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  • 08/08/15--23:55: 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.

    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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  • 08/08/15--23:56: Predictive dialer
  • What Is A Predictive Dialer?


    • "A predictive dialer is a computerized system that automatically dials batches of telephone numbers for connection to agents assigned to sales or other campaigns. Predictive dialers are widely used in call centers." - Wikipedia

    "Definitions of Predictive dialer on the Web:

    • A predictive dialer is an outbound call processing system designed to maintain a high level of utilization and cost efficiency in the contact center. The dialer automatically calls a list of telephone numbers, screens the unnecessary calls such as answering machines and busy signals, and then connects a waiting representative with the customer.

    note: above text may be copyrighted by it's respective owners)

    A VOIP Predictive Dialer, a.k.a. soft predictive dialer, is a software product capable of predictive dialing using VOIP service directly. Besides computer and internet connection, there is no equipment needed in order to use VOIP predictive dialer.


    Software Only Predictive Dialer

    New predictive dialing technology, together with faster computers and bigger broadband bandwidth, enables software only predictive dialers to work as good as or even better than hardware based dialers. Software based solution avoids expensive telephony board and associated hardware maintenance cost. It is easy to install and configure. For example, it is very easy to setup remote agent (at home agent).

    SOFTWARE ONLY PREDICTIVE DIALER

    • Unique Call center Suite Predictive/ Progressive/Auto Dialer and Web CRM Solution. For more details contact: avil@uniqueperform.com

    • Ameyo Predictive Dialer Ameyo Proactive Outbound is equipped with capabilities that power multi-channel campaigns and proactive outbound communications across multiple channels (voice, SMS, social, and web). Ameyo dialer is the heart of proactive outbound communications, that offers outbound capabilities that ensures communications compliance with legal regulations and harbours loyal and long-term relationship with customers.

    • Texo.cc auto dialer is an Agile Customer Interaction hub on cloud, enabling you to build Personalized engagements across the customer journey. Texo.cc empowers users to quickly innovate with minimal technical Intervention, thereby challenging the status quo of traditional inbound, outbound, or blended contact centers. Contact Texo.cc for the best voip predictive dialer. The Texo.cc auto dialer has been trusted by leading outbound sales teams worldwide, checkout Texo.cc predictive dialer reviews for more.


    Aavyukta Intel e Call Dialer and VoIP Solutions to Call Centers: Predictive Dialer (Unlimited Seats) + VoIP (US/UKLL/Canada) + Hosted/Cloud Server @ 1 US cent/Min, Reach us on www.dialerphilippines.com or catch us on skype on id avyukaindia +91-9549999916

    • 3CLogic Cloud-Based Contact Center Solutions - No hardware is required. Agents can be working from home or multiple locations. Automatically initiate contact with the next prospect before a rep finalizes a call, reducing call center queue times and operational costs. With 3CLogic, you will have complete visibility into call center operations including advanced scripting and reporting.
    • Astral Predective Dialer Solutions from

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  • 08/10/15--02:27: voip-info.org
  • Welcome to the VOIP Wiki - a reference guide to all things VOIP.


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

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


    NEWS


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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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  • 08/10/15--18:53: Old News
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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.

    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. ...

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  • 08/11/15--14:43: 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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  • 08/11/15--15:38: Asterisk config vxmld.conf

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  • 08/11/15--15:47: Asterisk cmd Vxml
  • Vxml()

    Synopsis


    Execute a VoiceXML document over Asterisk (Based on the i6net VXI* VoiceXML browser).
    The application use Asterisk internal API (Prompt / DTMF / Record) and installed applications.

    Description

    Vxml(url)

    Features :

    - Audio (play and record gsm, wav, WAV files)
    - Video (play and record h263, mp4, 3gp files)
    - DTMF (bargein support)
    - Transfer (use Dial/Transfer applications and to exchange with Asterisk function/variables too)
    - Text To Speech (most TTS supported with HTTP connector, and Festival/Flite and unimrcp applications)
    - Automatic Speech Recognition (Nuance, Lunenvox, Verbio, Vtech, VoiceInteraction, Vestec, use Asterisk Speech API or unimrcp )
    - Accounts for hosting (ranges, url, stats, max limitations)

    Documentations

    - Installation guide
    - Developer guide

    Configuration files

    - vxml.conf
    - vxmld.conf

    Variables

    If the variable VXML_URL(or VXML_URL2) has been set when vxml is run, the value of that variable will be used as the URL if not parameter is set to the application.
    If the variable VXML_ID has been set when vxml is run, the VoiceXML session ID variable called “telephone.id” is set with this value (in the VoiceXML execution session context).
    If the variable VXML_PARAM has been set when vxml is run, the value of that variable will be used as “telephone.param” (in the VoiceXML execution session context.

    After execution, the VoiceXML result passed with the <exit> tag and the property ‘expr’ is accessible with the variable VXML_RESULT.

    Asterisk Dialplan example

    [incoming]
    exten => s,1,Answer
    exten => s,n,Wait(1)
    exten => s,n,Vxml(http://links.i6net.com/index.vxml)
    exten => s,n,Hangup

    VoiceXML syntax


    CLI commands

    - vxml show version
    - vxml show license
    - vxml show configuration
    - vxml show statistics

    Return codes

    Always returns 0.

    See also



    Asterisk | Configuration | The Dialplan - extensions. ...

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  • 08/11/15--17:56: 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.



    DID Providers by country

    Algeria

    • 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.
    • BuyDDINumbers.com 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.
    • iTalkWorld.com 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.
    • divertmycalls.com 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.
    • Buyvirtualnumber.com 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. ...

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  • 08/11/15--22:06: DID
  • Direct Inward Dialing Number (Also known as DID or DDI)


    DID (DDI) Background

    Most businesses have several incoming telephone numbers used for specific purposes. For example customer service, sales, etc. Some have an individual telephone number for each user in the system. In a home setting on the other hand, each telephone number comes in on a different pair of wires typically. This is not practical in a business environment that has many telephone numbers.

    Why was DID actually Created?

    So DID ("direct inward dialing") was invented as a way to re-use a limited number of physical phone lines to handle calls to different published numbers. In a business with DID, the phone company uses DID signalling to identify the number they are about to connect to the business's PBX. Historically, this was done by pulsing the last 3 or 4 digits of the number being dialed before connecting the number. The PBX would use these DID digits to switch the call to the right recipient.

    In modern PBX's, typically, digital methods (example: PRI) are used to do the same thing, ie. supply the "called party" information. But many business's still have old PBX's which use the analog signalling I mentioned before. The type of telephone lines used for analog DID are different than regular home telephone lines. Usually, battery voltage is supplied by the business PBX instead of the telco. Also, the telco signals a new call by bridging the line instead of by ringing the line. The receiving PBX signals back that it's ready to take the call by momentarily reversing polarity of the voltage on the line (this is called "winking" the line)

    Old Fashion Way: (PSTN WORLD)

    Direct Inward Dialing is used when your PBX telco connection allows direct dialing to extensions within a PBX, using physical lines (or channels on a PRI) on a shared basis. DID service consists of identifying the "called party" by using DTMF or by digital means, before connecting each call. The service can be sent over an E&M Wink T-1 as DTMF and also as D-Channel information on a PRI.

    On a PRI connection, the telco can send only the digits that differ between the group number and the extension (often four digits) or the whole number - it depends on the connection to the telco.

    DID (DDI) in the new VOIP World

    Let's say you buy a phone line from Vonage or some other phone service provider who offers phone service over broadband. The number that they provide to you, in technical terms is a DID number. This is the number that they have assigned to you to connect you to the old PSTN Networks around the world. Any service provider who wants to offer a phone service over IP address, needs to buy DID numbers from his CLEC or any other large service provider like Level 3 in the United States or go to a consortium (company that will take large blocks from many providers and hand them out one at a time)

    If you are using an IP PBX like Asterisk, and you want to connect yourself to PSTN so people can call your office, you can either

    1) Buy an Analog or E1/T1 card from Digium, OpenVox, Rhino Equipment Corp or Sangoma

    2) Buy DID number from DID service provider

    DID Service Providers, convert the analog to digital and provide these DID numbers over the Internet, with SIP or IAX2.

    Service providers such as
    Wholesale DID Numbers, BuyDIDNumber Fax ,Asterisk Supported BuyVirtualNumber

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  • 08/12/15--11:37: WebRTC
  • Synopsis

    The practical implementation of VoIP was started on hardware based IP Phones. The idea was well received and was transferred into the concept of Soft Phones or software based IP Phones. These softwares always required some additional installation to the native Operating System. Most common examples of Softphones or Software based SIP client is Counterpath's X-Lite and Bria.

    The Evolution of Software Development made it possible to translate or formulate equivalent of almost every desktop based application to web based application. This brought major shift in Software Industry as the web browsers are integral part of almost every Operating System. SIP clients, were also transformed into Web Extensions. Most of the time, Flash was used to develop such extensions however, it always required extra plugin installation, thus decreasing system performance, and increasing chance to troubleshoot as it required additional resources to be deployed. And this problem gave rise to the concept of WebRTC.

    Overview

    customLogo.gif.png

    WebRTC provides the functionality of realtime multimedia applications without any installation of additional plugins, downloads or extensions. The ideal form of WebRTC describes such web based Real Time Communication independent of Browser being used by user. It's a Javascript based API originally being developed to develop browser to browser communication applications for Voice, Video and Peer to Peer File Sharing tasks.

    Architecture

    The architecture of WebRTC, as described by W3C looks something like this:
    WebRTCpublicdiagramforwebsite (2).png


    Design

    Major components of WebRTC include:

    • getUserMedia, which allows a web browser to access the camera and microphone
    • PeerConnection, which sets up the audio/video calls
    • DataChannels, which allow browsers to share data via peer-to-peer

    Support

    Chrome WebRTC Development Team

    Discussion List: https://groups.google.com/group/discuss-webrtc
    Google Plus Page: https://plus.google.com/113817074606039822053
    Chrome WebRTC Issue Tracker: http://code.google. ...

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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


    Alcazar Networks Inc - Free toll-free termination
    Alcazar Networks - VoIP Services
    • Providing FREE toll-free termination to the US48 800 - 855 - 866 - 877 - 888
    • Full accurate Caller-ID number (ANI) delivered
    • Codecs supported: G711 and G729
    • NO SIP REGISTRATION REQUIRED
    • If you require registration we accept ANY credentials so you can start passing calls immediately.

    ArcTele Communications, Inc - Toll Free Termination
    http://www.arctele.com
    ArcTele Communications, Inc offers a Toll Free termaintion gateway service free of charge.
    - We send your callerID
    - ULAW and G729
    - Send calls in the format of 18XXXXXXXXX
    - Unauthenticated to TOLL FREE NUMBER@tf.arctele.com port 5060

    Broadvox is a leading wholesale VoIP service provider that delivers reliable VoIP solutions for domestic and international businesses.

    Broadvox has 8YY number toll-free termination services that provide switching for 8YY calls originated by customers worldwide; conduct the database lookup and determine to which Interexchange carrier (IXC) the toll free number belongs; then route the call appropriately for call completion.

    For more information, to get a FREE quote on our rates or call us at 216.373.4800 for a FREE consultation


    Call With Us
    https://www.callwithus.com/ , under the heading on the page, "Toll free service:"
    • no registration required, simply send SIP messages to tf.callwithus.com
    • Codec is G.711u. ...

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  • 08/14/15--04:51: VOIP Event Calendar
  • 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.

    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

    • 23

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    High Availability (HA) is normally achieved through "clustering" - which means two machines acting as one for a specific purpose. There are many ways to create a cluster, each with its own benefits, risks, costs, and trade-offs. The terms "High Availability" (HA) and "Clustering" can be overused so beware of the hype. Clustering, and HA have specific (and different!) meanings. If you are responsible for creating a high availability cluster for Asterisk, below are the issues and concepts you should be aware of. This page is intended to be a starting point in the design, creation or selection of a High Availability or Clustering solution for Asterisk.

    Note that if you are designing a call center for PSAP (Public Safety Answering Point) / 911 call centers there are specific requirements you must consider. Some are noted below, others are specified by rules/orders from FCC (USA), CRTC (Canada), and similar country specific organizations. (eg: FCC 05-116 order 10). Even if you are not designing for a PSAP, these guidelines are excellent best practices often applied by large commercial call centers.

    Please do not add specific product names/links to this page, it is intended to be product neutral. Don't say "this is the best" because your product/your favorite product uses it.. Stick to facts please.

    Co-Dependence and Autonomy

    This criteria is among the most important (if not THE most important) criteria when designing/selecting/building a high availability telephony environment. In order to be a true cluster, the machines (or "peers") must be autonomous. Some HA solutions involve sharing hardware, software, a logical device, etc .The problem with this approach is that you create a single point of failure. For example, if a cluster shares a hardware channel bank (eg: connected to 2 machines via 2 USB cables), then if the channel bank fails the entire cluster fails. As another example, if a cluster shares a disk (eg: DRBD), then corruption of the disk content from a failing peer immediately corrupts the disk content of the other peer. In a true cluster the peers must be autonomous; i.e. not share any hardware, software, logical devices, etc. (Beware of some solutions which place a single device in front of Asterisk servers - creating a single point of failure).

    Telephony devices in true high availability environments do not share any logical/physical resources. For example, in emergency call centers/PSAP's nothing on the call path is shared: from clustered PBX's, to separate switches, to clustered routers (HSRP/VRRP) to the trunks. Each peer (whether PBX or router or other) must survive the destruction of its peer. (NG911 Section IV.C). ...

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  • 08/14/15--11:55: Call Center Monitoring
  • Companies offering call center monitoring services or software solutions offer concrete methods for assessing, assuring, and improving the performance of call center agents. Call center monitoring is achieved through metrics such as evaluations, comparative analyses, and feedback.


    Call Center Monitoring

    Call center monitoring is accomplished through three basic actions: measurement, analysis, and feedback. This creates a continuous loop of assessment and calibration based on objective metrics and scoring gleaned from call recordings, real-time analytics, and other forms of evaluation.

    For call center metrics to be most effective, the statistics and reporting must be considered relevant, practical, and above all, objective by the call center agent.

    Traditional call center metrics often rely on two key aspects:

    • Speech analysis
      • Phonetic indexing
      • Transcription
      • Multi-speaker speech separation (Role recognition)
      • Emotion detection
      • Talk-over analysis
    • Call flow
      • Call duration
      • Call time
      • Number of call transfers
      • Number of call holds
      • Hold duration

    Newer features of speech analysis, such as emotion detection, reflect a blending of competing viewpoints (acoustic features only vs. linguistics-only philosophies) and the technologies that can accommodate them both.

    Acoustic features include:
    • Volume
    • Tone
    • Pitch
    • Intensity
    • Inflection
    • Rate of speed

    Linguistic attributes include:
    • Words
    • Pauses
    • Stops
    • Hesitations
    • Laughter
    • Sighs

    Emotion detection can create a more layered, nuanced approach to call monitoring, ensuring that the overall context (rather than simple word frequency) provides a fuller picture, especially if that picture is one of customer frustration.

    Another new tool in speech analysis is talk-over analysis. Simultaneous crosstalk between customers and representatives is a source (and indicator) of frustration. Talk-over analysis can also pinpoint silences, which can imply a knowledge gap and a potential improvement area to target.

    Real-Time Analytics

    Real-time analytics are increasing in popularity with many call center managers. Recorded calls can take days to index, depending on factors such as how much data there is and where it's housed (on-premises or hosted). Real-time analytics can help call center agents to regulate and improve their CRM performance while it's most critical — as it's happening.

    The real-time call center monitoring and reporting offered by some call center monitoring services and software give call center managers quick access to data on groups and individuals. Some call center solutions take real-time analytics a step further by integrating real-time analytics into the call center routing procedures. ...

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  • 08/14/15--15:42: Sip Trunking Providers
  • 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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    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. ...

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  • 08/17/15--11:38: BULK SMS
  • Bulk SMS, also known as bulk messaging or bulk text messaging is the act of disseminating SMS messages in large numbers so they are able to be delivered to various mobile phone terminals. This form of messaging is typically utilized by consumer brands, banks, enterprises and media companies. The messages are generally used for mobile marketing, enterprise and entertainment. However, banks often use them for fraud control. For example, if criminals are circulating a fake email that is asking people who have accounts at a certain bank to provide their social security numbers or other confidential information, these text messages can alert people to the scam so they do not fall victim to it. Bulk messaging is often utilized for reminders and alerts. However, it is more frequently used to send communications and information between customers and staff of various companies. Bulk messaging enables the delivery of SMS messages to large numbers of mobile phones that are located all around the world.

    Bulk messaging software

    In order to receive and send bulk messages, software is needed. There are many types of software packages specifically designed for this task that are available. These packages give their users the ability to send messages to as many phone numbers as they want. There are many different ways in which these phone numbers can be managed.

    The vast majority of software applications that are designed to be used with SMS enable the user to upload mobile phone number lists with the use of a CSV or TXT file. Systems that are more advanced are capable of automatically deleting any numbers that are repeated. There are also systems that can be programed to validate all of the mobile phone numbers before the messages are sent to them.

    Enhanced software features are also currently available that allow users to schedule messages to be delivered at certain days and/or times. Bulk messages are also able to be sent on mobile networks that are international or national, assuming that the provider of the bulk messaging software sends internationally.

    Bulk messaging portal

    Bulk messaging features can be added to websites through the use of this specific online script. Unlimited mobile phone numbers can be added to the list of numbers to send messages to. There are a wide variety of ways that can be used to manage these numbers.

    Bulk messaging API

    The majority of services that handle bulk messaging use the API's (Application Programming Interface) listed below. These enable the addition of functionality to programs by their programmers:

    • Email
    • HTTP
    • SMPP (Short Message Peer to Peer)
    • FTP (File Transfer Protocol)

    Immediate benefits of using bulk SMS messaging

    When a particular business is not doing well financially, they need to utilize various tools that can help them gain a competitive advantage in their specific industry. One of the main reasons that bulk SMS is so popular is its ability to lower operational costs while also generating revenue at the same time. Bulk SMS might be the only medium that is able to show a return on investment that is able to be measured. Wholesale SMS messaging is targeted, which makes it extremely effective at getting people to respond and generate revenue.

    Reduces operational costs

    Bulk SMS message transmission is more effective than email and less expensive than voice calls. There are thousands of businesses located all around the globe that utilize wholesale SMS as a way to communicate with their suppliers, employees and customers. There is a significant cost savings as a result of time being saved because actual voice calls to suppliers, employees and customers do not need to be made. A single message can instantly be sent to many people at the same time, as long as the person is located in an area with mobile coverage. The ability to disseminate information so quickly to a large target audience reduces communication costs while also generating revenue if used for marketing purposes.

    Allows customers to be accessed easily

    More people have mobile phones than have access to email or landline phones. Every mobile phone supports the use of text messaging. All mobile phone users are comfortable using this technology because it is simple and easy to understand. This makes wholesale SMS the perfect medium to use for communication with customers. There are also no demographical or geographical restrictions. ...

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  • 08/17/15--11:59: Voicepulse
  • VoicePulse FIVE


    To learn more about VoicePulse FIVE and chat with a Live Representative click here.



    VoicePulse FIVE is our fifth generation VoIP platform serving the residential, SMB and Wholesale segments. FIVE is a completely rewritten platform designed from the ground up to provide Internet telephony in the most powerful, flexible and easy to use way possible. Sign up for a free, no obligation evaluation account and be up and running in just minutes.

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

    Company News

    2015-08-11 VoicePulse launches CallCode, a call flow feature that allows users to write Javascript code to integrate with third party platforms.
    2015-07-17 New Feature Release - Real-time 24 hour usage, Usage Limit Reset, Calling Access
    2014-11-06 VoicePulse Introduces VoicePulse FIVE, the Next generation in VoIP Services

    Pricing

    Business Gateway Pricing click here.

    Ideal for SMB and Enterprise

    Use your existing SIP enabled on-premise softswitch or PBX to make and receive phone calls regardless of call volume.

    Channels

    Channels are included with your Business Gateway at no additional cost

    Endpoints

    Endpoints are included with your Business Gateway at no additional cost

    Phone Numbers

    Choose your number from anywhere in the US or move your existing DIDs to VoicePulse FIVE. There is no fee to port your phone number to VoicePulse FIVE.

    Incoming Calls

    Incoming calls to a U.S. phone number are $.01 per minute

    Incoming Toll-Free Calls

    Incoming calls to a U.S. toll-free phone number are $.029 per minute.

    Outgoing US & North American Calls

    Outgoing calls to the US48 are $.02 per minute

    Outgoing International Calls

    Competitive International rates.
    Look up international termination rates here.


    Account Center


    • View your current Statement Balance
    • Monitor real time costs for usage, Endpoints, Trunks, Gateways, Call Apps, Channels, and E911
    • Make instant payments by credit card
    • Add unlimited Channels or call paths to your Trunk
    • View your active phone numbers
    • See our inventory of numbers
    • Instantly activate new numbers
    • Manage E911


    Customer Support



    Chat with a Live Representative. M-F 9am to 5pm EST


    Supported Protocols


    • Session Initiation Protocol (SIP)

    Supported Codecs


    • G.729a
    • G.711ulaw
    • G. ...

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