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  • 05/28/15--22:43: Yeastar - NeoGate TG
  • VoIP GSM/CDMA/UMTS Gateway Series—Yeastar TG Series

    Reduce costs for SOHO and SMBs


    Yeastar TG series is a series of VoIP GSM/CDMA/UMTS gateway connecting GSM/CDMA/UMTS Network to VoIP Network directly, which can support two-way communication: GSM/CDMA/UMTS to VoIP or VoIP to GSM/CDMA/UMTS. It is the best solution ever to connect IP-based telephone systems and softswitches to GSM/CDMA/UMTS network; and also the best fallback solution when landline goes down.

    Benefits

    1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
    2) Back up - Work as a cost-effective backup when the landline network goes down.
    3) Easy to install - Everything can be easily set up in the Web based management interface.
    4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

    Features

    IP Server and SIP Trunk supported
    SIP Peer Mode supported
    Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
    GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
    Incoming /Outgoing Routing rules
    SMS Sending and Receiving
    Send Bulk SMS
    Gain Adjustment
    USSD
    PIN Modify
    Carrier Selection: Auto/Manual
    Balance Alarm
    Caller ID/CLIR
    Black List
    Hotline
    Call Duration Limitation
    Call Transfer
    Call Back
    Call Status Display
    Call Detail Record (CDR)
    Call Progress Tone Generation
    Call Duration Limitation for SIM Card/Single Call
    LCR (Least Cost Routing)
    Top voice quality (EFR super sound)
    SIP Response Code Switch
    Open API for SMS and USSD
    Real Open API Protocol (Based on Asterisk)
    IP Blacklist
    Network Attack Alert
    System Logs
    Web based configuration


    Yeastar TG100


    Yeastar TG100 is a fully featured 1 port VoIP GSM/CDMA/UMTS gateway that provides GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It significantly reduces the costs of calls with two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. With friendly GUI, everything can be easily set up.

    TG100侧面副本副本.png


    Specification

    Number of GSM/CDMA/UMTS channels (Max): 1
    GSM Network type: 850/900/1800/1900MHz
    CDMA Network type: 800MHz
    UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz

    Protocol: SIP (RFC3261),IAX2
    Transport Protocol: UDP,TCP,TLS,SRTP
    DTMF: RFC2833, SIP INFO, In-band
    Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
    Echo Cancellation: ITU-T G.168 LEC

    LAN: 1 (10/100Mbps)
    Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
    QoS, OpenVPN
    NAT Traversal: Static NAT, STUN

    Size: 110x70x24mm
    Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

    Operation Range: 0°to 40°C, 32° to 104° F
    Storage Range: -20° to 65°C, 4° to 149° F
    Humidity: 10-90% non-condensing


    Yeastar TG200


    Yeastar TG200 is a VoIP GSM/CDMA/UMTS gateway with 2 channels providing GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It supports two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. Thus the calls costs could be significantly reduced by VoIP or GSM/CDMA/UMTS network.

    TG200正面副本副本.png


    Specification

    Number of GSM/CDMA/UMTS channels (Max): 2
    GSM Network type: 850/900/1800/1900MHz
    CDMA Network type: 800MHz
    UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
    Protocol: SIP (RFC3261),IAX2
    Transport Protocol: UDP,TCP,TLS,SRTP
    DTMF: RFC2833, SIP INFO, In-band
    Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
    Echo Cancellation: ITU-T G. ...

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  • 05/28/15--22:46: Yeastar - NeoGate
  • Yeastar TG100 - VoIP GSM/CDMA/UMTS Gateway

    Reduce costs for SOHO and SMBs

    Yeastar TG100 is a fully featured 1 port VoIP GSM/CDMA/UMTS gateway that provides GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It significantly reduces the costs of calls with two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. With friendly GUI, everything can be easily set up.

    TG100侧面副本副本.png


    Benefits

    1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
    2) Back up - Work as a cost-effective backup when the landline network goes down.
    3) Easy to install - Everything can be easily set up in the Web based management interface.
    4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

    Specification:

    Number of GSM/CDMA/UMTS channels (Max): 1
    GSM Network type: 850/900/1800/1900MHz
    CDMA Network type: 800MHz
    UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
    Protocol: SIP (RFC3261),IAX2
    Transport Protocol: UDP,TCP,TLS,SRTP
    DTMF: RFC2833, SIP INFO, In-band
    Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
    Echo Cancellation: ITU-T G.168 LEC

    LAN: 1 (10/100Mbps)
    Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
    QoS, OpenVPN
    NAT Traversal: Static NAT, STUN

    Size: 110x70x24mm
    Power Supply: AC 100~240V 50/60Hz (DC 12V, 1A)

    Operation Range: 0° to 40°C, 32° to 104° F
    Storage Range: -20° to 65°C, -4° to 149° F
    Humidity: 10-90% non-condensing

    Features:

    SIP Server and SIP Trunk supported
    SIP Peer Mode supported
    Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
    GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
    Incoming /Outgoing Routing rules
    SMS Sending and Receiving
    Send Bulk SMS
    Gain Adjustment
    USSD
    PIN Modify
    Carrier Selection: Auto/Manual
    Balance Alarm
    Caller ID/CLIR
    Black List
    Hotline
    Call Duration Limitation
    Call Transfer
    Call Back
    Call Status Display
    Call Detail Record (CDR)
    Call Progress Tone Generation
    Call Duration Limitation for SIM Card/Single Call
    LCR (Least Cost Routing)
    Top voice quality (EFR super sound)
    SIP Response Code Switch
    Open API for SMS and USSD
    Real Open API Protocol (Based on Asterisk)
    IP Blacklist
    Network Attack Alert
    System Logs
    Web based configuration



    Yeastar TG200 - VoIP GSM/CDMA/UMTS Gateway

    Reduce costs for SOHO and SMBs

    Yeastar TG200 is a VoIP GSM/CDMA/UMTS gateway with 2 channels providing GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It supports two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. Thus the calls costs could be significantly reduced by VoIP or GSM/CDMA/UMTS network.

    TG200正面副本副本.png


    Benefits

    1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
    2) Back up - Work as a cost-effective backup when the landline network goes down.
    3) Easy to install - Everything can be easily set up in the Web based management interface.
    4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

    Specification:

    Number of GSM/CDMA/UMTS channels (Max): 2
    GSM Network type: 850/900/1800/1900MHz
    CDMA Network type: 800MHz
    UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
    Protocol: SIP (RFC3261),IAX2
    Transport Protocol: UDP,TCP,TLS,SRTP
    DTMF: RFC2833, SIP INFO, In-band
    Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a. ...

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    Looking for India Cli and non Cli Skype : john.edward178

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  • 05/29/15--06:37: Asterisk config logger.conf
  • Logger.conf


    In this file, you configure logging to files or to the syslog system.

    "logger reload" at the CLI will reload configuration of the logging system.

    [general] section

    dateformat
    Customize the display of debug message time stamps. see strftime(3) Linux manual for format specifiers
    this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS)
    dateformat=%F %T

    appendhostname
    This appends the hostname to the name of the log files.
    appendhostname = yes

    [logfiles] section


    Format is "filename" and then "levels" of debugging to be included:
    debug
    notice
    warning
    error
    verbose
    dtmf

    Special filename "console" represents the system console
    syslog keyword : This special keyword logs to syslog facility. Follow it with a . and the syslog facility name to use

    Note: To see the cli output use the verbose log level.

    Examples:
    debug => debug
    console => notice,warning,error
    messages => notice,warning,error
    full => notice,warning,error,verbose
    syslog.local0 => notice,warning,error


    Rotating logs


    You can rotate logs by running "logger rotate" on the CLI.

    You can do this from cron by putting the following in a cron job:

    /usr/sbin/asterisk -r -x 'logger rotate'

    On Win32 platform only you can do the following:

    If you want to rotate your logs (for example starting a new log every day), use the windows scheduler. Add a new scheduled item, (eg: midnight every day / week / month) to run, and till it to run RotateLogs.vbs This script will tell Asterisk to create a new log, and then rename the old log in standard windows log format (eg: msg_YYYYMMDD.log).

    The RotateLogs.vbs script file should be downloaded from Telium and copied anywhere on your disk. Edit the script for to modify the constants at the top of the file, to tell where you placed asterix.exe, where your logs files are, etc.

    Then that's it!

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  • 05/30/15--18:32: 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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    Debugging and troubleshooting VOIP problems.

    (SIP, MGCP, H.323, RTP, Skinny etc.)

    One of the primary techniques is to view what is actually getting sent and received by VOIP devices. There are several ways to do this:
    • Monitor Ethernet Traffic
    • Debugging displays from a VOIP program

    It helps to understand whats supposed to be happening. Studying the relevant RFCs and other protocol documents and tutorials is helpful.

    Ethernet Monitoring Tools

    • ClarifiedNetworks
      • HowNetWorks - a free VMWare appliance
      • Tia - sophisticated monitoring and flow analysis tool with visualisation, multiple data sources, ...
    • List of Monitoring and sniffing software
    • ngrep (Available for Linux, Windows, Apple, BSD, etc.)
      • Dumps only the ASCII portion of packets, excellent for ASCII based protocols
    • sngrep (Available for Linux, Apple, BSD.) by Irontec
      • Curses based SIP dialogs monitoring tool.
      • Support UDP, TCP and TLS.
      • Graphical(curses) dialog summary and dialog flow detail.
      • Very useful for debugging and learning purposes.
      • GPL License
    • Packetyzer: User-friendly packet sniffer for Windows, supports SIP
    • Rate which provides real time packet-per-second and data transfer rates
    • SIP Workbench Displays SIP ladder diagram from WireShark/pcap captures
      • Displays STUN/TURN interactions
      • Allow users to filter on particular call flows
    • Spirent Communications - Test Solutions for VoIP networks and devices
    • STINGA SS7 Protocol Analyzer and Monitoring System
      • Call trace, mature SS7 protocol decodes
      • SIP, SS7, SIP-T, SIGTRAN, ISDN, MGCP and Megaco/H.248
      • Performance analysis
      • Open MySQL database
    • TamoSoft CommView - network analyzer for Windows
      • Real-time VoIP call monitoring
      • SIP and H.323 analysis and decoding, call playback
      • Jitter, QoS, Bandwidth charts
    • tcpdump (standard utility in most Linux distributions)
    • Touchstone
      • WinEyeQ
        • 100% software-based
        • monitors/analyzes/records/replays SIP and H. ...

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  • 05/31/15--08:58: Virtual PRI
  • Virtual PRI services is an optimum solution for businesses that require quality, high capacity local access numbers without having their own facilities. The Virtual PRI provides immediate turnup and allows expansion into new regions and markets with very minimal cost and investment.

    Typical applications for Virtual PRI service are:

    • Call Centers
    • CallingCard Applications
    • Conference Applications
    • High volume incoming calls businesses


    Virtual PRI Provider List (In Alphabetical Order):


    DID Number Virtual PRI We Offers DID + 30 incoming channels in 23 countries for 39.99$ a month, for Special offer contact support@buydidnumber.com .

    Alcazar Networks - Over 3,100 rate centers. Virtual PRI as low as $46.00 for 23 channels. DID numbers as low as $0.10/each. Stop buying from the middle man.

    Virtual Number Virtual PRI Offers DID + 30 incoming channels in 23 countries for 39.99$ a month, SPECIAL OFFERING Also Available 100 Channel DID's in Europe at $ 25 a Month , contact support@buyvirtualnumber.com to get details of Special Offering

    CallnFax PRI for business - We offer PRI services in several cities, and the customer support your business needs. We offer hands on service custom tailored to your business needs

    DIDLIVE Virtual PRI Offers DIDs + 24 incoming channelsor more in the USA and Canada with over 4000 cities available. Call 1-212-901-0800 or visit www.didlive.com for details.

    DIDWW Virtual PRI Offers DID + 30 incoming channels in 23 countries for 50$ a month, revenue sharing model available.

    DomesticNumbers Unlimited channels DID + unlimited incoming channels for €19.95/month covering 70+ countries. Also International Free Phone (Toll Free) and Premium numbers available.

    MyVNumber - Virtual Phone Numbers and Virtual PRI Numbers with 30 channels per DID. Amazing Cloud PBX phone system platform is included for each account.

    Phone2call Offers Virtual PRI with 30 channels (up 30 simultaneous incoming calls) around 25+ countries. You can forward your Virtual PRI to regular mobiles/landlines (PSTN), VOIP, Google Talk, Linphone, Virtual PBX. Only 2 USD/channel. Know why we keep VOIP leadership!

    SendMyCall Virtual PRI Offers DID + 30 incoming channels in 26 countries for only 45$ a month. ...

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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.

    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.

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

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  • 05/31/15--22:37: PORTech
  • PORTech is the original manufacturer in Taiwan with 36 years; therefore we’ll do our best to offer the best service and support

    PORTech major
    E1 /T1 GSM Channel Bank
    1/2/4/8/16/32 ports VoIP GSM/UMTS/3G/CDMA Gateway
    SIM Server
    SBK-32/128:32/128 SIMs Remote SIM Bank
    SMS Server
    SMS Gateway
    IP Speaker
    IP Broadcasting System
    RFID Reader
    Free Roaming Gateway(1 port Remote sim)
    VoIP Adapter
    2 ports GSM VoIP PCI Card (Asterisk GSM Card)
    1/4/8 ports GSM Fixed Wireless Terminals(Follow Me GSM Gateway)
    Smart Power Monitor,IP Power Switch /PSTN Power Switch(4 Ports /8 ports)-Remote power manage by PSTN and WEB
    http://www.portech.com.tw

    Worldwide use ( 2G-850/900/1800/1900 MHz ,3G compatible with all world and Japan,CDMA 2000)

    Frequecny:
    Quad Band:850/900/1800/1900MHZ
    CDMA 2000(800/1900MHZ)
    3G/UMTS:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900, 2100 MHz

    VoIP GSM Gateway/IP UMTS Gateway/IP 3G Gateway/Mobile VoIP

    MV-370:1 port
    MV-372:2 ports
    MV-374:4 ports
    MV-378:8 ports
    MV-3716:16 ports
    MV-3732:32 ports
    call termination (VoIP to Mobile ) and origination (Mobile to VoIP)
    Support Asterisk,Trixbox,SIP Proxy Server,VoipBuster
    Image

    Image


    Portech SIM Server

    SIM Server main handle MV-37X:VoIP GSM Gateway and SBK-32:Remote sim bank
    With SIM Server,you can set talk time per sim and assign several sims(4 or 8 sims) per gsm port

    SS-128: 128 sims SIM Server (manage 4 SBK-32 sim banks)
    SS-256: 256 sims SIM Server (manage 8 SBK-32 sim banks)
    SS-512: 512 sims SIM Server (manage 16 SBK-32 sim banks)

    Features:

    Set Talk Time per SIM
    Set GSM Group (Assign several SIMs Per GSM Port)
    Set Day of week
    Set Time Range

    Portech SBK-32 : 32 SIMs Remote SIM Bank

    SBK-32_200.jpg

    Remote sim access that connect with serial MV-37X via internet.
    SBK-32 : 32 SIMs Remote SIM Bank
    SBK-128 : 128 SIMs Remote SIM Bank

    With SBK-32,SIM cards are no longer need to be installed in MV-37X anymore; you can deploy your GSM gateways in different locations and countries. New concept of using SBK-32 solution, you can centralize and supervise all sims in one place.
    In that case, SIM management will be easy, quick and safe as well. All SIM cards can be stored and control in one secure place. Once in a while, MV-37X must change SIM cards to get the best rate on different GSM operators. With SBK-32, you can place physical sims in office or change SIM corresponding port directly on web any second.MV-37X connects with SBK-32 to read the SIM information and register it on the GSM network. ...

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  • 06/01/15--11:32: VoIP Gateways
  • Information about VoIP gateways, including VoIP media gateways, FXO gateways, and other VoIP gateways can be found on this page. If you want to add your company's products, please read the Posting Guidelines for Promoting Products and Services

    Media Gateways

    Media gateways, also commonly referred to as VoIP gateways are devices which bridge conventional telephone networks and equipment to VoIP telephone networks. A typical media gateway has at least one conventional telephone port and at least one ethernet port.

    Analog FXO gateways


    • Allywll-WIA2008 GSM/CDMA VoIP gateway
    • 1Telecom Ltd - FXO/FXS VoIP GSM Gateways
    • 2N Telekomunikace - FXO GSM Gateways
    • Aastra Aastra Venture FXO Gateway
    • Abilis Abilis the all-in-one VoIP gateway with ISDN backup
    • AirTouch - FXO/FXS Skype VoIP Gateways
    • ALLYWLL - FXO/FXS/SIP VoIP Gateways
    • Allwin Tech SIP/H.323 dual protocols, 2/4/8 FXO/FXS ports ,NAT, Router, register up to 4 servers simultaneously
    • Anketechnology - FXO gateway--VoicePixie-211 www.anketechnology.com
    • Atcom - FXO gateway for skype Au-600forward skype to your mobile phone
    • AudioCodes - FXS & FXO
    • Axtan
    • AZACALL200 - 2 port FXS, 1 port FXO, 1 Lan , 1 WAN. ...

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  • 06/01/15--11:34: CyberData
  • www.CyberData.net

    Products:

    ceilingspkr.png
    Ceiling Speaker V2.0
    indoorintercom.png
    Cyberdata Intercom
    intercom2.png
    Cyberdata Intercom





    Cyberdata PoE VoIP Paging Gateway

    Cyberdata VOIP Gateway


    The CyberData SIP-enabled Paging Gateway allows the use of legacy analog zone paging amplifiers when converting to a VOIP system. The Gateway is compatible with most SIP-based IP/PBX servers that comply with the SIP RFC 3261. The Gateway is powered via PoE 802.3af - no external power supply is needed. ~ description via website


    Notes on setup in FreePBX using a SIP Extension:
    Set Qualify = no to get it to work.

    WAudette...

    For more information contact;
    Bill Majerczak
    billm@cyberdata.net
    831-373-2601 x-102

    Cyberdata PoE VoIP Speaker


    An overhead paging speaker that connects direct to Ethernet.

    From the Data Sheet:

    The CyberData PoE VoIP Speaker is a
    Power-over-Ethernet (PoE) and
    Voice-over-IP (VoIP) public address
    loudspeaker that easily connects into
    existing local area networks with a single
    cable connection. The speaker is powered
    via a standard ethernet cable – no external
    power supply is needed. With its small
    footprint and low height, the speaker can be
    discreetly mounted almost anywhere.


    New in 2009 - VoIP Talk-Back Speaker

    ^
    The CyberData SIP-enabled Talk-Back Speakers is a Power-over-Ethernet (PoE802.3af) and Voice-over-IP (VoIP) public address two-way loudspeaker that is easily connects into existing local area network with a single CAT5 cable connection. The speaker is compatible with most SIP-based IP PBX solutions. In a non-SIP environment, the speaker is capable of broadcasting audio through a multicast. It’s small footprint and low height allows the speaker to be discretely mounted almost anywhere. Speakers can be mounted into existing ceilings or in one of our enclosure kits. Enclosure kits offered are beveled wall mount adapters and NEW in 2009 our wall mount adapter with a digital clock perfect for schools.


    See Also



    Gentek
    Westcon
    888VoIPstore
    Neobits
    Anixter
    VoiPmodo.com

    New since last update

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  • 06/01/15--13:03: Call Quality Metrics
  • Several standards are available for measuring telephone call quality.

    MOS measures subjective call quality for a call. MOS scores range from 1 for unacceptable to 5 for excellent.
    VOIP calls often are in the 3.5 to 4.2 range.

    The ITUP.861 (PSQM) and P.862 standards explain how to calculate MOS scores.
    The methods used require a test call in order to make the measurements. Kurittu and Mattila debate the practicality of P.862 for VoIP in an AES paper titled "Practical Issues in Objective Speech Quality Assessment with ITU-T P.862". I can't vouch for it because I couldn't find it online for free, but there may be a few similarities with Takahashi, Hideaki, and Kitawaki's Perceptual QoS Assessment Technologies for VoIP.

    ITUP.563 calculates call quality passively and calculates an R Factor that can be used to estimate a MOS score.

    Using the PESQ and PAMS call quality measurement methods requires a license from Psytechnics a spinoff from British Telecom.

    Newer versions of Cisco IOS has built-in tools for measuring call quality metrics including estimated MOS scores for test calls. A detailed description of their methods is in this document: Service Assurance Agent (SAA) VoIP Proactive Monitoring

    For products to measure call quality see: How To Debug and Troubleshoot VOIP

    See also


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    Fidem Technologies born out of the passion and dedication of a group of young entrepreneurs who have experienced and learned about business the most practical way doing it themselves. Founded in 2011 and based out of New Delhi, the capital city of India and a thriving technology and business hub.

    Fidem Technologies aims to simplify Voip, Networking and Surveillance technology by providing Products, Services, Support, Applications and Consulting under the same umbrella. Fidem Technologies is all about simplifying discovery and reducing the time to procure, thus saving the most precious resource - time.

    Fidem Technologies when formed was based on just a very simple principle. “SERVICE”. Being on par in terms of price and quality only gets you into the game. Service wins the game. Therefore Fidem Technologies foundation is always give people more than what they expect to get.

    Fidem Technologies's Online VoIP Store is Online Shop which representing technology partners i.e 3CX, 3CX Phone System, Avaya, Aastra Phones, ACTi, "AudioCodes", Axis, Allo, Cisco, ClearOne, CyberData, D-Link, "Dinstar", Dasscom, "Digium", Digium Switchvox, Elastix, Fanvil, "FreeMate", Freetalk Wireless Headsets, "Grandstream," Gateprotech, GoIP, Huawei, Jabra, Kaspersky, Konftel, Logitech, Moxa, Mediatrix, Mobotix, Multi-Tech, Matrix, Netgear, OpenVox, OBihai, Panasonic, Patton, Plantronics, Polycom, QNAP, Quintum Technologies, RedFone Communications, Rhino Equipment, RockBochs, Ruckus Wireless, SpectraLink, "Sangoma", Sangoma Session Border Controllers, Sennheiser, Siemens, "Snom," Sony, Ubiquiti, Vonia Headset, "Yealink", Zyxel.

    "Fidem Techologies" Products Categories :


    VoIP Phones
    o Audiocodes SIP IP Phones (6)
    o Avaya SIP IP Phone (11)
    o Cisco SIP IP Phones (26)
    o Fanvil SIP IP Phone (11)
    o Digium SIP IP Phones (3)
    o Grandstream SIP IP Phone (24)
    o Matrix SIP IP Phones (4)
    o Yealink SIP IP Phones (19)
    o Polycom SIP IP Phone (14)
    VoIP Analog Gateways
    o FXS Analog Gateways (29)
    o FXO Analog Gateways (18)
    o Combo FXO-FXS Gateways (8)
    VoIP Digital E1-T1-PRI Gateways
    o AudioCodes Media Gateways (3)
    o Digium Media Gateways (4)
    o Matrix Media Gateways (1)
    o SangomaVega Media Gateways (3)
    VoIP Asterisk Cards
    o Digium IP Telephony Card (19)
    o Sangoma IP Telephony Card (19)
    o OpenVox IP Telephony Card (12)
    VoIP Headsets & Dial-Pads
    o Avaya VoIP Phone Headsets (15)
    o Contact Center Headset (27)
    o Dial-Pads (5)
    o Cisco IP Phone Headset (15)
    VoIP PBX Systems
    o IP- PBX-Hardware (10)
    o IP- PBX- Software (18)
    VoIP ATA's
    o Audiocodes ATA's (1)
    o Cisco ATA's (0)
    o Grandstream ATA's (5)
    o Matrix ATA's (4)
    o Patton ATA's (1)
    VoIP GSM Gateway
    o Dinstar GSM Gateways (5)
    o Matrix GSM Gateways (5)
    o Yeastar GSM Gateway (4)
    VoIP IP Cameras
    o Grandstream (17)
    o Mobotix (2)
    Repairing of Headsets
    Wireless Solutions
    o Indoor Access Points/Client Bridges (42)
    o Outdoor Access Points/Client Bridges (56)
    • About Us
    • Specials
    • Our Brands
    • Contact Us


    Fidem Technologies is dealing in VoIP Phone, VoIP Gateway, VoIP Headset and Dial Pad, VoIP PBX, VoIP Camera, VoIP Solutios, VoIP FXS Gateways, VoIP FXO gateways, VoIP GSM Gateway, VoIP Asterisk Cards,VoIP Digital gateways,VoIP Hadware, VoIP Equipments, 3CX Phone System, Aastra Phones, ACTi Dome IP Cameras, ACTi Indoor IP Cameras, ACTi NVR, ACTi Outdoor IP Cameras, ACTi PTZ IP Cameras, ACTi Video Servers, Adtran Routers, Advanced Network Devices, Digital Message Boards, Audiocodes E-SBC, AudioCodes Gateways, AudioCodes Mediant 1000, AudioCodes Mediant 2000, AudioCodes Mediant 600, AudioCodes MediaPack, Audiocodes MSBG Gateways, AudioCodes Phones, Axis Accessories, Axis Camera Housing, Axis Camera Mounts, Axis Lenses, Axis Peripherals, Axis Power, Axis Software, Axis Dome IP Cameras, Axis Indoor IP Cameras, Axis IP Cameras, Axis Outdoor IP Cameras, Axis PTZ IP Cameras, Axis Thermal IP Cameras, Axis Video Servers, 1-Port Video Servers, 6-Port Video Servers, Rack Solutions, Axis Wireless IP Cameras, Cisco Adapters, Cisco Conference Phones, Cisco Phones, Cisco Phone Accessories, Cisco Phone Headsets, Cisco Wireless Phones, Cisco Routers, Cisco Security, Cisco Switches, Cisco 100 Series Unmanaged ...

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    This is a list of Asterisk system vendors in India.



    AAB Asterisk Consultant India


    Services Provide:

    • Predictive Dialer
    • Voice Broadcasting & Fax Blasting
    • Medium to Large Multi-Office Business Telephony Systems
    • Call Centers - Local and International
    • Asterisk Dialers and Bulk Calling Systems
    • International Office Telephony Systems
    • Specialty Calling Systems (Entertainment and Personals)
    • Healthcare Application Integration
    • Customer Relationship Management, CRM Application Integration
    • Distributed Server Architecture and Asterisk Load Balancing
    • SIP Express Router, SER Load Balancing
    • Hospitality Telephony Systems (Hotel PBX Integration)
    • Complete IVR Development
    • Local or Datacenter PBX Customization
    • Wireless Telephony Installations
    • Database Integration and Customization
    • Custom Application Development
    • vtiger vicidial integration

    IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices.

    Support for digium / openvox / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards /grandstream

    Asterisk Hyderabad Ph : 9392335385
    Asterisk Bangalore Ph : 9392335385

    Asterisk IPPBX India (Hyderabad)

    Phone: +91 9392335385

    Asterisk Consultant : Subramanyam (subbu)

    subbu6699@gmail.com

    Grandstream hyderabad Ph : 9392335385
    Asterisk Chennai Ph : 9392335385
    Asterisk Mumbai Ph : 9392335385
    Grandstream india Ph : 9392335385
    predictive dialer hyderabad Ph : 9392335385
    voip hyderabad Ph : 9392335385

    Asterisk Service and Solution

    • Asterisk Services and Solution Provider division of Ecosmob Technologies Pvt. Ltd. has expertise in open source and Asterisk PBX. The company has developed many custom software solutions for the global clients and enjoy the leading position in the Asterisk industry. The team of experienced Asterisk developers offers design, development, configuration and support services for following

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  • 06/02/15--04:19: Open Source Billing Systems
  • Billing Systems


    This page lists Open Source Billing systems.

    General purpose systems


    Asterisk2Billing (A2Billling):


    A2Billing, combined with Asterisk is a physical Telecom Platform and Soft-Switch providing a wide range of telecoms services using both traditional telephone technology or VoIP. It contains a real time billing engine which rates and bills and invoices calls, and supports payment gateways.


    This now gives any Telecom company a very good reason to consider the A2Billing Platform over the traditional offerings for TDM and VoIP Soft-Switches as well as wholesale billing.

    Solutions

    A2Billing can be used in a number of different roles, and our consultancy services can advise on the appropriate hardware and configuration of your Switch.

    • Calling Card service with either PIN or Caller ID recognition
    • Call-back service
    • VoIP Billing
    • VoIP termination for IP PBX systems
    • Wholesale VoIP Termination and Origination
    • Residential VoIP Termination and Origination.
    • Special Applications Platforms
    • Predictive Dialler and Sales Campaign Tool
    • Hosted PBX, IP Centrex and Multi-tenant systems.
    • VoIP reseller white label solutions.
    • and much more :)

    There is online demo available at http://demo.asterisk2billing.org/a2b/. For more information about any of the features and benfits of any of our products, please contact us at sales@star2billing.com. For A2Billing installation services, see http://www.star2billing.com/consultancy/managed-install/



    AstBill:

    AstBill is not only a web-based, user friendly billing interface for Asterisk and VOIP. It is also a Asterisk configuration and GUI management tool and a standardized implementation of Asterisk using REALTIME and static configuration as you please. AstBill is Open Source and under constant development. Prepaid, Postpaid and Calling Cards supported.


    Asterisell:


    Asterisell is a open source web application
    for rating, showing, and billing of VoIP calls.


    ASTPP

    Features:
    • Provide call rating for Asterisk.
    • Calling Cards
    • Integrates with AMP (amp.coalescentsystems.ca)
    • An "unlimited" amount of different price lists.
    • Batch billing
    • Realtime billing
    • Rates are billed to 6 decimal places.
    • Rates are highly configurable. ...

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  • 06/02/15--04:39: Sip Trunking Providers
  • This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

    Country specific pages:


    What Is SIP Trunking?

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

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

    The Benefits of Using SIP Trunking Services

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

    How SIP Trunking Can Take Your Business To The Next Level

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

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

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




    1comms VoIP provider for UK Businesses. ...

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  • 06/02/15--04:46: VOIP PBX and Servers
  • Please list information about VoIP PBX and Servers on this page. Please keep VoIP PBX and server provider information in alphabetical order, and below any other relevant information.

    Page Contents

    Numeric

    • 1comms.co.uk: Asterisk-based converged telephone system for UK Businesses
    • 1GATE VoIP PBX by Wangate: Cheap VoIP PBX with hardware DSP. Optional internal gateways for Analog/ISDN/PRI/GSM. VoIP resellers welcome.
    • 2daydirect: Brand NEW Small Business VoIP phones. Free 2 day shipping anywhere in the United States
    • 2N NETSTAR PBX, virtual PBX: VoIP PBX system
    • 2N Omega IP PBX: VoIP PBX system
    • 2N VoiceBlue Enterprise: Simple VoIP SIP PBX
    • 3CX: Windows IP PBX / VOIP Phone system
    • 4PSA VoipNow: Hosted PBX software for service providers and enterprises, accelerating SaaS deployment. It runs on Linux environments (RHEL, SuSE Linux, CentOS, Fedora) on x86 and Power PC architecture based servers.
    • 8ix Zenith: 8ix Zenith spells an Asterisk derived IP Telephony application with the most advanced calling and communication features.

    A

    • ALLO PSTN-IPPBX for SOHO with 30 IP extension, upto 6 Analog Extension & upto 4 PSTN trunk
    • ActivePBX™ | Turn-Key Business Phone System $149/mo.

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  • 06/02/15--04:50: VoIPSupply.com
  • sm_voipsupply_fb.GIF


    About VoIPSupply.com


    VoIPSupply.com is the retail website of North America's leading VoIP solutions provider, VoIP Supply. VoIPSupply.com offers a fast and easy way to find the VoIP hardware you are looking for.

    VoIPSupply.com has products for consumers, small medium businesses, enterprises, government/educational entities, resellers and service providers. You can visit VoIPSupply.com Here.

    About VoIP Supply


    VoIP Supply is North America's leading VoIP solutions provider. Since 2004, VoIP Supply has delivered valuable solutions for some 60,000 customers worldwide.

    With over 30 passionate employees, 2,500 products, 40,000 square feet of office space and an unlimited number of VoIP solutions to meet your needs VoIP Supply has everything you need for VoIP, whether you are a consumer, business, service provider or reseller.

    VoIPSupply.com Offerings


    VoIPSupply.com specializes in standards based VoIP equipment and VoIP systems. In addition to complete VoIP systems, VoIP Supply offers:


    Your Favorite VoIP Manufacturers


    VoIP Supply makes it easy for you to find products from your favorite manufacturers.

    For your convenience, some popular manufacturer categories include:

    • VoIP Headsets for Polycom Phones - Take the guesswork out of figuring out which VoIP headset pairs perfectly with your Polycom Phone
    • Cisco VoIP Phones - Save money on long distance charges and transfer calls to anyone at any location with Cisco VoIP phones. Deploy a phone system anywhere you have broadband Internet access
    • Grandstream SIP Video Phones - Grandstream SIP Video phones offer Skype certified compatibility, multiple SIP account support and up to a 7" color LCD display

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  • 06/02/15--08:15: Fidem Technologies
  • Fidem Technologies born out of the passion and dedication of a group of young entrepreneurs who have experienced and learned about business the most practical way doing it themselves. Founded in 2011 and based out of New Delhi, the capital city of India and a thriving technology and business hub.

    Fidem Technologies aims to simplify Voip, Networking and Surveillance technology by providing Products, Services, Support, Applications and Consulting under the same umbrella. Fidem Technologies is all about simplifying discovery and reducing the time to procure, thus saving the most precious resource - time.

    Fidem Technologies when formed was based on just a very simple principle. “SERVICE”. Being on par in terms of price and quality only gets you into the game. Service wins the game. Therefore Fidem Technologies foundation is always give people more than what they expect to get.

    Fidem Technologies's Online VoIP Store is Online Shop which representing technology partners i.e 3CX, 3CX Phone System, Avaya, Aastra Phones, ACTi, "AudioCodes", Axis, Allo, Cisco, ClearOne, CyberData, D-Link, "Dinstar", Dasscom, "Digium", Digium Switchvox, Elastix, Fanvil, "FreeMate", Freetalk Wireless Headsets, "Grandstream," Gateprotech, GoIP, Huawei, Jabra, Kaspersky, Konftel, Logitech, Moxa, Mediatrix, Mobotix, Multi-Tech, Matrix, Netgear, OpenVox, OBihai, Panasonic, Patton, Plantronics, Polycom, QNAP, Quintum Technologies, RedFone Communications, Rhino Equipment, RockBochs, Ruckus Wireless, SpectraLink, "Sangoma", Sangoma Session Border Controllers, Sennheiser, Siemens, "Snom," Sony, Ubiquiti, Vonia Headset, "Yealink", Zyxel.

    "Fidem Techologies" Products Categories :


    VoIP Phones
    o Audiocodes SIP IP Phones (6)
    o Avaya SIP IP Phone (11)
    o Cisco SIP IP Phones (26)
    o Fanvil SIP IP Phone (11)
    o Digium SIP IP Phones (3)
    o Grandstream SIP IP Phone (24)
    o Matrix SIP IP Phones (4)
    o Yealink SIP IP Phones (19)
    o Polycom SIP IP Phone (14)
    VoIP Analog Gateways
    o FXS Analog Gateways (29)
    o FXO Analog Gateways (18)
    o Combo FXO-FXS Gateways (8)
    VoIP Digital E1-T1-PRI Gateways
    o AudioCodes Media Gateways (3)
    o Digium Media Gateways (4)
    o Matrix Media Gateways (1)
    o SangomaVega Media Gateways (3)
    VoIP Asterisk Cards
    o Digium IP Telephony Card (19)
    o Sangoma IP Telephony Card (19)
    o OpenVox IP Telephony Card (12)
    VoIP Headsets & Dial-Pads
    o Avaya VoIP Phone Headsets (15)
    o Contact Center Headset (27)
    o Dial-Pads (5)
    o Cisco IP Phone Headset (15)
    VoIP PBX Systems
    o IP- PBX-Hardware (10)
    o IP- PBX- Software (18)
    VoIP ATA's
    o Audiocodes ATA's (1)
    o Cisco ATA's (0)
    o Grandstream ATA's (5)
    o Matrix ATA's (4)
    o Patton ATA's (1)
    VoIP GSM Gateway
    o Dinstar GSM Gateways (5)
    o Matrix GSM Gateways (5)
    o Yeastar GSM Gateway (4)
    VoIP IP Cameras
    o Grandstream (17)
    o Mobotix (2)
    Repairing of Headsets
    Wireless Solutions
    o Indoor Access Points/Client Bridges (42)
    o Outdoor Access Points/Client Bridges (56)
    • About Us
    • Specials
    • Our Brands
    • Contact Us


    Fidem Technologies is dealing in VoIP Phone, VoIP Gateway, VoIP Headset and Dial Pad, VoIP PBX, VoIP Camera, VoIP Solutios, VoIP FXS Gateways, VoIP FXO gateways, VoIP GSM Gateway, VoIP Asterisk Cards,VoIP Digital gateways,VoIP Hadware, VoIP Equipments, 3CX Phone System, Aastra Phones, ACTi Dome IP Cameras, ACTi Indoor IP Cameras, ACTi NVR, ACTi Outdoor IP Cameras, ACTi PTZ IP Cameras, ACTi Video Servers, Adtran Routers, Advanced Network Devices, Digital Message Boards, Audiocodes E-SBC, AudioCodes Gateways, AudioCodes Mediant 1000, AudioCodes Mediant 2000, AudioCodes Mediant 600, AudioCodes MediaPack, Audiocodes MSBG Gateways, AudioCodes Phones, Axis Accessories, Axis Camera Housing, Axis Camera Mounts, Axis Lenses, Axis Peripherals, Axis Power, Axis Software, Axis Dome IP Cameras, Axis Indoor IP Cameras, Axis IP Cameras, Axis Outdoor IP Cameras, Axis PTZ IP Cameras, Axis Thermal IP Cameras, Axis Video Servers, 1-Port Video Servers, 6-Port Video Servers, Rack Solutions, Axis Wireless IP Cameras, Cisco Adapters, Cisco Conference Phones, Cisco Phones, Cisco Phone Accessories, Cisco Phone Headsets, Cisco Wireless Phones, Cisco Routers, Cisco Security, Cisco Switches, Cisco 100 Series Unmanaged ...

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  • 06/02/15--09:11: Asterisk Traning
  • Companies that provide Asterisk training:












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