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  • 02/11/16--06:19: Voice Broadcasting
  • Voice Broadcasting

    The following is the definition for Voice Broadcasting:

    • "Voice broadcasting is a mass communication technique, begun in the 1990s, that broadcasts telephone messages to hundreds or thousands of call recipients at once. This technology has both commercial and community applications. Voice broadcast users can contact targets (whether they be members, subscribers, constituents, employees, or customers) almost immediately." - Wikipedia

    Voice broadcasting phone software manages a database of phone lists as well as digitized phone messages. Using analog, digital or VOIP telephony components, these computers can simultaneously broadcast thousands of phone messages. Personalized information can be included in the phone messages through the integration of text to voice software.

    Advanced systems include telephony boards or software that can detect the difference between an answering machine and a 'live' person answering the call. These systems employ the logic to properly play a unique message to answering machines without message truncation.

    There are few enterprise-level commercial solutions for Voice Broadcasting. They include such functions as:
    • Possibility to play personalized messages to every recipient which consist from library of predefined messages (pre recorder by every client) and some variable information (dates, numbers, digits, amounts with currency and so on). This method eliminates discomfort from text to voice modules (message looks completely like pre recorded by a human) and have high flexibility within predefined applications.
    • List of phone-numbers to call with parameters of call in DB such as Oracle DB, Microsoft SQL, IBM DB/2, MySQL, PostgreSQL.
    • Possibility to organize polls and to combine poll with statical or dynamical message. All results of such polls store within enterprise grade DB.
    • Possibility to transfer to call center by DTMF command from called party.
    • Possibility to opt-out by DTMF from called party.
    Such solutions always customizable for every client and easily integrates with ERP, CRM, billing systems. They usually used by banks, collectors and credit companies in order to remember their clients about payment day (with personalized amount to pay calculated to specific day).

    VoiceXML and Voice Broadcasting

    Phone messaging systems and services automatically send phone messages from a computer system to a remote phone systems using XML push logic. For example, alerts can be broadcast to tenants in a building if there is a fire or disaster. Heat sensors with IP connectivity that are installed in the tenant building can initiate an emergency voice broadcast by sending a VoiceXML message . Other applications may be as simple as wakeup calls or weather alerts that are triggered automatically from a computer system or websites.

    VoiceXML Gateway Information

    VoiceXML is a W3C standard for interactive telephone applications. The platform/server for VoiceXML is called a VoiceXML gateway. The gateway is like a web server, but it interprets VoiceXML command instead of HTML. You can test out VoiceXML applications with a hosted VoiceXML service, you could also setup your own VoiceXML gateway.



    See Also (Vendor Information)


    Voice Broadcast Applications


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  • 02/12/16--00:26: ICTBroadcast
  • ICTBroadcast is web based multi tenant unified communication autodilaer software solution. It features SMS messaging, Email marketing, Fax blasting and Voice broadcasting, suitable for SMB's , Enterprenuers and Internet Telephony Service Providers (ITSP). ICTBroadcast supports multiple type of Communication Engines including re-known open source Asterisk, Freeswitch and Kannel. ICT Broadcast is scalbalbe and integerated with RabbitMQ to achieve scalability and it can be scaled to blast thousands of simultaneous voice calls using either VoIP ( SIP or IAX ) or PSTN and Fax calls using using either FOIP (T.38 / G.711 pass through ) or PSTN.It is simple,multilingual support, reliable and user friendly web portal.

    ICT Broadcast platform support following type of campaigns

    Simple Voice Broadcasting )
    Voice Broadcasting with direct forwarding to Live agents support on answer)
    Interactive Voice Broadcasting / press 1 campaign )
    Survey / Polls )
    Inbound IVR campaigns)
    SMS Broadcasting )
    Fax Broadcasting )
    Email Marketing )
    Custom IVR voice broadcasting)

    How Voice Broadcasting works

    User upload a list of telephone numbers, upload audio message or record his voice message through telephone , configure outbound voice gateways and start a new campaign according to requirements using ICTBroadcast web interface and within seconds, ICT Broadcast starts broadcasting user's voice message to given list of telephone numbers with real time statistics.

    WHMCS integration with ICTBroadcast For Autodialer Billing

    ICTBroadcast has released a new billing module for ICTBroadcast Service Provider edition. This module will allow WHMCS to be used as Client Management and Billing front-end for Auto Dialer service, After integration with WHMCS system will be able to provide a complete business platform for broadcasting services including Website, package listing, automated order and Account provisioning. Following are few billing scenarios which can be achieved by ICTBroadcast for voice, fax, sms and email broadcasting business.

    ICTBroadcast Deployment Scenarios


    Automated Telemarketing
    Enterprise grade message Broadcasting
    Emergency notification system
    Interactive voice broadcasting / Smart Predictive dialer
    Customer surveys / Collections
    Polling Auto Dialler
    Mass Communications / notifications
    Political voice broadcast
    Robocall / call blasting,
    Phone reminders
    Community / Emergency alerts
    School Notifications
    Non-profit Fund Raising
    Wedding invitations
    cold calling
    mass broadcasting.
    Appointment reminders
    Retail sales / Buisness advertisment

    ICT Broadcast Features


    ICTBroadcast Trial version availale for Download


    For more information please visit this ICTBroadcast web site

    ICTBroadcast is developed by ICT Innovations

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    Company Profile:
    Ecosmob Technologies Pvt. Ltd. (commonly as known, Ecosmob) is India’s leading IT Company offering various IT software solutions and services. It was introduced in 2007 to provide complete IT based solutions and services. It has its headquarter in Ahmedabad, Gujarat. It has been delivering flexible, simple and affordable IT solutions to the renowned enterprises overseas.

    Ecosmob Technologies has secured a leading place in the VoIP industry with its next generation VoIP solutions and products. The team has the team of experienced developers who have rich expertise in developing customized VoIP software based on the client requirement. The company also offers open source consulting services. In nutshell, the company provides the design, development, deployment, consultancy and support services in the VoIP technologies such as:


    We provide custom design, development and deployment services for VoIP solutions. Some of them are briefed below:

    Conferencing Solution:
    The company provides a comprehensive conferencing software solution to conduct voice, video and web conferences. It provides custom development services for conferencing solution with selected features from the whole range of features the company offers including, Personalized meeting rooms, Web Phone, Conference wise Polling, Live Conference Viewer, and more. Being an environment friendly solution, the conferencing system is not just reducing the corporate carbon footprints, but it saves the travelling costs and the time to schedule the meeting without any location constraints.

    IP PBX Solution:
    Custom IP PBX solution allows media communication to take place with the help of a PBX combined with VoIP. The IP PBX software solution is not just limited to call features, but also includes an interactive directory listing, DND, conference bridging, IVR, privacy management feature, etc. Being a web/GUI based configuration, the IP PBX system eliminates the phone wiring and vendor lock-in that results in offering better customer productivity and services.

    Hosted PBX Solution:
    The company produces the Hosted PBX software solution to administer the communication that takes place without the need of any hardware. The development services that the company provides for Hosted PBX solution includes auto attendant, caller IDs, customized message alerts, fax to email, voicemail, find me, follow me, call waiting, conferencing and forwarding as their key features. Improved customer support, reduced cost of both phone bills and hardware are the added advantages to this system.

    Class 4 Solution:
    The Class 4 SoftSwitch solution routes the long distance, VoIP calls among the various IP networks. ...

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  • 02/22/16--21:28: SBO
  • sbo multipath logo.jpg

    What is SBO?

    Synchronous Bandwidth Optimizer is generally known as SBO. It is a bandwidth optimization solution for VoIP call termination developed by Synchronous ICT, a world famous VoIP software provider. It is a pioneer Bandwidth Optimization Technology fully loaded with amazing features.

    Why Bandwidth Saver is so important for VoIP?

    In VoIP termination, main expenditure is Bandwidth cost. Minimizing Bandwidth consumption is the main purpose of SBO Solution, at the same time it helps to reduce business operation cost. As a perfect bandwidth saver SBO reduces more than 80% internet cost directly. However, the quality of service is never compromised. To run a quality VoIP business congestion free, telco grade bandwidth connection is required . Such type of bandwidth connection is very costly as well as extremely rare. So, it was almost impossible to run quality VoIP business basically for small and medium entrepreneurs. But, now things are different, SBO Multipath make its possible to run IP telephony business with low cost share internet connection by maintaining supreme service quality. It can combine multiple internet connections together and use all the available bandwidth creating a single connection.

    How does SBO work?

    Well, there are mainly two parts in SIP communication, one is payload and another is RTP header. For a SIP call with G.729 it consumes almost 31.5 kbps bandwidth. But, noticeable matter is that the payload size is only 8 kbps. Rest of the bandwidth is consumed by RTP and other headers. SBO works here. It has own proprietary VoIP protocol which can replace the RTP and only transmit payload size thus reducing bandwidth consumption.

    Key Features of SBO:

    • The most important feature of SBO is, it can reduce bandwidth cost directly by 80% without degrading service quality.
    • SBO works behind any type of firewall and NAT. That means, it has anti-block feature.
    • Works with all commonly used codecs such as G.729, G.723 etc.
    • Multipath: SBO Multipath allows you to use multiple number of internet connections same time. These connections can be used simultaneously for your VoIP termination establishment which can balance load among available networks and it develop service quality expectedly.
    • Works with any type of internet connection i.e. GPRS, EDGE, 3G, 4G, Wi-Fi, Wi-Max, so it is possible to setup anywhere where mobile internet is available. ...

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    The purpose of this page is to list any provider who will give a free unique telephone number for incoming calls. Listings are by area code.

    To be listed on this site the provider must give a unique phone numbers. Services that offer gateways with extensions do not count. The number must be 100% and must not require signing up for other premium or paid services.

    Please list by area code. Include the city name if you know it, the provider and a link.


    North America

    416 - Toronto - Vbuzzer http://vbuzzer.com

    For a list of providers by Provider and not be area code, see http://www.voip-info.org/wiki-VOIP+Service+Providers




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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


    1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 60 countries. We have regional network facilities on five continents connected across our private network. Toll-Free termination is free and through direct connections.

    Mosaic NetworX, LLC
    www.mosaicnetworx.com
    Mosaic NetworX, LLC is a fully authorized US CLEC. One of our core offerings is Toll Free Termination. We have direct agreements in place with most of the RBOCs. Our collections percentages are 95% or greater. We offer compensation on Toll Free traffic with no minimums. Contact us at jay@mosaicnetworx.com

    Toll Free Dollars
    http://www.tollfreedollars.com
    Terminate your toll free traffic for free. We only use the best providers and our quality is always great. www.tollfreedollars.com


    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. ...

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  • 03/01/16--15:29: IPKall
  • IPKall is shut down effective May 1st, 2016

    Free personal Washington State (USA) PSTN number that forwards to any SIP or IAX destination you specify, including your own Asterisk server.


    Support



    Instructions on setting up a direct incoming IPKall phone number under Asterisk, bypassing FWD


    First, register an IPKall number at either of the web addresses provided above.
    IPKall will ask for the following information:

    SIP phone number
    SIP proxy
    Email Address
    Password (4 digit PIN)
    Voicemail preferences

    When specifying the SIP phone number, you can use any number, but you should try to avoid using
    a number that's already an extension in your extensions.conf file.
    Put the IP address (or hostname) of your Asterisk server into the SIP proxy field.
    If your SIP proxy is not running on the default port 5060, then you will need to specify the port.
    For example, if your IP is 123.254.254.1 and your SIP proxy is running on port 7777,
    then you would enter 123.254.254.1:7777 as the SIP proxy. Be sure to enter a valid e-mail address.

    Here is a sample IPKall configuration for sip.conf and extensions.conf.
    Place this under [inbound] in extensions.conf. (note, if [inbound] already exists, don't add it again):
    important note: your incoming context may be, and probably is different than [inbound].
    Replace [inbound] with the appropriate context in both extensions.conf and the "context=" line below.

    [inbound]
    exten => 508,1,Goto(your-main-menu|s|1)

    Where 508 is the SIP phone number you specified when setting up IPKall.
    Now, put the following information in your sip.conf file:

    [508] ;IPKall
    type=peer
    dtmfmode=rfc2833
    context=inbound
    insecure=very
    host=voiper.ipkall.com
    nat=no


    How it works

    1. When you dial your IPKall DID, IPKall sends a request (i.e. 508@123.254.254.1) to your SIP proxy
    2. Asterisk accepts the request, since the [508] context in sip.conf tells Asterisk to accept incoming calls with the number "508"
    3. Asterisk searches extensions.conf for the "inbound" context (since we specified context=inbound in the [508] context)
    4. Asterisk then matches extension "508" in the [inbound] context, jumping to "your-main-menu" in this example.


    Codecs Supported

    G.711
    GSM
    iLIBC
    G.729 (since Feb 2006)

    Troubleshooting

    If you just get a busy signal when calling your IPKall number, try the following:

    1. From the Asterisk CLI (asterisk -r) turn on sip debugging by typing "sip debug", try calling your IPKall number again, when you get the busy signal, turn off sip debugging by typing "sip no debug". Then read through the debugging information for clues.

    2. Make sure host=voiper.ipkall.com is specified in the IPKall context (in the above example, [508]), without this option you will likely get "SIP/2.0 404 Not Found" or "Found no matching peer or user for" sip debug errors. Since we are specifying a hostname, make sure DNS resolution works (nslookup voiper.ipkall.com)

    3. ...

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  • 03/03/16--08:59: VOIP Event Calendar
  • May 2016

    • 18-20 : Kamailio World Conference 2016, in Berlin, Germany - jubilee edition celebrating 15 years of development for Kamailio, the open source SIP server project
    • 10-12 : OpenSIPS Summit 2016 in Amsterdam, The Netherlands - 2 days of presentations and updates from the OpenSIPS core team, community and related projects + 1 day of Workshops and Design Clinics.

    March 2016

    • 24 – 25 TELECODAYS Prague 2016 - Global telecom meeting in the heart of Europe bringing together wholesale voice, mobile, data carriers, MVNOs and telecom solutions vendors from around the globe.
    • 15 – 16 Telecoms World Asia 2016 - With a 17 year track record, and the revamped theme “Exploding Bandwidth, Emerging Markets, Evolving Partnerships”, Carriers World Asia continues to be the stage, in this new digital age, where decision makers from leading international carriers, operators, authorities and suppliers throughout the entire wholesale communications chain meet to learn, strategise and partner for the common goal of core revenue sustenance and growth in a region of rising opportunities.
    • 14 - 18 : Join TrainingCity in Midtown Manhattan, "The Big Apple" for our ever popular "hands on" VoIP Training & SIP Training classes! VoIP Hands On Workshop * Hands On SIP Training These classes will sellout, Email TrainingCity today for more information.

    Feb 2016

    • 11 - 12 Phoenix, AZ SIP Training Course. Hands on SIP training with the trusted experts from TrainingCity. Join us in Phoenix & Learn SIP with hands on labs and expert instructors! Hands On SIP Training. For more information, Email TrainingCity.



    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

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  • 03/03/16--09:31: DID for Sale Reviews
  • DID for Sale has been a wholesale VoIP provider since 2008, offering its products to VoIP service providers and large businesses. These products include SIP trunking, A-Z termination, metered and unmetered DIDs, toll free numbers, and much more.

    Depending on the kind of minutes you need and how many direct inward dialing lines you have, their DID number offerings are low cost. DID for Sale also offers two affordable plans for DID numbers: a Flatrate Plan and a Metered Plan. Pricing depends on the number of inbound and outbound calls, as well as the number of DIDs. Visit their website for more details.

    With DID for Sale, you can get new numbers, keep your existing number, or purchase toll-free numbers.

    Additional features include:

    • Redundant systems for more uptime
    • A failover facility
    • Local presence in any USA state
    DID for Sale also allows for USA phone numbers on international equipment, and offers international availability as long as you have a strong internet connection.

    DID for Sale has a highly-rated team of customer service representatives and engineers to ensure that your system functions properly and with ease.

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    02/09/2016 - New version 7.5.1.0 of Thirdlane Business PBX, Thirdlane Multi Tenant PBX, and Thirdlane Elastic Cloud PBX has been released

    Thirdlane offers reliable, cost-effective, and highly customizable Asterisk-based PBX software. All Thirdlane PBX products feature the Thirdlane Communications Manager GUI for ease of configuration and administration. Thirdlane PBX product line includes:

    Thirdlane Elastic Cloud PBX


    Thirdlane® Elastic Cloud PBX Platform is a Unified Communications software platform for large scale hosted Multi Tenant PBX deployments by carriers and Internet Telephony Service Providers.

    Thirdlane Elastic Cloud PBX integrates Kamailio SIP Server and Asterisk® and offers the same advanced IP PBX functionality as the Thirdlane Multi Tenant PBX - and the next level of scalability and availability.

    Thirdlane Multi Tenant PBX


    Thirdlane® Multi-Tenant PBX is an IP PBX and unified communications software platform capable of hosting multiple virtual PBXs on a single server. Multi-Tenant PBX enables Internet Telephony Service Providers, telephony resellers, and VoIP entrepreneurs to quickly roll out reliable hosted telephony services with powerful features, flexibility, and cost-effectiveness to ensure satisfied customers, reduce churn, and provide a steady return on their investment.


    Thirdlane Business PBX


    Thirdlane Business PBX is a an IP PBX and unified communications software platform that forms the heart of a versatile Voice over IP (VoIP) enabled telephony system. It provides small businesses, larger companies, and multi-site enterprises a flexible combination of the best of open source and commercially developed solutions, offering an alternative to high-cost telephony systems.


    Thirdlane Call Center


    Thirdlane Call Center software delivers a reliable and highly-customizable contact center solution with sophisticated features and functionality for a broad range of call centers - on-premises or in the cloud. Thirdlane Call Center is an add-on that is integrated with Thirdlane PBX products and offers set of features supporting environments ranging from professional contact centers to workgroups requiring more advanced queuing, call distribution, and call metrics features.



    Try Thirdlane for Free!
    Try a demo of Thirdlane Multi Tenant or Business PBX, or download a free trial ISO version of either PBX product from the Thirdlane website.

    Find Out More
    Find out more about Thirdlane products by visiting the Thirdlane website.

    Contact Us
    For more information, or if you have any questions, please contact us!


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  • 03/07/16--08:00: Dialpad Reviews
  • Dialpad Reviews (Formerly Switch.co)

    As cloud-based communication steadily becomes more prominent among businesses, investors are funding the development of economic and versatile systems that are changing the standard for voice communication. One notable VoIP system that is capitalizing on this trend is Dialpad, a system that grants the employees of businesses the means to communicate across several devices and provides a trial for potential users. Headed by the reputable Craig Walker, who spearheaded the creation of Google Voice and other notable communication services, Dialpad can be set up between your smartphone, tablet, and desktop. The system can also be used with Android, iOS, Windows, Chromebook, and Linux.

    Available at $15 a month per user, Dialpad manages to be cost effective and ambitious by using its own voice engine and data center to provide high quality and expedient communication with unlimited calls and texts for its users. These frameworks, in addition to Dialpad’s expansion as a system, were made possible by investors such as SoftBank, Felicis Ventures, Google Ventures, and other firms and companies that raised millions in order to facilitate the establishment and growth of Dialpad as a system.

    Striving to be as versatile as possible for businesses, Dialpad finalized its integration with Microsoft Office 365 back in October 2015. Through Dialpad, users can expect to have their Outlook contacts immediately synced through an interface that provides easy customization tools for companies. These tools can be used to set up an entire company directory with users, chat options, departments, and even a virtual receptionist. This convenience also extends to your Google contacts, which is no surprise to those that are already are of Dialpad’s deep integration with Google Apps.

    After establishing Dialpad, Walker was intent on creating a powerful, voice-based substitute to Gmail, Google docs, and other Google services. As a result, Dialpad has many functions that are of great convenience to its users. Documents shared between your contacts are visible to you at any time. An ongoing call can be seamlessly transferred to another device. A user can even integrate their profile with their social media accounts, a feature likely made for those who work in a pertinent field for it.

    In order to include businesses that cannot divorce themselves from their traditional desk phones, Dialpad partnered with the VoIP phone manufacturer Obihai for a line of phones that were produced with the intent of being easy to install and highly available.

    Anyone looking into the history of Dialpad will have no difficulty being impressed by company’s track record. Founder Craig Walker, among others, helped establish a VoIP company prior to Switch Communications called Dialpad. In 2016, Switch.co renamed itself Dialpad. Yahoo would go on to acquire Dialpad (in the late 90s), and with it Yahoo established Yahoo Voice. It should also be mentioned that before Dialpad, Switch Communications developed a myriad of quality products and services such as Jotly, NoshList, Nosh, and UberConference.

    Beyond being a substitute for the cloud-based services established by Google, Dialpad is also targeted towards businesses that are still using traditional phone services with extensions. As cloud apps and services are expanding and seeing more use by the everyman, the demand for a more sophisticated, economic, and versatile phone service alternative is inherently apparent. It also pays to state that this demand is not exclusive to the U.S. market.

    Dialpad’s growth has been expedited, partially due to users that are comfortable with the other services that were developed by Dialpad. The transition to a cloud-based system has been particularly easy for Nicholas Gardner, the Sr. Director of Internal Systems for Weather. Being a satisfied user of UberConference, Gardner used the trial and is currently in the process of updating the internal employee systems of his company to Dialpad.

    With Dialpad currently in a stage of newfound availability, it has been intimated that other businesses will follow Gardner as cloud-based systems become the new standard for voice communication.

    See also


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  • 03/07/16--08:03: Dialpad
  • Dialpad is one of the older VOIP to PSTN providers. Their website claims over 14 million users.
    http://www.dialpad.com

    March 7, 2016: Switch.co has rebranded to Dialpad.

    They provide PC-phone service using their Chameleon software. They support all kinds of NAT and some of strict firewalls also.

    Recently they have added support for a Cisco ATA, SIPURA SPA-2000 and dpPhone using SIP protocol on a broadband connection. While this may continue briefly for existing customers, it is unlikely these will be supported by Yahoo.

    They used to offer unlimited US long distance calls for $11.99 monthly charge, but they are no longer taking new signups - attempts to do so redirect one to signup for Yahoo Voice, which offers only prepaid per-minute options.
    Currently their service is outbound only, website says inbound is coming soon.

    In June of 2005 Yahoo officially acquired Dialpad.

    Image

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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 then 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 anyways.

    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: connecting 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.

    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).

    Data Synchronization and Scalability

    In order for a cluster to remain useful, the data on the peers must remain in sync. This allows one peer to pick up where the other left off in the event of a failure. However, synchronization is one of the greatest challenges for clusters. This is the next most important criteria in designing/selecting/building a high availability telephony environment.

    There are 2 approaches to solving this problem. The simplest approach is letting the 2 peers share a disk (i.e. ...

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  • 04/05/16--03:56: SBO
  • sbo multipath logo.jpg

    What is SBO?

    Synchronous Bandwidth Optimizer is generally known as SBO. It is a bandwidth optimization solution for VoIP call termination developed by Synchronous ICT, a world famous VoIP software provider. It is a pioneer Bandwidth Optimization Technology fully loaded with amazing features.

    Why Bandwidth Saver is so important for VoIP?

    You know, in VoIP termination, main expenditure is Bandwidth cost. Only for highly bandwidth cost many companies are struggling day by day. Minimizing Bandwidth consumption is the main purpose of SBO Solution, at the same time it helps to reduce business operation cost. As a perfect bandwidth saver SBO reduces more than 80% internet cost directly. However, the quality of service is never compromised. To run a quality VoIP business congestion free, telco grade bandwidth connection is required . Such type of bandwidth connection is very costly as well as extremely rare. So, it was almost impossible to run quality VoIP business basically for small and medium entrepreneurs. But, now things are different, SBO Multipath make its possible to run IP telephony business with low cost share internet connection by maintaining supreme service quality. It can combine multiple internet connections together and use all the available bandwidth creating a single connection.

    How does SBO work?

    Well, there are mainly two parts in SIP communication, one is payload and another is RTP header. For a SIP call with G.729 it consumes almost 31.5 kbps bandwidth. But, noticeable matter is that the payload size is only 8 kbps. Rest of the bandwidth is consumed by RTP and other headers. SBO works here. It has own proprietary VoIP protocol which can replace the RTP and only transmit payload size thus reducing bandwidth consumption.

    Key Features of SBO:

    • The most important feature of SBO is, it can reduce bandwidth cost directly by 80% without degrading service quality.
    • SBO works behind any type of firewall and NAT. That means, it has anti-block feature.
    • Works with all commonly used codecs such as G.729, G.723 etc.
    • Multipath: SBO Multipath allows you to use multiple number of internet connections same time. These connections can be used simultaneously for your VoIP termination establishment which can balance load among available networks and it develop service quality expectedly.
    • Works with any type of internet connection i.e. ...

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  • 04/05/16--10:19: Old News
  • This page lists all the old VoIP news stories from the home page.


    March 2016

    February 2016

    January 2016


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  • 04/21/16--12:39: VirtualPBX Phone Systems
  • VirtualPBX offers flexible options that meet the needs of small and large businesses alike. Easy setup solutions are readily available, such as Dash Plans for ease of use or VirtualPBX Office Plans for total VoIP-based business phone service with unlimited minutes. VirtualPBX is an industry leader, having won numerous awards for their reliable and flexible products. Whether your business uses VoIP phones, softphones, or mobile phones, VirtualPBX has phone system solutions to help you achieve ultimate mobility. Check out their award-winning products and plans below.

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  • 03/03/16--09:31: DID for Sale Reviews
  • DID for Sale has been a wholesale VoIP provider since 2008, offering its products to VoIP service providers and large businesses. These products include SIP trunking, A-Z termination, metered and unmetered DIDs, toll free numbers, and much more.

    Depending on the kind of minutes you need and how many direct inward dialing lines you have, their DID number offerings are low cost. DID for Sale also offers two affordable plans for DID numbers: a Flatrate Plan and a Metered Plan. Pricing depends on the number of inbound and outbound calls, as well as the number of DIDs. Visit their website for more details.

    With DID for Sale, you can get new numbers, keep your existing number, or purchase toll-free numbers.

    Additional features include:

    • Redundant systems for more uptime
    • A failover facility
    • Local presence in any USA state
    DID for Sale also allows for USA phone numbers on international equipment, and offers international availability as long as you have a strong internet connection.

    DID for Sale has a highly-rated team of customer service representatives and engineers to ensure that your system functions properly and with ease.

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  • 04/27/16--09:13: Virtual PBX Reviews
  • VirtualPBX is a business virtual PBX and a business phone system provider based in the United States.

    See also


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    TERMINATING ACCESS TANDEM SOLUTION FOR TELECOM COMPANIES

    Our Terminating Access Tandem solution eliminates your need to maintain costly circuits and outdated technology in your network. With our solution your access traffic does not have to be routed through the ILEC tandem. Routing your codes behind West’s telecom services tandems allows you to set up an interconnection architecture that meets your needs–not the ILECs.

    Terminating Access Tandem allows network operators to select West Telecom Services as their access tandem in the LERG. When West is listed in the LERG as the access tandem, IXCs will route traffic through West–allowing partners to minimize transport costs.
    Our solution gives partners the freedom of choice–choose the interconnection method and network topology that suits you. We offer SIP or TDM handoffs at geographically diverse sites.
    Let us be your total tandem solution provider.

    West Telecom Services contact: dguglietta@west.com

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