Contact Information

    Do you need help with your Asterisk PBX Phone System? Well if you do you have come to the right place! Alpha Computer Group’s expert telecom engineers can support all versions of Asterisk and Linux. We provide 24/7 online, telephone, and chat support services for all types of Asterisk based applications. Our Telecom and Technical Support Engineers are trained professionals in all types of Asterisk applications, with troubleshooting experience ranging from basic dial plan to real time configuration management and complicated IVR integrations. Asterisk is a software implementation of a telephone private branch exchange (PBX); it was created in 1999 by Mark Spencer of Digium. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services, such as the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. Its name comes from the asterisk symbol, *.

    We love Asterisk at Alpha Computer Group. We even run Asterisk for our business! In our office we have a mixture of Cisco IP Phone 79xx running SIP and Chan_SCCP protocols. We can help with Configuration and Administration of Asterisk (All Versions), Elastix, TrixBox, PIAF, 3CX, AsteriskNow, Incredible PBX, FreePBX, IVR, Voice Logger, IP PBX and CallCentre Solutions (ViciDial). We also offer consultancy services for establishing Asterisk based Inbound / outbound Call Centre solutions, Enterprise IP PBX with 100 – 1000 extensions, IVR configuration and integration, CRM Integration, Voice Logger and other business telephony needs. We are your one stop source for everything Asterisk!

    Some of telephone system issues we can help you with:

    Emergency Support Issues Remote Extensions
    SIP Configuration Issues Firewall Issues
    IAX or IAX2 Configuration Issues Nat Issues
    Chan_SCCP Configuration Issues Install or Upgrade your Asterisk PBX or Server
    Extension Configuration Issues FreePBX/Any Distro or Linux password reset
    Ring Group and Queue Setup Setting up Network Time Server (aka NTP Server)
    Auto Attendent Configuration (aka IVR) One-way Audio Issues
    SIP Provider Issues Setting up Extension via the End Point Manger

    Much Much More!

    Phone System Features:

    ADSI On-Screen Menu System
    Alarm Receiver
    Append Message
    Authentication
    Automated Attendant
    Blacklists
    Blind Transfer
    Call Detail Records
    Call Forward on Busy
    Call Forward on No Answer
    Call Forward Variable
    Call Monitoring
    Call Parking
    Call Queuing
    Call Recording
    Call Retrieval
    Call Routing (DID & ANI)
    Call Snooping
    Call Transfer
    Call Waiting
    Caller ID
    Caller ID Blocking
    Caller ID on Call Waiting
    Calling Cards
    Conference Bridging
    Database Store / Retrieve
    Database Integration
    Dial by Name
    Direct Inward System Access
    Distinctive Ring
    Distributed Universal Number Discovery (DUNDi™)
    Do Not Disturb
    E911
    ENUM
    Fax Transmit and Receive
    Flexible Extension Logic
    Time and Date
    Transcoding
    Trunking
    VoIP Gateways
    Interactive Directory Listing
    Interactive Voice Response (IVR)
    Local and Remote Call Agents
    Macros
    Music On Hold
    Music On Transfer:
    Flexible Mp3-based System
    Random or Linear Play
    Volume Control
    Predictive Dialer
    Privacy
    Open Settlement Protocol (OSP)
    Overhead Paging
    Protocol Conversion
    Remote Call Pickup
    Remote Office Support
    Roaming Extensions
    Route by Caller ID
    SMS Messaging
    Spell / Say
    Streaming Media Access
    Supervised Transfer
    Talk Detection
    Text-to-Speech (via Festival)
    Three-way Calling
    Time and Date
    Transcoding
    Trunking
    VoIP Gateways
    Zapateller
    Voicemail:
    Visual Indicator for Message Waiting
    Stutter Dialtone for Message Waiting
    Voicemail to email
    Voicemail Groups
    Web Voicemail Interface

     

    Compatible Phones For Asterisk That We Support

    IP Phones/Desk Phones

    Aastra / Sayson phones
    ATCOM IP Phone
    ALLYWLL IP Phone
    Cisco 79xx series
    Cisco ATA 18x series
    Cisco 12SP+/VIP30
    Cisco Linksys / Cisco SPA phones – SPA devices (SPA941, SPA942…)
    D-Link DPH-540
    Digitmat GP1266
    Cortelco 2747
    GNET VP320
    Grandstream BudgeTone
    Grandstream GXP2020
    Linksys SPA-941
    Mitel Phones 5055

    Mitel Phones 5215
    Mitel Phones 5220
    Nortel Phones i2004
    ShoreTel 210
    Siemens HiNet LP5100
    Siemens OptiPoint 600 Office SIP
    Siemens Gigaset DECT
    Sipura SPA-2000
    Sipura SPA-3000
    Swissvoice IP
    Snom Phones
    Soyo G668
    Uni-Ta Professional
    UTP2000 Business IP phones
    UTP3000
    UTP1600
    UTP1200 Entry level IP Phone

    Uniden UIP200
    Polycom
    Sound Point IP
    Sound Station
    Video Phone
    Conference Phone
    Pulverinnovations WISIP
    Yealink
    IP Phone SIP-T19
    IP Phone SIP-T20
    IP Phone SIP-T21
    IP Phone SIP-T22
    IP Phone SIP-T26
    IP Phone SIP-T28
    YUXIN IP Phones
    YUXIN YWH600 IP Phone SIP
    YUXIN YWH201 IP Phone SIP,IAX2 Phones with 2RJ45 ports, POE
    Zultys IP Phones
    Zyxel P2000W
    VTA1000

    Soft Phones

    Idefisk
    iFon
    SJphone
    CounterPath X-Lite
    CounterPath eyeBeam
    CounterPath Bria
    Windows Messenger
    KPhone
    LIPZ4
    Firefly
    linphone
    MozPhone
    MGCP EyeP Phone
    SflPhone
    Ekiga
    KIAX
    Cisco IP Office Communicator

    ** We also support and install door phones and intercom systems **

     

    Computer-Telephony Integration

    AGI (Asterisk Gateway Interface)
    Graphical Call Manager
    Outbound Call Spooling
    Predictive Dialer
    TCP/IP Management Interface

    Scalability

    TDMoE (Time Division Multiplex over Ethernet)
    Allows direct connection of Asterisk PBX
    Zero latency
    Uses commodity Ethernet hardware
    Voice-over IP
    Allows for integration of physically separate installations
    Uses commonly deployed data connections
    Allows a unified dialplan across multiple offices

    Speech

    Cepstral TTS
    Lumenvox ASR

    Codecs

    ADPCM
    CELT (pass through)
    G.711 (A-Law & μ-Law)
    G.719 (pass through)
    G.722
    G.722.1 licensed from Polycom®
    G.722.1 Annex C licensed from Polycom®
    G.723.1 (pass through)
    G.726
    G.729a
    GSM
    iLBC
    Linear
    LPC-10
    Speex
    SILK

    VoIP Protocols

    Google Talk
    H.323
    IAX™ (Inter-Asterisk eXchange)
    Jingle/XMPP
    MGCP (Media Gateway Control Protocol
    SCCP (Cisco® Skinny®)
    SIP (Session Initiation Protocol)
    UNIStim

    Traditional Telephony Protocols

    E&M
    E&M Wink
    Feature Group D
    FXS
    FXO
    GR-303
    Loopstart
    Groundstart
    Kewlstart
    MF and DTMF support
    Robbed-bit Signaling (RBS) Types
    MFC-R2 (Not supported. However, a patch is available)

    ISDN Protocols

    AT&T 4ESS
    EuroISDN PRI and BRI
    Lucent 5ESS
    National ISDN 1
    National ISDN 2
    NFAS
    Nortel DMS100
    Q.SIG

    Did you know? The Asterisk software includes many features available in proprietary PBX systems: voice mail, conference calling, interactive voice response (phone menus), and automatic call distribution. Users can create new functionality by writing dial plan scripts in several of Asterisk’s own extensions languages, by adding custom loadable modules written in C, or by implementing Asterisk Gateway Interface (AGI) programs using any programming language capable of communicating via the standard streams system (stdin and stdout) or by network TCP sockets.

    Asterisk supports a wide range of Voice over IP protocols, including the Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), and H.323. Asterisk can interoperate with most SIP telephones, acting both as registrar and as a gateway between IP phones and the PSTN. The Inter-Asterisk eXchange (IAX2) protocol, RFC 5456, native to Asterisk, provides efficient trunking of calls among Asterisk PBXes, in addition to distributed configuration logic, and call completion to VoIP service providers who support it. Some telephones support the IAX2 protocol directly (see Comparison of VoIP software for examples).

    By supporting a mix of traditional and VoIP telephony services, Asterisk allows deployers to build new telephone systems, or gradually migrate existing systems to new technologies. Some sites are using Asterisk servers to replace proprietary PBXes; others to provide additional features (such as voice mail or voice response menus, or virtual call shops) or to reduce costs by carrying long-distance calls over the Internet (toll bypass).

    Asterisk was one of the first open source PBX software packages.

    In addition to VoIP protocols, Asterisk supports many traditional circuit-switching protocols such as ISDN and SS7. This requires appropriate hardware interface cards supporting such protocols, marketed by third-party vendors. Each protocol requires the installation of software modules. With these features, Asterisk provides a wide spectrum of communications options.

    World Wide Asterisk Appeal

    While initially developed in the United States, Asterisk has become a popular VoIP PBX worldwide because it is freely available under open source licensing, and has a modular, extensible design. The American English, French, Persian (Farsi) and Mexican Spanish female voices along with other new prompts like Australian English for the Interactive voice response and voice mail features of Asterisk are frequently updated with submissions from developers in many different languages and dialects. Additionally, voice sets are offered for commercial sale in different languages, dialects and genders.

    Products that use Asterisk

    Asterisk is a core component in many PBX’s in a box commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software as open source. Open-source examples include FreePBX, and Elastix.

    Asterisk is also included in the LinuxMCE home entertainment/automation system.

    Asterisk is released under a dual license model, using the GNU General Public License (GPL) as a free software license and a proprietary software license to permit licensees to distribute proprietary, unpublished system components.

    Originally designed for Linux, Asterisk also runs on a variety of different operating systems including NetBSD, OpenBSD, FreeBSD, Mac OS X, and Solaris. Asterisk is small enough to run in an embedded environment like Customer-premises equipment-hardware running OpenWrt.

    For all of your Asterisk needs contact Alpha Computer Group @ (877) 608 – 8647

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    Alpha Computer Group
    220A Franklin Avenue, Franklin Square, New York, 11010
    Phone: 1-877-608-8647 |

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