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Asterisk-based commercial PBX

June 19, 2015, 12:56 am
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Here is a list of producers of ready made, black box PBXs that are based on Asterisk (in no particular order):

Ecosmob Technology - Custom PBX Software Development Service Provider

Ecosmob Technologies is a VoIP solutions provider company which provides custom PBX software development, configuration and remote support services based on client requirements.

EBSolution - Custom IT Solutions

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

Wayatone Media Inc. - Communication

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


Svanto.net - Tailored Internet telephony solutions

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

AAB Asterisk Consultant Lithonia GA

  • IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices.
  • Support for digium / openvox / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards /grandstream
  • Asterisk IPPBX

4PSA VoipNow

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


Alpha Computer Group - IT and Telecommunication Engineers

  • Web Site:
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VoIP Termination

June 19, 2015, 5:06 am
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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 Origination

June 19, 2015, 5:22 am
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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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DID Service Providers

June 19, 2015, 6:09 am
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A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

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.
  • SIPMarket is providing DIDs in more than 150 countries. Direct numbers with incoming SMS available. Incoming SMS is free.


Algeria

  • Incorpus TeleNetworks Incorpus provides DID of 50+ countries. Just visit out website and live chat with us for details. Cheap DIDs available at low costs and discounts for bulk orders. No per minute charge. Only monthly and go on
  • 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.
  • DIDx.net Algerian | Algeria Virtual DID numbers whole sale pricing check out today.
  • DomesticNumbers.com Algerian Virtual Phone Numbers € 7.50/month including free PBX. Forwarding to Skype, Google Hangouts, FreePBX, Asterisk, Voipbuster or other VoIP provider, PSTN, or with free sip account.
  • 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. ...
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adtran

June 19, 2015, 10:00 am
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This should apply to US/North American Adtran TSU 600 + T100P. I've got 6x quad FXO cards, so this may not apply to the FXS models.

In the menuing system of the TSU 600, go to the config section and verify the following settings:
1) Network (NI)

format: ESF
code: B8ZS
yel alarm: ENA
xmit prm: Off
clock source: network
set lbo: 0.0 dB (if cable is short)
Inband Lpbck: ??
Bit Stuffing: ??


7) Port Config
Cycle through all available ports and verify they are FXO+. Mine came shipped from ebay in this mode.
Then hit enter on each one and verify that they are all in FXO_LS mode.

Now plug the T1 Crossover Cable into the "Network" slot in the back of the Adtran unit (I made one myself, it shipped with a staight-through T1). Plug all appropriate telephone cables into the FXO slots in each module, and connect them to the wall outlet.

I'll assume that you've already installed * and it is working from your SIP phones (i.e. you can connect to the sample server).

Edit your /etc/zaptel.conf and add the following lines:

span=1,0,0,esf,b8zs
fxsks=1-24

Edit your /etc/asterisk/zapata.conf and add the following lines (at the bottom):

group=1
context=default
signalling=fxs_ks
channel => 1-24

Now, you'll need make sure that the module is loaded. Do this by typing:

# modprobe zaptel
# modprobe wct1xxp

Verify that things are working by:

# ztcfg -vv

You should see something like:

Zaptel Configuration
======================

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Slaves: 08)
Channel 09: FXS Kewlstart (Default) (Slaves: 09)
Channel 10: FXS Kewlstart (Default) (Slaves: 10)
Channel 11: FXS Kewlstart (Default) (Slaves: 11)
Channel 12: FXS Kewlstart (Default) (Slaves: 12)
Channel 13: FXS Kewlstart (Default) (Slaves: 13)
Channel 14: FXS Kewlstart (Default) (Slaves: 14)
Channel 15: FXS Kewlstart (Default) (Slaves: 15)
Channel 16: FXS Kewlstart (Default) (Slaves: 16)
Channel 17: FXS Kewlstart (Default) (Slaves: 17)
Channel 18: FXS Kewlstart (Default) (Slaves: 18)
Channel 19: FXS Kewlstart (Default) (Slaves: 19)
Channel 20: FXS Kewlstart (Default) (Slaves: 20)
Channel 21: FXS Kewlstart (Default) (Slaves: 21)
Channel 22: FXS Kewlstart (Default) (Slaves: 22)
Channel 23: FXS Kewlstart (Default) (Slaves: 23)
Channel 24: FXS Kewlstart (Default) (Slaves: 24)

24 channels configured.

If you want to be able to dial out, then add this to your extensions.conf file:

exten => _91XXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})

Now start up asterisk using:

# asterisk -vvvvc

Congratulations! Asterisk should now be working with your new (or gently used) Adtran TSU 600 + Digium T100P.


Where to buy

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Advanced Network Devices

June 19, 2015, 10:19 am
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color logo small.bmp

Website: www.anetd.com
Support: www.anetdsupport.com
Forum: www.anetdsupport/forum.com

Products:

BI6I7812.png
Advanced Network Devices IPClock


Advanced Network Devices Power over Ethernet (PoE) Products That Easily Connect to Existing Data Networks:


IP Clock and IP Speaker are Power over Ethernet (PoE) synchronized clocks and intercoms that requires only an RJ-45 connector to connect to existing data networks. Simultaneously, broadcast to both phones and speakers. The clock auto synchronizes and can be used as a scrolling text display and a standard built in microphone allows two way voice communication. Works on existing data networks and inter-operates with Cisco and other VoIP phone systems. Full IP product line can be used together to meet specific needs. Send broadcasts to specific rooms, departments, or groups as needed. Bi-directional audio / intercom supported on most models.


For more information contact;
sales@anetd.com
847-463-2236

Technical support contact;
tech@anetd.com

Supported Software

IP Clockwise
Singlewire- Informacast
Syn-Apps- SA-Announce
IPcelerate- IPsession
Bell Commander
MessageNet

Where to Purchase

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Cisco

June 19, 2015, 10:26 am
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Cisco Systems, Inc

Cisco's IP telephony products offer great flexibility, a large feature set, and stand up well to every day corporate use. That being said, they're also not the cheapest in the field.

Products


Most likely you'll need access to firmware packages for upgrades or version changes, i.e. converting a phone from SCCP to SIP. The files are available on Cisco's website along with a good amount of documentation but you'll need a service contract. The service contract for a Cisco phone is around $8/year. The service contract price will be based on the value of the item you want covered. Cisco has several support options. You just need the most basic support in order to be able to get updated software.



GKTMP

GKTMP is Cisco's open, but proprietary, protocol used for communication between a call control device such as a Cisco Gatekeeper or SIP Proxy and an external application such as a route server. GKTMP, which stands for GateKeeper Transaction Message Protocol, was originally developed for H.323 gatekeepers but since it is a rich and lightweight Operations and Billing Support System (OSS/BSS) protocol, it is equally useful as an API to external applications for SIP proxies or B2BUAs. A major benefit of GKTMP is the ability to offload complex routing algorithms, such as least cost routing tables with hundreds of thousands of routes, to an external server.

An open source module which provides a GKTMP interface to open source OSP servers, such as OpenOSP and RAMS on www.sipfoundry.org/OSP, is available at sourceforge.net/projects/gktmp-to-osp.

Service Contract


Obtaining a service contract directly from cisco can be a lot of fun. When last checked, there's no online registration available, only a page with an email address that's no longer valid. ...
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Call Center Software

June 20, 2015, 1:38 am
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Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

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

Types of Call Center Software


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

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

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

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

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

Contact Center Software

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

Call Center Software Providers

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

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

June 20, 2015, 1:42 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

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.
  • 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 .
  • 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.
  • 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. Our ACDs include skills based routing and sophisticated operator productivity algorithms.
  • AVOXI Provider of
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CTI

June 20, 2015, 1:46 am
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What Is CTI?

The following is a definition for CTI from Wikipedia:

  • "Computer telephony integration (CTI) is technology that allows interactions on a telephone and a computer to be integrated or co-ordinated. As contact channels have expanded from voice to include email, web, and fax, the definition of CTI has expanded to include the integration of all customer contact channels (voice, email, web, fax, etc.) with computer systems."

This technology incorporates analog, digital and VOIP phone platforms.

See Also

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IVR

June 20, 2015, 1:55 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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Call Center Monitoring

June 20, 2015, 2:00 am
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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.

Page Contents


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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VOIP Service Providers Residential

June 20, 2015, 2:07 am
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voip service in India

June 20, 2015, 2:14 am
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VOIP SERVICE PROVIDER BANGLADESH

June 20, 2015, 2:18 am
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voip service providerVirtual Phone Number, B2B Lead Generation, Employee Tracking System, Customer Lead Management, Lead Identification, Lead Grading, Lead Distribution, Lead Nurturing, Toll Free Number, Virtual Receptionist Services,

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DID Service Providers

June 21, 2015, 10:15 am
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A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

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.
  • SIPMarket is providing DIDs in more than 150 countries. Direct numbers with incoming SMS available. Incoming SMS is free.


Algeria

  • Incorpus TeleNetworks Incorpus provides DID of 50+ countries. Just visit out website and live chat with us for details. Cheap DIDs available at low costs and discounts for bulk orders. No per minute charge. Only monthly and go on
  • 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.
  • DIDx.net Algerian | Algeria Virtual DID numbers whole sale pricing check out today.
  • DomesticNumbers.com Algerian Virtual Phone Numbers € 7.50/month including free PBX. Forwarding to Skype, Google Hangouts, FreePBX, Asterisk, Voipbuster or other VoIP provider, PSTN, or with free sip account.
  • 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. ...
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maple4VOIP

June 21, 2015, 10:38 pm
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MapleLeaf Technologies

June 21, 2015, 10:39 pm
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voip-info.org

June 22, 2015, 4:12 am
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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.


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