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  • 07/20/15--00:50: VOIP Event Calendar
  • 2015 VOIP Related Event:

    August 2015

    • 11 - 13 The Prepaid Press Expo 2015 - The ttp EXPO'15 is a very popular event that focuses on prepaid products and services. All prepaid service providers around the world are expected to attend this event which offers business networking sessions for striking new business opportunities. The tpp EXPO'15 is scheduled from August 11 – 13, 2015 at Planet Hollywood, Las Vegas.

    June 2015

    May 2015

    April 2015

    March 2015

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    What is CTMS?

    CTMS:abbreviation of Cloud Telephony Management System, is the perfect solution. CTMS consists of three parts: CTMC+ CTN+ IP Phone.

    CTMC: (Cloud Telephony Management Center) is a network node management center which has been independently developed by ZYCOO, and can be utilized by VoIP service providers and enterprises users to manage multiple CooVox CTNs (Cloud Telephony Nodes). CTMC provides a multitude of features for CTN including auto-provision, software/firmware upgrade management, status/performance monitoring, warning log diagnosis etc. CMTC is a powerful solution that delivers the features and functionality required to manage and maintain a highly dispersed telephony environment through the use of a single centralized management system.

    CTN: (Cloud Telephony Node) is the node of cloud telephony management system and handling the switch of telephony communication. ZYCOO CooVox series IP PBX can be taken as the node after upgrading. CTN is allowed for branch’s administrator to configure the local network connections.

    IP Phone:which support SIP protocol can be used in CTMS; especially the phones support auto-provision with ZYCOO CooVox Series IP PBX; all the phones located in different CTN are allowed to be auto-provisioned via CTMC directly.

    Benefits & Features:

    1. Centralized configuration and upgrading of CTNs
    2. Monitor system information, configuration and service status
    3. View and backup of system log, operation log and call log
    4. Manage multi-service and user groups based on template
    5. Manage configuration for individual or multiple devices based on user groups
    6. Flexible upgrading control strategy allowing for convenient software and firmware upgrades
    7. Based on TR069 protocol, allowing nodes to pass through private networks
    8. Adopting B/S managing mode to achieve multi-language GUI, humanized management process, and easy operation
    9. Based on Linux which ensures the device is secure and reliable
    10. Password change supported and license authentication available

    Contact us:

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

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

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


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  • 07/20/15--01:54: Paging Management System
  • QQ截图20150717111946.jpg

    Paging Management System is an innovative software which is primarily used to distribute public address via sip protocol over SIP Speakers. According to the simple interconnection with CooVox IP PBX, multiple SIP Speakers will be centrally managed, monitored and configured through paging management system.

    Paging Management System allows you to make announcement with specific voice file at the specific time/date. And all the announcement logs are available to check from the paging management system. Moreover, Paging Management System is easily expandable to add more SIP Speakers to the network.

    The SIP based audio system iSpeaker utilizes the built-in intercom and paging capability already inherent in most modern IPPBX systems and enhances this to improve end user experience by providing a dedicated high performance digital amplifier on which to distribute announcements or play background music. There are two models now: iSpeaker B20 and iSpeaker C20.

    Key Features

    1. Volume Control
    2. Group announcement
    3. Real-time announcement
    4. Public Address for specified area
    5. Public Address in specified time
    6. Remote settings /management/ upgrade
    7. Centralized management of audio files
    8. Centralized configuration of SIP speakers
    9. Real-time and remote monitor speakers status
    10. Friendly Web interface and easy to use
    11. Paging to up to 500 SIP Speakers
    12. Custom service is available (e.g.: emergency announcement)

    Paging Management System Application
    Applicable in mall, factory, office, school...


    Contact us:

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

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

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


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    No License Fee
    Advanced IP PBX features
    Automatic Call Distribution(ACD)
    Call Detail Records(CDR)
    Fax to Email, Email to Fax
    Voicemail to Email
    Industry SIP Trunk and Digital Trunks Supported
    Firewall Intrusion Detection
    HD Voice(G.722.) Supported

    Applicable Trunks: FXO/ FXS/ GSM/ BRI/ E1/T1

    Software: Developed based on OpenSource Asterisk 1.8.x

    Hardware: Dualcore CPU; Modular Design (flexbile to change the module)

    CooVox Series IP PBXs come in four sizes: U20 / U50 / U60 / U100

    CooVox-U20: Support 30 Ext. Users, Max. 10 Concurrent Calls
    SDRAM 128MB DDR2; Memory 4GB SD Card

    CooVox-U50: Support 100 Ext. Users, Max. 15 Concurrent Calls
    SDRAM 256MB DDR2; Memory 4GB SD Card

    CooVox-U60: Support 200 Ext. Users, Max. 80 Concurrent Calls
    SDRAM 1GB DDR3; Memory 32GB SSD

    CooVox-U100: Support 500 Ext. Users, Max. 80 Concurrent Calls
    SDRAM 2GB DDR3; Memory 500GB HD



    Contact us:

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

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

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


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  • 07/20/15--05:53:
  • 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.


    News Resources

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  • 07/20/15--12:32: Asterisk consultants USA
  • This is a comprehensive list of Asterisk consultants in the USA (United States). Add your entry here (alphabetical order, by state and company), but stick to states where you have actual presence!

    Feel free to add a few lines (max 5) describing your business. Don't forget to add VoIP telephone numbers, like a SIP URI. Use common courtesy with others' entries! No images!


    Asteria Solutions Group

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  • 07/20/15--13:25: Toll Free
  • To receive calls to USA and Canada TollFree numbers (800, 866, 877, 888, 855), you will need and Toll Free Number service provider.
    To send calls to USA and Canada TollFree numbers you can either send these calls to your usual long distance service provider (who may charge you by the minute) or send the calls to one of these free: Toll Free Termination Providers. If you have a large volume of calls to tollfree numbers, some Toll Free Termination Providers will pay you for the calls.

    Toll free (800, 866, 877, 888, 855) phone number providers:

    (In alphabetical order):
    • Alcazar Networks Inc is offering Wholesale VoIP Origination / Termination / Free Toll Free Termination - Get paid for your toll free traffic! Wholesale SIP
    • Anveo offers USA, Canadian and international Toll-Free phone numbers. Anveo provides web based Visual Call Flow Designer where you can easily create custom IVR Call Flow for your business. Anveo has many exceptional features including click to call, ringing multiple phones, call transfers to Skype or Gizmo5, conference calls, online FAX, voicemail, extensions, Business Control Panel to create and manage employee's sub accounts and more!
    • BinFone Telecom offers toll free service, delivered via SIP and IAX.
    • ComCanada CommunicationsCRTC Registered, Provider of Hosted PBX, DID service, Toll Free, equipment sales, consulting, and retail/wholesale origination & termination. Supported Protocols: SIP & IAX 1-877-697-VOIP
    • Conexiant Wholesale VoIP – direct connections to CLECs for best quality, inc. Level3.
      • Global Level 3 footprint Denver, LA, Hong Kong, China and Frankfurt, Germany
      • DIDs, Toll Free, E911 service, T.38,VoIP fax, quality termination and dedicated VoIP servers
      • DIDs per-channel or per-minute
    • DIDx.NET Buy Toll Free DID numbers at wholesale in more than 100+ countries and its cities from DIDx, The Wholesale DID Exchange Platform for DIDs.
    • Database Systems Corp.Toll Free Phone Services - IVR toll free services. Products include voice broadcasting, IVR systems, IVR software and IVR hosting services.
    • Easy Office Phone - Toll free services delivered via SIP or with our Hosted PBX platform.
    • Flowroute LLCWholesale VoIP, A-Z SIP Termination, Cheap DIDs, Toll-Free Origination, Free CNAM Storage, E911, T. ...

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    Toll-free termination are calls destined to 8YY destinations. These providers allow you to terminate toll-free calls from the US and Canada for free in some cases. If you have a large volume of calls to toll-free numbers, some providers may pay you for your calls. Carriers who have direct agreements have a higher success in collections. Calls must originate from your network to the Carrier bound to a Toll Free Number.

    There are several elements to a Toll Free Call.
    1) End User dials a Toll Free Number. (Voip or TDM)
    2) End User Provider Network must route this call. (End office elements)
    3) Resellers of this traffic must hand this call off to a TANDEM provider.
    4) Tandem Provider will DIP SMS800 for CIC instructions on every TFN. (see HyperCube)
    5) CIC (IXC) will receive traffic and route to RespOrg
    6) RespOrgs are the Responsible Organizations of Every Toll Free Number.
    7) End User / Owner of the Toll Free Number. (Final destination)

    Without registration required

    Tollfreedollars is now offering $.002 per minute for your toll free traffic. No monthly minimums.
    To receive the highest payout a contract is required.

    • HyperCube Toll Free termination ****WHOLESALE ONLY**** Toll free termination for Service Providers, Call Centers, Resellers.
    • Free SIP termination to all NA 8YY destinations.
    • Free SIP trunk from any GLOBAL IP.
    • Free Reporting and Stats on all traffic.
    • Contract required for Collection allowance.
    • CLECs are welcome, Keep your End Office CABS and use our TANDEM.
    • Tandem interconnection/
    • No Dialer Traffic Permitted.
    • Free Toll Free Numbers for Toll free services available as well. SEE TFT services H3 TFT - Free Toll-Free SIP Termination
    • Free SIP termination to NANPA toll-free destinations ( 1-800, 1-844, 1-855, 1-866, 1-877, 1-888 )
    • Caller-ID / ANI is delivered as sent
    • Crystal clear audio with minimal latency and jitter
    • G711u / ULAW with 10ms packetization
    • SIP Registration NOT required!
    • Customer Support Available via E-Mail
    • Compatible with postage meters!
    • Compatible with 10 digit, 11 digit, and full E.164 number formats

    Alcazar Networks Inc - Free toll-free termination

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  • 07/21/15--06:59: VOIP and VPN
  • Surprisingly, using VOIP across an SSL-based VPN can actually improve the call quality (as measured by MOS scores). The improvement seems to be due to encapsulating the UDP VOIP packets ( SIP and RTP ) in TCP/IP. NB Datagram-based VPNs, such as IPSec's ESP are still bad.

    According to a study by Sirrix VPN has no negative influence on latency, jitter and packet loss; in the case of the g7.11 codec and compressed VPN it is even possible to gain 10% bandwidth compared to non-VPN traffic. Apart from that, different common VPN solutions have big difference on the available throughput, which is due to the rather small packet sizes and greatly increased overhead:

    With enabling authentication, encryption, HMAC, anti-replay attack, and initialization vector, and use small RTP size for Codec, the vpn overhead is high:
    g723 with 30ms RTP size and using VPN tunneling: approx. 85% overhead;
    g729a with 20ms RTP size and using VPN tunneling: approx. 80% overhead;

    But when making some adjustments on the encryption/authentication settings and double the RTP size, the overhead can go down to about 20%-30%, which is affordable for most of cases.

    Comparing to SRTP as encryption method for VoIP: approx. 5% additional overhead.

    There is an OpenVPN-based service available on the net which resolves the excessive traffic consumption issue. Several voice packets are placed in the buffer before encapsulation. This minimizes VPN impact and traffic usage doesn't grow with VPN service. This can also help to prevent VoIP traffic detection by packet size, since the size of a single packet is comparable with MTU size (usually 1500 or less).

    VoIP and VPN Forums:

    Tunnel methods


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  • 07/21/15--12:27: 800 Number
  • Toll Free or "800" Numbers are programmed by a

    Responsible Organization: RESPORG.

    Toll Free Number and Service Providers (alpha sequence)

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  • 07/21/15--12:29: VoIP Origination
  • Please add information to this page about VoIP Origination.

    What is VoIP Call Origination?

    One of the terms most often used when talking about VoIP communications is call origination, which basically has to do with how a telephone call starts and how it travels to the receiver once it does. The following will provide an in-depth look at what call origination is, how it works and what type of hardware is required.

    What is Call Origination?

    VoIP stands for "Voice Over Internet Protocol". This means that phone calls utilize a technology that allows the calls to be sent directly over internet networks, which is a much cheaper way to make calls. Call origination refers to the point where the call starts, which takes place over the PSTN telephone network and transferred to their destination through the internet. It's important to note that a phone call through VoIP starts between the initiation point and the destination point, which are referred to as the originator and terminator respectively. There are typically different types of call origination depending on the services provided by the ITSP involved. The only way for call origination to work is if the VoIP originator has a call termination arrangement with a VoIP terminator.

    Required Hardware

    The best aspect of VoIP services is that there are hardly any noteworthy hardware requirements. All that is necessary is a gateway, which essentially transfers phone calls from the internet and onto PSTN lines. Since a gateway must interact with both the internet and standard PSTN lines, there are two interfaces necessary for a gateway, including a telephony interface that takes digital and analog lines and an Ethernet interface as a connection between the gateway and the internet. It's important to understand that a digital line can support a large amount of calls at one time, which can range anywhere from 20 to 30 depending on the type of line that is chosen. In comparison to a digital line, an analog line can only support one phone call at a time.

    How Call Origination Works

    Call origination is a fairly simple process that is a bit more complicated to explain. In essence, the gatekeeper mentioned previously will receive the calls and requests from the dialer. When a user makes a connection to the dialer, the gatekeeper will ask a Radius server to check if the user has input the correct password and username. The Radius server will then answer the gatekeeper with a yes or a no. If a yes answer is received from the Radius server, the user the has the ability to make a phone call from the origination point. Once the number has been placed, the gatekeeper once again receives a request for the phone call to be made. It is at this point where the gatekeeper will interact with the Radius server again to see if the user in question has enough money to make the phone call. The Radius server will then connect to the billing server to ascertain how much money has been provided by the user for this specific call, in order to nail down how long the call can last before being cut off.

    The billing server will then take a look at the location at which the call is originating from in order to measure the current rate of pay, which all depends on whether the call is being made in off peak or peak hours. Once this has been determined, the relevant information will then be sent back to the Radius server. Once the Radius server has received this information, it will be sent back to the gatekeeper. The user is then provided with the IP address for the gateway at the destination to which the call is being placed. This gateway will send the call to the final destination. If no other issue arises during this time, the call can take place and the two users can talk for however long the duration of the call is. If the call takes place once the maximum time allotment has been reached for the call, it will be disconnected and more money will need to be provided in order to make another call and continue the conversation. When the call has come to a conclusion, whether by the callers themselves or because they were disconnected, the initial user will have the total price of the call deducted from their payment source by the billing server.

    Types of VoIP services

    There are two basic types of VoIP services, including PC to phone and phone to phone. The provider for the PC to phone service will place a dialer on the internet that the user of the service can download and install. This dialer allows for an account to be created with any type of payment source that is allowed by the service provider. Once this is done, the user can make and receive calls. ...

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  • 07/21/15--12:36: Call Center Software
  • Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

    There are hundreds of different providers of call center software across the globe, and every call center software system has its pros and cons. When selecting the right call center software for your business, contact center, or call center, it's important to decide which features you want your phone system to have.

    Types of Call Center Software

    ACD helps productivity by assigning inbound agents to incoming calls. The automatic call distributor uses a set of instructions to determine who gets the call in the system. The algorithm can route calls based on agent skill or whoever has an idle phone. ACD can use caller ID or automatic number identification, but usually interactive voice response is enough to help the system determine the reason for the call.

    An automatic call distributor can also take advantage of computer telephony integration. Agents can receive relevant data on their computers along with the incoming call.

    Computer telephony integration is a broad category of software that connects telephone and computer systems. Computer telephony integration software can have both desktop and server functions. Various applications make up a system that can control phones, display call information, and route and report calls.

    Interactive voice response allows callers to route themselves to the appropriate department or use the company’s database for assistance. More sophisticated interactive voice response systems can access accounts and perform certain tasks, such as activating a credit card through a bank’s phone system. IVR involves using dial tone multi-frequency or voice commands. In the VoIP industry, a PBXauto attendant is near interchangeable with IVR. However, auto attendants are not capable of speech recognition.

    A predictive dialer calls a list of phone numbers at once. Outbound agents are then connected to the numbers that answer. A predictive dialer uses calculations to minimize the idle time of agents and the potential of losing answered calls when no agents are available.

    Contact Center Software

    For contact centers, software includes applications for chat, email, and web interaction in addition to telephony functions.

    Call Center Software Providers

    This is a list of call center software providers and developers. Please keep this list in alphabetical order.

    • Ameyo Contact Center Software is an all-in-one software based communication solution that manages end-to-end customer journeys and consistently delivers exceptional customer experiences. It is a powerful and highly flexible IP-based Call Center Software platform that lets you have a personalized interaction with every customer across multiple channels, thereby driving customer engagement to a level par excellence. ...

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  • 07/22/15--15:08: Asterisk Consultants Austria
  • 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.


    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. ...

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  • 07/23/15--05:41: VoIP Providers India
  • This page is a list of VoIP service providers in India. Please keep this list in alphabetical order. India VoIP providers looking to add their services can do so in the list below.


    • ALTOTELECOMVoIP provider for business and Call Centers - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK
    • Incorpus TeleNetworks- Incorpus is a VoIP provider based in India which provides pinless calling for both residential and business customers. Services provided by us are:- RESIDENTIAL VOIP, BUSINESS VOIP, CALLING CARDS, A-Z WHOLESALE TERMINATION, RESELLER/AFFILIATE PROGRAMS, DIDs, TOLLFREE, PBXs . Contact Asap for a free trial at, Skype id:-
    • 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 Contact Number: 800-231-9802
    • Voxvalley Technologies - provides VoIP based communication applications and customized solutions to SMBs, Enterprises and VoIP telecommunication providers worldwide. VoIP Products & Solutions from Voxvalley Mosip Mobile Dialer, Vox Suite, Vox Switch, Vox Bridge, Vox PC, Hybrid Dialer, Vox Con and Vox IM.
    • NOVANET - A leading Business-class VoIP & Cloud Communication provider to Contact Centers & Enterprise customers. Novanet offers a complete suite of VoIP and Cloud services specifically crafted to suit the needs of a modern day Contact Center.
    • i7 Solutions VOIP Provider International Calls service provider. Provides Voip calling to Home Users, Small Office, BPO, Call Centers, Exporters and corporates. To become resellers. please contact us.
    • Live-TechOffers TalkSwitch VoIP Hardware & Lynk VoIP Services | Remote Technical Support | VoIP Service Plans | Worldwide Support
    • Ozonetel Systems-Offering PBX, Call Center and IVR services.
    • barePacket - Offering VoIP/IP-PBX, IVR & Call Centre Solutions, Mail Server aimed at SMEs.
    • Call Centric India - VoIP, Free accounts, Pay as you go & Unlimited accounts, International DID's, BYOD. ...

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  • 07/23/15--14:34: VoIP Termination
  • Please add information to this page about VoIP call termination.

    What is VoIP Termination?

    VoIP call termination is used to refer to the procedures that are used for routing telephone calls from one provider to the next provider until the call has been routed to the last telephone company and has been received by the recipient. Voice termination is another term that is used for call termination. The telephone companies are also referred to as providers or carriers.

    Called Party

    The called party is the person who has received the telephone call. The end point of the route may be on the Internet or may be at a point that was reached by routing the call through the public switched telephone network. The procedures for routing the call stop when the call has been received by the recipient. The process may seem simple to individuals who do not experience problems with making telephone calls but is not so simple to individuals who make calls that will not connect such as calls to a different country.

    Calling Party

    The calling party is the person who has initiated the call and who wants to be connected to the called party. There may be problems with the telephone calls that begin on the Internet and end at a cellular phone. The sound of the voices may seem to be amplified with echoes and have a scratchy quality. The calls that are initiated with Google Voice may never connect or could be delayed.


    Voice over Internet protocol (VoIP) is a term that is used to describe a call that was initiated on the Internet such as Skype calls or calls made with Google Voice. The calls that were initiated on the Internet usually end at a point that is not on the Internet. Most of the recipients of telephone calls receive the calls with a landline phone or with a cellular phone. The route may begin on the Internet but will end at a point that was reached by routing the call from the public switched telephone network (PSTN), which is the common description for call termination.

    Internet Networks

    A tier-one operator is licensed and registered to operate an Internet protocol (IP) network for Internet telephony services. Tier-one operators can handle call origination and call termination. A tier-two operator can lease services from a tier-one operator. The tier-three operators can lease services from either a tier-one or two-two operator. There are also resellers of VoIP services and wholesalers in the market. The quality of the services is not very high because of the inconsistencies in the market such as fluctuations with demand, fraud and problems with doing business on an international level.

    Call Origination

    Call origination is used to refer to telephone calls that originate from the public switched telephone network and end the route on the Internet. Call termination is considered to be the opposite of call origination because the direction of the paths are reversed. The terms are associated with the starting point of the calls, path of the route of the calls and termination point of the calls. The operators of IP telephony services can handle calls that originate or terminate on the Internet.


    The fees for services are subject to the regulations of several countries because the routes of the calls will cross over more than one country. Those countries can use legislation to control the fees for the services. Termination rates are usually very high for the countries in the Middle East and Asia. The rates are intentional high because there are more incoming calls than outgoing calls from the countries, which is caused by the diaspora effect of a migrating population.

    VoIP Termination Providers

    Please list VoIP Termination providers here in alphabetical order.

    10gea 10gea's wholesale SIP termination provides exceptional quality routes and high volume switching capacity for all types of Wholesale end users. Very competitive rates for both dialer and conversational high volume traffic on tier one routes.

    • Extremely Competitive Pricing
    • Short Duration and Conversational Routes Available
    • Experienced 24/7 Network Monitoring and Technical Support
    • Quick & easy test and turn up process

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    Business VoIP Providers - Compare and Choose a Business VoIP Provider

    Quality business VoIP providers today offer a wide variety of feature packages, services and prices. Selecting the ideal provider and service options will depend on your type and size of business, features needed and projected volume of usage. Even when working with top-tier providers, your basic monthly service charges per line may begin at rates as low as $20. Before choosing your VoIP provider, it is essential to first determine your company's precise telecommunications needs to enable timely and cost-efficient initiation of your service. By consulting your chosen Voice over IP service team and seeking their expert advice in advance, you can be prepared to take the following steps to facilitate the smooth, productive startup of your services:

    • Evaluate Your Internet Connection. - Determine the strength and capacity of your Internet connection and bandwidth. You need to ensure that your system has adequate speed to best accommodate your new VoIP installation for top quality service.
    • Assess Your Company Budget and Needs. - With knowledge of your company's current budget and VoIP needs, you can more easily select the service provider and feature options that meet your requirements.
    • Determine Your Equipment Needs. - Evaluate your current and near future VoIP equipment needs. Phones can be purchased from around $50 to $500 or more. Once you decide which feature options are immediate requirements and which ones can be added later as needed, you are ready to choose your service provider.
    • Compare VoIP Providers. - By comparing VoIP company service options, advanced features and equipment along with user and industry reviews, you can best make a wise decision, selecting the ideal VoIP provider for your enterprise.

    Important Information to Request from Any Potential VoIP Provider

    Before signing a service contract with any business VoIP provider, be sure to request basic service information and practices in writing. You need to be certain of such details as startup costs and monthly fees, any limitations and costs on portable phone numbers and exactly which features are included in the service package you select. You also need to know if international calling is included, charges for adding extra features and the extent of customer care and technical services provided. Also important are such issues as whether your provider offers a money back guarantee and if there are any cancellation fees. It is also helpful to determine prior to signing up for VoIP services if there are any hidden fees assessed by your chosen provider.

    Take Full Control and Advantage of Your VoIP System

    Once your new business VoIP system and service are in place, you and your staff members will have full-control capabilities for use of your business communications system. Your service provider will ensure connection with your online portal for customizing your telecomm options. These modern digital portals are user-friendly, enabling feature changes and additions to be made for immediate availability. You and your staff can make decisions and changes in real-time that work for you right in the moment.

    You can manage your call settings remotely, directing calls to voicemail or having them transferred to another number or extension. You can also make exceptions to any chosen setting in your phone system. For example, if you are expecting an important business call and want to take that call, but hold all other calls for a few hours, you can set your phone to direct only the designated call to ring on your extension. This system allows and encourages you to take complete control of your telecommunications systems and settings so that the service works for your best interests and immediate needs at all times.

    Major Business Benefits and Advantages of Installing VoIP

    With an excellent quality VoIP system installed and running well in your company offices to provide remote access for you and your employees, you can work much more efficiently, achieving more in less time. You will enjoy the many benefits of knowing that you can leave the responsibility of your advanced office telecommunications system operations to your VoIP provider while you handle other important business matters. Other major benefits and advantages of your new business VoIP system enable you to accomplish the following:

    • Schedule Your Own Business Hours. ...

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

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

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

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

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

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

    Providers in other countries/continents can be found here:


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  • 07/23/15--18:43: Virtual PBX providers
  • Virtual BPX is a service offering functionality of a PBX without the need to install switching equipment at the customer location. Only VOIP phones need to be installed at the customer site. This makes supporting distributed workers very easy as each requires only and internet connection and a VOIP phone. A business virtual PBX phone system can reduce your monthly phone bill significantly compared to a traditional business phone system.

    What Is a Virtual PBX?

    A PBX, short for private branch exchange, is a telephone system with the capacity to switch calls between different users on local lines while still relying on the same number of external phone lines. With a virtual PBX system, the system is posted and software based without all of the traditional hardware of a physical PBX.

    Virtual PBX Primary Function

    A virtual PBX is used by businesses in a variety of ways. Primarily, companies utilize the system as an auto-attendant to establish preset call transfer options without needing an operator or receptionist. This type of system is capable of performing tasks that include auto-attendant settings, time of day or day of week functions, or even find or follow me sequences.

    One of the most important functions of a virtual PBX system for companies is the software’s ability to establish pre-determined sequences. For example, in some businesses it may be appropriate for the phone to ring to a receptionist or operator first. If the receptionist does not answer in a predetermined number of rings, however, the call is then transferred to a secretary. Again, if the call is unanswered, it can be set to forward to an assistant. Left unanswered by these two individuals, the call can be forward to a manager or even an owner. These call settings are completely customizable and can be based on any number of sequences.

    This type of software is also able to facilitate customized answering menus and sub-menus. The system can be modified to establish appropriate dial prompts leading to a number of different departments within the business, including different sequences on different days. PBXs are used by the vast majority of businesses to establish advanced call routing services.

    Virtual PBX Cost

    A virtual PBX is a complex service; however, that doesn’t mean that it is expensive. In fact, a virtual PBX is typically more cost effective than a physical PBX. The main reason that a virtual system saves on cost is that it does not require the same investment in capital to establish or set-up the call system. Because a virtual PBX is a software or hosted system, it is typically an operational cost, or a low monthly payment rather than a large upfront investment. This aspect alone generally makes a virtual or hosted PBX a less expensive, or at least more cost effective, option compared to the traditional PBX.

    Virtual PBX Benefits

    Aside from offering an effective call system, a virtual PBX presents a number of added benefits for users. As a whole, virtual PBXs lead the industry in business communication choices. This type of system seamlessly integrates the call management system with any existing phones to affordably and effectively deliver better call management. These systems also feature several innovative call features to meet the needs of any business. These systems offer various functions including call routing, follow and find me call forwarding, voicemail notifications, call recording, and more.

    The benefits aren’t limited to the features, though. Virtual PBXs offer virtually limitless application for one or hundreds and even thousands of employees. Likewise, there is not hardware to maintain or constantly upgrade. Considering that benefit, the system is also more cost effective and generally provides for a variety of flexible billing options. The limited maintenance, web-based management, and hassle-free setup alone are often enough to convince a company to switch over to this option.

    PBXs are an important tool in any business that makes and receives nearly any volume of calls. A virtual PBX can dramatically increase the efficiency of a business by effectively managing calls. This efficiency combined with the other numerous benefits of a virtual PBX can virtually transfer the communication capabilities of any company.

    List of Virtual PBX Providers


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  • 07/24/15--00:28: DID Service Providers
  • 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

    SMS enabled DID Providers

    • MultiTEL is providing retail and wholesale Worldwide DIDs - over 90 countries. Pick your own SMS enabled DIDs from over 40 countries. (US, UK, CA, Germany, etc). Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. SMS forwarded via e-mail , URL or as SMS/Text message. All calls are forwarded to SIP, PSTN or to our free Hosted PBX. Coverage and numbers always available in stock from more than 90 countries.


    • CarryMyNumber.comAlgeria DID /Virtual Phone Numbers at _wholesale rate@$ 4/month with free PBX with the Free forwarding to SIP, Softswitches ,Trixbox ,Asterisk or VOIP. Phone Numbers from over 70 countries available. Free PBX . Unlimited Channel numbers for call centers /Calling Card Providers__. Largest FootPrint worldwide. No Per Minute charges.
    • Provides Cheapest Algeria DID /Virtual Phone Numbers/DDI Numbers @_€ 6.95/month including free PBX. with the Free forwarding to Skype ,Gtalk , iTalkWorld, Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX __. Toll Free Number Available without Monthly commitments.
    • BuyDIDNumber We Provide Algeria Virtual Phone Numbers@ $ 7.99 / Month NO SETUP FEE , UNLIMITED CHANNELS available with the Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk , voipbuster , iTalkWorld ,any Betamax Delmont Voip or any other ITSP . Phone Numbers all over the world are available. Free PBX . Toll Free Number Available without Monthly commitments.
    • Currently only national Algerian numbers. Also 70 other countries available. Free forwarding to our PC & Mobile apps , Our PC and Mobile Apps also work in Countries where voip is Blocked . Further We have Lowest Call Forwarding Rates anywhere in world , starting as low as 1/2 cent.
    • Provides Cheapest Algeria DID /Virtual Phone Numbers @ $ 7.99 with the Free forwarding to Skype ,Gtalk , iTalkWorld, Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX . Toll Free Number Available without Monthly commitments.
    • Cheapest _Algeria Virtual Numbers @ Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk or VOIP, Free PBX , Any Betamax Voip_. Toll Free Number Available without Monthly commitments. _Cheap Forwarding rates starting 1.5 cent/ Minute_

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  • 07/24/15--07:14: Phoenix VoIP
  • This is a list of VoIP providers in Phoenix. Phoenix VoIP companies typically support multi-line telephone systems, small gateways and hosted VoIP. Please add VoIP Providers in Phoenix to the list below.

    • CloudNet Group offers carrier class Hosted VoIP, SIP Trunking and PRI services on an national level. With over 8500 installs across the country, CloudNet Group specializes in delivering a managed voice solution, personally installing and training every customer to ensure a smooth and seamless transition. All services are provided utilizing dual, mirrored Metaswitches. 24x7x365 support is standard for all customers. Call today to experience the CloudNet difference.
    • FluentStream Technologies FluentStream Technologies is a fantastic Cloud-hosted business phone system. We have a best-in-class web portal, an industry-leading WebRTC-based FluentCloud WebPhone, and world class 24x7x365 support. Get all the benefits of the cloud with service that you'll love!
    • SIPSRUS SIPSRUS provides cloud computing based business phone and fax systems designed for today's mobile and distributed business world. SIPSRUS provides services in USA, Canada, UK, Australia, Japan, Spain, India, UAE and many more!
    • Jive Communications - Jive offers Phoenix VoIP services business.
    • Lantelligence is an International provider of Business Communications solutions that include IP Phone Systems, Call Centers, Video and Web Conferencing services for multi-location corporations.
    • ConfCentral offers flat fee conferencing with local access number in the USA
    • ActiveServe PBX Hosting 3CX, Asterisk, Elastix, and Trixbox PBX Hosting. CAT-5 Data Center, Active NAT Assistance™, Fully Managed Cisco Network, Cloud Platform. Eczema Cure No hardware or software to purchase or maintain. 24/7/365 Support. Do-It-Yourself or Turn-Key
    • Connect Me Voice offers a full line of services from basic voicemail to full business systems
    • Digivoix is saving tons of money on all your internet telephony needs, Hosted phone services start at $34.95 (Unlimited Local and Long Distance calling), SIP Trunking starting at $12.95 (Unlimited incoming, 1.6 cents outgoing (US)),

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