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  • 01/14/16--03:47: 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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  • 01/14/16--18:32: voip-info.org
  • Welcome to the VOIP Wiki - a reference guide to all things VOIP.


    This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

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


    NEWS


    News Resources


    Getting Started


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  • 01/14/16--18:37: 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

    Algeria

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

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  • 01/14/16--18:59: Asterisk func cut
  • Synopsis:

    CUT(varname,delimiter,fieldspec)
    at least on asterisk 1.2.8 the parameters must be separated by "|" and not ","


    Description:

    CUT(varname,delimiter,fieldspec)

    • varname: variable you want cut (and not a string - see Asterisk func passthru of Asterisk 1.8)
    • delimiter: defaults to -
    • fieldspec: number of the field you want (1-based offset), may also be specified as a range (with -) or group of ranges and fields (with &).

    Notes

    • The delimiter must be a single character. When multiple characters are specified, only the first character is used.
    • To specify a comma or semicolon as a delimiter, escape it with a backslash: CUT(foo,\,,1)
    • When multiple fields are specified, they are joined back together using the delimiter specified.
    • Leaving off one end of the range will give you the full variable at the field and to the other end of the variable.
      • "CUT(somevar,,3-)" would give the 3rd field and everything past,
      • "CUT(somevar,,-2)" would give the 2nd field and everything before.
    • This command is most useful when dealing with fields that are variable width. For fixed width string processing, use the builtin variable substitution.
    • CUT is frequently used to trim the uniqueid section off a channel name. For instance, the channel name might be SIP/somehost-f387 and you might want to trim that to SIP/somehost.
    • This function replaces the application Cut, which is now deprecated.

    The varname parameter must be a variable name, not a string value. This is unusual syntax. So:


    exten => s,1,Set(foo=${CUT(bar,,2)}) ; This is correct syntax
    exten => s,1,Set(foo=${CUT(${bar},,2)}) ; This is invalid syntax (unless bar contains the name of another variable)

    Return value

    Returns the resulting string.

    Example

    exten => s,1,Set(foo=1-2-3-4-5)
    exten => s,2,Set(foo=${CUT(foo,,1-3&5)})

    foo is now set to 1-2-3-5

    Cut a comma separated list:

    exten => s,1,Set(myVar="one,two,three")
    exten => s,2,Set(cutVar=${CUT(myVar,\,,1)})


    cutVar is set to value 'one'

    See also

    • Asterisk func FIELDQTY
    • Asterisk func ARRAY
    • Asterisk func POP: Removes and returns the last item off of a variable containing delimited text
    • Asterisk func SHIFT: Removes and returns the first item off of a variable containing delimited text
    • Asterisk func LISTFILTER as introduced with Asterisk 1.6.2: Remove elements from a set list (by name)
    • Asterisk func REPLACE as introduced with Asterisk 1.8
    • Asterisk func passthru: Literally pass the same argument back as its return value. The intent is to be able to use a literal string argument to functions that currently require a variable name as an argument. (Asterisk 1.8)
    • Asterisk variables



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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):

    SuperSistemas - Switchvox -solution

    switchvoxbest2-lowres.jpg


    EBSolution - Custom IT Solutions

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

    Wayatone Media Inc. - Communication

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


    Svanto.net - Tailored Internet telephony solutions

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

    AAB Asterisk Consultant Lithonia GA

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

    4PSA VoipNow

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

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  • 01/14/16--22:52: 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

    • Intuitive Technology : Evolution PBX Appliance - One of the most widely used Asterisk distributions available since 2005. Highly Mature, Highly Stable, Great Servive
    • Atcom
    • BroadTel iPBX-400: 4-port FXS/FXO IP PBX embedded with Asterisk, plz contact broadtel@126.com 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.
    • PBXinaFlash: http://pbxinaflash.com PBX in a Flash™ allows you to easily setup your own Linux PBX. ...

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  • 01/15/16--10:58: Asterisk system vendors
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  • 01/17/16--01:46: VOIP Billing
  • Hosted Billing Services (in Alphabetical Order)


    • 4PSA VoipNow fully featured, carrier-grade, multi-tenant edition for service providers and businesses, that can be installed on their chosen infrastructure or delivered as a UCaaS. VoipNow provides a fast, competitively priced go-to-market solution, from deployment and provisioning all the way to selling and billing.
    • Adore VoIP Billing Adore VoIP Billing Software comes with the enhanced functionality along with the architecture with class. It is fully compatible and gets integrated with all other VoIP related products. It is designed with all the present and future demands of booming telecom industry kept in the mind. The telecom industry is changing and developing with rapid speed and the VoIP products such as VoIP Billing comes as an excellent product in this time.
    • A2BILLING - VoIP Billing Solution / AAA / Class 5 Softswitch.
    • Aradial AAA for Billing Solutions
    • benotos offers free callshop billing system 4-level billing system: reseller-subreseller-callshop-customer, 2 different routes, nice easy to use interface, intelligent ratemanager, online payment, detailled reports, receipt printing with own logo, white labelled, use your own brand and domain name and much more features. About 9000 callshops around the world are using our excellent callshop billing solution already. Free signup - best rates on market - low payment amounts
    • BillCall - Telecom Resource Management for wholesale Voip Carriers Panamax’s Telecom Billing Solution BillCall provides solutions for End-User billing, Carrier Access Billing (CABS), CDR Mediation, Rating & Routing.
    • CloudAstrix SPE CloudAstrix SPE is such a VoIP Switch. Build on the world renowned WHMCS Billing Suite, the Soft-switch module brings all necessary functions to perform and provide a top class VoIP service.As a Carrier Neutral soft-switch, CloudAstrix has already proven to be a firm favourite among ISPs all over the world.
      • Note:CloudAstrix SPE Module works with FreeSwitch.
    • Cybercallshop ultimate callshop server Incredible advanced online callshop server 100% standalone able to handle many shop, simply the best software to get customer loyalty because it's many more than simple online booth billing.
    • CRM Sipit Enterprise CRM system with fully integrated pdf invoice billing, customer balance, export, import of priceplans. This is a fullblood CRM system with invoice capabilities. Compatible with Kamailio.
    • Cyneric Fully Integrated Billing Solution Platform. Compatible with: Cisco, Radius, Mera, SIP, SER, Asterisk, Quintum, SNOM, Audiocode.
    • DORETEL Communications, Inc. Hosted Calling Card Solution, Hosted Wholesale VoIP Billing Platform, Hosted SoftSwitch Solution, Hosted Call Shop
    • dtlvoip is hosted VoIP billing and switching service provided by DataTechLabs since 2002. ...

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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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  • 01/17/16--23:19: ICT Innovations
  • images.jpg


    ICT Innovations is a software development company having experienced and dedicated professionals with expertise in LAMP Stack and computer telephony integration (CTI). ICT Innovations has strong knowledge of Open Source communications technologies and applications such as Asterisk, Freeswitch, Drupal, Plivo,voip, Elastix and OpenSIPS/ Kamailio.

    Products

    ICTBroadcast

    ICTBroadcast is a multi-tenant unified communications telemarketing autodialer software solution. It supports Voice broadcasting, SMS broadcasting, Email broadcasting & Fax broadcasting. It is suitable for small business owners, enterprises and Internet telephony service providers. ICTBroadcast is a smart auto dialer software with advanced autodialing multilingual supported features.

    ICTDialer

    ICTDialer is a Open Source unified communications marketing software. The ICTDialer is multi-tenant and allows voice, SMS & fax broadcasting. These capabilities have been developed using the Open Source content management system Drupal and the powerful Freeswitch and ICTCore Communication Framework. ICTDialer has the operating capacity to make thousands of simultaneous calls using VoIP, FoIP or PSTN.

    ICTFAX

    ICTFAX is an Open Source (GPL v 3) multi-user, web based software solution for service providers based on Open Source Spandsp, Drupal and ICTCore Framework. ICTFAX is an email to fax , web to fax gateway, it supports G.711 faxing, PSTN faxing and T.38 origination and termination.

    ICTInvoice

    ICTInvoice is an Open Source Elastix PBX module for invoice management. It converts Elastix PBX into a multi-tenant hosted PBX platform suitable for offering hosted PBX services to small business owners and enterprises. It is Open Source GPL v 3 software and is developed and maintained by ICT Innovations. ICTInvoice enhances the capabilities of Elastix billing and empowers Elastix administrators. It allows email invoices to be automatically generated and emailed to users on monthly basis.

    Services

    Consultancy services

    ICTInnovations offer consultancy services to its clients and work with them to fully understand their exact business requirements. ICTInnovations customize, develop, migrate, integrate and provide network architecture and deployment plans to our international client base.

    Support services

    ICTInnovations provide support services for Open Source communications technologies such as Asterisk, Freeswitch, OpenSIPS, Plivo and Drupal.

    Integration and Development

    ICTInnovations offer services to allow the integration of Open Source telephony projects and components into any existing network. ICT Innovations can provide a complete business solution per each individual client's requirement. We will develop tailored API's on request to integrate Open Source VoIP projects with your existing communications infrastructure.

    Monitoring and Support services

    ICTInnovations provide 24/7 support services to monitor VoIP/Linux Servers. This ensures that any outage causes minimal service disruption. We provide immediate support for any issue arising and our professional support team is always available during our client’s business hours. ...

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    Syntax:
    show queue <queue number>

    
    ^
    
     ---- queue Number
     | Agents & Extensions that will be part of this queue
     | |   Calls currently being processed                                          0*    1*    2*      3*           4*
     | |             |                             Ring Strategy   ?? (average?)    |     |      |      |            |
     | |             |                                |                 |           |     |      |      |            |
     V |             V                                V                 V           V     V      V      V            V  
     10V         has 1 calls (max unlimited) in 'ringall' strategy (49s holdtime), W:0, C:210, A:201, SL:0.0% within 0s
       Members:
          Local/1234@from-internal/n (Unknown) has taken no calls yet  <-------------- This extensions is currently doing nothing (?)
          Local/1920@from-internal/n (In use) has taken 12 calls (last was 543 secs ago) <----- This extension is processing a call
    )
       Callers:
          1. Zap/12-1 (wait: 1:02, prio: 0) <-------------------- This channel is waiting
    ^

    0: Queue weight defined in queues.conf
    1: Calls answered
    2: Calls unanswered (People who called, but hang up before getting answered)
    3: Service level (% of calls answered within X seconds)
    4: Time period to calculate service level (see above), specified in queues.conf

    in version 1.4.x, the show queue command is deprecated, and will be removed in a future release. use asterisk cli command queue show instead.


    See alse



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    Synopsis

    show the queue current status

    Syntax:

    Asterisk 1.4 or later:
    queue show [<queue name>]


    Note: the queue name is not necessary.
    If queue name is null, that command will show all queues status.



    
    queue Name
     | Agents & Extensions that will be part of this queue
     | |   Calls currently being processed                                          0*    1*    2*      3*           4*
     | |             |                             Ring Strategy   ?? (average?)    |     |      |      |            |
     | |             |                                |                 |           |     |      |      |            |
     V |             V                                V                 V           V     V      V      V            V  
    10 V         has 1 calls (max unlimited) in 'ringall' strategy (49s holdtime), W:0, C:210, A:201, SL:0.0% within 0s
       Members:
          Local/1234@from-internal/n (Unknown) has taken no calls yet  <-------------- This extensions is currently doing nothing (?)
          Local/1920@from-internal/n (In use) has taken 12 calls (last was 543 secs ago) <----- This extension is processing a call
    )
       Callers:
          1. Zap/12-1 (wait: 1:02, prio: 0) <-------------------- This channel is waiting


    0 Queue weight defined in queues.conf
    1 Calls answered
    2 Calls unanswered (People who called, but hang up before getting answered)
    3 Service level (% of calls answered within X seconds)
    4 Time period to calculate service level (see above), specified in queues.conf


    See alse



    Asterisk | Asterisk CLI | Applications | Functions | Variables | Expressions | Asterisk FAQ

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  • 01/18/16--03:10: IP PBX
  • IP PBX is a phone system that utilizes IP communications. Traditionally IP PBX's are located on site where they can also interface to traditional telco services such as analogue phone lines. The business end users connect via IP to the IP PBX for voice service.


    What is an IP PBX?

    An IP PBX can be referred to as a lot of things: a business phone system, a unified communication system, or simply as a "PBX. ...

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  • 01/18/16--03:51: Session Border Controller
  • A Session Border Controller is a device used in select VoIP networks to exert control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down calls. The SBC enforces security, quality of service and admission control mechanism over the VoIP sessions.

    The Session Border Controller is often installed in a point of demarcation between one part of a network and another. Most Session Border controllers will be installed between peering service provider networks, between the enterprise network and the service provider network, or between the service provider network and residential users.

    A Session Border Controller is like a Firewall for VOIP.
    They are often configured as a SIP Back-To-Back User Agent (See SIP RFC).

    In addition to firewall functions they also may provide services like NAT traversal.

    See: NAT and VOIP for more information and watch the video on The Anatomy of Session Border Controllers and To Couple or Decouple Routing Intelligence from SBC.

    Martyn Davies of Dialogic discusses Session Border Controllers as the often misunderstood "black magic" application for VoIP networks.

    Although carriers often use Session Border Controllers for signal translation and security, most do not include the hardware-based signal processing needed for media transcoding. For all-IP environments, new elements are required that can mediate signaling, transcode among different media formats, and handle basic security issues. The concept of Multimedia Border Element (MMBE) meets these needs.

    Products



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  • 01/19/16--01:19: Asterisk config sip.conf

  • Configuration file for AsteriskSIP channels, for both inbound and outbound calls.

    Starting with Asterisk v1.2.0: The global option "port" in 1.0.X that is used to set which port to bind to has been changed to "bindport" to be more consistent with the other channel drivers and to avoid confusion with the "port" option for users/peers.

    Starting with Asterisk v1.6.0: The previously deprecated options "insecure=very" and "insecure=yes" have now been removed. "insecure=invite,port" is the equivalent of "insecure=very"

    
    [tammari]
    type=friend
    callerid="Tuomas Tammisalo" <1000>
    username=tammari
    host=dynamic
    secret=********
    regcontext=tammari-internal
    regexten=1005
    dtmfmode=rfc2833
    insecure=invite,port
    canreinvite=yes
    nat=yes
    qualify=yes
    context=merus-sipphone
    pickupgroup=1
    callgroup=1
    mailbox=1000@default


    Asterisk as a SIP client

    In sip.conf under [general] add a register definition:

    Format:
    register => user[:secret[:authuser]]@host[:port][/extension]

    or
    register => fromuser@fromdomain:secret@host

    or
    register => fromuser@fromdomain:secret:authuser@host:port/extension

    See also: bug 14367 with a documentation fix for 1.6.

    If you have problems with your network connection going up and down (e.g. an unreliable cable connection) and you keep losing your sip registry, you may want to add registerattempts and registertimeout settings to the general section above the register definitions. Setting registerattempts=0 will force Asterisk to attempt to reregister until it can (the default is 10 tries). registertimeout sets the length of time in seconds between registration attempts (the default is 20 seconds).

    In case of DynDNS issues, for example with myasterisk.dyndns.org changing its IP, you might want to consider taking a look at ddclient to automate a "sip reload" in the CLI.

    P.S. Note for sipgate.co.uk users: /extension must be your sipgate number (this is not true; I am using "99" --jrc) - define one to accept this in your extensions.conf. An alternate port does not seem to work with sipgate.co.uk unless it is defined as the bindport in sip.conf without the [:port] syntax. ...

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  • 01/19/16--08:19: Asterisk consultants USA
  • This is a comprehensive list of Asterisk consultants in the USA (United States). Add your entry here (alphabetical order, by state and company), but stick to states where you have actual presence!

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


    ALABAMA


    Asteria Solutions Group


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  • 01/19/16--16:49: Virtual PBX providers
  • Virtual BPX is a service offering functionality of a PBX without the need to install switching equipment at the customer location. Only VOIP phones need to be installed at the customer site. This makes supporting distributed workers very easy as each requires only and internet connection and a VOIP phone. A business virtual PBX phone system can reduce your monthly phone bill significantly compared to a traditional business phone system.

    What Is a Virtual PBX?

    A PBX, short for private branch exchange, is a telephone system with the capacity to switch calls between different users on local lines while still relying on the same number of external phone lines. With a virtual PBX system, the system is posted and software based without all of the traditional hardware of a physical PBX.

    Virtual PBX Primary Function

    A virtual PBX is used by businesses in a variety of ways. Primarily, companies utilize the system as an auto-attendant to establish preset call transfer options without needing an operator or receptionist. This type of system is capable of performing tasks that include auto-attendant settings, time of day or day of week functions, or even find or follow me sequences.

    One of the most important functions of a virtual PBX system for companies is the software’s ability to establish pre-determined sequences. For example, in some businesses it may be appropriate for the phone to ring to a receptionist or operator first. If the receptionist does not answer in a predetermined number of rings, however, the call is then transferred to a secretary. Again, if the call is unanswered, it can be set to forward to an assistant. Left unanswered by these two individuals, the call can be forward to a manager or even an owner. These call settings are completely customizable and can be based on any number of sequences.

    This type of software is also able to facilitate customized answering menus and sub-menus. The system can be modified to establish appropriate dial prompts leading to a number of different departments within the business, including different sequences on different days. PBXs are used by the vast majority of businesses to establish advanced call routing services.

    Virtual PBX Cost

    A virtual PBX is a complex service; however, that doesn’t mean that it is expensive. In fact, a virtual PBX is typically more cost effective than a physical PBX. The main reason that a virtual system saves on cost is that it does not require the same investment in capital to establish or set-up the call system. Because a virtual PBX is a software or hosted system, it is typically an operational cost, or a low monthly payment rather than a large upfront investment. This aspect alone generally makes a virtual or hosted PBX a less expensive, or at least more cost effective, option compared to the traditional PBX.

    Virtual PBX Benefits

    Aside from offering an effective call system, a virtual PBX presents a number of added benefits for users. As a whole, virtual PBXs lead the industry in business communication choices. This type of system seamlessly integrates the call management system with any existing phones to affordably and effectively deliver better call management. These systems also feature several innovative call features to meet the needs of any business. These systems offer various functions including call routing, follow and find me call forwarding, voicemail notifications, call recording, and more.

    The benefits aren’t limited to the features, though. Virtual PBXs offer virtually limitless application for one or hundreds and even thousands of employees. Likewise, there is not hardware to maintain or constantly upgrade. Considering that benefit, the system is also more cost effective and generally provides for a variety of flexible billing options. The limited maintenance, web-based management, and hassle-free setup alone are often enough to convince a company to switch over to this option.

    PBXs are an important tool in any business that makes and receives nearly any volume of calls. A virtual PBX can dramatically increase the efficiency of a business by effectively managing calls. This efficiency combined with the other numerous benefits of a virtual PBX can virtually transfer the communication capabilities of any company.

    List of Virtual PBX Providers




    OnePipe


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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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  • 01/20/16--02:17: 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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  • 01/20/16--06:46: VoIP Training
  • Companies that provide training in VoIP solutions and technologies:

    • 1 on 1 TrainingCity Hands ON Training, Real Live Instructor Led VoIP & SIP Training. TrainingCity Delivers VoIP Hands ON classes, VoIP Hands ON Workshop and Advanced SIP & SIP Trunking Course. Public classes are scheduled in a variety of locations including San Diego, California every month and both classes are available for onsite customized delivery.







    • Alta3 Research provides worldwide training services to businesses and individuals in all things telecom. Alta3 Research specializes in SIP, IMS, 4G/VoLTE, MEGACO/H.248, Avaya, IPv6, Microsoft Lync, and VoIP.


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