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  • 12/30/15--06:54: SIP
  • 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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  • 01/02/16--08:30: International calling cards
  • International calling cards allow you to preload money onto a card, and then use the card to make international calls.

    PowerUP cards from AnnSystem

    By the end 2012 AnnSystem launched PowerUP line of prepaid calling cards from United States to India and South Korea. Later, 2013 AnnSystem also launched calling cards for Turkey and as December 2013 AnnSystem has launched calling cards for Latin America.

    Calling cards available from AnnSystem are:
    • PowerUP South Korea $5, $10, $20, $50
    • PowerUP Turkiye $2, $5
    • PowerUP India $2, $5
    • PowerUP Latin America $2, $5

    For latest rates visit AnnSystem PowerUP

    Company website is AnnSystem

    • Cheapest International Calling Cards - Making International calls has no longer been a costly affair owing to the availability of international calling cards in markets. Shifatel offer cheapest international calling cards and plans gathered from all the top phone network service providers all over the world. provides premium India Calling Card and Pre-Paid India Calling PIN Services for Indians living in the U.S., Canada and India. We also serve Indian Business Travelers and Indian Students requiring a reliable, low cost India card service to connect to their business or family in India. NEW!! Call to India – and Call from India with the most Powerful, Multi-Feature, PrePaid India Calling Card in the US, CANADA and INDIA!
    TelCan now offers 8 ways to Call India: Toll Free, Local Access, VoIP Apps, Access using Skype and MagicJack and even One Touch Calling. Try Now!
    TelCan make you possible to do cheap calls to India from USA, and cheap India calling from USA in lowest possible rates and without any compromise in quality. TelCan offers you the preeminent, reliable and most economical calling cards from any other.

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    This page is for all of the different Calling Card Solution Providers out there that have Turn-Key Solutions to offer. Please post info about your solution with links to the Data Sheets or White Papers as well as a way for companies to contact you if they are interested.

    Advancefone Calling Card & VoIP Provider Platform

    Advancefone provides the complete solution to become a Calling Card Provider
    • Bulk Calling Card Generation for printing or online sale
    • White-label, Front-end Portal for customers and admin
    • Multi currency
    • Managing Rate cards
    • Managing Dial plans
    • Managing LCRs
    • Creating Vouchers
    • Managing Customer Payments
    • Online payments through major payment gateways e.g.Paypal
    • Email alerts
    • Add Customers
    • Manage Rate Plans

    Advancefone provides the complete solution to become a VoIP Provider

    • Online Signup and Password retrieval
    • Manage Rates
    • APIs (HTML)
    • Receive Payments online
    • Payment History
    • Call History
    • Vouchers
    • Built in Payment gateways
    • Invoices
    • Pinless Dialing
    • Password Management
    • Real-time billing, rating and routing
    • Account balance monitoring
    • Call back module
    • SMS module
    • Calling Card module

    Advancefone provides Cheap Multi Channel DIDs for Calling Card Providers to use as Access Numbers

    • Pay per month
    • Increasable Channels
    • No per minute charge
    • Available in more than 40 countries

    Contact us for details.

    Advancefone Ltd.
    Advancefone provides inbound (DIDs) and outbound (A-Z Termination) services. Advancefone runs independent network. Unlike other operators we own the resources. Advancefone focuses on the quality end of the market rather than the LCR or grey route market. Advancefone offers services for Personal & Business use, Call Centres, Call shops, Calling Card Providers, VoIP Providers (inbound/outbound).


    Aloha Wholesale

    Free UK DIDs for calling cards. For customers who recieve more than 2,500/minutes a day we also pay out £0.001/min on calls received. Capacity is unlimited and welcome small and large calling card operators.

    Contact us today.

    Aloha Wholesale | Part of the Aloha Telecommunications Group, a UK National operator. Aloha Wholesale provide inbound (DID/DDIs) and outbound (UK National and A-Z Termination) services. As a UK national operator with our own independent network. Unlike other operators we own the resources we sell in regards to inbound (i.e all numbers are allocated to us via Ofcom the regulator). Aloha focuses on the quality end of the market rather than the LCR or grey route market. ...

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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:
    • Email:
    • 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:
    • Email:
    • 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:
    • Email:
    • 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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    Recently a few services have popped up that allow "free" calls to various international points. There are all based in the U.S. state of Iowa, and the way they work is that you call a regular number in Iowa, wait for a prompt, then dial the number you want to call. As best anyone can determine, there must be some quirk in Iowa telecommunications law that allows them to charge higher than normal termination charges on connected calls, and they use VoIP to complete the calls, so they actually wind up making anywhere from a fraction of a cent to a few cents on each minute of use. If the Federal Communications Commission ever reforms the ridiculous compensation system we have here, these companies will probably be out of business overnight.

    NOTE: on January 29, 2007 AT&T filed a lawsuit that "seeks to stop FuturePhone as well as the telcos who provide local infrastructure from continuing with their operations that use regulatory-fee arbitrage and VoIP to provide international calls for only the price of a long-distance call to Iowa." (quote from linked article). This has already resulted in some of these companies discontinuing or modifying their service.

    For now, if you have "free" calling within the USA, and can place calls to the Iowa numbers that are still in operation, you can then get connected through to numbers in other countries, including Mexico (which is a fairly high cost call on many U.S. long distance plans) at no additional charge.

    The challenge, then, is whether Asterisk/FreePBX can work in this manner. Assuming a user dials an international call to certain countries, select an outbound trunk (that allows completion of U.S. calls) and then do the following:

    1) Connect to a particular U.S. number (NOT the number dialed by the user),
    2) Wait for the other end to answer, then wait until it is ready to receive digits (a couple seconds timeout may work),
    3) Send DTMF tones, possibly interspersed with waits, to the other end, representing the number the caller originally dialed, but the number MIGHT have to be massaged a bit. For example, if the caller is in a country where you dial 00+country code+number and the U.S. system expects 011+country code+number, you'd need to strip the "00" and add the "011." Also I believe one of the systems has an initial prompt that is something like "press 1 for English, press 2 for Spanish" so in some cases there may be a need to prepend additional digits and/or wait states.

    Now, I have determined that it IS possible to have Asterisk send post-connect DTMF digits and waits. For example, if you were to add this to extensions_custom.conf in FreePBX or Trixbox:

    exten => *20,1,Dial(Local/18005558355@outbound-allroutes,,D(wwwwww932))
    exten => *20,2,Hangup

    Assuming you had a route for U.S. 1-800 numbers, when the user dials *20 it would place a call to the "TellMe" service, then wait three seconds, then send DTMF for 932 ("WEA") which would take you directly to the WEAther forecast section (to further automate it for a particular location, you could add a few more waits - I've found that wwwwwwww is about right - after the 932, and then a five-digit U.S. zip code). The point is that the caller never hears the tones that are being sent because the audio path doesn't connect until after the digits have been sent (which would make this technique useful for sending account codes, etc.)

    I will list a few of these "free" international call services below. Remember that the numbers you have to call are NOT toll free, so they are only "free" if you have unlimited U.S. calling, or are using a cell phone with "free" airtime, etc. Also, I have no connection to any of these companies, so I cannot say whether they might save the calling or called numbers for any reason - obviously if you are highly concerned about privacy of communications, you would want to investigate these services further before using them. Please note that the list of countries that can be called through each provider may change without notice. ...

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  • 01/06/16--01:26: How to start a VOIP Business
  • 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.

    Asterisk SIP Trunking - US — Wholesale Reseller User Portal and Admin Portal. With this 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 Account and subscribe to the CIM for remote PCI protection of credit card numbers. When your customers pay the money is sent to your bank for each nightly batch settlement. The funds are transferred directly to you and our services are billed to your account at a wholesale rate. Call now: 877-686-4787 or visit us:

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  • 01/06/16--02:41: 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

    • DIDWW - the source for wholesale International DIDs and Toll Free Virtual Numbers. We provide SMS enabled DIDs in Canada, Israel, Russia, UK, Ukraine and growing.
    • 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


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

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  • 01/06/16--04:46: VOIP Event Calendar
  • May 2016

    January 2016

    • 25 -28 ITEXPO Florida - ITEXPO is the business technology event bringing together service providers, enterprises, government agencies, resellers, vendors and developers to discuss the latest innovations that are changing the marketplace.
    • 20 -22 Convergence India 2016 Expo - Convergence India expo is the most influential and relevant trade event for the ICT, broadcast and digital media industry in South Asia. The expo is a powerful platform for telecom operators, service and content providers, equipment and device manufacturers, information & network security, broadcasters, broadcast equipment, cable & satellite companies, digital media etc.

    2015 VOIP Related Events:

    December 2015

    November 2015

    • 09 – 10 OpenSIPS Summit 2015 in Austin, TX - Come and help us "Keep Austin Wierd" at our eclectic venue, The Contemporary Austin Jones Center. We'll be presenting, via presentations and workshops, all the great additions we have in OpenSIPS 2.1 - webRTC, async queries, SIP compression, fraud detection and many others. The Summit is not only about 2.1, but also about tools, platforms or engines which are OpenSIPS interconnected (like billing, interfaces and others). We'll also have presentations from companies using OpenSIPS in their production environments, giving us insight into it's real world applications.
    • 11 – 13 OpenSIPS LIVE BOOTCAMP 2015 in Austin, TX - Enjoy a rare opportunity to train with the founder of the OpenSIPS Project Bogdan-Andrei Iancu! As part of OpenSIPS week in Austin, we'll be holding a LIVE training following the conclusion of the OpenSIPS Summit on Nov 10th. If you've ever wanted to become an OpenSIPS Certified Professional, this is you're chance! You'll have the rare opportunity to train live with the founder of the OpenSIPS Project Bogdan-Andrei Iancu! It's not a secret that there is a growing demand for OpenSIPS Certified Professionals all over the world. This training is intended to offer you a 50/50 view between theoretical and practical applications of OpenSIPS. It covers advanced topics that are critical to telecom network operators in regards to security, scaling, and integration. For participants that would like an introduction to OpenSIPS basics and to qualify for prerequisites to the advanced topics, there will be a 2 day online primer prior to the live training.

    October 2015

    • 06 – 08 ITEXPO Anaheim 2015 - ITEXPO brings together service providers, enterprises, government agencies, resellers, vendors and developers to discuss the latest innovations that are changing the marketplace. Conference topics include WebRTC, cloud computing, unified communications, software defined networking (SDN), network functions virtualization (NFV), channel strategies and more. ITEXPO will be held October 6 - 8, 2015, at the Anaheim Convention Center in Anaheim, California.
    • 18 - 22

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  • 01/06/16--11:01: STUN
  • STUN (Simple Traversal of UDP through NATs (Network Address Translation)) is a protocol for assisting devices behind a NAT firewall or router with their packet routing. RFC 5389 redefines the term STUN as 'Session Traversal Utilities for NAT'.

    Note: The STUN RFC states: This protocol is not a cure-all for the problems associated with NAT.

    • STUN enables a device to find out its public IP address and the type of NAT service its sitting behind.
    • STUN operates on TCP and UDP port 3478.
    • STUN is not universally supported by VOIP devices yet. Support as of 2015 is fairly good, but legacy devices may lack it.
    • STUN may use DNS SRV records to find STUN servers attached to a domain. The service name is _stun._udp or _stun._tcp

    Definitions (from the RFC)

    • STUN Client: A STUN client (also just referred to as a client) is an entity that generates STUN requests. A STUN client can execute on an end system, such as a user's PC, or can run in a network element, such as a conferencing server.
    • STUN Server: A STUN Server (also just referred to as a server) is an entity that receives STUN requests, and sends STUN responses. STUN servers are generally attached to the public Internet.

    Various types of NAT (still according to the RFC)
    • Full Cone: A full cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host, by sending a packet to the mapped external address.
    • Restricted Cone: A restricted cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Unlike a full cone NAT, an external host (with IP address X) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X.
    • Port Restricted Cone: A port restricted cone NAT is like a restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can send a packet, with source IP address X and source port P, to the internal host only if the internal host had previously sent a packet to IP address X and port P.
    • Symmetric: A symmetric NAT is one where all requests from the same internal IP address and port, to a specific destination IP address and port, are mapped to the same external IP address and port. If the same host sends a packet with the same source address and port, but to a different destination, a different mapping is used. Furthermore, only the external host that receives a packet can send a UDP packet back to the internal host.

    Closing words (also from the obsolete RFC 3489)

    14.6 In Closing

    The problems with STUN are not design flaws in STUN. The problems in STUN have to do with the lack of standardized behaviors and controls in NATs. The result of this lack of standardization has been a proliferation of devices whose behavior is highly unpredictable, extremely variable, and uncontrollable. STUN does the best it can in such a hostile environment. Ultimately, the solution is to make the environment less hostile, and to introduce controls and standardized behaviors into NAT. However, until such time as that happens, STUN provides a good short term solution given the terrible conditions under which it is forced to operate.

    Standard documents

    STUN RFC RFC 3489, now obsolete (Oct 2008)
    STUN RFC RFC 5389 (Current as per October 2008)

    Update to STUN protocol

    STUN standard is currently has been rewritten with RFC 5389. ...

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  • 01/06/16--11:05: Ecessa
  • Ecessa-Logo.png

    Ecessa Corporation manufactures network control hardware and software for data and telecommunications. Since 2000, the company has focused on Internet, cloud and wide area network technologies to give business networks greater reliability, redundancy and the ability to confidently use more affordable types of bandwidth.

    Ecessa did this initially with PowerLink™, which bundles multiple WAN lines (T1, xDSL, Cable, ISDN, Wireless, Satellite, etc.) into a greater pool of available bandwidth while providing business continuity through ISP failover and redundancy.

    Next they added ShieldLink™, which incorporates all the capabilities of PowerLink™ and brings network security to the next level with a built-in firewall and VPN gateway.

    ClariLink™ incorporates all of the above features, plus provides real-time call failover for hosted VOIP. ClariLink optimizes call performance by managing traffic among multiple Internet links to avoid jitter, latency and congestion.

    ClariLink monitors all VoIP calls (incoming and outgoing) in real time. In the event of a WAN failure or call degradation (i.e. high latency, excessive jitter, packet loss, etc) it automatically moves active VoIP calls to better performing WAN links, without the call dropping or the user even noticing.

    This means that business VoIP users and providers never have to worry about another dropped or low-quality VoIP call. Ecessa is the only service/product offering this advanced feature.

    Ecessa's most advanced products, WANworX™ software-defined wide area network (SD-WAN) solutions, are the most scalable and flexible technology for improving WAN performance and ensuring Never Down™ network availability. WANworX enables enterprises with multiple locations to seamlessly connect all their locations plus connect to the Internet and cloud-based networks.

    By integrating multiple WAN links from different carriers, WANworX supports multiple paths for data to travel, and continuously monitors links and routes traffic over the best-performing links, ensuring not only seamless uptime but also predictable application performance, even when carrier service level fluctuate. This is particularly important for real-time applications, such as VoIP, videoconferencing and VDI, where even a slight delay can result in dropped calls, glitchy video and dropped sessions - plus wasted time while all those services must reconnect.

    Ecessa products enable network engineers to strategically design wide area networks they way they want to, add links, add rules, and not shy away from complex multi-layered networks that will accomplish their performance goals. And it allows them to manage the whole thing confidently with cloud-based controls.

    Within the Ecessa suite of products, you'll find solutions for:
    • VoIP Failover and Monitoring
    • Multi-ISP WAN Failover
    • Intelligent Inbound and Outbound Load Balancing
    • Proactive monitoring and reporting of WAN network traffic
    • Network Traffic and Application Visibility, Traffic Shaping and Application Prioritization
    • Firewall and VPN
    • WAN Virtualization and VPN Tunnels between sites

    Company News




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  • 01/06/16--21:44: Asterisk consultants USA
  • This is a comprehensive list of Asterisk consultants in the USA (United States). Add your entry here (alphabetical order, by state and company), but stick to states where you have actual presence!

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


    Asteria Solutions Group

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  • 01/07/16--07:51: Asterisk consultants csa
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  • 01/07/16--21:47: Call Accounting Software
  • A Call Accounting System is a telecommunications software or hardware application that captures, records, and costs telephone usage events. Internationally call accounting systems may be referred to as call logging systems. Call accounting systems detect outbound and inbound calls, call ring outs, call routings, abandoned calls, and other activities..

    Common Applications of Call Accounting

    Service Billing and Provisioning

    Call accounting systems may provide packaging, pricing, provisioning, billing, and posting or presentment of telephone services for purposes of revenue generation. Professional services firms utilize call accounting software for account code or client based billing of their phone usage. The hospitality industry uses call accounting to resell phone services to visiting guests and groups. These call accounting systems often provide accessible application-specific rating and provisioning capabilities found generally on carrier-level operational support systems (OSS) and business support systems (BSS).

    Departmental and Employee Chargeback

    The original purpose of call accounting systems was within corporate entities for purposes of cost allocations within the enterprise. Enterprises use call accounting to allocate costs back to divisions, departments, and even individual employees. Such systems may also provide data directly to corporate accounting and human resource systems.

    Cost and Revenue Optimization

    Call accounting software can reconcile multiple telecom carrier's billing reports by integrating telecom invoices, wireless billing, long distance charges and calling cards into a single platform, allowing your business to use the convergent expense software to provide management reports, analytic reports, alerts and robust presentations

    Staff Productivity

    Call accounting systems provide visibility into the calling patterns and activity of employees and can be used to minimize productivity losses through non-business calling activity. They can also be used to evaluate the effectiveness of revenue-generating staff and sales processes, and manage the responsiveness of customer service staff.

    Network Optimization

    Companies also use call accounting systems to determine whether their voice and data networks are being utilized efficiently, in a cost effective manner, or to capacity. Call accounting applications are used to monitor network activity and bandwidth, identify over- and under-used trunks to optimize trunking configurations, pinpoint the root cause of circuit outages, monitor call routing effectiveness, queue times, abandoned calls and other information, and report usage trends and statistics. Call accounting can help companies more efficiently allocate telecom resources and make better planning decisions affecting a telecommunications network.

    Security and Compliance

    Call accounting applications enable IT departments to shield companies from a variety of internal and external security threats by monitoring for network attacks, intrusion attempts and telecom activity that exceeds acceptable or established thresholds.

    The Sarbanes-Oxley Act established new or enhanced standards for financial reporting by public companies and accounting firms, obliging them to examine and revise their internal governance, accounting and reporting controls. Examples of those "internal controls" include procedures for securing, protecting and ensuring the availability of critical infrastructure (like telecommunications systems), monitoring those systems for performance issues, and preventing or quickly detecting unauthorized attempts to acquire company records or assets.

    Call accounting applications help facilitate regulatory compliance by providing tools to aggregate, monitor and analyze telecommunications data in order to correctly track and report expenses. They can be used to enhance corporate accounting practices by documenting fraud-related costs to substantiate telephony-related disputes with carriers. And they can automate the monitoring of internal communications to improve visibility into sales and financial processes.

    Asterisk compatible call accounting software

    • Asterisell: support complex organization hierarchies, and can merge CDRs from two or more collaborating VoIP servers, creating a single logical call

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  • 01/07/16--21:51: VOIP Billing
  • Hosted Billing Services (in Alphabetical Order)

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

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  • 01/07/16--21:59: New Software Releases
  • This page is to inform on various VoIP related software releases.

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

    January 2016

    • 2016-01-08 - miniSIPServer LTS versions are upgraded to V24, and stable versions are upgraded to V25.

    December 2015

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  • 01/08/16--01:22: PBX in a Flash
  • Web Site:

    From the web site

    We've tried to collect in one place everything you will need to create your own PBX in a Flash system in under an hour! Once you have your system installed, you'll have a fully functional server with the CentOS 5.x Linux operating system including an Apache web server, a SendMail server, a MySQL database server, an IPtables firewall, PHP, and WebMin plus Asterisk 1.4, FreePBX, phpMyAdmin, and more. You'll then be ready to choose (or not choose) from dozens of Add-On's that make PBX in a Flash a unique offering in the IP telephony marketplace. There's the Asterisk 1.6 beta, automatic backups, CallerID lookup services, X-Windows, SSL Keys, Gtalk, Cepstral with Allison for text-to-speech applications, fax support, and on and on. If you don't need the extra features, don't load 'em. But every Add-On is designed to install with one click in under a minute! For some applications, you may have to perform a bit of customization, but we've strived to make that an easy process with excellent documentation. What really separates PBX in a Flash from the competition is its painless upgradeability... and our #1 Goal is No Bloat, No Bugs! Whether it is the underlying CentOS operating system, the Asterisk telephony platform, the FreePBX web-based user interface, or Bug Fixes, upgrades are always one button click away. Enjoy and welcome to the PBX in a Flash family!

    See Also

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  • 01/08/16--02:57: Autodialer
  • What Is An Autodialer?

    The following definition for Autodialer appears on Wikipedia.

    • '"An autodialer is an electronic device that can automatically dial telephone numbers to communicate between any two points in the telephone, mobile phone and pager networks. Once the call has been established (through the telephone exchange) the autodialer will announce verbal messages or transmit digital data (like SMS messages) to the called party."'

    Autodialing Techniques

    Autodialing can take on many forms including connecting calls to agents versus simply playing a recorded message. The following Auto Dialer Types are available from companies such as 3CLogic and Database Systems Corp.:

    • Voice Broadcasting delivers a pre-recorded message to live answers and answering machines. If another call status is detected (busy, etc.), the phone systems can reschedule the call for a later time. Simple messages can be delivered or the call recipient can be presented with an IVR script that accepts touch phone responses.

    • Preview Dialer allows a phone agents to view the call information prior to the call being placed. The agent can decide not to initiate the call.

    • Progressive Dialer passes the call information to the agent at the same time the number is being dialed by the phone dialer. The agent usually has a few seconds to view the call information, but cannot stop the call process. Often referred to as Forced Preview Dialing.

    • Predictive Dialer is more sophisticated because the phone dialer automatically calls several numbers and only passes a call to an agent when a person has been contacted. This eliminates busy signals, answering machines, etc.

    • Smart Predictive Dialer places calls, plays recorded messages and prompts, and passes the calls to agents only when the called individual requests a contact. "

    Autodialing Equipment

    Autodialers are usually desktop computers equiped with either special telephony voice boards or off-the-shelf voice modems. Traditionally, voice modems lack many of the features telephony boards offer. For example, call progress detection, call transfer, etc. However, with the ever improvement in today's computer speed, many of the telephony board features are implemented in software, which drastically reduces the cost and improves flexibility. Examples of such equipment are 3CLogic's Autodialer and Voicent BroadcastByPhone Autodialers, which offer all features offered by telephony board based autodialer systems and many more.

    Another advantage of using a voice modem based system is that it can turn a laptop computer to an autodialing system. There are good quality USB based voice modems, such as those from Creative Lab and Zoom.

    See Also

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  • 01/08/16--05:56: Asterisk billing
  • 0 0
  • 01/08/16--05:56: CNAM
  • CNAM is an acronym which stands for Caller ID Name.

    When phone calls are made, there are usually two user-facing identifiable pieces of information: a phone number and a Caller ID Name (usually a 15-character string). CNAM can be used to display the calling party's name alongside the phone number, to help users easily identify a caller.

    There are numerous CNAM lookup services which allow you to pay a small fee to lookup the CNAM of a specified caller (by phone number).

    CNAM Lookup Services List: Cost: Only $0.005 per query for carriers or $0.009 for hobbyists! No catch, guaranteed with easy paypal integration. Sign Up for a FREE Account and we will credit you 30 FREE CNAM queries to try No monthly fees or account minimums and 20 free queries to test our service when you open an account ( instant setup ). Simple HTTP API or Fast AGI that can be placed in your Asterisk dial plan. Also native support for Switchvox PBX systems. Results are never cached so you get up to the minute real-time results. Retail prices are $.006 per query and bulk pricing is available with a volume commitment of at least 25,000 queries per month. Free support and installation assistance is available. offers both CNAM ($0.006) and LRN ($0.0003) look ups. No minimums and monthly charges. Simple HTTP API, easy to integrate to Asterisk dial plan.

    CID(name) Professional CNAM (Caller name) delivery

    • EVERY LOOKUP IS LIVE FROM THE SS7 (direct from the carrier owning the number)
    • NO 3rd party data sources
    • NO monthly fees
    • NEVER pay full price for unavailable results
    • Carrier grade, multi-redundant platform
    • Simple to integrate HTTP API
    • 99.7% caller id name accuracy
    • Lightning fast query responses (under 500ms)
    • Volume pricing as low as $0.002 per query
    • Try before you buy, 100 free dips with every new account
    • You choose the output, TEXT/JSON/XML
    • Track sub-accounts
    • Easy integration with Freeswitch, Asterisk, OpenSIPS, and other open source voip platforms
    • Easy access and daily downloads to your account activity
    • Thousands of happy customers

    Get CARRIER GRADE CNAM at offers both CNAM and a pseudo-CNAM service at a fraction of the cost. Integration with asterisk is as easy as downloading the AGI and adding a single line to your dial plan. NEW SPECIAL PRICING; 0.00247 per transaction.

    • $0.00247 per lookup - that's less than 1/4 penny each!
    • Futher discounts for larger volumes
    • Easy-to-use RESTful API
    • SIP / HTTPS support
    • Asterisk integration tools available.
    • Free trial available
    • Real-time (never cached) data
    • NO contracts necessary
    • NO minimum volume commitments necessary
    • Includes access to our other services such as carrier lookup, phone append, etc.
    • First-rate customer support!

    For more information, call us at 877-805-3282, or ...

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