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How to start a VOIP Business

August 17, 2015, 9:06 pm
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The first thing to do is decide what part of VOIP marketplace you want to serve. Here are some possibilities:

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

Some general suggestions:

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

Value Added services

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

  • Message broadcasting / Call Center Solutions

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

VoIP Business - Startup No Money Down:

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

Become a VoIP provider without heavy investments. Speedflow offers hosted Class 5 Softswitch at a special price. You can use it either with our high-quality termination or with your own VoIP routes. No setup fee, no hidden payments.

VoIP Business: Buy, Build or Resell?

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

August 17, 2015, 11:22 pm
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QoS

August 18, 2015, 12:26 am
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QoS (Quality of Service) is a major issue in VOIP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due interference from other lower priority traffic.

Things to consider are
  • Latency: Delay for packet delivery
  • Jitter: Variations in delay of packet delivery
  • Packet loss: Too much traffic in the network causes the network to drop packets
  • Burstiness of Loss and Jitter: Loss and Discards (due to jitter) tend to occur in bursts

For the end user, large delays are burdensome and can cause bad echos. It's hard to have a working conversation with too large delays. You keep interrupting each other. Jitter causes strange sound effects, but can be handled to some degree with "jitter buffers" in the software. Packet loss causes interrupts. Some degree of packet loss won't be noticeable, but lots of packet loss will make sound lousy.

VOIP QoS Requirements

Latency

Callers usually notice roundtrip voice delays of 250ms or more. ITU-T G.114 recommends a maximum of a 150 ms one-way latency. Since this includes the entire voice path, part of which may be on the public Internet, your own network should have transit latencies of considerably less than 150 ms.

Most network SLAs specify maxium latency
  • Axiowave SLA 65ms maximum latency
  • Internap SLA 45ms maximum latency
  • Qwest SLA 50ms maximum latency - Measured Actual for Oct 2004: 40.86ms
  • Verio SLA 55ms maximum latency
The SLA numbers above are for backbone providers, the total latency for a VOIP call may also include additional latency in the VOIP provider's and the user's local ISP networks.

Jitter

Jitter can be measured in several ways. There are jitter measurement calculations defined in:
  • IETF RFC 3550 RTP: A Transport Protocol for Real-Time Applications
  • IETF RFC 3611 RTP Control Protocol Extended Reports (RTCP XR)
But, equipment and network vendors often don't detail exactly how they are calculating the values they report for measured jitter. Most VOIP endpoint devices (e.g. VOIP phones and ATAs) have jitter buffers to compensate for network jitter. Quoting from Cisco:
  • Jitter buffers (used to compensate for varying delay) further add to the end-to-end delay, and are usually only effective on delay variations less than 100 ms. Jitter must therefore be minimized.

Whats an acceptable level of jitter in a network? Several network providers now speciify maximum jitter in their SLAs.
  • Axiowave SLA 0.5ms maximum jitter
  • Internap SLA 0.5ms maximum jitter
  • Qwest SLA 2ms maximum jitter - Measured Actual for Oct 2004: 0.10ms
  • Verio SLA 0.5ms average, not to exceed 10ms maximum jitter more than 0.1% of time
  • Viterla SLA 1ms maximum jitter
The SLA numbers above are for backbone providers, the total jitter for a VOIP call may also include additional jitter in the VOIP provider's and the user's local ISP networks.

Detailed jitter reading

  • More detailed overview
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Mobile VoIP

August 18, 2015, 1:11 am
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Mobile VoIP is an efficient, low-cost way to communicate using your cell phone and the services provided by your home or business VoIP provider.

How Does Mobile VoIP Work?


Mobile VoIP works with a cell phone’s 3G, 4G, GSM, or other Internet service to send voice calls as digital signals over the Internet using voice over IP technology. Mobile VoIP phones can also take advantage of WiFi hotspots to eliminate the calling costs of a cellular voice or data plan.

Page Contents


By using VoIP, mobile VoIP phone users — especially smartphone users — can benefit from lower costs when calling, texting, or other common smartphone activities. Digital data transmission using VoIP is also typically faster, as the data is spread out over multiple packets, each taking the fastest route to its intended destination.

Using a mobile VoIP phone with WiFi hotspot access can also reduce a mobile VoIP phone user's costs by sidestepping the carrier's expensive 3G service altogether. For instance, with a cellular carrier's monthly data plan, callers can easily exceed bandwidth maximums, incurring overage charges. Tapping into WiFi hotspots with mobile VoIP software reduces that risk and extends the lifespan of the monthly data allotment.

A mobile VoIP phone service can eliminate the need for a basic voice plan, as well as optional (and costly) text add-ons. With a mobile VoIP phone, cell phone users can enjoy more flexibility in calling times than a cellular voice plan provides, with fewer restrictions. VoIP mobile phone service means that a mobile VoIP user can make unlimited inexpensive or free calls using voice over IP technology at any time.

Mobile VoIP users don't need to worry about the limitations associated with cell phone calling plans, such as:

  • Anytime minutes
  • Night or weekend minutes
  • Rollover minutes
  • Roaming charges
  • Incoming call charges
  • Messaging limits
  • Mobile-to-mobile calling (check with your mobile VoIP provider, some do treat in-network calls differently)

Mobile VoIP phone users can also take advantage of the additional, integrated features a mobile VoIP app supports. This includes high-bandwidth activities such as group chat and video chat. Accessing these functions without mobile VoIP software (by fring or Talkonaut, for instance), typically requires a separate app, and using it could impact or exceed monthly text and bandwidth maximums.

Accessing Mobile VoIP

Cell phone users can use mobile VoIP service on their phone with the addition of mobile VoIP software. These are apps offered by VoIP phone service providers customers may already be using at home or at work, such as Vonage, or standalone mobile VoIP apps such as Skype, Vyke, or Truphone.

Some services, such as Truphone, also offer an entire mobile VoIP network by combining a SIM (Subscriber Identity Module) card and an app together. (The SIM card contains all the information needed to identify network subscribers. ...
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Automatic Call Distributor

August 18, 2015, 7:00 am
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Automatic Call Distributors

Automatic Call Distribution or ACD, is a tool commonly used in the telephony industry. ACD systems are commonly found in any office that handles a large volume of inbound calls. The primary purpose of an Automatic Call Distributor is to disperse incoming calls to contact center agents or employees with specific skills.

The ACD system utilizes a rule based routing strategy, based on a set of instructions that dictates how inbound calls are handled and directed. These rules are often simply based on guiding a caller to any agent as fast as possible, but commonly multiple variables are added, all with the end goal of finding out why the customer is calling. Matching and routing literally thousands of calls to the correct agent is a difficult task, and is often done in concert with Interactive Voice Response and Computer Telephony systems. ACD servers can cost anywhere between a few thousand dollars to close to millions of dollars for a very large call center handling thousands of calls per day.

Automatic Call Distributor Vendors

  • 3CLogic Cloud-Based Contact Center Software 3CLogic is a leading provider of cloud contact center solutions based on an innovative approach, designed to deliver modern-day contact center features to meet the challenges of a modern world. With 3CLogic's ACD functionality, you can set, manage, and adjust call priorities to automatically ensure the most urgent inquiries are always answered first.
  • ICTBroadcast Automatic Call Distributor: Is a Unified automatic call distribution software solution from ICT Innovations . Feature- unifed Auto Dialing, Custom IVR Designer ,Survey Campaign , SMS blasting & marketing , Fax blasting , Voice blasting ,AMD supported, Email marketing and appointment reminder solution.
  • Virtual Phone Number IVR GURU providing ivr service for call center to automatic distribute call to multiple number and we louche new DND software to filter data.
  • Vocalcom Intelligent distribution of calls is something that Vocalcom has been re-inventing for many years, refining and perfecting to ensure the optimum solution to connect customer and agent.
  • Voicent ACD Software is designed to be configurable to the user. We offer default 'round robbin' call distributions, to the more advanced 'rule & skill based' transfers. Voicent is the leading provider of the Managed Call Center Software.
  • DooxSwitch DooxSwitch provides one of the most comprehensive and state-of-the-art cloud-based call centre software solutions.
  • Five9 ACD Software is designed so that any business user can configure it, yet it has all the sophisticated routing features any enterprise requires. Five9 is the leading provider of cloud contact center software.
  • Foehn - We are the experts in IP Communications with over 12 years of successful deployment of Asterisk and open source technology solutions. ...
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IVR

August 18, 2015, 7:01 am
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What Is IVR?


IVR, or interactive voice response, is a what allows phone systems to process touch tones or voice waves during a telephone call. IVR technology is responsible for the menus people hear and respond to when they call up a company or business and hear the words: "press 1 for sales, press 2 for marketing, press 0 to speak to the operator," for example. IVR systems can be fully customized to play back dynamic audio, or pre-recorded menu options.

IVR is not necessarily related to VOIP, however, a VOIP IVR is. Most VOIP IVR systems or software support SIP based VOIP, but Skype IVR also support non-standard based Skype service.

Computer Telephony Component

IVR is an automated computer telephony integration CTI system which allows providers to create complex menus which the caller can navigate by using touch-tone key-presses or via spoken commands. IVR systems can be used as a Voice portal to access remote information such as bus scheduling where the caller can select the route for which they require information, or for billing or customer service systems which allow the caller to enter information such as their account number or credit card details without the need for operator assistance.

IVR and ACD Integration

IVR solutions are often integrated with an ACD, which routes incoming phone calls to agent work groups. This integration can be both a front end and back operation.

  • Most typically, an ACD system can route callers to an IVR program based upon DNIS or other parameters such as time of day or day of the week.
  • A smart IVR can transfer callers back to an ACD system to route the call to the next available agent within an agent hunt group.

One important task of an integrated IVR and ACD is to display Screen Pop information from the caller on the agent's workstation so that the agent has caller information readily available without the need to prompt the caller again.

IVR and Voice Broadcasting

IVR applications are typically associated with inbound calling programs. However, IVR technology can be applied to outbound calling campaigns and are most commonly used with Voice Broadcasting and touchtone responses. Examples of the application of this technology include the option to speak with an operator, opt out of a calling campaign, or taking an outbound survey.

Here is an example of IVR implementation in Voice broadcasting

Graphical Design Tool for IVR Applications

Recent IVR systems usually use high level scripting languages such as VoiceXML, an open standard for interactive voice response systems. For most users who lack technical training, developing an IVR system using scripting language, even high level language, are not feasible. The good news is there are design tools that are based on graphical user interface for the techies and none-techies alike. By using a GUI tool, a user can simply drag-and-drop components and create and deploy an IVR system in minutes. The whole design is a call flow diagram, much like a voicemail system user manual.

See Also (Vendor Information)

IVR Information


  • CCXML standard markup language for IVR / call control applications
  • IVR System Simulation Model - estimates resources required for an inbound calling campaign.
  • IVRS World - Blog about IVR
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Modem over VOIP

August 18, 2015, 9:32 am
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If you are replacing an analog PSTN line with VOIP service, in addition to ordinary phones, there may be other devices that contain data modems that were using the PSTN line. Examples are: Tivo, POS Terminals, FAX machines, etc.

The problem is that the codecs used by VOIP ATAs are designed to compress voice, not the analog signals sent and received by modems. A second problem is if a non-compressing codec is used, the transmission will be very sensitve to network QoS, i.e. packet loss, jitter, and latency will be issues. To sucessfully use data modems over a VOIP connection you will need a minimum of:
  • A non-compressing codec - ITU G.711 is the usual choice
  • A very high quality network connection

In order to make modem connections less sensitive to network QoS problems, rather than passing through the modems signals over the VOIP connection, the signals can be converted (de-modulated) at the VOIP ATA and sent as data over the network to the far end where they are converted (re-modulated) back to their original form. This method has the advantage that the data transmission over the network does not require a high QoS.

For FAXs T.38 is the standard for relaying FAX data accross IP networks.
Many VOIP endpoints/ATAs now support T.38 but many VOIP Service Providers do not support T.38 or are in the process of implementing it.

For Modems the ITU approved ITU V.150.1 (also know as V.MOIP) in January 2003. This standard defines how to relay modem data accross IP networks. This standard is not implemented yet in most VOIP equipment.

Resources



See Also







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T.38

August 18, 2015, 9:55 am
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T.38 is an ITU standard for sending FAX across IP networks in a real-time mode.
FAX messages are sent as UDP or TCP/IP packets.

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

From RFC 3362:

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

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

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



The Importance of T.30 ECM Error Correction Mode (ECM) in T.38 Deployments

One of the most important features of the traditional facsimile standard (T.30) is ECM error correction. The inventors of fax recognized that the audio quality of PSTN phone lines could not be trusted to be 100% error free - it was not uncommon to hear static, or the occasional pop or crackle on some fax calls. ECM error correction allows the sending and receiving terminals to compare notes at the end of each page, and selectively retransmit any data that was not received the first time around. This retransmission process is continued until the received page is certified error free, and the transmission of the next page begins.

Remarkably, many T.38 implementations, including those of top tier carriers such as Level3, XO, Verizon etc have explicitly disabled ECM error correction. This problem is discussed in detail here. Don't make this mistake! Be sure to enable ECM and insist your provider or PBX vendor do the same!

Want to check to make sure you have ECM enabled? This handy ECM self-test tool can tell you in minutes.


T.38 with Fax Voip T38 Fax & Voice

FaxVoip Software develops solutions for the transmission of a fax via the Internet Telephony (FOIP). The main emphasis has been placed on the transfer T.38 and audio via SIP, H.323 and ISDN CAPI 2.0.
Fax Voip application operates with T.38 faxes via standart COM port interface.
What is Voip? For your fax or voice application, it's a Voice Fax Modem. ...
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Sip Trunking Providers

August 18, 2015, 12:23 pm
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This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

Country specific pages:

What Is SIP Trunking?

Traditional phone calls go over phone lines, but with the rise of the internet, Voice Over IP communication systems give you much more functionality to receive a high volume of calls. This is called SIP (session initiated protocol) trunking. These services are offered by the majority of VoIP providers and can be tailor made to suit the needs of a small business all the way up to a large corporation with a gigantic workforce and everything in between.

One of the main advantages of going with SIP trunking is that it will integrate everything from data, the internet, your video, and your voice services all into a single line. The term "trunking" is a bit dated and goes back to a time when traditional telephone services were used and it described how a wide variety of telephone users could share a much smaller pool of communication paths, thus making it a much more efficient option for businesses than getting a separate phone line for each worker. That was in the past, though, as today's SIP trunking capabilities will even allow all calls to be considered local calls and won't even require the business to have a single phone line at all.

The Benefits of Using SIP Trunking Services

Choosing to use a good SIP trunking service will significantly level the playing field for smaller business to compete with large corporations and appear much more professional and bigger than they actually are. For larger business, one of the main advantages is being able to much more easily maintain that local presence. Regardless of the size of the business itself, SIP trunking's biggest benefit overall is simply the reduction in cost to the company's overall expenses. This is particularly true for SIP VoIP services that are specifically crafted for a business in particular.
  • It offers very low cost calling.
  • It's much easier to scale than other options, making it very future proof.
  • SIP trunking makes moving a breeze, as you can easily take your phone number with you while moving offices or even while traveling.
  • Network outages are much less impactful, as incoming calls can easily be routed to other locations.
  • It's ideal for any sized business with at least 25 physical phones.
  • It's a fantastic choice for any business that has an international location.
  • It offers the same sort of emergency usages and will even offer enhanced 911 dialing options.

How SIP Trunking Can Take Your Business To The Next Level

It used to be that you had to have a PRI (Primary Rate Interface) in order to have video, voice, and data all on a single line. This was a physical device that had to be purchased separately but along with your chosen phone provider's calling plans. This is no longer the case with SIP, as it allows your provider to set up a proxy server (commonly referred to as a SIP proxy), which will provide the outgoing, incoming, and PBX calls for you. Everything is located off site and handled by a professional IT team or contractor through your provider, meaning it saves you a ton of time and money on the technical support side of the equation.

Being temporarily shut down means simply having the SIP reroute calls to a different office and also allows you to much more easily outsource to international locations, such as with customer service centers.

All of these benefits are much better than what traditional PSTN (public switched telephone network) have been able to provide through their services and will give your business just what it needs to take it to the next level.

1Call,Inc 1Call understands the callcenter and SMB business demands and seek to under promise and over deliver. ...
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VOIP Resellers

August 18, 2015, 12:28 pm
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VoIP Resellers


This is a list of VoIP resellers in Alphabetical Order:

1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
  • Custom portal for your customers
  • Actual US CLEC
  • Set your own pricing for your customers

8774e4voip.com 8774e4voip.com - Contact Us Toll Free @ 877.434.8647

Air21group.co.uk - UK BASED/ 247 SUPPORT - Call us on 0121 314 1114
  • Add to your telecoms business or start your own business reselling business telecoms today with the Air 21 Group
  • Whitelabel reseller program
  • High Commissions
  • Wholesale VoIP origination
  • Termination SIP Trunking
Click to enquire now

Alcazar Networks Inc. is offering Wholesale Origination / Termination / Free Toll Free Termination - Get paid for your toll free traffic! We provide high quality, dependable access to over 3100 rate centers and instant access to over 1,200,000 DIDs including T38 and local number portability. Wholesale SIP

Asterisk SIP Trunking - US — Offers SIP Trunking for Asterisk. Over 500,000 DID's available in 9,500 rate centers. You can activate and setup service in minutes. TDM Enterprise quality. Fully qualified Asterisk consultants ready to assist you. Live customer service with 24/hr ticketing system. No per channel fees, we offer unlimited channels in with your SIP trunk. FREE API for your website. Use our API to leverage the power of our customer user portal on your own website. You can build your own back office admin panel with our API and also provide your customers the ability to order DIDs in REAL TIME along with setting up SIP Trunks. Automate everything and increase customer base.Our Asterisk SIP Trunks are not only for Asterisk. We work with every IP Ready telephone system. Try us out today! No contract and low rates. DIDs are $0.50 per month, Toll Free DIDs are $0.50 per month US/Canada calling rates per minute are $0.009 for TIER1 quality calls. Nothing will route out low quality networks. International calling is available immediately upon registration. Auto refill on your credit card will ensure your account never runs out of funds.
      • New *** Wholesale Reseller User Portal and Admin Portal. With this new service you can customize your own Admin and User portal to sell SIP Trunk Services to your end users. This means that you can set your own rates to your customers, your admin portal will charge your customers in real-time so you never have to chase your money. All services are turned up in real-time for your end users. This includes DID ordering, SIP Trunks, vFAX and all other telecom related services. This is the ultimate reseller portal program. As for the requirements you must have an Authorize. ...
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VoIP Termination

August 18, 2015, 12:51 pm
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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.

VoIP

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.

Fees

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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VOIP Service Providers

August 19, 2015, 1:12 am
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For a list of VOIP to PSTN service providers, indexed by country, please see:


VoIP and VoIP Service Providers

What is VoIP?

VoIP (which stands for "voice over internet protocol" and is commonly referred to simply as an internet phone) is a highly cost effective and reliable way for businesses or even homeowners to make calls across the world or even just across town. The majority of major cable companies that offer bundled internet, television, and phone services already utilize this newer technology, but there are tons of other independent companies that specialize in providing this service to their customers at reasonable rates and with tons of extra features.

More Than Just Computers

When people think of VoIP, they generally think of computers due to the popularity of the numerous free communication services like FaceTime and Skype, but this is truly just one aspect of what VoIP can truly offer. It is true that VoIP technology transmits voice communication that's been converted into digital data across a packet-switched network or the internet (what this means, in essence, is that a user making phone calls over high speed internet lines rather than phone lines). With that in mind, users are not confined to only using it on a computer. VoIP technology can connect through the internet using traditional telephone equipment just like a regular line. The phone itself is connected to the internet using an adaptor that's plugged straight into a home or business's internet network. Most major services offer a softphone option as well, which allows the user to use their computer directly as a telephone service. In addition to all that, VoIP providers will generally also offer mobile or tablet apps that allow their customers to make calls on the device (assuming it's connected to Wi-Fi at the time). ...
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Asterisk Paid Support

August 19, 2015, 11:21 pm
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This is a list of companies that offer Asterisk Paid Support.

Important notice to posters entering new companies to this page:


Artvister

  • http://www.artvister.com
  • +44 2037348345
  • office@artvister.com
  • Asterisk PBX Installation and Support
  • PBX - VOIP integration and maintenance
  • Telco Support integration - ISDN PRI/BRI
  • Call Center, Contact Center Support Services,Data Entry.
  • Professional Technical Support Services
  • Auto Dialer, Predictive Dialer, ACD Services,Billing Services.
  • Mediant,AudioCode,Digium,Topex,2n System Integrator.
  • IVR,TTS.
  • VirtualPBX, Hosting PBX, CTI and CRM Integrator.
  • Cloud PBX, Cloud Vicidial, ViciBox, AgileDial
  • Remote Support.


Asterisk Agent (U.S. Based)

  • http://www.AsteriskAgent.com
  • World Wide Asterisk Support
  • support@AsteriskAgent.com
  • 800-763-2908
  • Specializing in Asterisk based solutions.
  • Asterisk Support & Staffing
  • Live Website Support Available

*astTECS

provides 24/7 expert support services in installation, configuration, troubleshooting, administration and management for all Asterisk based products, onsite and remotely .
Our areas of expertise are:
  • Asterisk Installation & Configuration
  • Custom Asterisk Development
  • Dial plan programming
  • IVR designing & Servicing
  • Custom Scripting
  • Call monitoring and recording (Voice Logger)
  • Custom API and CRM Integration
  • Database Integration (ODBC)
  • Custom Queue Systems
  • Conference Systems
  • Vicidial Installation & Configuration
  • Click to dial Applications
  • Support AsterCC
  • Support for A2 Billing (Calling Cards)
  • Free PBX configuration and Installation
  • Trix Box Configuration and Installation
  • AsteriskNow Configuration and Installation
  • Support for Free Switch
  • Remote server administration and management

For assistance please contact : 080-6640 6666
E-mail : info@asttecs.com
Please visit:
http://www.asttecs.com
http://www.asterisksupport24x7.com

Asterisk'Pros

  • Local installs in Southern California, USA - Support Worldwide
  • Provides design and support of any Asterisk-based PBX of all sizes, Polycom and Queuemetrics
  • Specializes in interfacing with external database systems, and call centers
  • Also has full-IT staff to troubleshoot end-to-end network setups
  • Web site: http://www.asteriskpros. ...
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VOIP Event Calendar

August 20, 2015, 3:40 am
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2015 VOIP Related Event:

September 2015

  • 01 – 04 Asian Carriers Conference 2015 (ACC) - The best and the biggest information technology and telecommunications companies from Asia and the rest of the world. The ACC 2015 is scheduled from 01 – 04 September at Shangri-La’s Mactan Resort and Spa in Cebu, Philippines.
  • 16 - 17 Wholesale World Congress - Large event for VoIP providers, ISPs, wireless and mobile operators, as well as other companies from the voice, data and SMS industries.

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


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New Software Releases

August 20, 2015, 10:43 am
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Search

Asterisk consultants Canada - Ontario

August 20, 2015, 1:12 pm
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This is a list of Asterisk consultants in Ontario, Canada. Also check out VoIP Providers Canada for VoIP service providers in Canada.



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
  • VoIP Support Services: Asterisk, Trixbox, Elastix, FreeSwitch, Vicidial, A2Billing, Avaya IP Office, Call Center solutions, High Availability (HA) and clustering systems.
  • Hourly and Contract Support, Consulting, Hosting, Office PBX, etc.
  • Asterisk full-featured PBX installation/configuration.
  • Voicemail, fax2mail, PSTN integration, Conferencing.
  • IP-Phone provisioning and support, softphones, TFTP.
  • Network(router,switch,firewall) design and support. BGP, MPLS, OSPF, B2B VPNs 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.

Active Access Communication Systems

  • Web Site: http://www.visionvoip.com
  • Email: mailto:mail@visionvoip.com
  • Location: Ottawa, Canada
  • Phone Number: 1-800-VVoIP-15 (1-800-886-4715)
  • Type of Support: Telecommunications, VoIP, Networking, Asterisk, Asterisk@Home / Trixbox, Web Development, AGI, ARA, AEL
  • Phone, Email and SMS Reminders: Using text-to-speech engine, your own recorded voice or just text
  • Hourly and Contract Support, Consulting, Hosting, Office PBX, etc.

Abel Technology Services

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Nomado VoIP

August 20, 2015, 10:43 pm
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Nomado VoIP

Nomado is a telephony, pbx and SMS solution for businesses and private customers. With its equitable and progressive rates, Nomado provides simplicity and quality as effective communication tools for companies .

Nomado was recognized as "Best quality / Price ratio" according to Datanews.be

We offer global service support with our 7/7 online chat support.

Why more companies are switching to Nomado?
Companies found Nomado as complete VoIP package, it provides business communication solution both for incoming and outgoing calls. Companies realized the importance of call management with the help of pbx system. With Nomado, you do not need a separate company and separate set up for the pbx system because it is already incorporated with Nomado Plans.

What Services Do nomado Provide?

• First Class VoIP termination
• Cisco Ip phones SPA502G 505G and 525G2 (WiFi IP phone) Hardware page : https://www.nomado.eu/shop/category/add-ons-hardware-5
• SMS
• Virtual Phone numbers or DID , page https://www.nomado.eu/shop/category/add-ons-phone-numbers-4
• VoIP access gateway for mobile calls (Access or Callback services)

New: nomadoSMS API

Nomado website : https://www.nomado.eu/

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VOIP Service Providers Business Africa

August 20, 2015, 10:48 pm
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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:

Africa

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ITSP

August 21, 2015, 3:16 am
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ITSP = Internet Telephony Service Provider

A provider of VOIP Telephone service.
Also known as VOIP Service Providers.


See Also


White Paper on Starting an Internet Telephony Service Provider:

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

Unified Communications Telemarketing Services :

If you are interested in setting up a unified communications telemarketing b2b platform to offer hosted broadcasting services , Following page might help you.
B2B solution for SMS, Fax , Email and Voice Broadcasting

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SIP

August 21, 2015, 8:09 pm
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SIP, the session initiation protocol, is the IETF protocol for VOIP and other text and multimedia sessions, like instant messaging, video, online games and other services.

Abstract from the RFC 3261 (formatted_and_explained version) - SIP: Session Initiation Protocol

This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.

SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.


SIP is very much like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for phone calls are defined in SDP -the session description protocol.

  • SIP is a text-based protocol that uses UTF-8 encoding
  • SIP uses port 5060 both for UDP and TCP. SIP may use other transports

SIP offers all potentialities of the common Internet Telephony features like:
  • call or media transfer
  • call conference
  • call hold

Since SIP is a flexible protocol, it is possible to add more features and keep downward interoperability.

SIP also does suffer from NAT or firewall restrictions. (Refer to NAT and VOIP)

SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The following features of SIP play a major role in the enablement of IP telephony and VoIP:

  • Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
  • Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported recognizing that not all the parties can support the same level of features. For example video may or may not be supported; as any form of MIME type is supported by SIP, there is plenty of scope for negotiation.
  • Call Participant Management: During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
  • Call feature changes: A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as 'voice-only', but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call
  • Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call, enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.

The SIP protocol

The SIP protocol defines several methods. ...
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