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