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  • 06/16/17--10:57: 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.


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  • 06/16/17--20:58: maple4VOIP
  • maple4voip voipinfo.jpg

    Digium OpenVox Sangoma VOIP Appliances, Gateways, IP PBXs, and Voice Cards

    www.voip4EU.com

    When it comes to internet telephony simply the best

    Manufacturers

    Product Catgeories

    maple4VOIP store
    MAPLELEAF TECHNOLOGIES
    Hauptstrasse 9
    83052 Bruckmühl, Germany
    Tel: (+49) 8062-726993-0
    Fax: (+49) 8062-726993-9
    Email: sales@voip4EU.com
    Internet: www.voip4EU.com

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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):

    Arrowtel
    -Specializing in asterisk based solutions
    -Local installs in Southern California, USA - Support Worldwide
    -Provides design and support of any Asterisk-based PBX of all sizes
    Location: CA 90025, USA.
    Telephone:800-284-3842
    E-mail :info@arrowtel.net
    Web : http://www.arrowtel.com

    Asterisk PBX Solution

    Since the globalization has enforced many enterprises and thus, it is essential to upgrade enterprises with advanced yet modern systems. Ecosmob renders various cost-effective IT services to its clients worldwide. Its VoIP services and solutions are highly scalable and engineered on two core elements – quality and reliability.

    Key Advantages for Asterisk IP PBX Solution:

    - Easy to setup, configure and maintain
    - Offers better customer productivity and services
    - Significant cost savings using VoIP providers
    - Web/GUI based configuration
    - Scalable, reliable and efficient
    - Eliminates phone wiring and vendor lock-in
    - Wide array of features

    For Asterisk IP PBX Solution Contact: https://www.ecosmob.com/, 1-303-997-3139


    Bicom Systems


    Bicom Systems is the creator of Telco-in-a-Box, the most complete IP PBX Unified Communications solution for ITSPs, Service Providers, CLECs, and more. Telco-in-a-Box meets all of your telecom needs including telephony, billing, mobility, security, and more in one tidy package. With unmatched compatibility, stability, and reliability, Telco-in-a-Box is the tool to build and grow a telecom. The product suite includes:

    • SERVERware is a Cloud IP Services Delivery Platform
    • PBXware is an IP-PBX turnkey business communications platform
    • TELCOware is a provisioning platform that performs billing, accounts, and more
    • sipPROT is an advanced SIP security module
    • sipMON is a SIP monitoring module that analyzes and improves VoIP calls
    • gloCOM is a desktop and mobile Unified Communications app

    A Zycoo Asterisk 13 System



    A Zycoo Asterisk V.13 base VoIP phone systems is user friendly and license free IP PBX system, includes optional telephony interfaces: PSTN, GSM, WCDMA, E1/T1, ISDN BRI. ...

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  • 06/17/17--19:07: REVE Systems


  • REVE Systems ensures the best returns on technology investments and strengthens the service providers' market presence by providing them with best-in-class VoIP solutions. We have a large pool of engineers who are experienced and well trained on varied environments and cross vendor platforms. This enables us to provide 24x7 Platinum Level Support to our clients and to ensure that their services are always available to their end customers.

    Headquartered in Singapore, REVE has its major development center in Bangladesh and India, branch offices in HongKong, USA and United Kingdom. We currently service customers in over 78 countries, where more than 2600 VoIP and telecommunication service providers have placed their trust on us.

    History


    REVE Systems started in 2003 with a focused approach to serve the IP based communication industry. A Telecommunication and Software Solution provider, REVE Systems has a wide assortment of products, ranging from backbone infrastructure to peripheral products, including middleware. The company today holds a leadership position in Mobile VoIP, Softswitch& Billing and Bandwidth Optimization solutions.



    REVE Systems Blog Posts






    Products


    • ALL-In-One Solutions

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  • 06/19/17--03:42: VoIP Origination
  • Please add information to this page about VoIP Origination.

    What is VoIP Call Origination?


    One of the terms most often used when talking about VoIP communications is call origination, which basically has to do with how a telephone call starts and how it travels to the receiver once it does. The following will provide an in-depth look at what call origination is, how it works and what type of hardware is required.

    What is Call Origination?

    VoIP stands for "Voice Over Internet Protocol". This means that phone calls utilize a technology that allows the calls to be sent directly over internet networks, which is a much cheaper way to make calls. Call origination refers to the point where the call starts, which takes place over the PSTN telephone network and transferred to their destination through the internet. It's important to note that a phone call through VoIP starts between the initiation point and the destination point, which are referred to as the originator and terminator respectively. There are typically different types of call origination depending on the services provided by the ITSP involved. The only way for call origination to work is if the VoIP originator has a call termination arrangement with a VoIP terminator.

    Required Hardware

    The best aspect of VoIP services is that there are hardly any noteworthy hardware requirements. All that is necessary is a gateway, which essentially transfers phone calls from the internet and onto PSTN lines. Since a gateway must interact with both the internet and standard PSTN lines, there are two interfaces necessary for a gateway, including a telephony interface that takes digital and analog lines and an Ethernet interface as a connection between the gateway and the internet. It's important to understand that a digital line can support a large amount of calls at one time, which can range anywhere from 20 to 30 depending on the type of line that is chosen. In comparison to a digital line, an analog line can only support one phone call at a time.

    How Call Origination Works

    Call origination is a fairly simple process that is a bit more complicated to explain. In essence, the gatekeeper mentioned previously will receive the calls and requests from the dialer. When a user makes a connection to the dialer, the gatekeeper will ask a Radius server to check if the user has input the correct password and username. The Radius server will then answer the gatekeeper with a yes or a no. If a yes answer is received from the Radius server, the user the has the ability to make a phone call from the origination point. Once the number has been placed, the gatekeeper once again receives a request for the phone call to be made. It is at this point where the gatekeeper will interact with the Radius server again to see if the user in question has enough money to make the phone call. The Radius server will then connect to the billing server to ascertain how much money has been provided by the user for this specific call, in order to nail down how long the call can last before being cut off.

    The billing server will then take a look at the location at which the call is originating from in order to measure the current rate of pay, which all depends on whether the call is being made in off peak or peak hours. Once this has been determined, the relevant information will then be sent back to the Radius server. Once the Radius server has received this information, it will be sent back to the gatekeeper. The user is then provided with the IP address for the gateway at the destination to which the call is being placed. This gateway will send the call to the final destination. If no other issue arises during this time, the call can take place and the two users can talk for however long the duration of the call is. If the call takes place once the maximum time allotment has been reached for the call, it will be disconnected and more money will need to be provided in order to make another call and continue the conversation. When the call has come to a conclusion, whether by the callers themselves or because they were disconnected, the initial user will have the total price of the call deducted from their payment source by the billing server.

    Types of VoIP services

    There are two basic types of VoIP services, including PC to phone and phone to phone. The provider for the PC to phone service will place a dialer on the internet that the user of the service can download and install. This dialer allows for an account to be created with any type of payment source that is allowed by the service provider. Once this is done, the user can make and receive calls. ...

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  • 06/19/17--04:35: 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.jpg
      DIDWW - the source for wholesale International DIDs and Toll Free Virtual Numbers. We provide Voice and SMS enabled DIDs in many countries. SMPP, HTTP and Email SMS forwarding.
    • MultiTEL is providing retail and wholesale Worldwide DIDs - over 90 countries. Pick your own SMS enabled DIDs from over 40 countries. (US, UK, CA, Germany, etc). Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. SMS forwarded via e-mail , URL or as SMS/Text message. All calls are forwarded to SIP, PSTN or to our free Hosted PBX. Coverage and numbers always available in stock from more than 90 countries.

    DID Providers by country

    Angola

    • MultiTEL is providing retail and wholesale Angola (Luanda) DIDs . Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. Payment by card, paypal or bank transfer. All calls are forwarded to SIP, PSTN or to our freehttps://www.multitel.net/pbx|Hosted PBX] . Coverage and numbers always available in stock from more than 90 countries.
    • TeleCallMart Local and Toll Free Numbers from 1$, Voip calls, SIP Phone, Auto Attendant, DTMF, REST API. No monthly fees and 0$ setup.

    Argentina

    • BuyDDINumbers.com Provides Cheapest Argentina DID /Virtual Phone Numbers/DDI Numbers @_€ 2.95/month including free PBX. with the Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX __. Toll Free Number Available without Monthly commitments.
    • BuyDIDNumber We Provide Argentina Virtual Phone Numbers@ $ 4.49 / Month NO SETUP FEE , Unlimited Channels available with the Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk , voipbuster , iTalkWorld , any Betamax/ Delmont Voip or any other ITSP . Phone Numbers all over the world are available. Free PBX . Toll Free Number Available without Monthly commitments.
    • CarryMyNumber.comArgentina DID /Virtual Phone Numbers at wholesale rate_ @$ 2.75/month with free fully hosted PBX. ...

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  • 06/19/17--11:19: VoIP Recording Software
  • Many companies find it necessary to record all or some of the calls in their contact centers. This might be because of regulatory requirements such as the Security and Exchange (SEC) and Sarbanes-Oxley (SOX) regulations, requirements of the Financial Services Authority (FS) in the United Kingdom, the Markets in Financial Instruments Directive (MiFID) in Europe, or the Customer Service decree in Brazil. Many companies also call recordings to train and review their employees' performance or to ensure accurate record keeping against the possibility of a legal claim by an unsatisfied customer.

    SIP Call Recording Options

    • PCBest Network VOIP Recorder Record SIP calls into wav or mp3 files by sniffing network traffic. The software also save call info into xml files, which can be used by other applications.
    • SIPfish Voice-Recorder A call recording appliance with a built-in web interface. It can record between 15 and 300 extensions, and also allows managers to rate and leave notes on calls as they listen to them.
    • TekTape is an audio recorder and call detail records (CDR) generator for Windows (XP, Vista, 7/8/10, 2003/2008/2012 Server). Simple, easy to use HTTP interface. Real time monitoring of SIP calls. Recording of audio streams of SIP calls. Recorded calls are saved in 16 bits, 8 Khz mono format wave files. Support for TZSP (TaZmen Sniffer Protocol). CDR generation for monitored calls. TekTape also creates CDR for failed calls. Real time listening of audio for a selected SIP call and call termination. TLS decoding with SRTP decryption for SIP calls. TekTape can record audio conversations, with supported codecs, between Lync endpoint if Lync server certificate with its private key is installed on to TekTape installed system. Multi-site recording support.
    • FonTel VoIP FonTel VoIP is a device for recording VoIP calls. The recorder supports up to 100 simultaneous calls in the most popular SIP protocol.
    • StarTrinity VoIP Recorder freeware VoIP recorder for your IP PBX or softswitch (like Asterisk or FreeSwitch). It runs on Windows as a service, captures VoIP traffic (SIP+RTP) via mirroring port, decodes it and saves to CDR and WAV files. The CDR data is saved via ODBC driver, it can be your MSSQL, MySQL, PostgreSQL or other database. G.729 and G.711 RTP frames are saved to WAV files to hard disk, recorded file name is saved into CDR. The recorder is free to use by commercial and non-commercial organizations.
    Performance: 1700 concurrent G.711 calls, 420 G.729 calls on a 4x3.5GHz i7, 8GB RAM server

    Oreka GPL and Oreka TR - Enterprise call recording system with both open source and commercial versions available. www.orecx.com.

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  • 06/20/17--07:30: Bicom Systems


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

    History


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

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

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

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

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

    Bicom Systems published How to Grow an ITSP.


    Products



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  • 06/21/17--19:28: 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.


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  • 06/15/17--21:48: Virtual PBX
  • Virtual PBX is a budget-friendly form of hosted VoIP (Voice over Internet Protocol) that usually only handles inbound calls. A virtual PBX is typically intended for small business VoIP customers with fewer than 10 employees and low-volume telephone traffic.


    What Is Virtual PBX?

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

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

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

    Virtual PBX Features


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

    Standard features offered with most virtual PBX plans are:

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

    Limitations


    Virtual PBX phone systems generally:

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

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

    Cost

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

    Virtual PBX Service Providers


    Some virtual PBX providers include:

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  • 06/23/17--01:31: Call Center Software
  • Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

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

    Types of Call Center Software


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

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

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

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

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

    Contact Center Software

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

    Call Center Software Providers

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

    • Dialer360 is Call Center Software which provides you inbound, outbound and blended services on high voltage and low cost for your call center. It is the complete telecom solution including Omni Channel, communication with your customers by using Email, SMS, Web chat and Social Media (Facebook, Twitter, LinkedIn, Google) etc. It can be integrated with the CRM of your choice. It gives you services for 24/7 and make sure your data save. It will make your call center to compete with the modern technology in the market. ...

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  • 06/23/17--06:19: BULK SMS
  • Bulk SMS, also known as bulk messaging or bulk text messaging is the act of disseminating SMS messages in large numbers so they are able to be delivered to various mobile phone terminals. This form of messaging is typically utilized by consumer brands, banks, enterprises and media companies. The messages are generally used for mobile marketing, enterprise and entertainment. However, banks often use them for fraud control. For example, if criminals are circulating a fake email that is asking people who have accounts at a certain bank to provide their social security numbers or other confidential information, these text messages can alert people to the scam so they do not fall victim to it. Bulk messaging is often utilized for reminders and alerts. However, it is more frequently used to send communications and information between customers and staff of various companies. Bulk messaging enables the delivery of SMS messages to large numbers of mobile phones that are located all around the world.

    Bulk messaging software

    In order to receive and send bulk messages, software is needed. There are many types of software packages specifically designed for this task that are available. These packages give their users the ability to send messages to as many phone numbers as they want. There are many different ways in which these phone numbers can be managed.

    The vast majority of software applications that are designed to be used with SMS enable the user to upload mobile phone number lists with the use of a CSV or TXT file. Systems that are more advanced are capable of automatically deleting any numbers that are repeated. There are also systems that can be programed to validate all of the mobile phone numbers before the messages are sent to them.

    Enhanced software features are also currently available that allow users to schedule messages to be delivered at certain days and/or times. Bulk messages are also able to be sent on mobile networks that are international or national, assuming that the provider of the bulk messaging software sends internationally.

    Bulk messaging portal

    Bulk messaging features can be added to websites through the use of this specific online script. Unlimited mobile phone numbers can be added to the list of numbers to send messages to. There are a wide variety of ways that can be used to manage these numbers.

    Bulk messaging API

    The majority of services that handle bulk messaging use the API's (Application Programming Interface) listed below. These enable the addition of functionality to programs by their programmers:

    • Email
    • HTTP
    • SMPP (Short Message Peer to Peer)
    • FTP (File Transfer Protocol)

    Immediate benefits of using bulk SMS messaging

    When a particular business is not doing well financially, they need to utilize various tools that can help them gain a competitive advantage in their specific industry. One of the main reasons that bulk SMS is so popular is its ability to lower operational costs while also generating revenue at the same time. Bulk SMS might be the only medium that is able to show a return on investment that is able to be measured. Wholesale SMS messaging is targeted, which makes it extremely effective at getting people to respond and generate revenue.

    Reduces operational costs

    Bulk SMS message transmission is more effective than email and less expensive than voice calls. There are thousands of businesses located all around the globe that utilize wholesale SMS as a way to communicate with their suppliers, employees and customers. There is a significant cost savings as a result of time being saved because actual voice calls to suppliers, employees and customers do not need to be made. A single message can instantly be sent to many people at the same time, as long as the person is located in an area with mobile coverage. The ability to disseminate information so quickly to a large target audience reduces communication costs while also generating revenue if used for marketing purposes.

    Allows customers to be accessed easily

    More people have mobile phones than have access to email or landline phones. Every mobile phone supports the use of text messaging. All mobile phone users are comfortable using this technology because it is simple and easy to understand. This makes wholesale SMS the perfect medium to use for communication with customers. There are also no demographical or geographical restrictions. ...

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  • 06/23/17--06:28: 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.

    .e4 VoIP Hardware



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  • 06/27/17--07:59: VOIP Service Providers
  • For a list of VOIP to PSTN service providers, indexed by country, please see:


    VoIP and VoIP Service Providers

    What is VoIP?

    VoIP is the acronym for Voice over Internet Protocol. To put it in absolutely simple terms, VoIP could be defined as the technology that helps you to make phone calls over the internet. Since the calls would basically be measured in bits and not in minutes, VoIP rates are much cheaper.

    What makes VoIP More Cost Effective?


    One of the biggest advantages of using VoIP is that you don’t need a lot of specific hardware to use VoIP. This could be compared to the analog mobile network, where you need to get a lot of specialized equipment. This reduces the cost of setting up the system. The cost of installation is also less compared to an analog telephone system. Some of the other things that make VoIP more cost-effective are.

    More Than Just Computers

    When people think of VoIP, they generally think of computers due to the popularity of the numerous free communication services like FaceTime and Skype, but this is truly just one aspect of what VoIP can truly offer. It is true that VoIP technology transmits voice communication that's been converted into digital data across a packet-switched network or the internet (what this means, in essence, is that a user making phone calls over high speed internet lines rather than phone lines). With that in mind, users are not confined to only using it on a computer. ...

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  • 06/28/17--10:56: Intermedia Review
  • Intermedia Overview

    Since 1995 Intermedia has been providing their customers with the best in enterprise level features with their Cloud PBX product. The 24/7 customer service team with Intermedia knows that you don't have time to waste, and that it is crucial for your business to always be serving at its highest capacity. That is why they follow these five pillars; security, reliability, support, compliance and on boarding & migration. Intermedia boasts about their worry-free experience that customers like Stranded Oil rave about!

    Benefits

    Intermedia offers their customers with 99.999% uptime SLA. You may ask yourself why they advertise all "five nines", and it's because those extra nines matter. With five nines instead of the standard three, Intermedia offers your team a better quality product. Imagine only being offline 5.289 hours per year (99.999%) rather than the standard 17.68 hours (99.9%). That's a big difference, and with companies that understand that time is money, Intermedia knows that you can't afford to lose that much time.

    Awards

    Intermedia is the recipient of the Utilites & Telecommunications Award for Best Cloud Business Applications Specialist in 2017.

    Irina Shamkova, Senior Vice President of Product Management at Intermedia, was recognized by CRN in their "Power 100: The Most Powerful Women of The Channel 2017" article for her key role in Intermedia's events strategy.

    Jim Kruger, CMO at Intermedia, was recognized by Bizx Journals as a "person on the move" because of his prestige expertise in marketing that is guaranteed to elevate Intermedia's position as a leading cloud applications provider.




    See also



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    Htek Logo-med.jpg


    HTEK UC862 EXECUTIVE HD COLOR GIGABIT IP PHONE


    860.png


    Physical Features


      • 3.5”TFT-LCD, 480 x 320 pixel, 262K colors
      • Ethernet: Dual Gigabit Ethernet ports
      • Keys: 47 keys including 14 programmable keys
      • Full-duplex speakerphone with AEC
      • Handset: 1 RJ9 (4P4C)
      • Headset: 1 RJ9 (4P4C)
      • Power adapter (included): 5V/1.2A
      • Power over Ethernet(PoE): IEEE 802.3af

    VoIP Protocols

      • SIPv2, SDP(RFC 2327), RTP(RFC 1889,1890), RTCP
      • RFC 2833 X-NSE Tone Events for SIP/RTP, AVT Tone
      • Events for SIP/RTP

    Voice Codecs

      • HD wideband codec: G.722
      • HD Codec, HD speaker, HD handset
      • G.711u-law/a-law, G.723.1, G.726, G.729A/B.
      • DTMF(In-Band, RFC2833, SIP Info)
      • Acoustic Echo Cancelation(AEC)
      • Acoustic Gain Control(AGC)
      • Voice Activity Detection(VAD), Comfort Noise insertion

    Security

      • HTTPS Server/Client
      • Transport Layer Security (TLS)
      • SRTP (RFC3711), SIPS
      • VLAN QoS (802. ...

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  • 06/28/17--12:34: Hanlong HD IP Phone-UC804P
  • 804.png


    Physical Feature


      • Dual color LCD, 128 x 96 pixel
      • Ethernet: Dual-port Gigabit Ethernet ports
      • Keys: 37 keys including 4 programmable keys
      • Handset: 1 RJ9(4P4C)
      • Headset: 1 RJ9(4P4C)
      • Power adapter: 5V/1.2A
      • Power over Ethernet(PoE)

    VoIP Protocol

      • SIPv2, SDP(RFC 2327), RTP(RFC 1889,1890), RTCP
      • RFC 2833 X-NSE Tone Events for SIP/RTP, AVT Tone
      • Events for SIP/RTP

    Voice Codecs

      • HD wideband codec:
      • G.722. HD Codec, HD speaker, HD handset
      • Full-duplex speakerphone with AEC
      • G.711u/a-law, G.723.1, G.726, G.729A/B.
      • DTMF(In-Band, RFC2833, SIP Info)
      • Acoustic Echo Cancelation(AEC),
      • Acoustic Gain Control(AGC)
      • Voice Activity Detection(VAD), Comfort Noise insertion

    Security

      • HTTPS Server/Client
      • Transport Layer Security (TLS)
      • SRTP (RFC3711), SIPS
      • VLAN QoS (802.1pq)

    Telephone Interfaces

      • 3 VoIP Accounts
      • Menu-driven user interface, XML Idle Screen, Theme,
      • Screen Sleep
      • Call hold, Call waiting, Call forward, Call return,
      • Redial,Call transfer
      • Caller ID display, DND, Auto-answer, 5-Way Conference
      • Mute, Speed dial, SMS, Voicemail, Message Waiting
      • Indication (MWI) LED, Call history
      • BLF/BLA
      • Tone /Volume control
      • Ring tone selection/Import/Delete
      • Broad and Deep Interoperability
      • Soft keys programmable
      • Phonebook, Black list XML/LDAP phonebook

    Network Protocol

      • TCP, UDP, ICMP, RARP, ARP, DNS, NTP,SNTP, STUN, UPnP,SNMP
      • Static/DHCP/PPPoE
      • TFTP/DHCP/PPPoE client
      • Telnet/HTTP/HTTPS server

    QoS

      • Layer 2(802.1Q, 802. ...

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  • 06/28/17--12:35: 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®, FreePBX®, 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. HTek is the new standard for quality and value in IP phones.

    Sold in over 50 countries worldwide, HTek products offer high-quality, cost-effective solutions. HTek IP Phones can easily be 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 UC900 series IP phones continue the tradition of focusing on world-class quality, cutting-edge features, and competitive pricing.

    HTek R&D Team


    HTek has an experienced in-house R&D and engineering team to create the highest quality product designs, efficiently 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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  • 06/30/17--10:24: VOIP Event Calendar
  • August 2017


    • 7-10: CHICHAGO ClueCon 2017 A 4-Day conference full of demos and presentations from industry leaders. ClueCon features a full-day hack-a-thon devoted to IoT, Making, Coding Telephony and media applications.
    http://cluecon.com

    • 11: FreeSWITCH and OpenSIPS Training After the ClueCon conference there will be a training session available with the choice of FreeSWITCH or OpenSIPS training
    http://cluecon.com/training.html

    May 2017


    • 2-5: AMSTERDAM Opensips three exciting days filled with VoIP and RTC presentations, workshops and design clinics bringing the latest updates from the OpenSIPS community
    https://www.opensips.org/events/Summit-2017Amsterdam.html

    • 8-10: BERLIN - Kamailio World Conference & Exhibition - real time communications event with sessions covering SIP, VoIP, VoLTE/IMS and WebRTC - https://www.kamailioworld.com


    • 17-18: LONDON Discover the solutions designed to provide flexible ways of delivering, managing and supporting communication that include major UC&C technologies: Cloud, Mobile, Customer, Video, Networks, Collaboration.
    http://www.ucexpo.co.uk

    November 2016

    • 16-17: VoIP2DAY 2016 | Santiago Bernabéu (Madrid) - Discover the european meeting of Telephony and VoIP Communications that brings together each year the main international industry leaders. Enjoy two full days of papers and workshops featuring the most important professionals from international scene. More information at http://www.voip2day.com/en/2016

    October 2016

    • 25: MediaCore SMS Solution Free Webinar by Speedflow.
    • 19-20: ElastixWorld 2016 - The seventh edition of the event will be held at Buenos Aires, Argentina. More information at ElastixWorld
    • 12: JeraSoft Experts provides FREE WEBINAR about new VCS features!
    • 12-13: Middle East 2016 GCCM - Annual carriers' meeting with 500+ atendees. Meet Speedflow Team at GCCM to discuss their telecom services and VoIP software solutions.
    • 11 European Voip Summit Amsterdam

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