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  • 09/21/15--00:36: VoIP Hardware
  • This page lists information about VoIP hardware and VoIP hardware products. For phones and hardware to use with Asterisk, including VoIP phones (both hard and soft phones) and Analog Telephone Adapters, see Asterisk phones.

    PSTN Interface cards (analog, GSM, ISDN-PRI and R2/MFC)


    This section contains VoIP hardware for connecting analog or digital phone lines from the Public Switched Telephone Network to your Asterisk server. Please keep VoIP hardware providers in alphabetical order.

    2-Day Direct

    • Cisco SPA303 3-line business-class IP phone; Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
    • Cisco SPA504G Full-featured 4-line business-class VoIP phone supporting Power over Ethernet (PoE)
    • Cisco SPA525G2 5-Line Business IP Phone with Enhanced Connectivity and Media for a New Level of Small Business User Experience; includes wifi and bluetooth connectivity

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  • 09/21/15--02:35: Open Source VOIP Software
  • Open Source VOIP applications, both clients and servers.

    Open source means all source code is available!! Do not post any "free but not open" software here!

    SIP Proxies


    • JAIN-SIP Proxy
    • Mini-SIP-Proxy A very tiny perl POE based SIP proxy
    • MjServer cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
    • MySIPSwitch SIP Proxy server which allows using multiple SIP accounts with a single SIP login
    • NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
    • Net-SIP A Perl SIP framework that includes a stateless proxy
    • OpenJSIP Opensource distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. ...

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  • 09/21/15--02:45: sipX
  • Image

    sipX - The SIP PBX for Linux


    Original project: sipXecs
    Original Website: http://www.sipfoundry.org

    fb-cover-7122-3463412.jpg

    This project has also been forked and continues with the original developers under the sipXcom name.
    New project: http://www.sipxcom.org
    Wiki: http://wiki.sipxcom.org
    User Forums: https://groups.google.com/forum/#!forum/sipxcom-users


    See sipX Solution Summary for further details.



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  • 09/21/15--03:02: Small Business VoIP
  • Small business VoIP service can provide a host of benefits for small businesses that are interested in saving on costs and upgrading their business telephone system. Small business VoIP benefits include:

    • Phone service that is significantly less expensive than traditional telephone service
    • A small business VoIP telephone system is geared towards small businesses
    • Plans are scalable for small businesses that have the potential to grow
    • Small business VoIP providers usually offer no contract plans
    • Small business VoIP phone systems offer advanced features that can make a small business seem like a Fortune 500 company

    How does small business VoIP work?

    Small business VoIP typically works with a hosted PBX model. This is to say that the PBX is hosted at the facilities of the small business VoIP provider. By hosting the PBX at the provider's location, the small business does not need to spend money, resources, and man-hours on expensive PBX hardware, software, and maintenance.

    Small Business VoIP Providers

    This is a list of VOIP Service Providers who offer full service products primarily small businesses. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP. 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.

    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. ...

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  • 09/21/15--03:18: sipX Solution Summary
  • sipX – Easy to use, scalable, and feature rich open source unified communications and IP PBX solution for Linux


    Image
    SIPfoundry info:
    Original Website: http://www.sipfoundry.org
    SIPfoundry Wiki: http://wiki.sipfoundry.org
    Github: https://github.com/SIPfoundry

    fb-cover-7122-3463412.jpg

    sipXcom info:
    Website: http://www.sipxcom.org
    sipXcom Wiki: http://wiki.sipxcom.org
    Github: https://github.com/sipXcom


    sipX is a complete open source SIP based unified communications solution. With a comprehensive plug & play management system fully available in open source, sipXecs is made for non-technical people. Installation and configuration is all automated and GUI based, phones from over 10 manufacturers are plug & play managed including auto-discovery, full configuration, directory management, speed dial, BLF, and firmware management, and users are provided with a feature rich personal Web portal for self-service administration. Typically the company receptionist takes on the task of system administration being responsible for all adds, moves and changes to the system.

    sipX runs on standard Linux and does not require any proprietary add-on cards or kernel drivers. It is a native SIP solution built around a fully standards based SIP proxy call server at its core. Load-sharing redundancy can be established easily by adding additional call servers with typical configurations using 1, 2, or 4 redundant call servers. Should a server fail no calls are interrupted since media is not routed through the call server proxies. Not routing calls through the call server also enhances voice quality and allows the solution to scale easily. Additional services are added as SIP based application servers for voicemail, IVR and auto-attendant, call park, group paging, presence, conferencing, and SIP trunking gateway and NAT traversal services. PSTN gateway redundancy and failover are easily possible adding more than one external gateway. Such gateways can be setup remote in branch offices or centralized in a single location. sipXecs is a Web services enabled application offering a Web Services SOAP interface northbound for all its management and configuration capabilities allowing Web mashups and IT integration. Other IT integration capabilities include Microsoft Active Directory and Exchange integration, LDAP synchronization, and SNMP diagnostics using an alarm server.

    sipX cooperates with the FreeSWITCH project where sipX now incorporates FreeSWITCH as its new media server initially offering highly scalable and feature rich conferencing and IVR capabilities. The FreeSWITCH server runs as an additional application server alongside all the other sipXecs components. It is fully integrated with the sipX Web based management system offering centralized configuration, process control and diagnostics. Every user is automatically configured with a personal conferencing bridge based on group permission settings and dynamic conference controls are offered to every user over the personal sipX Web portal.

    sipX is installed in a lot of places, large and small. Deployments include professional offices, universities, school district and local government, manufacturing, retail, and parts of the financial and insurance industry.

    In 2007 Nortel Networks created the Nortel Software Communications System SCS500 based on sipXecs. ...

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  • 09/21/15--07:18: FAX
  • FAX is often carried over VOIP networks.

    ITU standards T.37 and T.38 define how FAXes are sent over IP networks.


    See Also


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  • 09/21/15--07:49: Asterisk Expressions
  • Asterisk dialplan expressions are special expressions that can be used in the dialplan of Asterisk.

    Syntax

    $[expr1 operator expr2]


    The high-level view of variable evaluations in Asterisk:

    Since most user input will come via config files to Asterisk, some filtering and substitutions are done
    as the config files are read.

    Next, as the dialplan is executed, the ${ ... } variables and functions are evaluated and substituted.

    And lastly, the contents of $[ .. ] expressions are evaluated and substituted.


    Parameter Quoting:

    exten => s,5,BackGround,blabla

    The parameter (blabla) can be quoted ("blabla"). In this case, a comma does not terminate the field. However, the double quotes will be passed down to the Background command, in this example.

    Also, characters special to variable substitution, expression evaluation, etc (see below), can be quoted. For example, to literally use a $ on the string "$1231", quote it with a preceding \. Special characters that must be quoted to be used, are [ ] $ " \. (to write \ itself, use \\).

    These double quotes and escapes are evaluated at the level of the asterisk config file parser.

    Double quotes can also be used inside expressions, as discussed below.

    Spaces Inside Variable

    UPDATE: with the latest Asterisk Beta (1.2.0_2) there is the following notice:

    Dialplan Expressions:

    • The dialplan expression parser (which handles $[ ... ] constructs) has gone through a major upgrade, but has one incompatible change: spaces are no longer required around expression operators, including string comparisons. However, you can now use quoting to keep strings together for comparison. For more details, please read the doc/README.variables file, and check over your dialplan for possible problems.

    If the variable being evaluated contains spaces, there can be problems.

    For these cases, double quotes around text that may contain spaces will force the surrounded text to be evaluated as a single token. The double quotes will be counted as part of that lexical token. ...

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  • 09/21/15--17:44: Yeastar Reviews
  • Read reviews about Yeastar products, or write your own. Yeastar specializes in the developing and manufacturing IP-PBX products and is committed to the distribution of new generation technology products in the field of enterprises' communications.

    See also


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  • 09/21/15--18:18: Yeastar - MyPBX
  • IP PBX for SMBs

    The new compact, feature rich PBX for every-day use!

    MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices of larger organizations (2-300 users per site). MyPBX also offers a hybrid solution (a combination of VoIP applications using legacy telecom equipment) alternative for enterprises who are not yet ready to migrate to a complete VoIP solution.



    MyPBX-SOHO

    SOHO_右侧副本.png
    MyPBX_back_SOHO.gif

    Specification:

    Users: 32
    Concurrent Calls(Max): 15
    Voicemail: 3000min

    Interface:
    Up to 4 Analog Ports (FXO/FXS)
    Up to 4 BRI Ports
    Flash: 512 MB Onboard Flash
    RAM: 256 MB Onboard RAM
    LAN: 1 (10/100Mbps)

    Size: 160x160x30 mm
    Weight: 500g
    Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

    Protocol: SIP(RFC3261), IAX2
    Transport Protocol: UDP, TCP, TLS, SRTP
    Codec: G.711, GSM, SPEEX, G.722, G.726, ADPCM, G.729 A, H261,
    H263, H263p, H264 ,MPEG4.
    DTMF: In-band, RFC2833, SIP INFO
    LED: Red for FXO, Orange for BRI, Green for FXS.
    Network: Firewall, VLAN, DDNS, QoS, DHCP Server, Static IP, VPN, PPPoE
    Multiple Languages Support:
    Chinese, English, French, German, Hebrew, Italian, Portuguese, Russian,
    Spanish, Swedish, Turkish and more.

    Features:

    Auto Provision
    Blind Transfer
    BLF Support
    Blacklist
    CDR (Call Detail Records)
    Call Forward
    Call Parking
    Call Pickup
    Call Routing
    Call Transfer
    Call Waiting
    Caller ID
    Conference
    Define Office Time
    DISA (Direct Inward System Access)
    DIDs
    Distinctive Ringtone
    DND (Do Not Disturb)
    Firewall
    Follow Me
    IVR (Interactive Voice Response)
    Intercom/Zone Intercom
    Mobility Extension
    Multi-language Prompt
    Music On Hold
    Music On Transfer
    One touch record
    PIN User ( PIN Code Control)
    Paging/Zone Paging
    Queue
    Ring Group
    Route by Caller ID
    Skype Integration (Skype Connect)
    Speed Dial
    Three Way Calling
    Voicemail
    Voicemail to Email
    Voicemail Forwarding
    Web Based Control Panel
    Spy functions (Normal Spy, Whisper Spy, Barge Spy)



    MyPBX-Standard

    std.2707副本.png
    MyPBX_back.gif

    Specification:

    Users: 100
    Concurrent Calls: 25
    Voicemail: 3000min

    Interface:
    Up to 16 Analog Ports (FXO/FXS)
    Up to 8 GSM Ports(Quad-Band GSM/GPRS850/900/1800/1900MHz)
    Up to 8 UMTS Ports(UMTS 900/2100MHz or 850/2100MHZ or 850/1900MHZ)
    Up to 8 CDMA Ports
    Up to 8 BRI Ports
    Flash: 512 MB Onboard Flash
    RAM: 512 MB Onboard RAM
    USB: 1 (USB2.0)
    LAN: 1 (10/100Mbps)
    WAN: 1 (10/100Mbps)

    Size: 290x180x33 mm
    Weight: 700g
    Power Supply: AC 100~240V/50~60Hz(DC 12V, 5A)

    Protocol: SIP(RFC3261), IAX2
    Transport Protocol: UDP,TCP,TLS,SRTP
    Codec: G.711, GSM, SPEEX, G.722, G.726, ADPCM, G.729 A, H261,
    H263,H263p, H264 ,MPEG4. ...

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  • 09/21/15--22:21: 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


    Roytel E1/T1 trunking gateway, RT-EIMS2002

    EIMS2002.jpg

    Description
    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.
    Features
    ● 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


    Roytel.jpg

    Home page:http://www.roytel.com.cn/index.php?lang=en
    Headquarters: Shenzhen, China
    E-Mail:kc@roytel.com.cn
    Skype ID: yong_chen5
    Telephone: +86 755 26743712

    OpenVox E1/T1 Trunk Gateways, DGW-1001R/DGW-1002R/DGW-1004R

    Home page:OpenVox
    Products: DGW-1001R/DGW-1002R/DGW-1004R
    The DGW-1001R/DGW-1002R/DGW-1004R digital VoIP gateway supports 1-4 software-selectable T1/E1/PRI interface and supports up to 120 concurrent calls. The "R" means that the device supports redundant power supply.

    -1-4 T1/E1/PRI w/ RJ-48
    -2 10/100M Ethernet ports
    -2 USB 2. ...

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  • 09/21/15--22:51: Yeastar
  • Image

    Easy Communication!


    Yeastar specializes in the developing and manufacturing IP-PBX products and is committed to the distribution of new generation technology products in the field of enterprises' communications. In the mean time, Yeastar provides the cost- efficient solutions for ITSP to develop the enterprises ultimate purchase market. With professional and high quality team, Yeastar designs products to worldwide applications and maintains the long-term stability of products to greatly benefit users. Yeastar team gave an insight into users' requirements and ready to listen to their new demands for implementing in the product design and services. Yeastar welcomes the cooperation from various kinds of companies and will sincerely treat them to create the multi-win situation together. Yeastar is working hard to make better office communication experience for you.

    Team
    With professional and high quality team, many of us are seasoned experts in telecom network and VoIP/PBX. A close-knit organization that we are, we aspire, work and achieve together. At Yeastar, we have built a tradition of teamwork. Innovation being our chief motivation, as a Yeastar, every one strives to come up with newer, more stable software and hardware products suitable for the users.

    Our Products:

    1. MyPBX - IP PBX for SMBs

    Yeastar - MyPBX

    The new compact, feature rich PBX for every-day use.

    MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices of larger organizations (2-500 users per site). MyPBX also offers a hybrid solution (a combination of VoIP applications using legacy telecom equipment) alternative for enterprises who are not yet ready to migrate to a complete VoIP solution.

    ImageImage

    ImageImageImage
    ImageImageImage

    ImageImage


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  • 09/22/15--05:09: Unified Communications
  • Unified Communications, often abbreviated as UC, is the combination of different types of communication mediums including telephone service (VoIP), chat, video and web conferencing, messaging, email, fax, and other types of communications. Unified Communications systems can be sold and implemented with a select amount of individual communication mediums, or many integrated into a single unified system.


    Unified communications consist of integrating various real-time and non-real time communication services into a convenient package for the user. Examples of real-time services commonly used include instant messaging, telephony and data sharing. Non-real time communication services focus mainly on messaging services like e-mail, SMS, voicemail and fax.

    With a unified communications solution, the user can receive a message through one communication medium and use another one to access it. One common example of this would be receiving a voicemail message on an office landline phone and using a mobile phone to retrieve it. ...

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  • 09/22/15--14:38: 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


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    This is a list of Asterisk consultants in Ontario, Canada. Also check out VoIP Providers Canada for VoIP service providers in Canada.



    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
    • VoIP Support Services: Asterisk, Trixbox, Elastix, FreeSwitch, Vicidial, A2Billing, Avaya IP Office, Call Center solutions, High Availability (HA) and clustering systems.
    • Hourly and Contract Support, Consulting, Hosting, Office PBX, etc.
    • Asterisk full-featured PBX installation/configuration.
    • Voicemail, fax2mail, PSTN integration, Conferencing.
    • IP-Phone provisioning and support, softphones, TFTP.
    • Network(router,switch,firewall) design and support. BGP, MPLS, OSPF, B2B VPNs 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.

    Active Access Communication Systems

    • Web Site: http://www.visionvoip.com
    • Email: mailto:mail@visionvoip.com
    • Location: Ottawa, Canada
    • Phone Number: 1-800-VVoIP-15 (1-800-886-4715)
    • Type of Support: Telecommunications, VoIP, Networking, Asterisk, Asterisk@Home / Trixbox, Web Development, AGI, ARA, AEL
    • Phone, Email and SMS Reminders: Using text-to-speech engine, your own recorded voice or just text
    • Hourly and Contract Support, Consulting, Hosting, Office PBX, etc.

    Abel Technology Services


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  • 09/22/15--23:34: OpenSIPS
  • Breaking News


    The next OpenSIPS Summit is to be held in Austin, TX, 9th to 10th of November 2015 - 2 days of presentations, workshops and roundtables on OpenSIPS 2.1 .

    The Summit event is followed by the OpenSIPS Advanced Training Class - 11th to 13th of November 2015, in Austin, TX, USA. The the first training with providing 3 days of in-class advanced training, with an included intro of 2 days online training for beginners. Everything providing in depth coverage of OpenSIPS 2.1.

    About OpenSIPS


    OpenSIPS (Open SIP Server) (former OpenSER) is a an Open Source SIP proxy/server for voice, video, IM, presence and any other SIP extensions.

    OpenSIPS is a multi-functional, multi-purpose signaling SIP server - it can act as SIP Router/Switch, SIP Registrar, Application Server, Redirect Server, Load Balancer / Dispatcher, Back-to-Back User Agent, Presence Server, IM Server, Session Border Controller, SIP Front-End, NAT traversal Server, IP Gateway (SMS, XMPP) and others - see the full Set of Features.


    OpenSIPS is recommended for any kind of SIP scenario / service by:
    • the high throughput - tens of thousands of CPS, millions of ‏simultaneous calls (see official tests)
    • the flexibility of routing and integration - routing script for implementing custom routing logics, several interfacing APIs (see the Manual)
    • the effective application building - more than 120 modules to provide features, for SIP handling, for backend operations, for integration, for routing logics (see List of Modules)

    OpenSIPS Summits


    What is OpenSIPS Summit ?

    The OpenSIPS Summit is a conferencing event that aims to provide the necessary knowledge to understand and use OpenSIPS. It is an 100% OpenSIPS event focused on:
    • presenting what OpenSIPS is
    • what is OpenSIPS place in SIP environments, integration considerations
    • usage scenarios for OpenSIPS
    • real case studies with OpenSIPS
    • open discussions on OpenSIPS matters
    • open talks on future directions for OpenSIPS

    Who should attend ?

    If you are looking to discover, understand and evaluate OpenSIPS, the OpenSIPS Summit is the right place to be. We will answer to your questions like:
    • what OpenSIPS can do for my SIP network
    • what are the OpenSIPS capabilities, features and limits
    • how to extend, expand and enhance my SIP service with OpenSIPS.

    Details and Schedule

    Check the official web page of the OpenSIPS Summits.

    Next event is scheduled for OpenSIPS Summit on 9th-10th of November 2015, Austin, TX, US.

    Registration is open and you can register at: Register for the Summit on 9th-10th of November 2015, Austin, TX, US.

    OpenSIPS 2. ...


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  • 09/23/15--05:28: SMS
  • SMS = Short Message Service

    SMS is a technology for passing short messages (mostly text) from one device to another. It is supported on a variety of mobile network systems, including GSM, UMTS, CDMA and iDEN. Most recently support has also started to appear on fixed networks.

    In Asterisk SMS is supported in several ways:

    • Asterisk cmd Sms is a mechanism of exchanging SMS using ETSI ES 201 912, which is a protocol for encoding SMS on landlines. This is only supported on a few selected carriers though.

    • chan_mobile can send SMS via the connected mobile phone.

    • FastSMS is an Asterisk command that exchanges SMS through a commercial gateway providing worldwide support for SMS.

    • ZIM-SMS is an Asterisk application which allows to send SMS from a dialplan as well as voicemail notifications based on information available in voicemail.conf. The application communicates with ZIM-SMS using XML-POST and is distributed under GPL as a sourcecode with detailed instructions. Please note, that you will need an account with ZIM to use their service. During the initial, test phase, only Canadian destinations will be covered.

    • jkSMS lets you send "sms" from your cell phone (or email client) to your Asterisk box. The message is read aloud via Swift (or festival) and you can reply or delete queued messages.

    See Also
    • Alaris SMS Platform— comprehensive SMS management solution comprising a 3 in 1 system for SMS trafffic switching, billing, and routing (read more...)
    • Bright Pattern 2-way SMS business texting— SMS text messaging as part of omni-channel contact center solution (voice, text, chat, e-mail, video and more)
    • MediaCore SMS Solution - solution for handling carrier-to-carrier SMS traffic that consists of switching, routing and billing modules. It supports unlimited number of SMSs within a session.
    • CallMax retail SMS - convenient retail SMS platform within the CallMax Class 5 Solution. It enables end-users to send SMS to external mobile operators and is perfect for SMS marketing campaigns.
    • Amarrelo provides SMPP and GSM gateway Outlook like SMS client for coporate usage
    • Kannel provides an SMS & WAP gateway
    • PlaySMS: Free and Open Source SMS Gateway
    • SMSD (found in most Linux distributions)

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  • 09/23/15--18:50: Hanlong Technology
  • HTek (Hanlong Technology Co. Ltd.)


    Htek Logo.jpg




    HTek Profile


    HTek™ is a name brand of Hanlong Technology Co., Ltd., a world-class designer and manufacturer of enterprise IP phones and VoIP products. HTek IP phones deliver superb sound quality, a rich set of SIP telephony features, and broad interoperability with leading phone system providers, including Broadsoft®, 3CX®, Elastix®, Asterisk®, Bicom®, and Alcatel-Lucent®. All HTek IP phone products feature the Texas Instruments® (TI) chipset for crystal-clear HD sound, and are backed by an industry-leading two-year warranty.

    Sold in over 50 countries worldwide, HTek products offer high-quality, cost-effective solutions that can also be easily rebranded or customized to meet OEM or ODM requirements. Since 2005, Hanlong Technology has provided enterprises, OEMs, and ITSPs worldwide with millions of advanced VoIP products. The latest HTek UC800 series IP phones continue the tradition of focusing on world-class quality, cutting-edge features, and competitive pricing. Hanlong Technology Co., Ltd. is a private company headquartered in Nanjing, China.

    HTek R&D Team


    HTek has an experienced in-house R&D and engineering team to create the highest quality product designs, easily respond to OEM and ODM requests, and provide superior customer technical support. With an average of over 8 year's experience in the communication field, the R&D team's expertise and professionalism guarantee superior products and support. HTek is also proud to partner with

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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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  • 09/24/15--07:53: Washington State VoIP
  • This is a list of VoIP providers in Washington State. Washington State VoIP companies are telephone providers that typically support multi-line telephone systems, small gateways and hosted VoIP. Please add VoIP Providers in Washington State and telecommunications providers in Washington State, WA to the list below.

    • FluentStream Technologies FluentStream Technologies is a fantastic Cloud-hosted business phone system. We have a best-in-class web portal, an industry-leading WebRTC-based FluentCloud WebPhone, and world class 24x7x365 support. Get all the benefits of the cloud with service that you'll love!
    • 1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways. Priced by trunk not by seat. Have 100 phones and 2 lines if you want. Numbers available in over 50 countries. Custom pricing available based on your project.
    • Jive Communications - Jive offers Washington State VoIP services business.
    • ActiveServe PBX Hosting 3CX, Asterisk, Elastix, and Trixbox PBX Hosting. CAT-5 Data Center, Active NAT Assistance™, Fully Managed Cisco Network, Cloud Platform. No hardware or software to purchase or maintain. 24/7/365 Support. Do-It-Yourself or Turn-Key
    • Connect Me Voice offers a full line of services from basic voicemail to full business systems
    • Digivoix is saving tons of money on all your internet telephony needs, Hosted phone services start at $34.95 (Unlimited Local and Long Distance calling), SIP Trunking starting at $12.95 (Unlimited incoming, 1.6 cents outgoing (US)), Turnkey IP PBX Solutions (offering variety of IP PBX systems with personalized turnkey solutions for small and medium sized markets as per customers needs), Great pricing on Hardware (IP Phones, IP Adapters, IP PBX Systems), Wholesale IP Termination (Offering US Flat Rate route as well as Standard/Premium A-Z routes).
    • FastPBX - FastPBX offers you HD quality service with our easy to use phone solution. We offer brand new, certified business class Cisco handsets with loads of great features that your business can take advantage of right away. All of our systems come with a 1-year Cisco manufacturer's warranty and 30-day money-back guarantee. Best of all, there are no contracts! To learn more about VoIP check us out at http://fastpbx.com/
    • FFB VoIP Services | FFB Services specializes in small business VoIP solutions helping your business make a big impression. FFB Services works with you to get the right solution every time. ...

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  • 09/24/15--07:57: Spokane VoIP
  • This is a list of VoIP providers in Spokane, Washington. Spokane VoIP companies are telephone providers that typically support multi-line telephone systems, small gateways and hosted VoIP. Please add VoIP Providers in Spokane and telecommunications providers in Spokane, WA to the list below.

    • FluentStream Technologies FluentStream Technologies is a fantastic Cloud-hosted business phone system. We have a best-in-class web portal, an industry-leading WebRTC-based FluentCloud WebPhone, and world class 24x7x365 support. Get all the benefits of the cloud with service that you'll love!
    • Jive Communications - Jive offers Spokane VoIP services business.
    • Lantelligence is an International provider of Business Communications solutions that include IP Phone Systems, Call Centers, Video and Web Conferencing services for multi-location corporations.
    • ActiveServe PBX Hosting 3CX, Asterisk, Elastix, and Trixbox PBX Hosting. CAT-5 Data Center, Active NAT Assistance™, Fully Managed Cisco Network, Cloud Platform. No hardware or software to purchase or maintain. 24/7/365 Support. Do-It-Yourself or Turn-Key
    • Connect Me Voice offers a full line of services from basic voicemail to full business systems
    • Digivoix is saving tons of money on all your internet telephony needs, Hosted phone services start at $34.95 (Unlimited Local and Long Distance calling), SIP Trunking starting at $12.95 (Unlimited incoming, 1.6 cents outgoing (US)), Turnkey IP PBX Solutions (offering variety of IP PBX systems with personalized turnkey solutions for small and medium sized markets as per customers needs), Great pricing on Hardware (IP Phones, IP Adapters, IP PBX Systems), Wholesale IP Termination (Offering US Flat Rate route as well as Standard/Premium A-Z routes).
    • FastPBX - FastPBX offers you HD quality service with our easy to use phone solution. We offer brand new, certified business class Cisco handsets with loads of great features that your business can take advantage of right away. All of our systems come with a 1-year Cisco manufacturer's warranty and 30-day money-back guarantee. Best of all, there are no contracts! To learn more about VoIP check us out at http://fastpbx.com/
    • FFB VoIP Services | FFB Services specializes in small business VoIP solutions helping your business make a big impression. FFB Services works with you to get the right solution every time. Over 50 standard features on every line and no matter if you're 1 or 1,000 seats, you get the same quality customer service 24/7. ...

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