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  • 02/10/17--13:13: GoIP
  • goip gateway

    GSM VoIP Gateway ( goip gateway ) for call termination, HyberTone Technology

    business card.jpg
    Models of GSM VoIP Gateway ( goip gateway )
    Series 1 SIM port 4 SIM ports 8 SIM ports 16 SIM ports 32 SIM ports
    External Antenna GoIP-1 GoIP-4 GoIP-8 GoIP-16 GoIP-32
    Internal Antenna GoIP-1i GoIP-4i



    1, 4, 8, 16, 32 ports GSM VoIP Gateway ( goip gateway ) ( External Antenna series )

    GoIP-1_300.jpg
    GoIP-4_300.jpg
    GoIP-8_300.jpg
    GoIP-16_300.jpg

    GoIP-32-5.jpg


    1, 4 ports GSM VoIP Gateway ( goip gateway ) ( Internal Antenna series )

    GoIP-1i_300.jpg
    GoIP-4i_300.jpg



    Specification


    Basic Features:

      • For call termination (VoIP to GSM) and origination (GSM to VoIP) }
      • Standard SIP & H. ...

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  • 02/10/17--13:13: 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

    Angola

    • MultiTEL is providing retail and wholesale Angola (Luanda) DIDs . Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. Payment by card, paypal or bank transfer. All calls are forwarded to SIP, PSTN or to our freehttps://www.multitel.net/pbx|Hosted PBX] . Coverage and numbers always available in stock from more than 90 countries.

    Argentina

    • BuyDDINumbers.com Provides Cheapest Argentina DID /Virtual Phone Numbers/DDI Numbers @_€ 2.95/month including free PBX. with the Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX __. Toll Free Number Available without Monthly commitments.
    • BuyDIDNumber We Provide Argentina Virtual Phone Numbers@ $ 4.49 / 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.
    • CarryMyNumber.comArgentina DID /Virtual Phone Numbers at wholesale rate_ @$ 2.75/month with free fully hosted PBX. with the Free forwarding to SIP, Softswitches ,Trixbox ,Asterisk,FreePBX 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.
    • Divertmycalls.com Provides __Cheapest Argentina DID /Virtual Phone Numbers @ $ 4.99 with the Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX . ...

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  • 02/10/17--13:17: T.38
  • T.38 is an ITU standard for sending FAX across IP networks in a real-time mode.
    FAX messages are sent as UDP or TCP/IP packets.

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

    From RFC 3362:

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

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

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



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

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

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

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


    T. ...


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  • 02/13/17--13:16: ICD
  • Asterisk ICD

    Asterisk ICD (Intelligent Call Distribution) is an advanced mechanism for handling queues, conferences and agents and provides a unified API for interfacing to external dialer systems.

    The goal of this application is to provide a flexible, thread-safe infrastructure to the Asterisk PBX for distributing calls. that is robust and customizable for larger scale deployments.
    ICD maintains a Finite State Machine for each call. Each state is managed by a pointer to a function so each state can execute a custom function, this architecture is really designed for call center developers that want to build customized call routing strategies that fit into a standard frame work.

    There are two major components to this system, the application itself and the infrastructure pieces that support the functionality of the application.

    ICD handles queueing differently than the normal asterisk queuing system. Queues are defined in configuration files, the dialplan defines customers and agents access to the queues and in which way they are being bridged to the queue.
    Agents can for example be dynamically connected to queues on login so you could for example have a queue without members, a defined agent without a queue and connect him to a queue on login.
    Customers can also maintain their place in line and request a callbacks with out waiting inline.

    ICD is now integrated into CallWeaver http://Callweaver.org as the standard agent and queue system as of Jan 1, 2008



    Installation instructions can be found in the file "README"
    Architecture info is found in "modules.txt".

    This is a work in progress

    if you have questions dev's guys are

    See also




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  • 02/13/17--13:17: H.263-Sorenson
  • Sorenson Spark is an implementation of H.263 for use in Flash Video and Adobe Flash files. FFmpeg uses FLV1 FourCC and Adobe frame identifiers of 0x21, 0x22 and 0x23.

    As Apple began to embrace MPEG-4 and move away from other proprietary codecs, Sorenson Media licensed Sorenson Spark (Sorenson H.263) to Macromedia, which was included with Macromedia Flash MX v6 on March 4, 2002. Sorenson Spark is the required video compression format for Flash Player 6 and 7.

    Macromedia later tried to find a better video codec. Starting with Flash Player 8 (released in September 2005), the preferred video codec became VP6. Sorenson Spark can be still used in the Adobe Flash CS4 Professional (2008) for Flash Video files (alongside H.264 and VP6).[17] According to Adobe engineer Tinic Uro, Sorenson Spark is an incomplete implementation of H.263. It differs mostly in header structure and ranges of the coefficients.

    FFmpeg in 2003 added encoding and decoding support for Sorenson H.263.
    Asterisk can support it with the chan_rtmp developped by ulex.

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  • 02/13/17--20:58: Asterisk at large
  • Planning a large installation with SER or OpenSER


    Improving performance

    • Store the voice prompts in RAM instead of harddisk: Take a look at ramfs. All you need then is to create a link (ln -s) in /var/lib/asterisk/sound to then ramdrive you created using ramfs.
    • add "fs.file-max = 131072" to /etc/sysctl.conf - run "sysctl -p"
    • add "ulimit -n32768" to /etc/init.d/asterisk

    SIP proxy before Asterisk (e.g. SER)

    SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic.

    Howto: Getting to know SER

    If you want a good explaination of SER and how to use it start here:
    Latest Guide http://www.iptel.org/ser/doc/gettingstarted
    Old Guide http://siprouter.onsip.org/doc/gettingstarted/
    They have GREAT pre-written configs and walk you through every part of SER. I was about to scrap SER before I found these tutorials.

    And then there's the OpenSER documentation on how to integrate with Asterisk in realtime for VoiceMail and MWI. Find even more on the same topic here.

    Asterisk, OpenSER and mediaproxy in a WAN environment

    Asterisk, OpenSER and WAN
    This article discusses some of the problems encountered with OpenSER and Asterisk features when implementing a Enterprise VOIP system - and how to solve them.

    Benefits for Asterisk

    We are using both, SER and Asterisk, in heavy production environment. SER is the BEST SIP proxy that i found. But it is just sip proxy. It can serve a _LOT_ connections (10,000 users, 20 cals per second). Asterisk is more like telephone switch with lot of features, but far slower.

    One nice feature of SER is that users can set up their own SIP accounts using a web interface and not needing to edit *.conf files.

    Q: In a large installation (many users) - how do you off-load Asterisk by managing RTP traffic effectively?
    A: There's no single truth here, but here's my opinion:

    • If you are all on the same internal network, make sure the SIP phones support re-invites and use that.
    • If you have users all over the Internet, use a SIP proxy (like SER) as a front-end to Asterisk. You will still be forced to handle a lot of RTP streams (because of NAT), but can distribute that over a SIP-proxy network with SRV records, DNS round-robin techniques or forcing the users to register with different proxies.

    Example: FWD and SIPGATE both run SER coupled with Asterisk

    There's been a couple of suggestions that we should make Asterisk a good SIP proxy. If you spend some time learning to understand Asterisk's architecture, you'll also understand that this would not really work. I'm not saying the SIP channel can't be improved, I'm just saying that it has to work with the rest of Asterisk's architecture. Asterisk in combination with a separate SIP proxy is a very powerful solution. ...

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    valid for kamailio (openser) 1.5.x and asterisk 1.4.x — work in progress


    • latest release in 1.5.1 series of Kamailio (OpenSER) is v1.5.1
    • latest release in 1.4 series of Asterisk is v1.4.24.1

    Kamailio is a pure VoIP signaling server using Session Initiation Protocol - SIP. It is flexible and highly configurable but cannot be used to provide media services as voicemail, announcements or conferencing. For such services, Asterisk is the most suitable open source product. In this document we present how to configure Asterisk to use Kamailio's subscribers database to provide voicemail service. A basic configuration file for Kamailio is posted down the page, allowing to have a functional system by following the steps in this tutorial.

    Requirements


    UnixODBC Installation

    Install unixodbc-dev package via your package manager. Alternative is to get the sources from http://www.unixodbc.org, compile and install them on your system, but Asterisk will have hard time to detect them and enable ODBC option for Voicemail application.

    In Debian:

    apt-get install unixodbc-dev


    MySQL Installation

    You can install MySQL using the packaging system from you Linux distribution. The only requirements is to be MySQL 5.0+. Most of the distributions deliver v5.0+ by default now.

    After installation, you can set the MySQL root password with a command like:
    /usr/bin/mysqladmin -u root password 'your-new-password'


    Asterisk Installation

    Get Asterisk sources from http://www.asterisk.org. At this moment Asterisk 1.4.22 is the latest stable version.

    cd /usr/local/src
    wget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.24.1.tar.gz
    tar xvfz asterisk-1.4.24.1.tar.gz
    cd asterisk-1.4.24.1
    ./configure


    Run make menuselect to enable ODBC storage for Voicemail, choose option 9. Voicemail Build Options, enable option 1. ...

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  • 02/14/17--06:58: Inaani Pte Ltd


  • Inaani is a preeminent retail and wholesale VoIP Service Provider with its diligent presence across the globe. This company is offering VoIP Service which enables the users to make international calls in a low cost.
    Inaani has been providing supreme quality 24X7 dedicated customer support to ensure that the customers can relish the best service for their business. We believe in perfection and authenticate our service quality before providing our services to our consumers. Inaani has been dealing with some of the most renowned carriers, such as Singtel, TATA Communications Ltd, Reliance Globalcom Ltd, Bharti Airtel Ltd etc.

    History


    Inaani was founded in 2006 with the ultimate focus of providing utmost service to its consumers and now it one of the leading VoIP service providing companies in the VoIP market. Substantiating premium quality service to its client has always been the infrastructure of Inaani. Headquarter of Inaani is established in Singapore and has offices in Hong Kong, India, Bangladesh and UAE. Possessing a versatile structure, Inaani is competent to terminate VoIP calls, simultaneously for the retail and wholesale VoIP business.



    Inaani Blog posts




    Products




    Services


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  • 02/14/17--07:47: VOIP Service Providers B2B
  • Here is a list of VOIP Service Providers focusing on Business-To-Business services. This includes VoIP origination and VoIP termination, plans aimed at call centers, IVR providers and generic Asterisk users. See also:


    Services which require the use of locked ATA devices should not be listed on this page. Nor should services which do not permit simultaneous calls — most services here support at least 4 simultaneous incoming calls. Please list only services which support Asterisk connections, via SIP or IAX2, to the PSTN.


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  • 02/14/17--10:46: GSM Codec
  • GSM (Global System for Mobile communications) is a cellular phone system standard popular outside the USA. You can find more info on GSM in general here

    GSM includes a codec, often just referred to as the GSM when discussing codecs.

    The original 'Full Rate' GSM speech codec is named RPE-LTP (Regular Pulse Excitation Long-Term Prediction). This codec uses the information from previous samples (this information does not change very quickly) in order to predict the current sample. The speech signal is divided into blocks of 20 ms. These blocks are then passed to the speech codec, which has a rate of 13 kbps, in order to obtain blocks of 260 bits. You can compare GSM cell phone plans using Wirefly.

    Newer GSM systems use a couple of newer codecs however these are heavily patent encumbered:

    • EFR (Enhanced Full Rate) uses ACELP (Algebraic Code Excited Linear Prediction)
    • HR (Half Rate) uses CELP-VSELP (Code Excited Linear Prediction - Vector Sum Excited Linear Prediction)


    GSM Gateways

    OpenVox VoxStack Series GSM Gateway is an industry 1st open source asterisk-based GSM VoIP Gateway solution for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily setup their customized Gateway. Also secondary development can be completed through AMI.

    Codecs

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  • 02/14/17--13:26: Asterisk Consultants Germany
  • Asterisk consultants: Germany


    Add your entry here (Alphabetical order by country and company):
    This page is growing large. Please don't post logos!!
    Add your geo-position on Asterisk-users Counter.

    ADDIX Internet Services GmbH, Kiel

    ADDIX bietet auf Asterisk basierende Telefonanlagen, Datenbank gestuetztes Administrationstool fuer Admins und User, Echtzeitmanagement von Telefonkonferenzen, Vermittlungsarbeitsplätze, Mobile Applicationen für Smartphones. Anbindung an Junghanns BMS Callcenter Applikation, Master/Slave Systeme zur zentralen Verwaltung in groesseren VPN/MPLS Netzen mit x Systemen. Internet Services und eigene Data Center. Asterisk Hosting mit PSTN Gateway, VPN/MPLS (Filialvernetzung) und Programmierung.

    ADES GmbH, Burscheid

    ADES GmbH bietet von der Konfigurationsunterstuetzung bis zur kompletten ASTERISK-Anlage komplette Dienstleitungen an.
      • Home page:: http://www.ades.de
      • Telephone: PSTN +49.2174.64043
      • Email:asterisk@ades.de
      • Contact: Bent Weichert

    Adimus GmbH, Bochum

    Adimus bietet Beratung, Konzeption, Implementierung und Support von Asteriskbasierenden Telefonanlagen sowie VPN-Lösungen für klein- und mittelständische Unternehmen.


    aixvox GmbH, Aachen

    Die aixvox GmbH ist ein international tätiges Beratungs- und Dienstleistungsunternehmen. Unser Fokus ist es, Telekommunikationsinfrastruktur in Unternehmen neu aufzubauen bzw. vorhandene Strukturen zu profitablen Systemen auszubauen; Asterisk bzw. Asterisk-basierte Systeme sind oft die passende Lösung. aixvox ist auch Herausgeber des unabhängigen Kompendiums voice compass 2007, der ausführlichen Übersicht über den deutschsprachigen Voice Markt.


    AMOOMA GmbH, 56566 Neuwied

    AMOOMA bietet sowohl Consulting wie auch Schulungen zu den Themen Asterisk, VoIP und Trouble-Ticket-Systeme an. ...

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  • 02/14/17--13:28: Asterisk fax
  • Asterisk and fax calls

    Fax over IP

    Across the Internet even a G.711 codec fax transmission is unpredictable. An excellent discussion of why faxing and modems don't work well over VoIP can be found here. However, people often get perfectly good results on lightly loaded LANs. It still isn't perfect, as a burst of data on the LAN can still upset things, but some people get results they can live with.

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  • 02/14/17--13:28: CLEC
  • CLEC (Competitive Local Exchange Carrier)

    A phone company that buys facilites and services from ILECs and resells service to its customers. The ILECs are required by law to sell serivce and access to their customers to the CLECs.

    Well known CLECs in the USA


    How to start a Clec


    How to start a CLEC as an ITSP


    See also


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  • 02/15/17--08:17: Virtual PBX
  • Virtual PBX is a budget-friendly form of hosted VoIP (Voice over Internet Protocol) that usually only handles inbound calls. A virtual PBX is typically intended for small business VoIP customers with fewer than 10 employees and low-volume telephone traffic.


    What Is Virtual PBX?

    A virtual PBX is an economy-class version of hosted PBX. Hosted and virtual PBX systems are business VoIP PBX phone systems that transmit calls over the Internet as data.

    A virtual PBX offers inexpensive business VoIP telephone service to small businesses. As with a hosted PBX phone systems, a virtual PBX is owned and maintained off-site by a VoIP service provider. A virtual PBX enables a small business telephone system to access enterprise-level features such as auto attendants and voicemail. With virtual PBX small business telephone systems, small start-ups, mom-and-pop shops, freelancers, and entrepreneurs can all present a professional image to vendors, investors, and customers.

    Depending on the service provider, a virtual PBX phone system may require a separate phone service for outbound calls.

    Virtual PBX Features


    Virtual PBX phone systems offer lower costs and fewer features than hosted PBX phone services. Compared to hosted PBX small business telephone systems, virtual PBX service is limited to the most basic fundamentals of business-oriented call controls. Virtual PBX is geared toward simple inbound call-routing for SoHo offices with few personnel, small budgets, and limited calling needs. As with many hosted PBX calling services, most virtual PBX phone systems do not require a contract or term commitment.

    Standard features offered with most virtual PBX plans are:

    • Voicemail
    • Auto attendant
    • Unlimited call handling (no busy signals)
    • Call forwarding

    Limitations


    Virtual PBX phone systems generally:

    • Handle only inbound calls
    • Offer a limited number of extensions
    • May not include Fax over IP (FoIP) services
    • Include a set amount of free minutes
    • May not offer voicemail-to-email
    • May not include international long-distance coverage
    • May not offer Internet fax service
    • May charge extra for conference calling

    The features offered vary by virtual PBX VoIP provider. As VoIP service becomes a more common solution for small business telephone systems, many virtual PBX plan features are incorporating the more advanced features of hosted PBX phone systems. Compare plans and prices to determine the best virtual PBX solution.

    Cost

    Virtual PBX phone prices depend on a variety of factors, such as the features included. Virtual PBX phone service plans can start as low as $9.95 (Grasshopper) per month.

    Virtual PBX Service Providers


    Some virtual PBX providers include:

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  • 02/15/17--13:23: Thirdlane Business PBX


  • 12/11/2016 - Thirdlane releases Thirdlane Connect for unified communications with support for voice, video, private and group chat, integrations, screen and file sharing powered by the new version of Thirdlane Business PBX and Thirdlane Multi Tenant PBX platform.

    Thirdlane Business PBX is a highly-reliable, cost-effective 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.

    In today’s fast-paced business environment, a reliable integrated communications system is critical to success. Thirdlane’s Business PBX delivers by including all the standard telephone system features you expect, plus advanced unified communications capabilities such as integration with email, messaging, and mobile platforms at no extra cost. Thirdlane systems have been field-proven for over ten years across thousands of customers worldwide, and are widely regarded for their unique combination of diverse features, flexibility, and bulletproof reliability.

    Thirdlane provides system administrators with all the tools required for easily making changes and adding users or devices. A unique deep customization ability allows each user or group their own feature set to meet virtually every business need. Thirdlane’s Business PBX also boasts an expandable open architecture and an extensive API (application programming interface) to allow integration with third-party applications such as CRM, ERP, accounting systems, and other business tools. The result is that your company can benefit from an easy-to-use yet highly adaptable set of advanced features with a low total cost of ownership.


    Key Features:

    • Advanced Call Features: Among the PBX features included are: IVR (auto attendant), conference bridges, call forwarding, transfer, call screening, call parking, call presence, ring groups, hunt groups, find me/follow me, call queues (ACD), direct dial (DID), fax handling, selective call screening and blocking, call recording, intercom, paging, voicemail to email, and much, much more.
    • Auto-Provisioning: Thirdlane Business PBX includes templates for auto-provisioning of devices such as Aastra, Cisco, Linksys, Polycom, Snom and Yealink phones and ATAs. This allows you to easily add phones and devices, individually or in bulk. Entire groups of users can be easily added and provisioned. Templates can be readily customized or added to support new devices.
    • Fine-grained permissions: Allow your administrators to easily control each user’s call permissions and features as appropriate for your business requirements, and to configure dialing rules on a per-user or per-route basis for operational economy and flexibility.
    • Highly Customizable: Choose from a number of Communications Manager and user portal GUI themes, and even customize menus and configuration files. Select from 12 supported languages, or add your own language translations or voice prompts. Easily add custom scripts to support user-requested features and integrate with most third party program’s API.
    • Proven Industry-Standard Components: Thirdlane systems are built with standard, proven components, including the Asterisk® telephony engine and CentOS® Linux, without custom patches. This allows you to easily update them if needed. Thirdlane also supports versions of Digium®-certified Asterisk and Red Hat® Linux for additional peace of mind in critical applications.
    • System Control and Updating: Unlike other solutions, Thirdlane doesn’t deny you root access to control your server for troubleshooting, integrating with third-party components, and performing system maintenance and customization. ...

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  • 02/15/17--13:24: Thirdlane Multi-Tenant PBX


  • Thirdlane® Multi Tenant PBX is a reliable, feature rich, scalable and cost-effective IP PBX and unified communications software platform capable of hosting multiple virtual PBXs on a single Asterisk® server. It enables Internet Telephony Service Providers (ITSPs), telephony resellers, and VoIP entrepreneurs to offer a fully-featured hosted PBX to their customers, providing them an economical alternative to premise-based PBX hardware.

    Thirdlane Multi-Tenant PBX provides an easy-to-deploy, hosting-ready solution. The software is provided as a complete package, ready to install with all the components needed to build a powerful PBX hosting platform. With Thirdlane’s field-proven technology, you can offer great value to your customers by providing reliable hosting services and a customizable set of advanced features at a very competitive per-user cost compared to equivalent solutions.

    Thirdlane also provides system administrators tools for easily provisioning and managing PBX tenants, and the ability to provide each customer their own customized feature set to meet virtually every need. Thirdlane’s Asterisk-based Multi Tenant PBX boasts an open architecture and an extensive API (application programming interface) to provide interoperability and ease of integrating with third-party components such as service provider portals, billing systems, CRMs, and other programs.

    Thirdlane Connect adds unified communications including private and group chat, voice and video conferencing, integrations, screen and file sharing to the latest version of Thirdlane Multi Tenant PBX platform .


    Key Features:

    • High- Availability: Redundancy, automatic failover, and clustering capabilities are built into the core of all Thirdlane PBX solutions. The high availability features can be configured during installation or added later to address your requirements
    • Auto-Provisioning: Thirdlane PBX is available with templates for auto-provisioning of devices such as Aastra, Cisco, Linksys, Polycom, Snom and Yealink phones and ATAs. This allows you and your customers to easily add phones and devices, individually or in bulk. Templates can be customized or added to support new devices.
    • Database-driven engine: To access the power of the Asterisk® real-time telephony, Thirdlane can utilize a database-driven engine for configuration and administration. This provides robust scalability and clustering options, and allows provisioning of “adds and changes” without reloading Asterisk.
    • Fine-grained permissions: Allow your administrators or customers to easily control each user’s call permissions per user or per group as appropriate, and to configure dialing rules on a per-route basis for operational economy and flexibility.
    • Customizable for your Company and Users: Upload your company name and logo to rebrand the PBX, select from Communications Manager GUI themes and languages, add language translations, and even customize GUI menus and configuration files. Since scripts are built in as part of the Thirdlane architecture, you can easily add custom scripts to support customer-requested features.
    • Retain Full Control of your Server: Unlike other solutions, Thirdlane doesn’t deny you root access to your server for troubleshooting, integrating with third-party components, and performing system maintenance and customization. Software updates are managed by a Thirdlane repository and are easy to install, to keep your system updated with the latest security features.
    • Industry-Standard Components: Thirdlane employs standard versions of the Asterisk telephony engine and CentOS® Linux, and doesn’t require custom patches. This allows you to update components of your system if needed. Thirdlane also supports versions of Digium®-certified Asterisk and Red Hat® Linux for additional peace of mind in critical applications. ...

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  • 02/15/17--15:44: ThirdLane


  • Third Lane Technologies creates unified communications solutions to businesses, government and public organizations, Internet Telephony Service Providers, and Call Center Operators. Reliability, advanced features, open architecture and great value made Thirdlane products the clear choice for thousands of customers and partners worldwide.

    Recently released Thirdlane Connect and other Thirdlane unified communication applications are free and available for download.




    Thirdlane Products


    Thirdlane offers professional unified communications software solutions for hosted and on-premises deployment. Thirdlane software solutions include:

    • Thirdlane Multi Tenant PBX - Multi Tenant PBX and Unified Communications platform for Unified Communications Service Providers
    • Thirdlane Elastic Cloud PBX - Unified Communications platfom for large scale multi site cloud or hosted deployment by Unified Communications service providers
    • Thirdlane Business PBX - advanced Unified Communications platform for businesses and Internet Telephony Service Providers for both on-premises or cloud deployment
    • Thirdlane Connect - Unified Communication app that adds messaging, voice, video, applications and CRM integrations to Thirdlane Multi Tenant and Thirdlane Business PBX platforms and can be deployed in modern browsers, mobile, and Windows, Mac, and Linux desktops
    • Thirdlane Call Center for businesses and call center operators
    • Thirdlane Mobile Dialer is a simple dialer for mobile devices. Thirdlane Mobile dialer can be installed from Google Play
    • Thirdlane Web Dialer Chrome Extension integrates Thirdlane Connect with Salesforce, Zoho, Zendesk and other CRMs and is available from Chrome Web Store.


    Thirdlane software solutions are built with standard, proven open-source components, including Kamailio SIP Server, Asterisk® PBX platform and CentOS® Linux. Thirdlane gives you the the freedom to choose your own devices and customize your system to fit your needs, ability to deploy anywhere, plus the quality, security, and reliability that comes from professionally-designed, managed, and supported software.

    Thirdlane Forums


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  • 02/16/17--07:28: Bicom Systems


  • Bicom Systems provides the Communicating World with the most Complete Turnkey Communication Systems available by Creating, Unifying and Supporting the Most Advanced of Current Technologies.

    History


    Bicom Systems Ltd. was founded in 2005 to exploit its PBXware product.
    PBXware was the first Commercial Turnkey Telephony System to use Open Source software including Asterisk.

    Among the first customers to use PBXware was Redhat. The business model of Bicom Systems does hold similarities to Redhat in the manner by which it wraps Open Source software in a professional and charge for model, warranted to work.

    In 2008 Bicom Systems delivered a custom built conferencing solution to NASA to facilitate the holding of scientific study groups such as the Inter Planetary Conference.

    In 2009 Bicom Systems launched its Multi-Tenant Edition of PBXware.

    In 2010 Bicom Systems began a relationship with NEC to provide a hosted Telephony Platform to businesses across Australia.

    Bicom Systems published How to Grow an ITSP.


    Products



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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):

    A Zycoo Asterisk 13 System



    A Zycoo Asterisk V.13 base VoIP phone systems is user friendly and license free IP PBX system, includes optional telephony interfaces: PSTN, GSM, WCDMA, E1/T1, ISDN BRI. UC Pro& CooCall Softphone expand user's mobility, security, productivity and collaboration.

    Visit our website Zycoo to drop us an email or give us a call for more information.

    • VoIP Phone Systems
      • CooVox-U20 2FXO/FXO+FXS/2FXS/1GSM/2xISDN BRI IP PBX, 30 users & 15 concurrent calls.
      • CooVox-U50 PSTN, GSM, WCDMA IP PBX, 100 users & 30 concurrent calls.
      • CooVox-U80 PSTN, GSM, WCDMA, E1/T1, ISDN BRI IP PBX, 200 users & 60 concurrent calls.
      • CooVox-U100 PSTN, GSM, WCDMA, E1/T1, ISDN BRI IP PBX, 500 users & 100 concurrent calls.

    • Mobility (Remote worker)
      • CooCall Softphone Free, G.729, Voicemail, Call recording, Presence, Conference, Phonebook, Multi-language, Android & iOS.

    • UC&C
      • UC Pro Windows and MAC desktop, full control of handsets from PC, integrate with wide range of 3rd party CRM and applications. Window pop-up, click-to-call, presence, messaging.

    • Billing
      • CooBill Enterprise phone billing system

    AvatarDialler




    Avatar Asterisk base PBX is combination of Hosted PBX and IP PBX, to communicate and connected with other end users, sometimes it work like premise base PSTN and IP PBX system. Integrate with CRM technology, expand user’s productivity, local caller identification, conference calling and emails to voice mails, messages and fax, calls with recording with single click and automated call distribution to ease agent workload.

    For more queries, give us a call at: (+1)6467571041 & (+44)2037696777 or
    Login for the demo at

    Demo



    Bicom Systems



    Bicom Systems is the creator of

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  • 02/22/17--09:26: VoIP Service for Schools
  • In schools, effective communication is essential for assisting administrators in managing the learning environment, helping teachers deliver quality education, and providing accurate information that can nurture the minds of students. Through a reliable phone solution like VoIP, administrators can provide students and teachers with the best communication tools they need to improve the learning process and to achieve success.

    Whether you're searching for a reputable VoIP provider in your area or are looking for authentic user-submitted reviews on VoIP services, Voip-Info can provide you with relevant and helpful information on various VoIP phone solutions for schools. You can search our site for tips, configurations, software, hardware, and guidelines while also visiting our VoIP forums to help make your transition to VoIP smooth and easy.

    How Can Schools Benefit from Using VoIP?

    A VoIP phone system is a dynamic communication platform that can help improve the learning process and enhance the safety of students, teachers, and faculty members.

    Schools need a reliable phone system with handy features to make communication accessible for teachers, staff members, parents, and students. Through an efficient communication solution like VoIP, schools can improve their mobility and productivity, build stronger teacher-student relationships, and enhance the overall learning experience.

    VoIP features such as voicemail, instant messaging, and video conferencing can make collaboration easier for teachers, outside field experts, and students. Through video conferencing, students can go on virtual tours to enhance their knowledge on certain subjects. Students can also work with other students (from different schools) or talk to field experts without leaving the school campus.

    Also, using a VoIP phone system can improve teacher-parent relationships. Through a VoIP phone solution, teachers can pick up e-mail messages or listen to voicemail over a mobile phone and then send a message or response to parents. Features such as hunt groups, call routing, and find me/follow me can also enable teachers to answer calls from parents even when they are off campus. This way, parents can easily keep track of their children's performance in the classroom and teachers can respond to inquiries in real-time.

    Additionally, a VoIP phone system enables teachers to communicate with students and outside field experts remotely. Because of this, teachers do not need to use a physical room or special network during a discussion or training because a virtual classroom can be established via mobile phones, desktops, tablets, laptops, and other devices.

    Plus, with the use of the call recording feature, students can listen to lectures in a virtual setting. They can easily record calls and store them in an audio format, and then use these files during study groups and mentoring programs. In this way, students can expand their knowledge and enhance their educational experiences.

    How Can VoIP Increase School Safety and Security?

    A robust VoIP system can help administrators to better manage their schools and to build safer learning environments for students, teachers, and other personnel.

    With the use of a VoIP phone system, school administrators can make the learning environment safer and more secure. One great way to enhance the safety and security of students, teachers, and other personnel inside the school campus is to integrate an emergency notification system. By making use of a VoIP-enabled emergency notification system, schools can increase their ability to manage any potential threats. With the help of this system, schools can immediately alert authorities to potentially dangerous situations.

    Aside from this, using instant messaging and group paging features of VoIP, administrators can also help notify students, teachers, and other school personnel during emergencies by sending messages to their desk phones or email. This way, school administrators can communicate with students, teachers, and staff members even when there is a power outage.

    Additionally, advanced VoIP features such as group paging can heighten school security because a single call can alert all offices and classrooms. Through this function, school administrators can page students, teachers, and other personnel by pre-programming a set of extension numbers and then dialing one extension number. In this way, administrators can relay messages or broadcast announcements throughout the entire campus to alert students, teachers, and staff members in case of an emergency. ...

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