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  • 05/03/15--04:47: Asterisk
  • Image

    Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows (emulated) and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

    Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. Check the Features section for a more complete list.

    Asterisk needs no additional hardware for Voice-over-IP, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism (for certain applications such as conferencing). A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers)


    For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsor, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks . In addition, single to quad port analog FXO and FXS cards are available and are popular for small installations. ...

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  • 05/03/15--15:25: CyberData
  • www.CyberData.net

    Products:

    ceilingspkr.png
    Ceiling Speaker V2.0
    indoorintercom.png
    Cyberdata Intercom
    intercom2.png
    Cyberdata Intercom





    Cyberdata PoE VoIP Paging Gateway

    Cyberdata VOIP Gateway


    The CyberData SIP-enabled Paging Gateway allows the use of legacy analog zone paging amplifiers when converting to a VOIP system. The Gateway is compatible with most SIP-based IP/PBX servers that comply with the SIP RFC 3261. The Gateway is powered via PoE 802.3af - no external power supply is needed. ~ description via website


    Notes on setup in FreePBX using a SIP Extension:
    Set Qualify = no to get it to work.

    WAudette...

    For more information contact;
    Bill Majerczak
    billm@cyberdata.net
    831-373-2601 x-102

    Cyberdata PoE VoIP Speaker


    An overhead paging speaker that connects direct to Ethernet.

    From the Data Sheet:

    The CyberData PoE VoIP Speaker is a
    Power-over-Ethernet (PoE) and
    Voice-over-IP (VoIP) public address
    loudspeaker that easily connects into
    existing local area networks with a single
    cable connection. The speaker is powered
    via a standard ethernet cable – no external
    power supply is needed. With its small
    footprint and low height, the speaker can be
    discreetly mounted almost anywhere.


    New in 2009 - VoIP Talk-Back Speaker

    ^
    The CyberData SIP-enabled Talk-Back Speakers is a Power-over-Ethernet (PoE802.3af) and Voice-over-IP (VoIP) public address two-way loudspeaker that is easily connects into existing local area network with a single CAT5 cable connection. The speaker is compatible with most SIP-based IP PBX solutions. In a non-SIP environment, the speaker is capable of broadcasting audio through a multicast. It’s small footprint and low height allows the speaker to be discretely mounted almost anywhere. Speakers can be mounted into existing ceilings or in one of our enclosure kits. Enclosure kits offered are beveled wall mount adapters and NEW in 2009 our wall mount adapter with a digital clock perfect for schools.


    See Also



    Gentek
    Westcon
    888VoIPstore
    Neobits
    Anixter
    VoiPmodo.com

    New since last update

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    PORTech Communications lnc.

    PORTech is the original manufacturer in Taiwan with 30 years; therefore we’ll do our best to offer the best service and support.
    http://www.portech.com.tw

    PORTech offer
    • E1 /T1 GSM Channel bank
    • VoIP GSM Channel bank
    • 1/2/4/8 Ports VoIP GSM Gateway /3G UMTS Gateway/CDMA Gateway
    • 2 ports GSM VoIP PCI Card
    • SIM Server
    • SBK-32:Remote SIM Bank
    • Free Roaming Gateway
    • VoIP Adapter
    • 1/4/8 ports GSM Fixed Wireless Terminals(Follow Me GSM Gateway)
    • Skype Gateway
    • IP/PSTN Power Switch
    • Worldwide use ( 2G,3G with all world and Japan,CDMA 2000)
    http://www.portech.com.tw

    VoIP GSM / CDMA / UMTS Gateway


    Portech MV-370 - 1 channel VoIP GSM Gateway

    MV-370 : 1 channel VoIP GSM Gateway
    Image

    Look at the MV-370 configuration howto

    Portech FTA-102L - Free Roaming Gateway

    fta-102L_200.jpg

    When you go abroad with FTA-102L, all incoming calls of your original mobile phone are free of charge. FTA-102L provide free roaming fee for international inbound calls. When you are on business travel oversea, your coworkers, friends, family, and clients can just call your original mobile phone number to reach you. This is of enormous benefit for user, you don’t’ need to use others country phone access number. Also, it’s simply and easy to send from your original mobile phone while traveling/abroad.
    FTA-102L must connect with MV-370.

    PORTech PG-101 VoIP adapter ATA with FXS, FXO interface

    ATA-200.jpg

    PG-101 VoIP adapter ATA with FXS, FXO
    PG-101 can connect MV-370 VoIP GSM Gateway:
    User can easily carry away to receive calls from your original mobile phone and make outgoing calls to home country with original rate while traveling/abroad.
    No more international call charge. And no miss calls while going abroad
    User can eliminate international call charge and roaming fee.


    Portech MV-372 - 2 channels VoIP GSM Gateway

    MV-372 : 2 channels VoIP GSM Gateway
    Image

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  • 05/04/15--02:27: VOIP Billing
  • Hosted Billing Services (in Alphabetical Order)


    • Incorpus TeleNetworks Incorpus provides Class 5 and Class 6 softswitches on very affordable monthly rental plans. These are carrier grade switches suitable for Big Enterprises as well as small companies and individuals who are trying to build their own voip company. All switches comes with strong firewalls and bandwidth optimizer and the plans start from as low as just 80$ monthly.Please visit our website and have a live chat with our sales team for any guidance you need. email us for more information at sales@incorpus.in or info@incorpus.in
    • Zoom Soft's dedicated easy VoIP billing server provides services for VoIP solution from small to cluster solution for handling thousands of concurrent solution. Our VoIPSwitch is class 5 softswitch which is integrated with billing VoIP Switch. This system also integrated with high features and facilities. Our billing server started with only $50 monthly with 24/7 support. Directly order could be possible with our VoIP billing portal system.
    • CloudAstrix SPE CloudAstrix SPE is such a VoIP Switch. Build on the world renowned WHMCS Billing Suite, the Soft-switch module brings all necessary functions to perform and provide a top class VoIP service.As a Carrier Neutral soft-switch, CloudAstrix has already proven to be a firm favourite among ISPs all over the world.
      • Note:CloudAstrix SPE Module works with FreeSwitch.
    • Adore VoIP Billing Adore VoIP Billing Software comes with the enhanced functionality along with the architecture with class. It is fully compatible and gets integrated with all other VoIP related products. It is designed with all the present and future demands of booming telecom industry kept in the mind. The telecom industry is changing and developing with rapid speed and the VoIP products such as VoIP Billing comes as an excellent product in this time.
    • 4PSA VoipNow fully featured, carrier-grade, multi-tenant edition for service providers and businesses, that can be installed on their chosen infrastructure or delivered as a UCaaS. VoipNow provides a fast, competitively priced go-to-market solution, from deployment and provisioning all the way to selling and billing.
    • A2BILLING - VoIP Billing Solution / AAA / Class 5 Softswitch.
    • Adore All-in-One SIP Server and Client v2.2.1 - new released with Class5 features
    • Aradial AAA for Billing Solutions
    • benotos offers free callshop billing system 4-level billing system: reseller-subreseller-callshop-customer, 2 different routes, nice easy to use interface, intelligent ratemanager, online payment, detailled reports, receipt printing with own logo, white labelled, use your own brand and domain name and much more features. About 9000 callshops around the world are using our excellent callshop billing solution already. ...

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    Here is a list of producers of ready made, black box PBXs that are based on Asterisk (in no particular order):


    EBSolution - Custom IT Solutions

    • Web Site: http://www.ebsolution.ca
    • Email: mailto:info@ebsolution.ca
    • Location: Toronto, Canada
    • Phone Number: +1.905.695.5485
    • Type of Support: Telecommunications, VoIP, Cisco Networking, Asterisk, Asterisk@Home / Trixbox, Elastix and more
    • Hourly and Contract Support, Consulting, Hosting, Office PBX, etc.
    • Residential and commercial phone and internet services.

    Wayatone Media Inc. - Communication

    • Web Site: http://www.wayatone.com
    • Email: mailto:info@wayatone.com
    • Location: Toronto, Canada
    • Phone Number: +1-647-247-8004
    • Type of Support: Telecommunications, VoIP, Networking, Asterisk, Asterisk@Home / Trixbox, Elastix and more
    • Hourly and Contract Support, Consulting, Hosting, Office PBX, etc.
    • Residential and commercial phone and internet services.


    Svanto.net - Tailored Internet telephony solutions

    VoIP provider for residential, wholesales and business. Providing international DID's via high quality SIP Trunks, IP PBX (WSP) and hosted PBX (OpenPBX)

    AAB Asterisk Consultant Lithonia GA

    • 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

    4PSA VoipNow

    4PSA VoipNow is a fully featured, carrier-grade, multi-tenant edition for service providers and businesses, that can be installed on their chosen infrastructure or delivered as a UCaaS. VoipNow provides a fast, competitively priced go-to-market solution, from deployment and provisioning all the way to selling and billing.


    Alpha Computer Group - IT and Telecommunication Engineers

    • Web Site: http://www.AlphaComputerGroup.com
    • Location: Long Island, New York
    • On-Service Areas: New York, New Jersey, Connecticut, Los Angeles, San Diego, Arizona
    • Phone Number: +1-877-608-8647
    • Type of Support: Telephone Systems, VoIP, Computer Repair, Virtualization, Networking, Security Cameras, Alarm Systems, Access Control, Asterisk, Freepbx, Elastix and anything to do with Technology
    • Hourly Rates and Maintenance Contracts available
    • IT and Unified Communication Solutions for the home and office

    ansit-com GmbH

    http://www.ansit-com. ...

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  • 05/04/15--05:12: New Software Releases
  • This page is to inform on various VoIP related software releases.

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


    April 2015


    March 2015

    • 2015-03-25 -

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  • 05/04/15--05:17: ICTFAX
  • Release Note


    Released new version of ICTFAX Ver 3.2 on April 15, 2015 based on ICTCore, a new communicatiosn framework, fixing reported issues

    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
    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

    .

    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

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    ICT FAX, a unique open source faxing solution

    News


    Released new version of ICTFAX Ver 3.2 on April 15, 2015 based on ICTCore, a new communicatiosn framework, fixing reported issues

    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 2.0 beta on Jun 15, 2012 , complete rewritten in Drupal 7.0 and ported to use Plivo Communication Framework using Freeswitch as communication engine as backend instead of Asterisk , Old version of ICTFAX was based on Drupal 4.7 and was not compatible with PHP 5.3 causing compatibility issues those are now fixed

    We are pleased to announce that ICTFAX Version 3.0 is released. New release completely removes Plivo Framework from ICTFAX. Now ICTFAX no longer depends on Plivo for communication with FreeSWITCH. Instead, ICTCore has been introduced as a new lightweight communication library. ICTFAX uses ICTCore to communicate with FreeSWITCH. Apart from this major change, other features that are included in this release includes GUI based trunk provider configuration, multiple trunks allowed, documentation converted to markdown syntax, attachement file name with spaces issue has been fixed, error on retry issue fixed along with other minor bugs.

    ICTFAX


    ICT FAX is an open source (GPL v 3), multi-user and web based business solution with advance billing capabilities featuring duration as well as per unite billing , ICTFAX is an email to fax gateway, supports G.711 faxing , PSTN faxing and T.38 origination and termination .ICTFAX is complete faxing solution and does not need to be integerated with other open source projects to function that makes ICTFAX a unique and innovative faxing solution .

    ICTFAX, a Faxing solution

    ICT FAX can be used in following faxing scenarios

    Email to fax / web to fax / fax to email
    G.711 Origination / Termination / Gateway
    T.38 Origination / Termination
    PSTN Origination / Termination / Gateway

    please visit http://www.ictfax.org for more information




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  • 05/04/15--05:27: ICTDialer
  • News

    ICT Innovations Released new version of ICTDialer Version 2.0 on April 15, 2015 based on ICTCore, a new communicatiosn framework, fixing reported issues


    ICTDialer Is an Open Source Unified Communications auto dialer Software

    ICTDialer is an open source Unified Communications marketing Software. ICTDialer is multi-tenant with Voice, SMS & Fax broadcasting capabilities developed over re-known open source Content Management System Drupal and Freeswitch based powerful ICTCore Communication Framework . It can be scaled to blast thousands of simultaneous calls using either VoIP, Foip or PSTN. ICTDialer capable to fit in many broadcasting and telemarketing scenarios. It empowers user with capabilities of Drupal CMS and ICTCore Communication Framework

    ICTDialer is released as Open Source GNU AGPLv3

    ICTDialer is developed and promoted by ICT Innovations with in-depth experience in open source ICT's , Unified Communications and Telemarketing technologies.


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  • 05/04/15--17:09: voip-info.org
  • 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.


    NEW


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  • 05/04/15--20:21: Yeastar - NeoGate
  • Yeastar TG100 - VoIP GSM/CDMA/UMTS Gateway

    Reduce costs for SOHO and SMBs

    Yeastar TG100 is a fully featured 1 port VoIP GSM/CDMA/UMTS gateway that provides GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It significantly reduces the costs of calls with two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. With friendly GUI, everything can be easily set up.

    TG100侧面副本副本.png


    Benefits

    1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
    2) Back up - Work as a cost-effective backup when the landline network goes down.
    3) Easy to install - Everything can be easily set up in the Web based management interface.
    4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

    Specification:

    Number of GSM/CDMA/UMTS channels (Max): 1
    GSM Network type: 850/900/1800/1900MHz
    CDMA Network type: 800MHz
    UMTS Network type: 900/2100MHz or 850(800)/2100MHz or 850/900/1900MHz or 850/900/1800/1900MHz
    Protocol: SIP (RFC3261),IAX2
    Transport Protocol: UDP,TCP,TLS,SRTP
    DTMF: RFC2833, SIP INFO, In-band
    Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
    Echo Cancellation: ITU-T G.168 LEC

    LAN: 1 (10/100Mbps)
    Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
    QoS, OpenVPN
    NAT Traversal: Static NAT, STUN

    Size: 110x70x24mm
    Power Supply: AC 100~240V 50/60Hz (DC 12V, 1A)

    Operation Range: 0° to 40°C, 32° to 104° F
    Storage Range: -20° to 65°C, -4° to 149° F
    Humidity: 10-90% non-condensing

    Features:

    SIP Server and SIP Trunk supported
    SIP Peer Mode supported
    Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
    GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
    Incoming /Outgoing Routing rules
    SMS Sending and Receiving
    Send Bulk SMS
    Gain Adjustment
    USSD
    PIN Modify
    Carrier Selection: Auto/Manual
    Balance Alarm
    Caller ID/CLIR
    Black List
    Hotline
    Call Duration Limitation
    Call Transfer
    Call Back
    Call Status Display
    Call Detail Record (CDR)
    Call Progress Tone Generation
    Call Duration Limitation for SIM Card/Single Call
    LCR (Least Cost Routing)
    Top voice quality (EFR super sound)
    SIP Response Code Switch
    Open API for SMS and USSD
    Real Open API Protocol (Based on Asterisk)
    IP Blacklist
    Network Attack Alert
    System Logs
    Web based configuration



    Yeastar TG200 - VoIP GSM/CDMA/UMTS Gateway

    Reduce costs for SOHO and SMBs

    Yeastar TG200 is a VoIP GSM/CDMA/UMTS gateway with 2 channels providing GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It supports two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. Thus the calls costs could be significantly reduced by VoIP or GSM/CDMA/UMTS network.

    TG200正面副本副本.png


    Benefits

    1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
    2) Back up - Work as a cost-effective backup when the landline network goes down.
    3) Easy to install - Everything can be easily set up in the Web based management interface.
    4) Easy to integrate - High compatibility with major IP PBX and softswitch brands. ...

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  • 05/05/15--02:29: Asterisk Consultants Germany
  • Asterisk consultants: Germany


    Add your entry here (Alphabetical order by country and company):
    This page is growing large. Please don't post logos!!
    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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  • 05/05/15--09:57: Old News
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  • 05/05/15--12:27: Voicepulse
  • 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.

    To view a demo of the VoicePulse FIVE portal, click here.

    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

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  • 05/05/15--13:38: Virtual PBX
  • Virtual PBX is a budget-friendly form of hosted VoIP (Voice over Internet Protocol) that usually only handles inbound calls. A virtual PBX is typically intended for small business VoIP customers with fewer than 10 employees and low-volume telephone traffic.


    What Is Virtual PBX?

    A virtual PBX is an economy-class version of hosted PBX. Hosted and virtual PBX systems are business VoIP PBX phone systems that transmit calls over the Internet as data.

    A virtual PBX offers inexpensive business VoIP telephone service to small businesses. As with a hosted PBX phone systems, a virtual PBX is owned and maintained off-site by a VoIP service provider. A virtual PBX enables a small business telephone system to access enterprise-level features such as auto attendants and voicemail. With virtual PBX small business telephone systems, small start-ups, mom-and-pop shops, freelancers, and entrepreneurs can all present a professional image to vendors, investors, and customers.

    Depending on the service provider, a virtual PBX phone system may require a separate phone service for outbound calls.

    Virtual PBX Features


    Virtual PBX phone systems offer lower costs and fewer features than hosted PBX phone services. Compared to hosted PBX small business telephone systems, virtual PBX service is limited to the most basic fundamentals of business-oriented call controls. Virtual PBX is geared toward simple inbound call-routing for SoHo offices with few personnel, small budgets, and limited calling needs. As with many hosted PBX calling services, most virtual PBX phone systems do not require a contract or term commitment.

    Standard features offered with most virtual PBX plans are:

    • Voicemail
    • Auto attendant
    • Unlimited call handling (no busy signals)
    • Call forwarding

    Limitations


    Virtual PBX phone systems generally:

    • Handle only inbound calls
    • Offer a limited number of extensions
    • May not include Fax over IP (FoIP) services
    • Include a set amount of free minutes
    • May not offer voicemail-to-email
    • May not include international long-distance coverage
    • May not offer Internet fax service
    • May charge extra for conference calling

    The features offered vary by virtual PBX VoIP provider. As VoIP service becomes a more common solution for small business telephone systems, many virtual PBX plan features are incorporating the more advanced features of hosted PBX phone systems. Compare plans and prices to determine the best virtual PBX solution.

    Cost

    Virtual PBX phone prices depend on a variety of factors, such as the features included. Virtual PBX phone service plans can start as low as $9.95 (Grasshopper) per month.

    Virtual PBX Service Providers


    Some virtual PBX providers include:

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  • 05/05/15--13:54: VoIP Hardware
  • This page lists information about VoIP hardware and VoIP hardware products. For phones and hardware to use with Asterisk, including VoIP phones (both hard and soft phones) and Analog Telephone Adapters, see Asterisk phones.

    PSTN Interface cards (analog, GSM, ISDN-PRI and R2/MFC)


    This section contains VoIP hardware for connecting analog or digital phone lines from the Public Switched Telephone Network to your Asterisk server. Please keep VoIP hardware providers in alphabetical order.



    2-Day Direct

    • Cisco SPA303 3-line business-class IP phone; Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
    • Cisco SPA504G Full-featured 4-line business-class VoIP phone supporting Power over Ethernet (PoE)
    • Cisco SPA525G2 5-Line Business IP Phone with Enhanced Connectivity and Media for a New Level of Small Business User Experience; includes wifi and bluetooth connectivity
    • Cisco SPA514G Advanced, Affordable, Feature-Rich VoIP Phone for Business or a Home Office; Full-featured 4-line business-class VoIP phone supporting Power over Ethernet (PoE)
    • Cisco SPA508G Full-featured 8-line business-class IP phone supporting Power over Ethernet (PoE); Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)

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  • 05/05/15--14:11: SIP Trunk Providers India
  • This page is a list of SIP trunk providers in India. Please keep this list in alphabetical order. India SIP providers looking to add their services can do so in the list below.

    • ALTOTELECOM - 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

    • Incorpus TeleNetworks Your one stop solution to all voip needs, Call termination, resellers, DIDs, Softswitches, Tollfree, residentail voip, Call center voip.
    Contact our executive today for more details visit our site and have a live chat or mail us at info@incorpus.in

    • AVOXI AVOXI Virtual Call Center Solutions - VoIP Service Provider, provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers. Contact Number 1-800-462-8694.

    • CallForwarding - Be present anywhere in the world with toll free forwarding services from CallForwarding.com. Contact Number: 800-231-9802

    • i7 Solutions - VOIP Wholesale and Retail i7 Solutions is looking for VoIP Resellers in markets all around the world. We provide the Dialers and Rates your customers want, and the tools for you to control all aspects of your business. i7 Solutions is a one of the very few global VoIP solution providers with direct contracts with local incumbent telecommunications operations across the globe. Our primary business includes A2Z termination services, retails & reseller solutions for VoIP services and also providing calling cards. India TATA CLI is our premium route. Please contact us @ Gtalk and Email ID info@i7solutions.in

    • UtterU UtterU is leading online voice calling business service provider. UtterU provide Wholesale option for reseller to start international calling business around the world. UtterU also provide SIP trunk facility for all business to making call over the internet at very low cost.

    • http://www.doorVaani.com DoorVaani.com is a Business, Residential and Personal VOIP services provider offering VOIP Call Minutes to destinations worldwide, Local Phone Numbers or DIDs in about 60 countries and Toll-free numbers in 5 countries. DoorVaani.com is also an automated web application where ordering of services, making payments and provisioning is done all in real time with no wait time or human interaction needed. Online payments are accepted in 25 currencies including Indian Rupees (INR). Offline payments or Bank Transfers accepted in INR and USD.



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  • 05/05/15--17:48: VOIP GSM Gateways
  • What's a VoIP GSM Gateway?

    A VoIP GSM Gateway enables direct routing between IP, digital, analog and GSM networks. With these devices (fixed cellular terminals) companies can significantly reduce the money they spend on telephony, gp-especially the money they spend on calls from IP to GSM. The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).

    Through least cost routing the gateways select the most cost-effective telephone connection. They check the number which is dialed as well as rate information which is stored in an internal routing table. Because several SIM cards and GSM modules are integrated within the VOIP GSM Gateway it is able to make relatively cheaper GSM to GSM calls instead of expensive IP to GSM calls.

    Who offers VoIP GSM Gateways?



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  • 05/05/15--17:51: CDMA
  • CDMA Gateways from Hypermedia Systems


    VOIP CDMA GATEWAY

    The HyperGateway VoIP CDMA Gateway line of products provides a tailor-made solution for alternative carrier call termination and for call origination providers.
    The VoIP CDMA Gateway is a complementary product to IP based equipments and supports up to 72 CDMA ports. By using HG-4000 there is no need for any external VoIP Gateways such as Cisco or Quintum.

    VoIP CDMA Gateway main features:

    • From 16 to 72 CDMA ports
    • Advanced LCR and routing groups
    • IP address and DDI pattern restrictions
    • Remote pre-paid SIM recharge
    • Integrated antenna splitter
    • CDMA worldwide use
    • Redounded power supply - Option
    • WEB management and control


    CDMA RUIM-Enabled Gateways NEW

    The new CDMA RUIM Gateways are professional, heavy duty CDMA Gateways that offer great new business opportunities. When connected to alternative carrier's equipment/networks, these CDMA Gateways increase profits for call termination vendors.

    Featuring seamless VOIP/PRI CDMA interoperability, service providers and system integrators can deploy total end-to-end IP telephony solutions to end-users, and alternative carriers can take advantage of the lower cost minute-based allotments offered by different RUIM cards.

    Available:
    VoIP CDMA Gateway HG-4000C series - supports up to 72 CDMA ports with RUIM cards
    PRI CDMA Gateway HG-3000C series – supports up to 30 (+2) CDMA ports with RUIM cards

    Major Advantages:

    • Fast ROI
    • Practical, easy to implement
    • Flexible and scalable
    • Value-added product
    • Reliable and proven technology
    • Compatible with PBX / IP-PBX systems of leading manufacturers worldwide
    • Online technical support and training
    • One point of contact for customer service
    • 24 months warranty


    CDMA E1/T1 PRI GATEWAY for PBX

    The Hypermedia Systems PRI CDMA Gateway provides superior voice technology for connecting PBX systems and IP-based systems with the cellular networks.

    Main Features:
    • Up to 30+2 cellular ports
    • CDMA worldwide use (850/1900)
    • Comply with Cisco, Quintum, Asterisk and others
    • Integrated antennas combiner
    • WEB Remote management
    • High quality of voice
    • VoIP - Option
    • Fast and easy installation


    Other related products:
    3G VoIP Gateway
    3G UMTS Analog Gateway
    GSM Analog Gateway


    For more information:

    Contact:

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  • 05/05/15--17:59: GSM
  • GSM stands for Global System for Mobile Communications. It is the dominant mobile phone comminucation protocol in Europe and also popular elsewere. The name is also used to describe some of the voice codecs used in the GSM standards.

    This page regards connecting an IP-based phone system directly to the GSM cellular network in addition to connecting a VoIP PBX to the GSM network & PSTN; all-in-one-box. For more information about the codec, see the GSM codec page.

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