur in Unternehmen neu aufzubauen bzw. vorhandene Strukturen zu profitablen Systemen auszubauen; Asterisk bzw. Asterisk-basierte Systeme sind oft die passende Lösung. aixvox ist auch Herausgeber des unabhängigen Kompendiums voice compass 2007, der ausführlichen Übersicht über den deutschsprachigen Voice Markt.


AMOOMA GmbH, 56566 Neuwied

AMOOMA bietet sowohl Consulting wie auch Schulungen zu den Themen Asterisk, VoIP und Trouble-Ticket-Systeme an. ...
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Asterisk consultants Africa

May 26, 2013, 12:10 am
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Aptus FonB

May 26, 2013, 12:44 pm
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Page Contents

back to top
The Phonebook Solution For Asterisk – Aptus FonB is a software product to be released soon by Aptus Telecom, developed to provide integrated contacts from Highrise, Google and Mobile right on your IP Phone registered with the Asterisk Telephony Engine. It makes easier for a user to merge and sync contacts so that a unified phonebook is provided in IP Phone. FonB features contacts from different platforms (Google, Highrise, Mobile) along with the comprehensive Call History. FonB is expected to work with major Asterisk distros like FreePBX, Elastix etc.

Brief details of FonB features are listed as under.


1. Contacts

Aptus presents 'Contacts' as the most unique and strongest feature of FonB. The idea of FonB basically revolves around syncing multiple contact lists from different devices in one unified directory. In technical terms, sophisticated use of XML flow and event triggering binds all the main constituents of the system ('Google Contacts', 'Highrise CRM' and Asterisk). Security tokens by Google and Highrise CRM also play a vital role in the efficient handling of contacts.

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1.1. ...

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Asterisk system vendors

May 27, 2013, 2:28 pm
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AX-1600P

May 27, 2013, 2:29 pm
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  • 4 ports FXO/FXS card

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AX1600P Asterisk card is the telephony PCI card that support up to sixteen FXO and FXS ports.
Using AX1600P analog card, open source Asterisk PBX and stand alone PC, users can create their SOHO telephony solution which include all the sophisticated features of traditional PBX, and extend features such as voicemail in IP PBX. The FXO and FXS modules are interchangeable to suit various requirements.

Features:

    • Support 1-16 FXO/FXS with PCI Interface
    • Support Asterisk ,Freeswitch , Dahdi , Zaptel
    • Support Trixbox , Elastix , Askozia
    • Application: PBX/Voicemail/IVR/Call Center/Call Park/Call Pickup/Call Transfer/Call Forward/Caller ID/Call Waiting/Call Conference
    • Elastix certified

Specifications:

  • Configuration
    • Motherboard: AX1600P
    • Dual ports FXS module: AX210S
    • Dual ports FXO module: AX210X

  • Hardware requirement
    • 500-Mhz Pentium III
    • 64MB RAM
    • 3.3V or 5V PCI 2.2 slot

  • PCI card dimension
    • 264mm (length) × 121mm (height)

Contact ATCOM:

ATCOM Technology co., LTD.
Address: A2F , Block 3 ,Huangguan Technology Park , #21 Tairan 9th Rd, Chegongmiao , Futian District , Shenzhen China
Tel: +(86)755-23487618
Fax: +(86)755-23485319
E-mail: sales@atcomemail.com
Website: http://www.atcom.cn

European Distributors:

Maple4VoIP ATCOM Authorized Channel Partner
Alsacecom | VoIP Product Reseller - France and EU Distribution.
IPChitChat UK and European Distributor for Atcom cards and Atcom AX1600P
UK Atcom Distributor - Netcloud


See Also



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AXE-1600P

May 27, 2013, 2:30 pm
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  • 4 ports FXO/FXS card

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AXE1600P Asterisk card is the telephony PCI-E card that support up to sixteen FXO and FXS ports .
Using AXE1600P analog card, open source Asterisk PBX and stand alone PC.
users can create their IPPBX telephony solution which include all the sophisticated features of traditional PBX, and extend features such as voicemail in IP PBX. The FXO and FXS modules are interchangeable to suit various requirements.

Features:

    • Support 1-16 FXO/FXS with PCI-EInterface
    • PCI Express interface
    • Support Asterisk ,Freeswitch , Dahdi , Zaptel
    • Support Trixbox , Elastix , Askozia
    • Application: PBX/Voicemail/IVR/Call Center/Call Park/Call Pickup/Call Transfer/Call Forward/Caller ID/Call Waiting/Call Conference
    • Elastix certified

Specations:

  • Configuration
    • Motherboard: AXE1600P
    • Dual channel FXS module: AX-210S
    • Dual channel FXO module: AX-210X

  • Hardware Requirement
    • 500-Mhz Pentium III
    • 64MB RAM
    • PCI-Express interface

  • PCI card dimension
    • 264mm (length) × 121mm (height)

Contact ATCOM:

ATCOM Technology co., LTD.
Address: A2F , Block 3 ,Huangguan Technology Park , #21 Tairan 9th Rd, Chegongmiao , Futian District , Shenzhen China
Tel: +(86)755-23487618
Fax: +(86)755-23485319
E-mail: sales@atcomemail.com
Website: http://www.atcom.cn

European Distributors:

Maple4VoIP ATCOM Authorized Channel Partner
Alsacecom | VoIP Product Reseller - France and EU Distribution.
IPChitChat UK and European Distributor for Atcom cards and Atcom AXE1600P

See Also

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AX-2G4A

May 27, 2013, 2:31 pm
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  • 2 ports GSM and 4 ports analog card

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AX-2G4A Asterisk card is a hybrid card, which supports two GSM channels and 4 FXO/FXS analog ports.
Using AX-2G4A, open source Asterisk PBX and stand alone PC, users can create their IPPBX telephony solution which includes all the sophisticated features of traditional PBX, and extended features such as voicemail in IP PBX.

Features

    • Bulk SMS sending
    • SMS receiving and storage
    • Multi-language for SMS content
    • IMEI changeable
    • PIN-Number unlock for the SIM card
    • Hardware echo cancellation in GSM module
    • Support Elastix 2.0.3, Trixbox 2.8.0.4
    • Monitor GSM signal strength by command and LEDs
    • Monitor communication status by command line
    • Send SMS by Asterisk CLI, and dial-plan
    • 4 GSM quadbandchannels(850/900/1800/1900MHz)
    • DTMF detection
    • Suitable for 3.3 volts and 5.0 volts 32 bit PCI slots
    • GSM connectivity for PBX
    • Mobile PBX
    • GSM VoIP gateway
    • SMS gateway
    • GSM callback services
    • GSM service operation
    • Billing system
    • Elastix certified

Specifications:


  • Configuration
    • Motherboard: AX2G4A
    • GSM module: GX01
    • FXO module: AX210X
    • FXS module: AX210S

  • Hardware Requirement
    • 500-Mhz Pentium III
    • 64MB RAM
    • 3.3V or 5V PCI 2.2 slot

  • PCI Card Dimension
    • 174mm (Length)*104mm (height)


Contact ATCOM:

ATCOM Technology co., LTD.
Address: A2F , Block 3 ,Huangguan Technology Park , #21 Tairan 9th Rd, Chegongmiao , Futian District , Shenzhen China
Tel: +(86)755-23487618
Fax: +(86)755-23485319
E-mail: sales@atcomemail.com
Website: http://www.atcom.cn

European Distributors:

Maple4VoIP ATCOM Authorized Channel Partner
Alsacecom | VoIP Product Reseller - France and EU Distribution.
Neotiq | VoIP Product Reseller and software editor - EU Distribution - support in French and English. see product at this link.
IPChitChat UK and European distributor for
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Sangoma

May 27, 2013, 2:34 pm
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Sangoma's data cards, voice cards, interface boards, telephony cards, gateways and connectivity software are used in leading PBX, IVR, contact center and data-communication applications worldwide. Sangoma has offered premium telephony solutions since 1984.

View Sangoma's Products | Solutions | Warranty
Compatible with Asterisk, Freeswitch, CallWeaver, Yate, 3CX, pbxnsip and others.


Media Gateways

Multiplexers

Transcoding Cards (300–400 or 400–2,000 Sessions)

NetBorder™ Suite for Call Centers

ISDN BRI Cards
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Allo.com

May 27, 2013, 2:36 pm
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Digium

May 27, 2013, 2:38 pm
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2009-07-16_085739.png
Digium®, Inc., the Asterisk® company, is the original creator and primary developer of Asterisk®, the industry's first open source telephony platform. Digium provides hardware and software products, including AsteriskNOW™, the complete open source software appliance; Asterisk Business Edition™, the professional-grade version of Asterisk; and the Asterisk Appliance™, hardware-based telephony solution, to enterprises and telecommunications providers worldwide. Digium also offers a full range of professional services, including consulting, technical support, and custom software development.

Used in combination with Digium's telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium's offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.

History

Mark Spencer founded Linux Support Services in 1999 while still a Computer Engineering student at Auburn University. When faced with the high cost of buying a PBX, Mark simply used his Linux PC and knowledge of C code to write his own! This was the beginning of the world-wide phenomenon known as Asterisk, the open source PBX, and caused Mark to shift his business focus from Linux support to supporting Asterisk and opening up the telecom market. Linux Support Services is now known as Digium, and is bringing open source to the telecom market while gaining a foothold in the telecom industry.

Digium is based in Huntsville, Alabama.

Philosophy


Digium is a young and fast-moving company with an energetic and hip culture that reflects the philosophy of Asterisk and the open source revolution. Mark strongly believes that every technology he creates should be given back to the community. This is why Asterisk is fully open source. Today that model has allowed Asterisk to remain available free of charge, while it has become as robust as the leading and most-expensive PBXs.

Digium was founded on the notion that the customer should have control over the technology that goes into his telecommunications systems, a rare notion in the proprietary world of Telecom. Since the inception of Asterisk, the idea of open source communications technology has been revitalizing an industry which was at one time crippled by the dominance of monolithic dinosaurs.

Digium employees find an added joy in their work knowing that they are pioneering this revolution. The Digium offices are filled with innovative problem-solvers in every department, not just in engineering, because fresh thought has been key to Digium's success from the very beginning.

Website: http://www.digium.com
Store: Digium Store
Sales: http://www.digium.com/en/forms/contact_sales.php
Support: http://www.digium.com/support
Telephone (North America and Worldwide): +1 256 428 6000
Telephone (EMEA): +44 207 183 7577
Facsimile: +1 256 864 0464
Toll Free (North America): 1 877 344 4861 / 1 877 DIGIUM1
IAXTel: 700 428-6000
Skype: digium

Job Openings:


Work@Digium

Products:


Complete List

Hardware

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Openvox

May 27, 2013, 2:41 pm
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http://www.openvox.cn

Manufacturer of asterisk hardware products includes:


Analog Cards:
  • A400P Series 4-port FXO/FXS PCI card (compatible to TDM400P)
  • A400E Series 4-port FXO/FXS PCI Express card
  • A400M Series 4-port FXO/FXS Mini-PCI card
  • A800P Series 8-port FXO/FXS PCI card
  • A800E Series 8-port FXO/FXS PCI Express card
  • A810P Series 8-port FXO/FXS PCI card with Lifetime Warranty
  • A810E Series 8-port FXO/FXS PCI Express card with Lifetime Warranty
  • A1200P Series 12-port FXO/FXS PCI card
  • A1610P Series 16-port FXO/FXS PCI card with Lifetime Warranty
  • A1610E Series 16-port FXO/FXS PCI Express card with Lifetime Warranty
  • A2410P Series 24 -port FXO/FXS PCI card with Lifetime Warranty
  • A2410E Series 24-port FXO/FXS PCI-E card with Lifetime Warranty

ISDN BRI Cards (with Lifetime Warranty):
  • B100M 1-port BRI Mini PCI card
  • B100P 1-port BRI PCI 5V/3.3V Universal Card supports bri-stuff and mISDN
  • B100E 1-port BRI PCI Express card
  • B200P 2-port BRI PCI 5V/3.3V Universal Card supports bri-stuff and mISDN
  • B200E 2-port BRI PCI Express 1.0 card
  • B200M 2-port BRI Mini PCI card
  • B400P 4-port BRI PCI 5V/3.3V Universal Card supports bri-stuff and mISDN
  • B400E 4-port BRI PCI Express 1.0 card
  • B400M 4-port BRI Mini PCI card
  • B800P 8-port BRI PCI 5V/3.3V Universal Card supports bri-stuff and mISDN
  • PFM100 Power Feeding Converter for ISDN BRI Mini Card

T1/E1 PRI Cards:
  • D110P 1-port T1/E1/J1 PCI Card(compatible to TE110P)
  • D110E 1-port T1/E1/J1 PCI Express 1.0 card
  • DE115P 1-port T1/E1/J1 PCI Card with hardware echo cancellation module EC100-32
  • DE115E 1-port T1/E1/J1 PCI Express Card with hardware echo cancellation module EC100-32
  • D210P 2-port T1/E1/J1 PCI Card(compatible to TE210P/TE205P), PCI 5V/3.3V Universal Bus Master
  • DE210P 2-port T1/E1/J1 PCI Card with hardware echo cancellation module EC100-64
  • D410P 4-port T1/E1/J1 PCI Card(compatible to TE410P/TE405P), PCI 5V/3.3V Universal Bus Master
  • DE410P 4-port T1/E1/J1 PCI Card with hardware echo cancellation module EC100-128
  • D210E 2-port T1/E1/J1 PCI Express 1.0 card
  • DE210E 2-port T1/E1/J1 PCI Express Card with hardware echo cancellation module EC100-64
  • D410E 4-port T1/E1/J1 PCI Express 1. ...
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Redfone

May 27, 2013, 2:42 pm
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Redfone Communications is a pioneer and leader in delivering high-performance, high-reliability T1/E1/PRI connectivity solutions. Formed in 2003 in Miami, Florida, Redfone was founded by a team of veteran engineers specializing in the very demanding field of avionics data communications. Today, Redfone's unique engineering pedigree delivers extremely robust, reliable and unique hardware and software solutions to the open-source telephony market, with thousands of satisfied customers worldwide.

Product Line

  • foneBRIDGE2- Single, Dual or Quad T1/E1/PRI to Ethernet Bridge.
  • VGate- a turn key T1/E1 to SIP Gateway VMWare Appliance

foneBRIDGE2 Overview

foneBRIDGE2 T1/E1 Bridge products provide the ability to terminate (or originate) T1/E1 circuits directly into a PC-based server running Asterisk, Asterisk-based distributions such as Elastix, Trixbox, etc., or Freeswitch. foneBRIDGE2 products are a good external alternative to PCI-based cards and to expensive, hard-to-configure Gateways. foneBRIDGE2 also provides for fast, automatic failover from a primary to a backup PBX server, and supports load balancing and clustering capabilities when deployed with clusters of supported PBX servers — all without additional hardware.

foneBRIDGE2 Features

  • Priced similarly to comparable internal PCI cards
  • Eliminates typical PCI card issues such as opening servers, differing PCI standards and voltages, and interrupt conflicts
  • No additional drivers to install - Fully compatible via standard DADHI, TDMoE, and libPRI drivers
  • Compatible with ISDNPRI connections, channel banks, legacy PBX integration, SS7 links and MFC/R2 lines
  • More economical, easier to install, and more reliable than most complicated, expensive T1/E1 Gateways
  • Easy to scale and easy to add high availability, automatic failover, or clustering capabilities - now or in the future
  • Reliable 100% solid state, FPGA-based - High MTBF, no software to crash
  • All products designed, manufactured, and supported in the USA

foneBRIDGE2 Product Portfolio

  • foneBRIDGE2 - 750-4000 - Single Port T1/E1 Bridge
  • foneBRIDGE2 - 750-4000-EC - Single Port T1/E1 Bridge with Echo Cancellation
  • foneBRIDGE2 - 750-5050-EC - Dual Port T1/E1 Bridge
  • foneBRIDGE2 - 750-5050 - Dual Port T1/E1 Bridge with Hardware Echo Cancellation
  • foneBRIDGE2 - 750-5000-EC - Quad Port T1/E1 Bridge
  • foneBRIDGE2 - 750-5000 - Quad Port T1/E1 Bridge with Hardware Echo Cancellation
  • foneBRIDGE2 - 750-1U - 1U Rackmount Kit (supports up to two foneBRIDGE2 products)


VGate Overview

The Redfone VGate is a ©VMWare compatible turn-key software appliance. When combined with foneBRIDGE2 hardware, it quickly and easily provides a complete T1/E1 (ISDN PRI) to SIP gateway with advanced* SIP features including SIP load balancing and auto fail-over to provide always on capabilities.

VGate allows customers to easily deploy Asterisk, Elastix, Trixbox, SwitchVox, 3CX, FreeSWITCH, and other voice systems in a virtual environment and still maintain superior call quality and scalability.
  • Supports all major E1 and T1 ISDN switch types. EUROISDN, AT&T 4ESS,
DMS 100, Lucent 5E, NI1/NI2, ETSI EuroISDN TBR4 PRI, Japan INSnet150. ...
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Rhino Equipment Corp

May 27, 2013, 2:43 pm
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Rhino Equipment Corp

Website: http://www.rhinoequipment.com
Toll-free: 1-877-RHINO-T1 (744-6681)
Phone: +1 (480) 621-4000
Fax: +1 (480) 961-1826
FWD: 633686
Tech-support: Knowledge Base & Trouble Tickets
Sales: sales@rhinoequipment.com
Address: 8240 S. Kyrene Rd.
Suite 107
Tempe, Arizona 85284
USA


Products


CB24 (Channel Bank)


Image
Website

Rhino Equipment is the manufacturer of the ever-popular Rhino Channel Bank.
  • Rhino Channel Banks are fully compatible with not only Asterisk but any standard T1 equipment. They support either 24 FXS or FXO channels, or a combination with their modular bank. In addition to 24 analog channels, they support V.35 data on channelized T1s. The units are quickly configured using AutoT1™ through their 4x40 LCD that displays real-time status of the unit, T1, and analog channels. The units support either 110V, 220V, or -48V input power and with their custom brackets are both rack-mountable (2U) in 19" and 23" racks, and wall-mountable. All neccesary cables and documentation are included.
  • There are currently 3 models: a 24 channel FXS, a 24 channel FXO, and a modular unit with each card incrementing either 4 FXS or FXO channels. All work great with Asterisk and are quite popular due to their easy, self-configuration. If you get stuck, their technical support is more than willing to help, no matter where you purchased from.
  • It is important to make sure you are running the latest firmware as some distributors will buy the banks and store them for many months while in the meantime Rhino has made improvements or fixes in subsequent firmware. The upgrade process is painless and their support is more than willing to talk you through it if you need assistance.

Ceros, Ceros Desktop and Ceros 1U


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IPKall

May 28, 2013, 8:51 am
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http://www.ipkall.com/
http://www.kallfree.com/

Free personal Washington State (USA) PSTN number that forwards to any SIP or IAX destination you specify, including your own Asterisk server.


Support



Instructions on setting up a direct incoming IPKall phone number under Asterisk, bypassing FWD


First, register an IPKall number at either of the web addresses provided above.
IPKall will ask for the following information:

SIP phone number
SIP proxy
Email Address
Password (4 digit PIN)
Voicemail preferences

When specifying the SIP phone number, you can use any number, but you should try to avoid using
a number that's already an extension in your extensions.conf file.
Put the IP address (or hostname) of your Asterisk server into the SIP proxy field.
If your SIP proxy is not running on the default port 5060, then you will need to specify the port.
For example, if your IP is 123.254.254.1 and your SIP proxy is running on port 7777,
then you would enter 123.254.254.1:7777 as the SIP proxy. Be sure to enter a valid e-mail address.

Here is a sample IPKall configuration for sip.conf and extensions.conf.
Place this under [inbound] in extensions.conf. (note, if [inbound] already exists, don't add it again):
important note: your incoming context may be, and probably is different than [inbound].
Replace [inbound] with the appropriate context in both extensions.conf and the "context=" line below.

[inbound]
exten => 508,1,Goto(your-main-menu|s|1)

Where 508 is the SIP phone number you specified when setting up IPKall.
Now, put the following information in your sip.conf file:

[508] ;IPKall
type=peer
dtmfmode=rfc2833
context=inbound
insecure=very
host=voiper.ipkall.com
nat=no


How it works

1. When you dial your IPKall DID, IPKall sends a request (i.e. 508@123.254.254.1) to your SIP proxy
2. Asterisk accepts the request, since the [508] context in sip.conf tells Asterisk to accept incoming calls with the number "508"
3. Asterisk searches extensions.conf for the "inbound" context (since we specified context=inbound in the [508] context)
4. Asterisk then matches extension "508" in the [inbound] context, jumping to "your-main-menu" in this example.


Codecs Supported

G.711
GSM
iLIBC
G.729 (since Feb 2006)

Troubleshooting

If you just get a busy signal when calling your IPKall number, try the following:

1. From the Asterisk CLI (asterisk -r) turn on sip debugging by typing "sip debug", try calling your IPKall number again, when you get the busy signal, turn off sip debugging by typing "sip no debug". Then read through the debugging information for clues.

2. Make sure host=voiper.ipkall. ...
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Asterisk Consultants Austria

May 28, 2013, 12:35 pm
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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. ...
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