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  • 07/30/15--05:09: Web Hosting
  • Please add information about web hosting and web hosting providers and companies to this page.

    Web Hosting Providers


    Please keep this list in alphabetical order


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  • 07/30/15--08:39: Voicepulse
  • VoicePulse FIVE


    To learn more about VoicePulse FIVE and chat with a Live Representative click here.



    VoicePulse FIVE is our fifth generation VoIP platform serving the residential, SMB and Wholesale segments. FIVE is a completely rewritten platform designed from the ground up to provide Internet telephony in the most powerful, flexible and easy to use way possible. Sign up for a free, no obligation evaluation account and be up and running in just minutes.

    To view a demo of the VoicePulse FIVE portal, click here.

    Company News

    2015-07-17 New Feature Release - Real-time 24 hour usage, Usage Limit Reset, Calling Access
    2014-11-06 VoicePulse Introduces VoicePulse FIVE, the Next generation in VoIP Services

    Pricing

    Business Gateway Pricing click here.

    Ideal for SMB and Enterprise

    Use your existing SIP enabled on-premise softswitch or PBX to make and receive phone calls regardless of call volume.

    Channels

    Channels are included with your Business Gateway at no additional cost

    Endpoints

    Endpoints are included with your Business Gateway at no additional cost

    Phone Numbers

    Choose your number from anywhere in the US or move your existing DIDs to VoicePulse FIVE. There is no fee to port your phone number to VoicePulse FIVE.

    Incoming Calls

    Incoming calls to a U.S. phone number are $.01 per minute

    Incoming Toll-Free Calls

    Incoming calls to a U.S. toll-free phone number are $.029 per minute.

    Outgoing US & North American Calls

    Outgoing calls to the US48 are $.02 per minute

    Outgoing International Calls

    Competitive International rates.
    Look up international termination rates here.


    Account Center


    • View your current Statement Balance
    • Monitor real time costs for usage, Endpoints, Trunks, Gateways, Call Apps, Channels, and E911
    • Make instant payments by credit card
    • Add unlimited Channels or call paths to your Trunk
    • View your active phone numbers
    • See our inventory of numbers
    • Instantly activate new numbers
    • Manage E911


    Customer Support



    Chat with a Live Representative. M-F 9am to 5pm EST


    Supported Protocols


    • Session Initiation Protocol (SIP)

    Supported Codecs


    • G.729a
    • G.711ulaw
    • G.711alaw
    • GSM
    • ADPCM
    • ILBC

    Supported User Agents




    • Asterisk - The Open Source PBX, AsteriskNOW, AA50, SwitchVox
    • Fonality PBXtra
    • trixbox CE, SE, EE, CCE
    • FreeSWITCH

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  • 08/02/15--17:22: VoIP Providers India
  • This page is a list of VoIP service providers in India. Please keep this list in alphabetical order. India VoIP providers looking to add their services can do so in the list below.


    India


    • 91Networks Communication 91Networks is a technology rich networking & consulting company with proven products and solutions.We deliver efficient and reliable business-critical products and connectivity solutions in IT-Telcom, Voice, TDM, IP, VOIP. such as VoIP Gateway,GSM-Gateway,IP-PBX, VOIP Phone,Headsets,IP Camera to help strengthen the networking infrastructure & productivity. call +91 9899735310 or email sales@91networks.com
    • ALTOTELECOMVoIP provider for business and Call Centers - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK www.altotelecom.com
    • Incorpus TeleNetworks- Incorpus is a VoIP provider based in India which provides pinless calling for both residential and business customers. Services provided by us are:- RESIDENTIAL VOIP, BUSINESS VOIP, CALLING CARDS, A-Z WHOLESALE TERMINATION, RESELLER/AFFILIATE PROGRAMS, DIDs, TOLLFREE, PBXs . Contact Asap for a free trial at info@incorpus.in,sales@incorpus.in Skype id:- incorpus.support
    • AVOXI AVOXI Virtual Call Center Solutions - VoIP Service Provider, provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers. Contact Number 1-800-462-8694.
    • CallForwarding - Be present anywhere in the world with toll free forwarding services from CallForwarding.com. Contact Number: 800-231-9802
    • Voxvalley Technologies - provides VoIP based communication applications and customized solutions to SMBs, Enterprises and VoIP telecommunication providers worldwide. VoIP Products & Solutions from Voxvalley Mosip Mobile Dialer, Vox Suite, Vox Switch, Vox Bridge, Vox PC, Hybrid Dialer, Vox Con and Vox IM.
    • NOVANET - A leading Business-class VoIP & Cloud Communication provider to Contact Centers & Enterprise customers. Novanet offers a complete suite of VoIP and Cloud services specifically crafted to suit the needs of a modern day Contact Center.
    • i7 Solutions VOIP Provider International Calls service provider. Provides Voip calling to Home Users, Small Office, BPO, Call Centers, Exporters and corporates. To become resellers. please contact us.
    • Live-TechOffers

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    This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


    Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

    Marketing is NOT ALLOWED on this page. Please describe services in neutral language and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in alphabetical order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!


    Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

    If you like this page, please link to it, so Google and other search engines will consider it more important.

    Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page.

    Providers in other countries/continents can be found here:

    Australia


    • 1300 Numbers Australia - We are specialist carriers of 1300 phone numbers and smartnumbers with a strong focus on providing excellent customer service. Our services include live answering service which is the perfect solution to compliment your 1300 number, full service telecommunications including fixed landlines, business-grade VOIP and hosted PBX, business broadband internet solutions, virtual office services, Fax2Email and Voice2Email. ...

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  • 08/02/15--23:49: VOIP Consultants
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  • 08/03/15--22:16: voxvalley technologies
  • voxvalley-facebook-logo.png

    Company Profile:
    Voxvalley Technologies is a leading VoIP solutions provider that has been servicing to the VoIP based needs of SMBs, Enterprises and VoIP Telecommunication providers. Headquartered in Singapore, with R&D Center in India, and with sales offices in the US, Middle East, Southeast Asia, Bangladesh and India, Voxvalley is known for its excellence in technology-leveraged products and client-focused services. Since its inception in 2010, we have delivered myriad VoIP based communication applications and customized solutions to businesses. Over the years, we have gone from strength-to-strength by incorporating state-of-the-art products, solutions, and applications to our strong business line.

    Voxvalley Products:
    • MoSIP
    • MoSIP+
    • MoSIP Hybrid
    • MoSIP C5
    • Vox Switch
    • Vox Suite
    • Vox Bridge
    • Vox PC
    • Vox Transcoder

    Products Description:

    MoSIP:
    Mobile VoIP Dialer: We provide comprehensive mobile VoIP solutions for VoIP Providers to manage their VoIP business efficiently. It also includes our white label branding solutions where providers can get customized mobile dialers as per their requirements. This will enhance their visibility and market value.
    • MoSIP is a SIP based application that allows VoIP calls over the internet to any landline or mobile worldwide.
    • It supports OS such as iOS, Android, Windows, Symbian and Blackberry.
    • Easy to configure and works excellent when linked with Vox Bridge.
    • Uses limited internet connectivity via 2G/3G/4G/Edge/GPRS/Wi-Fi.
    • Auto switches to stronger signals in case of weakening of existing signals.
    • Unlimited business opportunities to VoIP, ISP, MVNO, MVNE and TELCO businesses.

    MoSIP+:
    • MoSIP+ is improved MoSIP with messaging tools such as IM & SMS for more sociable VoIP Communication.
    • It’s a cross-platform mobile app supporting OS such as iOS, Android, Windows, Symbian and Blackberry.
    • Its messaging tools offer users the cheapest means of communication with their near and dear ones.
    • Uses limited internet connectivity via 2G/3G/4G/Edge/GPRS/Wi-Fi.
    • Unlimited business opportunities to VoIP, ISP, MVNO, MVNE and TELCO businesses.

    MoSIP Hybrid:
    • MoSIP Hybrid is a mobile dialer that enables IP calling and automates cellular dialing through calling cards.
    • Supports internet connectivity via 3G & Wi-Fi.
    • No need to enter PIN while making calls through calling cards.
    • User-friendly interface for easy navigation.
    • Fully automated functionalities.
    • Benefits for both End Users and VoIP Providers.
    • White label branding solutions to VoIP Providers.

    MoSIP C5:
    • MoSIP C5 is an IP PBX mobile client that includes the functionalities of PBX on the mobile device.
    • Now switch calls on the go.
    • Increases productivity as employees can easily communicate among themselves and with customers. ...

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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:

    • WebRTC
    • Asterisk
    • FreeSWITCH
    • Kamaliio
    • OpenSIPS
    • OpenSERs

    To get more details about us or to discuss your specific requirements, contact us:
    Website:https://www.ecosmob.com
    Phone: +1 303 997 3139 or +91 079 40054019
    Email:sales@ecosmob.com

    Stay connected with us on social media!
    LinkedIn:https://www.linkedin.com/company/ecosmob
    Google+:https://plus.google.com/+ecosmob
    Twitter:https://twitter.com/ecosmob
    Facebook:https://www.facebook.com/Ecosmob
    YouTube:https://www.youtube.com/theecosmob


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  • 08/04/15--03:08: Home phone Service Providers
  • This is a list of home phone service providers that use VoIP technology. A digital home phone system can reduce your monthly phone bill significantly compared to a traditional residential phone system. Most home phone service providers listed below offer hosted VOIP solutions. Learn more about home phone service, and view home phone service providers below.

    Residential VoIP services providers
    Vonage
    Magicjack
    Rangatel

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    High Availability (HA) is normally achieved through "clustering" - which means two machines acting as one for a specific purpose. There are many ways to create a cluster, each with its own benefits, risks, costs, and trade-offs. The terms "High Availability" (HA) and "Clustering" can be overused so beware of the hype. Clustering, and HA have specific (and different!) meanings. If you are responsible for creating a high availability cluster for Asterisk, below are the issues and concepts you should be aware of. This page is intended to be a starting point in the design, creation or selection of a High Availability or Clustering solution for Asterisk.

    Note that if you are designing a call center for PSAP (Public Safety Answering Point) / 911 call centers there are specific requirements you must consider. Some are noted below, others are specified by rules/orders from FCC (USA), CRTC (Canada), and similar country specific organizations. (eg: FCC 05-116 order 10). Even if you are not designing for a PSAP, these guidelines are excellent best practices often applied by large commercial call centers.

    Please do not add specific product names/links to this page, it is intended to be product neutral. Don't say "this is the best" because your product/your favorite product uses it.. Stick to facts please.

    Fault Tolerance vs Clustering vs High Availability

    Before you get started, you should know the difference between these three things. They all mean different things, and you need to be careful what you refer to.

    Fault Tolerance

    When a fault is detected, it repairs itself quickly. This must NOT result in an outage (for example, a failed CPU that requires a reboot), but the machine must continue working without an interruption. This is normally achieved with Redundant power supplies, network cards, and RAID.

    Clustering

    When something that is NOT fault tolerant breaks, you must be able to continue on, on different hardware. This may be a CPU failing that requires a reboot., or a kernel panic. At this point, another node should immediately start the services that were on the failed node.

    High Availability

    HA is a concept, rather than a thing. HA refers to the combination of both Fault Tolerance and clustering, as well as network and power design that removes as many single points of failure as possible. ...

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  • 08/05/15--06:06: Advanced Network Devices
  • color logo small.bmp

    Website: www.anetd.com
    Support: www.anetdsupport.com
    Forum: www.anetdsupport/forum.com

    Products:

    BI6I7812.png
    Advanced Network Devices IPClock


    Advanced Network Devices Power over Ethernet (PoE) Products That Easily Connect to Existing Data Networks:


    IP Clock and IP Speaker are Power over Ethernet (PoE) synchronized clocks and intercoms that requires only an RJ-45 connector to connect to existing data networks. Simultaneously, broadcast to both phones and speakers. The clock auto synchronizes and can be used as a scrolling text display and a standard built in microphone allows two way voice communication. Works on existing data networks and inter-operates with Cisco and other VoIP phone systems. Full IP product line can be used together to meet specific needs. Send broadcasts to specific rooms, departments, or groups as needed. Bi-directional audio / intercom supported on most models.


    For more information contact;
    sales@anetd.com
    847-463-2236

    Technical support contact;
    tech@anetd.com

    Supported Software

    IP Clockwise
    Singlewire- Informacast
    Syn-Apps- SA-Announce
    IPcelerate- IPsession
    Bell Commander
    Layered Solutions Inc.
    MessageNet

    Where to Purchase


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  • 08/05/15--08:48: Asterisk SIP media path
  • Asterisk SIP media path


    In a normal SIP proxy, the server is not involved in the media between the phones. With Asterisk, sometimes Asterisk stays in the path. It depends on many variables and configurations.

    Asterisk mostly sets up the SIP phone call with itself in the media path. When the phone call is connected, Asterisk normally sends SIP reinvites to both clients to redirect the media path so that Asterisk does not have to handle the media stream any more.

    If the phones do not support reinvite

    Some clients do not support re-invites. If this is the case, you have to configure asterisk NOT to re-invite.
    See Asterisk sip canreinvite for more information.

    If the phones support reinvites

    Asterisk will bridge a call in some cases and not in others. If codec conversion is required between phones, its stays in the middle. If the two phones can agree upon a common codec, etc, * is not in the middle from a pure communications perpective. In that particular case, what the phone does when the # key is press is totally a function of how the phone was programmed (and not asterisk). If the phone, as an example only, has an implementation bug that says I'm not going to forward the # key to asterisk during a conversation, obviously * can't interpret it.

    If the Dial() statement forces the path thru Asterisk

    Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: t, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more.

    If the call is routed through a queue

    When directing calls through a queue to an extension, asterisk will stay in the media stream if the extension is included in the queue in the usual "0123" format. Using "SIP/0123" will allow a direct RTP media connection to be established between SIP devices.


    See also



    Go back to Asterisk


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    Toll-free termination are calls destined to 8YY destinations. These providers allow you to terminate toll-free calls from the US and Canada for free in some cases. If you have a large volume of calls to toll-free numbers, some providers may pay you for your calls. Carriers who have direct agreements have a higher success in collections. Calls must originate from your network to the Carrier bound to a Toll Free Number.

    There are several elements to a Toll Free Call.
    1) End User dials a Toll Free Number. (Voip or TDM)
    2) End User Provider Network must route this call. (End office elements)
    3) Resellers of this traffic must hand this call off to a TANDEM provider.
    4) Tandem Provider will DIP SMS800 for CIC instructions on every TFN. (see HyperCube)
    5) CIC (IXC) will receive traffic and route to RespOrg
    6) RespOrgs are the Responsible Organizations of Every Toll Free Number.
    7) End User / Owner of the Toll Free Number. (Final destination)

    Without registration required


    Alcazar Networks Inc - Free toll-free termination
    Alcazar Networks - VoIP Services
    • Providing FREE toll-free termination to the US48 800 - 855 - 866 - 877 - 888
    • Full accurate Caller-ID number (ANI) delivered
    • Codecs supported: G711 and G729
    • NO SIP REGISTRATION REQUIRED
    • If you require registration we accept ANY credentials so you can start passing calls immediately.

    ArcTele Communications, Inc - Toll Free Termination
    http://www.arctele.com
    ArcTele Communications, Inc offers a Toll Free termaintion gateway service free of charge.
    - We send your callerID
    - ULAW and G729
    - Send calls in the format of 18XXXXXXXXX
    - Unauthenticated to TOLL FREE NUMBER@tf.arctele.com port 5060

    Broadvox is a leading wholesale VoIP service provider that delivers reliable VoIP solutions for domestic and international businesses.

    Broadvox has 8YY number toll-free termination services that provide switching for 8YY calls originated by customers worldwide; conduct the database lookup and determine to which Interexchange carrier (IXC) the toll free number belongs; then route the call appropriately for call completion.

    For more information, to get a FREE quote on our rates or call us at 216.373.4800 for a FREE consultation


    Call With Us
    https://www.callwithus.com/ , under the heading on the page, "Toll free service:"
    • no registration required, simply send SIP messages to tf.callwithus.com
    • Codec is G.711u. ...

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  • 08/06/15--16:22: VoIP Wholesale
  • Wholesale VoIP Market:


    There is no doubt today that VoIP is taking over the telecom market, and every month increases penetration into services and industries. Competitive carriers are looking at the numerous ways to make money from this exploding technology, but there's a lingering question as to whether it is profitable to deliver VoIP in a wholesale model? Their customers, typically Service Providers, are looking for their ‘competitive advantage' into this ‘lowest price' race, leveraging within three key alternatives for packet telephony : “build” , “buy” or “rent”. Business aspect, there’s no need to invest tens of millions of dollars in wholesale VoIP to join in. Many Telecom Companies have done the work for you. They offer a complete, turnkey VoIP business service and equipment. Now you can start wholesale VoIP business with virtually no investment and yet reap great dividends.

    Wholesale VoIP Resellers:


    In today’s world, Service providers seeking to deliver VoIP to as wide a customer base as possible may find that becoming a wholesale VoIP reseller is the way to go. Wholesale VoIP may be sold to both other service providers and to enterprises or residential customers.

    Reselling IP telephony as a wholesale VoIP company is becoming an increasingly popular business model. For many companies, becoming a wholesale VoIP provider hits the sweet spot between profit and market control. Any firm with a well-established customer base is a good candidate for reselling wholesale VoIP.
    Becoming a wholesale VoIP reseller is not a decision that should be taken lightly. It does, however, offer the potential of being very lucrative if done right.

    Wholesale Consumer Demand:


    An important characteristic of the industry is the complex segmentation of consumer demand and rapid change in the characteristics that are being demanded, both at the end customer and in the intermediate ones (wholesale customers).
    Demand coming from ‘packed customers'? will be significantly different of the conventional telecommunications one, were telephony was the unique service to provide and differentiation was based on tariff-distance paradigm, being today's service offerings closer to data applications rather than telephony. Voice communication (and not old POT telephony) becomes the common feature into several communications applications and devices, but not the unique one.
    Messaging, conference, collaboration, web contact centres, etc … requires a common communication format between parties, which is voice, implemented through VoIP technologies. Heterogeneous and rapidly changing customer demands and products are important dynamic influences on the evolving structure of the telecom industry, resulting into a new value-chain.
    Telecommunication markets evolution will be driven by ‘packed customers' demand rather than networks, technology or finance, changing many decades rules into this industry.

    Finance in Telecommunication Industry:

    Finance institutions had been influencing Telecomm Industry since the beginning, due the business itself was characterized by huge investments, big market shares and bigger capitalization, influencing in many cases top management, who addressed their strategy towards ‘stock' opportunities rather long term and solid business models. WorldCom crash has been an example of this ‘financial market' pressure and wrong business management.
    Today, the networks has been deployed. New scenario in Telecoms enable new players to deploy services over broadband without proprietary network and this new generation business will not be anymore capital intensive, let's say these will be innovation intensive.

    U.S. VoIP Market:

    The US market for VoIP advanced dramatically in 2006-2007, adding 3.8 million VoIP households in 2006, reports In-Stat: As a result, wholesale VoIP revenues grows quickly, as MSOs, Skype, and a myriad of new entrants most lacking network facilities enter the market and drive demand for telephony features and applications, the high-tech market research firm says.
    As retail VoIP expands, wholesale VoIP will accelerate quickly, says Bryan Van Dussen, In-Stat analyst. The largest segment remains international VoIP, but we expect the market for local services to surge from 12% of all revenues to 27% by 2010.
    Recent research by In-Stat found the following:

    • Consumer VoIP adoption will drive wholesale VoIP revenues to $3.8 billion by 2010 from $1.1 billion in 2006.
    • In-Stat finds small businesses are driving the growth of hosted services in the U.S. Hosted VoIP seats in the U.S. ...

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  • 08/07/15--06:13: SIP Trunking
  • From the SIP RFC 4904:


    A Session Initiation Protocol (SIP) to PSTN gateway may have trunks that are connected to different carriers. It is entirely reasonable for a SIP proxy to choose — based on factors not enumerated in this document — which carrier a call is sent to when it proxies a session setup request to the gateway. Since multiple carriers can transport a call to a particular phone number, the phone number itself is not sufficient to identify the carrier at the gateway. An additional piece of information in the form of a trunk group can be used to further pare down the choices at the gateway. As used in this document, trunks are necessarily tied to gateways, and a proxy that uses trunk groups during routing of the request to a particular gateway knows and controls which gateway the call will be routed to, and knows what trunking resources are present on that gateway.


    In an architecture where calls can be terminated on multiple gateways it is wise to consider routing the call to a destination based on some significant criteria such as cost, quality or proximity. Where a proxy has the ability to evaluate a call based on one or more of these criteria, as well as knowledge of the TDM trunk resources available, the proxy can "tag" the call using the tgrp and trunk-context values in the SIP Contact field of the INVITE. It is important to note that the tgrp and trunk-context values can only be used with a TEL URI, not with a SIP URI.

    Unlike in traditional telephony, where bundles of physical wires were once delivered from the service provider to a business, a SIP trunk allows a company to replace these traditional fixed Public Switched Telephony Network (PSTN) lines with PSTN connectivity via a SIP trunking service.

    What is SIP Trunking and how will it help my business?

    Basically, SIP Trunking is a service that provides VOIP or Voice over Internet Protocol. In other words, it is a form of communicating by transmitting telephone calls over the Internet. This communication through the Internet is done by connecting the private branch exchange (PBX) to the Internet. The Internet actually replaces the telephone trunk allowing for communication by users with both fixed and mobile telephone subscribers throughout the world.
    Your voice, data and videos are all combined into a single line with Session Initiation Protocol (SIP) Trunking. This allows for your local, long distance and broadband Internet service to be combined into one line. You will be able to keep your real time traffic off the internet as well as off the public switched telephone network (PSTN) as much as conceivable.

    Advantages of SIP Trunking

    The advantages of SIP Trunking over traditional telephone lines and older VOIP protocols are several:
    • Whereas before SIP you needed to carry voice, video and data over one line by using a Primary Rate Interface (PRI), the SIP Trunking eradicates the necessity for gateways of Basic Rate Interfaces (BRI), Primary Rate Interfaces (PRIs) and PSTN.
    • The provisions of incoming, outgoing and Private Branch Exchange (PBX) are made by your VOIP business provider setting up a proxy server also known as a SIP proxy.
    • Your provider also does all technical support. This saves you both time and money since you will no longer need an IT team or an IT contractor.

    SIP Trunking Saves You Money

    SIP Trunking allots lower costs without sacrificing quality. When it comes to pricing, SIP Trunks are significantly cheaper than the customary analog circuits. What is the saving? The cost of SIP trunks will range from approximately $20 to $30 per trunk, whereas the analog circuits cost roughly $30 for each circuit. There are also significant savings with charges of long distance terminations with SIP Trunks costing considerably less than TDM rates or customary analog rates. All calls are local with SIP. The result for your customers is that both incoming and outgoing calls have an area code that is local. This gives you a lower cost for your business, and your customers get a feeling of familiarity and closeness with your business. The cost of SIP calls per minute are only a fraction of a penny. In addition, SIP numbers that are toll-free are also available to you.

    Another factor that can be costly for your business is if you want to up the number of Primary Rate Interfaces (PRI) from 23 to 24 channels, you must buy a second PRI that has 24 channels. ...

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    This page is a list of SIP trunking providers in the Netherlands. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

    • ALTOTELECOM - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK www.altotelecom.com
    • 1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 50 countries. We have regional network facilities spread across the globe.
    • AVOXI AVOXI Virtual Call Center Solutions - VoIP Service Provider, provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers. Contact Number 1-800-462-8694.
    • CallForwarding - Be present anywhere in the world with toll free forwarding services from CallForwarding.com. Contact Number: 800-231-9802
    • Dynamic Telecom - Dynamic Telecom offers affordable online phone solutions through VoIP telephone systems in the cloud.
    • LocalTelecom.nl - Sip solutions for everyone. Dutch, English, German and Spanish support. Virtual Phone Numbers.
    • MKB Voice - Specialist in Voice-over-IP for SMB.
    • Redworks Best Telecom operator offers high availability, unlimited SIP trunking and hosted VoIP business telephony
    • Tritel BV - Affordable, Reliable and Accessible
    • Svanto.net - Tailored Internet telephony solutions VoIP provider for residential, wholesales and business. Providing international DID's via high quality SIP Trunks. Easy to connect your Asterisk, freepbx, Elastix, Trixbox, Alcatel-Lucent Pbx, CUCM, Avaya and many more.
    • VoipDigit - High quality business telephony, low prices. Online or local PBX, SIP trunking and virtual telephone numbers. Specialist VoIP knowledge.
    • Voiped Telecom - Tailormade complete SIP trunking solutions with competitive price to all destinations with tier1 direct interconnects for the best quality you could wish.
    • Voys Telecom - We change business telephony. High quality VoIP Trunks and Hosted VoIP accounts with the best service.
    • Verbonden Sip solutions for Dutch companies.

    • Voyced | Voyced is a VoIP (telephone) solution provider with the ambition to grow to becoming one of the main pan-European players with our low-cost, commercial grade IPPBX and VoIP services. ...

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  • 08/07/15--10:02: Asterisk cmd ExecIf
  • ExecIf

    Synopsis

    Executes dialplan application, conditionally.

    Usage:

    Version 1.2 and 1.4
    ExecIF (<expr>|<app>|<data>)

    Version 1.4.24.1
    ExecIF (<expr>?<app>|<data>)

    Version 1.6
    ExecIf(expression?appiftrue(args)[:appiffalse(args)])

    Description

    If <expr> is true, execute and return the result of <app>(<data>).
    If <expr> is true, but <app> is not found, then the application will return a non-zero value.

    It is important not to have any spaces as it does not seem to like that (gives a could not find application if spaces are present). Also, if you don't use quotation marks and the first value contains spaces (ex.: SYSTEMSTATUS="internal server error"), it'll always evaluate to true (ExecIf will evaluate as ExecIf(internal?Background)).

    Also note that the separator between the expression is a comma or a vertical bar, not a question mark (as is the case with GotoIf)(Version 1.2 and 1.4).
    As of 1.6 the syntax is inline with GotoIf and the 1.2/1.4 syntax will give you errors.


    Example

    Version 1.2 and 1.4
    exten => s,n,ExecIf($["${SYSTEMSTATUS}" = "SUCCESS"]|Background|dictate/playback)

    Version 1.4.24.1
    exten => s,n,ExecIf($["${SYSTEMSTATUS}" = "SUCCESS"]?Background|dictate/playback)

    Version 1.6
    exten => s,n,ExecIf($["${SYSTEMSTATUS}" = "SUCCESS"]?Background(dictate/playback))



    See also



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

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    Atcom Systems Inc. (Calgary)

    • Web Site: www.atcomsystems.ca
    • E-mail: support@atcomsystems.ca
    • Phone: (403) 212-5276
    • Twitter: @atcomsystems
    • Online Store
    • VoIP & TDM sales & service for businesses from 5 to 200 phones in the Calgary area
    • We offer support for Nortel/Toshiba/Panasonic/Trixbox/FreePBX/IP-PBX/Asterisk/Aastra/Polycom/Snom/Linksys/Sangoma


    eGuest Inc. (Calgary, Edmonton, Vancouver)

    • Web Site: www.eguest.ca
    • E-mail: info@eguest.ca
    • Phone: (403) 451-1400
    • Asterisk VoIP Experts, Digium Registered Partner, Sangoma/Mitel(Aastra)/Snom/Polycom/Yealink Reseller, Trixbox/Elastix/FreePBX/PBX in a Flash Support, Linux and Windows, Infrastructure Consulting, Network support

    Everything IP. (Calgary)

    • Web site: www.Everything-IP.com
    • E-mail: sales@Everything-IP.com
    • Phone: (403) 208-1941
    • Asterisk VoIP, Integrated Telephony, E-mail and Web Services, Custom Development

    Florian Consulting Inc. - Tomas Florian (Calgary)

    • Web site: www.florian.ca
    • E-mail: info@florian.ca
    • Phone: (403) 775-6949
    • Asterisk , VoIP & Networking

    Fort Technology Inc. (Parent company of Terrace Communications Inc.)

    (Updated August 16, 2004)
    • Web site: http://www.forttechnology.ca/
    • Email: ftgrp@forttechnology.ca
    • Phone: (780) 992-1896 x102 <Shawn Lawrence>
    • Reseller of Digium equipment and Asterisk PBX/Gateway Systems

    Globehaus Technology Group (Calgary)

    • Web Site: http://www.globehaus.com
    • E-mail: newclient@globehaus.com
    • Phone: (403) 475-2993
    • Telephony Solutions | IT Services | Network Solutions | Infrastructure Services


    IT Canada International (Calgary)


    0 0
  • 08/07/15--23:19: Dialshree Dialer
  • Elision TechnoLab LLP is the leading supplier and pioneer of VoIP solutions all over the world. We provide career-class solutions with powerful communication technology. Our company consists of a team of very skilled individuals who offer and implement solutions in network, data, video and voice infrastructure – hence, we are a one-stop provider for other companies’ technology needs.

    Here at Elision, we have the expertise, experience and the latest technologies to make your business communication simple, better, and faster.

    Online VoIP Solutions (OVS) is part of the Elision TechnoLab LLP, which was founded in December 2007. This means to say that the world’s pioneer and supplier of VoIP solutions is a division of ETL.

    Working with a lot of innovative and leading service providers in the core domain VoIP, our company pampers the telecommunication market, which range from start-up companies to established and well-known industry leaders.

    With a lot of areas of expertise such as Interactive Voice Response (IVR), VoIP Billing, Least Cost Routing, CallCenter Predictive Dialer, IP/PBX, Voice Broadcasting and different telecommunication solutions are all innovative products that we take pride against our competitors.

    In addition, the product portfolio of our company includes Video Telephony, Voice Logger and Click2Call Web and Calling Card solutions for worldwide calling that will lessen the cost to almost 80 percent.

    Predictive Dialer –has the ability to increase sales by as much as double per hour for each agent. It eliminate much of the wasted time an agent would normally spend waiting for a call to be answered, ensuring that the agent is spending his or her time actually selling.
    Click2Call Web – an innovative technology in order to engage your online visitors over a telephone.
    IVR System – a technology that automates communications and contacts with telephone callers.
    IP/PBX – helps in minimizing the monthly telephone bills of corporate homes while offering a more effective way of communication.
    Voice Broadcasting – a mass communication method that sends pre-recorded message to different call recipients at once.
    Video Telephony – you can receive and make unlimited quality video phone calls worldwide.
    Voice Logger – provides full-featured sophisticated voice recording together with everyday report of entire call logs.
    Our partners and clients are our most important asset as Elision TechnoLab LLP takes pride in developing long-term and strategic commitment with them. The diverse solutions and focuses approach of Elision TechnoLab LLP, along with experience, addresses the needs of service providers on a regular basis. We offer 24/7 support to our clients and make sure that their services are available at all times.

    - See more at: http://www.elisiontec.com/about-us/#sthash.MxfS7zKq.dpuf

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    High Availability (HA) is normally achieved through "clustering" - which means two machines acting as one for a specific purpose. There are many ways to create a cluster, each with its own benefits, risks, costs, and trade-offs. The terms "High Availability" (HA) and "Clustering" can be overused so beware of the hype. Clustering, and HA have specific (and different!) meanings. If you are responsible for creating a high availability cluster for Asterisk, below are the issues and concepts you should be aware of. This page is intended to be a starting point in the design, creation or selection of a High Availability or Clustering solution for Asterisk.

    Note that if you are designing a call center for PSAP (Public Safety Answering Point) / 911 call centers there are specific requirements you must consider. Some are noted below, others are specified by rules/orders from FCC (USA), CRTC (Canada), and similar country specific organizations. (eg: FCC 05-116 order 10). Even if you are not designing for a PSAP, these guidelines are excellent best practices often applied by large commercial call centers.

    Please do not add specific product names/links to this page, it is intended to be product neutral. Don't say "this is the best" because your product/your favorite product uses it.. Stick to facts please.

    Co-Dependence and Autonomy

    This criteria is among the most important (if not THE most important) criteria when designing/selecting/building a high availability telephony environment. In order to be a true cluster, the machines (or "peers") must be autonomous. Some HA solutions involve sharing hardware, software, a logical device, etc .The problem with this approach is that you create a single point of failure. For example, if a cluster shares a hardware channel bank (eg: connected to 2 machines via 2 USB cables), then if the channel bank fails the entire cluster fails. As another example, if a cluster shares a disk (eg: DRBD), then corruption of the disk content from a failing peer immediately corrupts the disk content of the other peer. In a true cluster the peers must be autonomous; i.e. not share any hardware, software, logical devices, etc. (Beware of some solutions which place a single device in front of Asterisk servers - creating a single point of failure).

    Telephony devices in true high availability environments do not share any logical/physical resources. For example, in emergency call centers/PSAP's nothing on the call path is shared: from clustered PBX's, to separate switches, to clustered routers (HSRP/VRRP) to the trunks. Each peer (whether PBX or router or other) must survive the destruction of its peer. (NG911 Section IV.C).

    Data Synchronization and Scalability

    In order for a cluster to remain useful, the data on the peers must remain in sync. This allows one peer to pick up where the other left off in the event of a failure. However, synchronization is one of the greatest challenges for clusters. ...

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