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  • 11/12/15--08:43: Asterisk Appliances
  • An Asterisk appliance is a computer with Asterisk installed or “embedded” along with an Asterisk GUI, and FXO and FXS ports. Generally, Asterisk appliances have smaller form factors than desktops or some server models, and are highly efficient at running an Asterisk PBX.

    Asterisk appliances are useful for small businesses wanting a complete PBX that is efficient in size, performance speed, and energy usage. Asterisk appliances are less expensive than typical PBXs, ranging from $200-$1000.

    Asterisk appliances can be custom-made or pre-built from proprietary sources, or a user can build their own. Because Asterisk can run on nearly any computer, there is a wide variety of choice when it comes to the individual components of an Asterisk appliance. For example, Asterisk can be configured, depending on the modules, for 32 or 64-bit architecture. Some Asterisk appliances can use netbook chipsets such as the Intel Atom for a smaller form factor.

    When building an Asterisk appliance, compiling modules can restrict chipset and operating system. For example, the Astlinux project must use the uClibc compiler rather than the more common GCC compiler, which makes Astlinux incompatible with Digium’sG.729a codec.

    Proprietary Asterisk appliances come in many forms and do not require the user to compile code. Rhino’s Ceros uses Intel Core i3, 160GB hard drive and 1GB RAM with whatever Asterisk distribution the user chooses. The Phonebochs Telephony Appliance from Rockbochs, on the other hand, uses Intel Core Duo Mobile and Trixbox.

    Asterisk Appliance from Digium

    Digium, Inc has its own product called Asterisk Appliance 50. The server is designed for small businesses (up to 20 users) and can function as a hybrid phone solution or complete VoIP PBX.

    Other Proprietary Asterisk Appliances

    • Atcom
    • BroadTel iPBX-400: 4-port FXS/FXO IP PBX embedded with Asterisk, plz contact for OEM/ODM
    • dmlink : DMLINK Telephony Appliance
    • beroNet: beroNet Telephony Appliance
    • Fonality Reviews Reviews)): Trixbox Appliance
    • Linksoft : low-power, high-performance IP-PBX appliance with redundant disks and industry-leading 30-month warranty, supporting up to 300 extensions and 180 concurrent calls.
    • Pika Technologies: Warp Appliance

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  • 11/12/15--11:02: VOIP Phones
  • This page is for listing brief details of VoIP Phones including details, where to buy, specifications, and any other relevant VoIP Phone information. Please read the Posting Guidelines for Promoting Products and Services before adding to it.

    Hard Phones

    Standalone Ethernet Hard Phones (voice only)
    An Ethernet hard phone is a self contained IP telephone that looks just like a conventional phone but instead of a conventional phone jack, it has an Ethernet port through which it communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone it does not require any personal computer nor any software running on a personal computer to make or receive VoIP phone calls. It can be used independently, all that is required is an internet connection. While PC based software solutions are cheaper, a hard phone is the best solution for IP telephony.


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  • 11/12/15--20:07: Sip Trunking Providers
  • This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

    Country specific pages:

    What Is SIP Trunking?

    Traditional phone calls go over phone lines, but with the rise of the internet, Voice Over IP communication systems give you much more functionality to receive a high volume of calls. This is called SIP (session initiated protocol) trunking. These services are offered by the majority of VoIP providers and can be tailor made to suit the needs of a small business all the way up to a large corporation with a gigantic workforce and everything in between.

    One of the main advantages of going with SIP trunking is that it will integrate everything from data, the internet, your video, and your voice services all into a single line. The term "trunking" is a bit dated and goes back to a time when traditional telephone services were used and it described how a wide variety of telephone users could share a much smaller pool of communication paths, thus making it a much more efficient option for businesses than getting a separate phone line for each worker. That was in the past, though, as today's SIP trunking capabilities will even allow all calls to be considered local calls and won't even require the business to have a single phone line at all.

    The Benefits of Using SIP Trunking Services

    Choosing to use a good SIP trunking service will significantly level the playing field for smaller business to compete with large corporations and appear much more professional and bigger than they actually are. For larger business, one of the main advantages is being able to much more easily maintain that local presence. Regardless of the size of the business itself, SIP trunking's biggest benefit overall is simply the reduction in cost to the company's overall expenses. This is particularly true for SIP VoIP services that are specifically crafted for a business in particular.
    • It offers very low cost calling.
    • It's much easier to scale than other options, making it very future proof.
    • SIP trunking makes moving a breeze, as you can easily take your phone number with you while moving offices or even while traveling.
    • Network outages are much less impactful, as incoming calls can easily be routed to other locations.
    • It's ideal for any sized business with at least 25 physical phones.
    • It's a fantastic choice for any business that has an international location.
    • It offers the same sort of emergency usages and will even offer enhanced 911 dialing options.

    How SIP Trunking Can Take Your Business To The Next Level

    It used to be that you had to have a PRI (Primary Rate Interface) in order to have video, voice, and data all on a single line. This was a physical device that had to be purchased separately but along with your chosen phone provider's calling plans. This is no longer the case with SIP, as it allows your provider to set up a proxy server (commonly referred to as a SIP proxy), which will provide the outgoing, incoming, and PBX calls for you. Everything is located off site and handled by a professional IT team or contractor through your provider, meaning it saves you a ton of time and money on the technical support side of the equation.

    Being temporarily shut down means simply having the SIP reroute calls to a different office and also allows you to much more easily outsource to international locations, such as with customer service centers.

    All of these benefits are much better than what traditional PSTN (public switched telephone network) have been able to provide through their services and will give your business just what it needs to take it to the next level. ...

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  • 11/14/15--21:14: Elastix
  • Elastix is an appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. It also adds its own set of utilities and allows for the creation of third party modules to make it the best software package available for open source telephony.

    The goals of Elastix are reliability, modularity and ease-of-use. These characteristics added to the strong reporting capabilities make it the best choice for implementing an Asterisk-based PBX.


    Elastix has multiple features and functionalities related with all the services available: IP Telephony, Mail Server, Fax Server, Conferences, Instant Messaging Server, among others. New features, functionalities and Services are added at the development of new versions.
    Below you will find a detail list of features and functionalities:


    • Call recording
    • Conference center with virtual rooms
    • Voicemail
    • SIP and IAX support, among others
    • Voicemail-to-Email functionality
    • Supported codecs: ADPCM, G.711 (A-Law & μ-Law), G.722, G.723.1 (pass through), G.726, G.728, G.729, GSM, iLBC (optional) among others.
    • Flexible and configurable IVR
    • Support for analog interfaces as FXS/FXO (PSTN/POTS)
    • Voice synthesis support
    • Support for digital interfaces (E1/T1/J1) through PRI/BRI/R2 protocols
    • IP terminal batch configuration tool
    • Caller ID
    • Integrated echo canceller by software
    • Multiple trunk support
    • End Point Configurator
    • Incoming and outgoing routes with support for dial pattern matching
    • Support for video-phones
    • Support for follow-me
    • Hardware detection interface
    • Support for ring groups
    • DHCP server for dynamic IP
    • Support for paging and intercom
    • Web-based operator panel
    • Support for time conditions
    • Call parking
    • Support for PIN sets
    • Call detail record (CDR) report
    • Direct Inward System Access (DISA)
    • Billing and consumption report
    • Callback support
    • Channel usage reports
    • Support for bluetooth interfaces through cell phones (chan_mobile)
    • Support for call queues
    • Elastix Operator Panel (EOP)
    • Distributed Dial Plan with dundi
    • Voip Provider configuration
    • Asterisk Real Time
    • Distributed Dial Plan with dundi
    • Voip Provider configuration
    • Asterisk Real Time


    • Fax server based on HylaFax
    • Fax to email customization
    • Fax visor with downloaded PDFs
    • Access control for fax clients
    • Fax to email application
    • Can be integrated with Winprint Hylafax
    • SendFax Module
    • Fax send through Web Interface
    • SendFax Module - Fax send through Web Interface


    • Online embedded help
    • Centralized updates management
    • System resources monitor
    • Backup/restore support via Web
    • Network configurator
    • Support for skin
    • Server shutdown from the web
    • Configurable server date, time and timezone
    • Access control to the interface based on ACLs
    • Backups on a FTP server
    • Heartbeat Module
    • Elastix Modules at RPMs
    • DHCP Client List Module
    • Automatic Backup Restore
    • Backup Restore Validation
    • DHCP by MAC
    • Elastixwave
    • New Dashboard
    • Elastix News Applet
    • Hardware detector enhancement
    • Telephony Hardware Info
    • Communication activity applet
    • Process Status Applet


    • PBX-integrated calendar with support for voice notifications
    • Phone Book with click-to-dial capabilities
    • Two CRM products integrated to the ...

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  • 11/15/15--15:19: Grandstream Retailer
  • List of Grandstream Retailer

    The list is split by continent to make it easier to find a retailer.

    North America

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  • 11/15/15--15:31: Grandstream

  • Grandstream is a manufacturer of SIP VOIP products.

    The current product range includes hard- and software VoIP phones, analog FXS and FXO adapters, IP PBX as well as video conferencing equipment and IP based video cameras.

    The list of retailers has been moved to a separate page. Please see Grandstream Retailer for the retailer list.

    At the end of the page is a list of older, out of manufacturing products. Some of them are still available through retailers and should still be fine to use.


    List of current hardware phones
    Model video SIP accounts screen PoE highlights
    GXP1630 no 3 132 x 64 yes 2 port Gigabit switch
    GXP1628 no 2 132 x 48 yes 2 port Gigabit switch
    GXP1620 no 2 132 x 48 no 2 port 100 mbps switch
    GXP1625 no 2 132 x 48 yes 2 port 100 mbps switch
    GXP1610 no 1 132 x 48 no 2 port 100 mbps switch
    GXP2160 no 6 480 x 277 4.3" color yes 2 port Gigabit switch, Bluetooth
    GXP2140 no 4 480 x 277 4.3" color yes 2 port Gigabit switch, Bluetooth
    GXP2130 v2 no 3 320 x 240 2.8" color yes 2 port Gigabit switch, Bluetooth
    GXV3240 yes 6 480 x 277 4. ...

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    Business VoIP Providers - Compare and Choose a Business VoIP Provider

    Quality business VoIP providers today offer a wide variety of feature packages, services and prices. Selecting the ideal provider and service options will depend on your type and size of business, features needed and projected volume of usage. Even when working with top-tier providers, your basic monthly service charges per line may begin at rates as low as $20. Before choosing your VoIP provider, it is essential to first determine your company's precise telecommunications needs to enable timely and cost-efficient initiation of your service. By consulting your chosen Voice over IP service team and seeking their expert advice in advance, you can be prepared to take the following steps to facilitate the smooth, productive startup of your services:

    • Evaluate Your Internet Connection. - Determine the strength and capacity of your Internet connection and bandwidth. You need to ensure that your system has adequate speed to best accommodate your new VoIP installation for top quality service.
    • Assess Your Company Budget and Needs. - With knowledge of your company's current budget and VoIP needs, you can more easily select the service provider and feature options that meet your requirements.
    • Determine Your Equipment Needs. - Evaluate your current and near future VoIP equipment needs. Phones can be purchased from around $50 to $500 or more. Once you decide which feature options are immediate requirements and which ones can be added later as needed, you are ready to choose your service provider.
    • Compare VoIP Providers. - By comparing VoIP company service options, advanced features and equipment along with user and industry reviews, you can best make a wise decision, selecting the ideal VoIP provider for your enterprise.

    Important Information to Request from Any Potential VoIP Provider

    Before signing a service contract with any business VoIP provider, be sure to request basic service information and practices in writing. You need to be certain of such details as startup costs and monthly fees, any limitations and costs on portable phone numbers and exactly which features are included in the service package you select. You also need to know if international calling is included, charges for adding extra features and the extent of customer care and technical services provided. Also important are such issues as whether your provider offers a money back guarantee and if there are any cancellation fees. It is also helpful to determine prior to signing up for VoIP services if there are any hidden fees assessed by your chosen provider.

    Take Full Control and Advantage of Your VoIP System

    Once your new business VoIP system and service are in place, you and your staff members will have full-control capabilities for use of your business communications system. Your service provider will ensure connection with your online portal for customizing your telecomm options. These modern digital portals are user-friendly, enabling feature changes and additions to be made for immediate availability. You and your staff can make decisions and changes in real-time that work for you right in the moment.

    You can manage your call settings remotely, directing calls to voicemail or having them transferred to another number or extension. You can also make exceptions to any chosen setting in your phone system. For example, if you are expecting an important business call and want to take that call, but hold all other calls for a few hours, you can set your phone to direct only the designated call to ring on your extension. This system allows and encourages you to take complete control of your telecommunications systems and settings so that the service works for your best interests and immediate needs at all times.

    Major Business Benefits and Advantages of Installing VoIP

    With an excellent quality VoIP system installed and running well in your company offices to provide remote access for you and your employees, you can work much more efficiently, achieving more in less time. You will enjoy the many benefits of knowing that you can leave the responsibility of your advanced office telecommunications system operations to your VoIP provider while you handle other important business matters. Other major benefits and advantages of your new business VoIP system enable you to accomplish the following:

    • Schedule Your Own Business Hours. ...

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  • 11/16/15--06:25: 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; $12 per month membership fee plus 0.00247 per transaction. That's less than 1/4 penny per lookup. SIP / HTTPS support. Free trial, Real-time (never cached) data. NO contracts, NO monthly minimums, and membership gives you access to our other services such as carrier lookup, phone append, etc. This special pricing requires that you call us to verify you are planning on using the CNAM data for it's intended purpose; caller-id name for inbound phone calls. More info: We have multiple SS7 interconnects and are able to provide you with some of the most accurate and up to date results. Pricing is $0.004 per query for our Free tier and it goes to $0. ...

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  • 11/16/15--10:14: MacOS X
  • MacOS X and Darwin

    MacOS X is Apple Computer Inc's operating system, powered by a BSD subsystem called Darwin.


    Open standards based SIP VoIP technologies available on MacOSX

    • Asterisk - the open source PBX and multi-protocol VoIP server
    • Blink - A fully featured SIP client for MacOSX (GPLv3) with support for IM, File Transfer and Desktop Sharing based on MSRP
    • REMWAVE - A SIP client for MacOSX (GPLv3) geared towards ITSPs
    • iaxComm - the open source IAX softphone
    • ZoIPer - Multiplatform IAX soft phone (universal binary) (formerly Idefisk)
    • JackenIAX - IAX Soft Phone for Mac OS X.
    • LoudHush - nagware IAX softphone for OS X
    • SJphone - SJlabs' H.323 and SIP softphone
    • SylkServer - Multiparty multimedia conference server, free for Linux, OSX version is a paid application in the Mac App Store
    • Telephone - the open source SIP softphone written in Objective-C/Cocoa
    • VDK - Voip Development Kit - the cross-platform, c++ framework to create VoIP applications in a breeze.
    • X-Lite - Counterpath's free SIP softphone
    • Bria 4 - Counterpath's Full Featured SIP and XMPP client ($)
    • Xmeeting - H323/SIP Audio Video Max OSX client , universal binairies.
    • VoixManager Is an an IAX/IAX2 switchboard call manager for Operators and Receptionists, is able to manage and display information about your Asterisk PBX activity in real time. Has been thought with simplicity in mind, all feature needed by the attendant, fast and easy usable, with the minimum configurations, just fill the phone and manager login information and play. ...

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  • 11/16/15--17:32: VOIP GSM Gateways
  • What's a VoIP GSM Gateway?

    A VoIP GSM Gateway enables direct routing between IP, digital, analog and GSM networks. With these devices (fixed cellular terminals) companies can significantly reduce the money they spend on telephony, gp-especially the money they spend on calls from IP to GSM. The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).

    Through least cost routing the gateways select the most cost-effective telephone connection. They check the number which is dialed as well as rate information which is stored in an internal routing table. Because several SIM cards and GSM modules are integrated within the VOIP GSM Gateway it is able to make relatively cheaper GSM to GSM calls instead of expensive IP to GSM calls.

    Who offers VoIP GSM Gateways?

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  • 11/16/15--20:08: Asterisk CRM Integration
  • Asterisk, being open source, can be integrated with a whole lot of different software solutions to enhance, improve and facilitate general users. One of such technological marriages is Asterisk-CRM integration. Asterisk-CRM Integration can provide your Asterisk with the ability to manage your customers directly. The CRM integration to Asterisk provides notes uploading, call recording, customer management, number lookups and a number of other benefits that a standalone Asterisk or independent CRM can not provide.

    There are number of products available in the market to achieve this goal. This page lists all of such products and solutions with alphabetical sorting.

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  • 11/16/15--22:07: VOIP Event Calendar
  • 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 Event:

    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 GITEX 2015 - 35th annual GITEX Technology Week in Dubai, UAE, 18-22 October. Speedflow will be among the recognizable exhibitors at the show, which acts as a hub for the Middle East, Africa and South Asia technologies industry.

    September 2015

    • 28 – 29 INDO AFRICA ICT EXPO 2015 - At Indo - Africa ICT Expo 2015, the latest innovative ICT technologies covering wide range of products, services, and software applications will be on display and open for technology transfer.

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  • 11/16/15--22:27: Call Center Software
  • Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

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

    Types of Call Center Software

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

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

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

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

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

    Contact Center Software

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

    Call Center Software Providers

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

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

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  • 11/16/15--22:57: trunking gateway
  • A trunking gateway is an interface between VoIP and PSTN. It is a device whereby the VoIP line and PSTN line are connected so that an end user can use PSTN phones to make a call over VoIP.

    Protocols On E1/T1 side:SS7, PRI, QSIG, R2, V5.2; on IP side: SIP, MGCP, H.248

    PDH system includes two major communication systems, ITU-T E1 system and ANSI T1 system.
    The E1 system is dominant in Europe and some non-Europe countries. The T1 system is dominant in USA, Canada and Japan.
    One major difference between them is :
    E1 provides 2.048 Mbps bandwidth but T1 provides 1.544 Mbps bandwidth.

    Trunking Gateways

    Synway E1/T1 Trunk Gateway, SMG2000/SMG3000


    The SMG2000/3000 trunk gateway connects IP and hybrid networks via telephony and Ethernet links in a compact 1U form factor appliance. It also transforms media and signaling to support efficient and reliable voice, fax and multimedia sessions for mobile, fixed and cloud-based applications.
    ● Support 1/2/4/8//16 E1/T1 in 1U Space
    ● Support SIP and SS7/ISDN/R2 and More
    ● Telco-grade Redundancy & Dual Power Supply

    Online Demo
    Account: admin; Password: admin

    Home page:
    Skype ID: x.alex_huang.x
    Telephone: +86 571 88860561

    Roytel E1/T1 trunking gateway, RT-EIMS2002


    Roytel RT-EIMS2002 is a digital trunking gateway which connects PSTN/PBX to IP network. It has carrier-class performance and adapts slot insert frame which can be easily configured as requirements. It provides a maximum of 8 E1 ports in one single device.
    RT-EIMS2002 offers 5 modes in one single device, 1/2/4/6/8 E1/T1 ports.
    ● Carrier-class performance
    ● Optional and extendable interfaces boards
    ● 1.5U standard rack-mounted
    ● Web console
    ● Flexible call routing
    ● Support interconversion among multi-protocols, E1<-->IP, E1<--->E1, IP<-->IP

    Application networking diagram
    The trunking gateway is deployed for telecom carriers.
    TG2002 for Carrier application.jpg

    The trunking gateway is deployed for enterprise.
    TG2002 for enterprise application.jpg

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  • 11/17/15--09:07: Asterisk Paging and Intercom
  • Paging and Intercom

    On legacy phone systems you can find the following kinds of paging:

    • Dial a code to connect to a separate overhead paging and announcement system (like in an airport)
    • Dial a code and connect directly to a built-in one-way announcement speaker on one or more phones
    • Dial a code and connect directly to a built-in two-way announcement and talkback function on one or more phones

    Some overhead paging systems also provide a talkback system so that the person being paged can just speak to respond. Background noise issues limit where this feature can be used. The talkback function is usually setup to be hands free. That means that the person responding to the page does not need to take any action other then speaking.

    New in Asterisk 1.2: The new dialplan command Page utilizes MeetMe to page one or more phones.

    New in Asterisk 1.8: A new RTP engine and channel driver have been added which supports Multicast RTP.
    The channel driver can be used with the Page application to perform multicast RTP paging. The dial string format is:

    MulticastRTP/<type>/<destination>/<control address>

    Type can be either basic or linksys. Destination is the IP address and port for the RTP packets. Control address is specific to the linksys type and is used for sending the control packets unique to them.

    Advanced Paging / Intercom

    There is also another system available since many years, the best one, combining paging and intercom. Here the talback system is limited to only one phone. The paging is done in one way mode through a group of phones, and the person being paged can respond pressing a digit to switch the nearer phone to two-way mode, simultaneously hanging-up all other phones speakers.
    This mode combine the best of the two world, eliminate the noise problems, and keep the communication private as soon as the paged person pressed the right digit on a phone.
    It should be possible to implement this mode on Asterisk with a managed conference and a feature map application.

    Multicasting begin to be supported at all major phones manufacturers, Aastra firmware v2.2, Snom v7, Linksys,... allow the setting of a multicast listening address. This will permit to reduce the generated trafic for an extensive paging.

    If a phone is in use when a page arrives, some systems can do a "whisper page" so that only the person being paged can hear the page.

    SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. The phones most often mentioned supporting this are:

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  • 11/17/15--10:10: trixbox
  • Image

    Up and running in about 15 minutes

    This page has been viewed {HITCOUNTER} times since being created on {CREATIONTIME}

    trixbox is an open telephony platform utilizing the best of the open source telephony tools into one easy-to-install package. Based on an enhanced LAAMP (Linux, Apache, Asterisk, mySQL, PHP) the trixbox dashboard provides easy to use web-based interfaces to setup, manage, maintain, and support an complete IP PBX system.
    Succeeding Asterisk@Home 2.8, trixbox has been loosely described as 'Asterisk@Home 3.0'. It offers improved stability and the promise of an upgrade process that doesn't require you to wipe the entire install and start over.

    Polycom support for trixbox now includes Paging and Intercom, Buddy Lists, and BLF functionality.


    trixbox 2.8 contains the following:
    • CentOS 5.3
    • Asterisk 1.6
    • DAHDI
    • mySQL
    • Apache
    • PHP
    • PBXconfig 5.5
    • VoIP Setup Wizards
    • Admin status screen
    • Network configuration tool
    • Telephone provisioning for Linksys, Polycom, Snom, Grandstream, Cisco, and Aastra
    • much much more

    Official Resources

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  • 11/17/15--11:08: Asterisk call notification
  • Call notification tools

    Possible instant messaging (and other) tools you could employ:

    1. smbclient (SAMBA)
    2. YAC (windows & TiVo listener)
    3. Jabber (various methods), ICQ, MSN, ...
    4. Covide (CRM software)
    5. Agile CRM (CRM software)
    6. FOP (Flash Operator Panel)
    7. app_notify network caller notification. Mac OSX client available (3rd party tool)
    8. app_broadcast: Works with YAC etc. (3rd party tool)
    9. asteroid (Open source, JAVA)
    10. iBCallMan (Windows Desktop Call Manager)
    11. PL Call Notify
    12. MythTV OSD
    13. NetCID (talking callerid)
    14. Emerald Caller-ID Search
    15. ADM - Asterisk Desktop Manager
    16. U-Rang II (Screen Pop Utility for Windows)
    17. Asterisk Dial & Announce Tool (ADAT). Free call-notifier (Outlook/CRM integration, event handling, click-to-dial, BLF panel and more).
    18. OutCall (Pop-Up as well as Outlook integration, free and open source)
    19. Email
    20. CallerIDpop Perl Script
    21. DialApplet, Windows, Mac & Linux
    22. AsteriskCallNotification Mac
    23. Asterisk+Twitter
    24. VoipOperator Free Windows call notification and dialer for Asterisk. Very easy to use and install.
    25. InGenius Connector Search-and Dial PC Application, Caller ID pop, integration with MS Outlook, Active Directory and Google. Optional Softphone.
    26. Thirdlane Dialer Free dialer and screen-pop application for Asterisk based PBXs. Integrated with Thirdlane Business PBX and Thirdlane Multi Tenant PBX , MS Outlook, any web based CRM, dual-mode - floating or Outlook toolbar, supports callto URLs.
    27. pyCalledMe Simple Caller ID popup script for Linux. ...

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  • 11/17/15--20:37: Synway
  • synway.gif

    Synway's achievement simply bases on continuous improvement of product quality and program-focused management, both of which are original from customers' demands and efficient synergy between customers and Synway. Program-focused management enables Synway to become a non-boundary and across-organizational corporation who simply cater to your needs.

    VoIP Gateway








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  • 11/18/15--02:32: Asterisk
  • Image

    Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows (emulated) and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

    Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. Check the Features section for a more complete list.

    Asterisk needs no additional hardware for Voice-over-IP, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism (for certain applications such as conferencing). A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers)

    For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsor, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks . ...

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