IP PBX (Private branch exchange) is a business telephone system designed to deliver voice or video over a data network and interoperate with the normal Public Switched Telephone Network (PSTN). Like Trixbox, Asterik, SIP PBX etc. VoIP (Voice over Internet Protocol) gateways can be combined with traditional PBX functionality enabling businesses to use their managed intranet to help reduce long distance expenses, enjoy the benefits of a single network for voice and data and advanced CTI features or be used on a pure IP system which in most cases give greater cost savings, greater mobility, and increased redundancy. An IP-PBX can exist as a hardware object, or virtually, as a software system.

Archive for the 'IP PBX' Category

SIP PBX Welltech

Written by admin on Thursday, November 6th, 2008 in IP PBX.

SIPPBX IP-PBX series 6200S, 6200GS and 6200N :

Introduction
    
SIPPBX 6200x IP-PBX product series are IP based IP-PBX which including
legend digital PABX telephony services, Auto Attendant, Voice Mail, Enterprise
Coloring Ring Back Tone, Conference and Announcement features together. It
also integrates Visual Door Phone unit at reception desk for visitor service. By
using WVP-800 Video Phone and SIP IP Camera through Web browser and
dial-in as extension number, SIPPBX 6200N, 6200GS and 6200S provide not
only office visual communication between coworker and visitors but also
provide enterprise/office local and remotely Security monitoring.

It supports up-to 1000 subscribers ( SIPPBX 6200N ), up to 400 users at
SIPPBX 6200GS and is a cost effective solution for small to medium
enterprise. Also the traveler soft-phone (SP362), operator console ( software
or hardware LP-600 with up to 96 DSS keys ) and billing software provides
you a complete evolution from traditional digital PABX to new generation IP
based PABX.
   
Benefits and application
   
    * Office Voice and Data Communication based on one IP network
    * Upgrade from 50 user up to 1000 users by license file (with different
      hardware )
    * Support SIP/RTP Encryption/Decryption for security purpose
    * One extension number can use different device at different location.
    * Flexible to make VoIP call or local PSTN call
    * Integrate SIP Video Door Phone at Reception Desk for Visitor
      Service
    * Support Video Phone to talk with Video SIP Door Phone
    * Support SIP based IP Camera for office surveillance and
      reception desk monitoring
    * Efficiency communication with voice PBX, Video Phone and
      replace reception desk operator

Key Features
   
          o SIP RFC 3261 Compliance
          o Voice Codec G.711, G .729A and GSM voice codec
          o Support SIP/RTP Encryption/Decryption
          o Support RADIUS Server or Enterprise Billing via TCP
          o Support High Available active/standby Redundant (optional)
          o Support SIP ENUM
          o Support VLAN & QOS tagging ( 6200N does not support VLAN )
          o Support Setup Wizard for easy installation to IP Phone
          o Built-in CDR Report
    * Auto-attendant
          o Web-Base Auto Attendant Flow Editor
          o Scheduled Special Announcement
          o Holidays Working Time Support
          o Multiple Language Support
          o Support Branch Office
          o Support Transit Call
    * Voice Mail
          o Web-Base Voice Mail Flow Editor
          o Personal Greeting
          o Multiple Language Support ( Customized Local Language )
          o Native TTS (Chinese & English & Japanese) Support
          o Support Additional Customized TTS Language
          o Message Waiting Indication
          o Email Notify
          o Web Retrieve
          o Phone Retrieve
    * Conference Bridge
          o Support RFC 4579 (without XML)
          o Ad-Hoc Conference
          o Virtual Conference (Meeting Me)
          o Virtual Conference (Ad-hoc)
          o Event Tone Notice
          o Up-to 8 parties for software DSP version (6200S/6200GS)
          o Quick Conference by Soft-phone (SP362)
    * Enhanced Service
          o System Announcement Service
          o Company-wide Coloring Ring Back Tone Service
          o Provided Server Hold Tone
    * Voice Router
          o Public and Private IP Legs
          o SIP-Aware RTP Routing
          o Natural VOIP Firewall/NAT
    * Optional Features
          o Traveler Soft-phone (SP 362)
          o Operator Console Software
          o Operator Console LP-600 up to 96 DSS keys ( coming soon )
          o Enterprise Billing Software
          o Web Caller Module
          o Microsoft Office Communicator/Exchange 2007 Module (Option)
    * Selected Telephony Features
          o Call Transfer
          o Call Forward
          o Call Forwarded Notice
          o Call Screening
          o Caller ID Privacy
          o Call Waiting
          o Call Hold
          o Call Pickup (Global, Group)
          o Specified Call Pickup
          o Find Me
          o Short Code
          o Do Not Disturb
          o Miss Call Notify by Email
          o ANI Replacement
          o Call Return
          o Hide ANI/Show ANI Selection
          o Call Park/Retrieve
          o Call Camp on
          o Display Name Replacement
          o PSTN Number
          o Ring PSTN & IP Device Simultaneously
          o Broadcasting Service
          o Wake-Up Call
          o Reject Anonymous Call
          o Support SIP TAPI
          o Busy Lamp Filed (RFC 4235)

Optional Features
   

    * Traveler Soft-phone (SP 362)
    * Operator Console Software ( PC Windows environment )
    * Enterprise Billing Software
    * Web Caller Module
    * Microsoft Office Communicator / Exchange 2007 Module ( OCS )

ePBX-100A-128, Embedded Asterisk IP-PBX

Introduction
   

Welltech ePBX-100A-128 is the new generation IP PBX system for small
enterprise. It is also designed to operate on a variety of VoIP applications, such
as auto-attendant, voice conference, call transfer, call pick up and IP-based
communications. Beside voice switch over IP, it also supports Video switch over
IP-PBX. It integrates Video phone WVP-800, SIP based IP Camera for Security
and Surveillance purpose. SIP IP based Door Phone with Video are also
integrated to establish an voice, Video, Security and office monitoring application.
It is suitable for small Enterprise/Office or homes application.

Customers can select different suite and optional products to meet their request.
To Integrate with FXO gateway ( i.e. WellGate 2680 and 3804A ) to provide PSTN
lines access. Install IP Phone LP-388, LP-388A(PoE) and LP-600 to provide IP
phone or attendant console. To install FXS gateway to connect analog phone sets
such as WellGate 2424S, 2608 and 3504A , 3502A.

To install WVP-800 Video Phone, SIP IP Camera to provide front desk Lobby
monitoring and Security / IP Surveillance purpose. To Install SIP Door Phone to
welcome visitor from any extension of ePBX-100A-128 without allocating one staff at
front desk.

With flexible and full functionality, ePBX-100A-128 give a complete transition from
traditional digital PABX to the new generation IP-PBX.
   
Specifications
   

   1. SIP RFC 3261 Compliance
   2. Voice Codec : G.711, G .729A, GSM and G.723 Pass-through
   3. H.263 Video Codec pass-through
   4. Automated Attendant
   5. Branch Office
   6. System Announcement Service
   7. Call Routing
   8. Speed Dial
   9. Call Transfer
  10. Call Forward
  11. CLIR
  12. Do Not Disturb
  13. Call Pickup (Global and Group)
  14. Specified Call Pickup
  15. Call Park
  16. Camp-On (Call Back on Busy)
  17. SIP Trunk
  18. Call Monitor
  19. CDR - Call Detail Record for Billing Purpose
  20. Busy Lamp Filed
  21. Music On Hold
  22. Music Ring Back Tone
  23. Voice Mail to E-mail
  24. Access Voice Mail by phone set
  25. NAT Traversal
  26. Export and Import Configuration
  27. Built-in 1 GB voice mail storage

The differences between ePBX-100A-128 and ePBX-100A :

   1. Built-in 128MB RAM.
   2. Linux Kernel Version 2.6
   3. Asterisk V1.4.
   4. Support 15 concurrent calls.
   5. Support Broadcast (3 Broadcast Groups, 8 members per group).
   6. Support Meet-Me Conference (2 Conference Rooms, 6 members per room).
   7. Support T.38 FAX.
   8. Support DDNS.
   9. Support Video with MPEG4 pass-through.
  10. Support Music to play with MP3 format.
  11. Support personal password for outbound call.
  12. Support NTP server for LAN port.
  13. Not save Voice Mail to CF card - Option

More Details on Welltech SIP PBX

VoIP PBX Aastra

Written by admin on Wednesday, November 5th, 2008 in IP PBX.

A member of the AastraLink family of PBX solutions, the AastraLink Pro 160 is a Linux-based appliance that hosts Asterisk® open source PBX software. Targeted for 25 users and under, the AastraLink Pro provides all the standard PBX/Key System features and functionality while leveraging a host of new IP-based services. Specifically designed to meet the unique requirements of small business, AastraLink Pro enables users to quickly and easily get phones up and running by simply plugging the system into an existing LAN.

Key Features and Benefits:

    * Feature Rich IP PBX
      The AastraLink Pro delivers large PBX functionality to the small business environment. Featuring an Auto-Attendant, which automatically configures on initial startup, the AastraLink Pro manages and directs incoming calls to individuals or groups by extension dialing or name search. In addition, individual Voicemail boxes are also automatically configured and operational at start up. PBX telephony features include: call forward, hold and hold alerts, call log, call transfer, 3 way conferencing, corporate and personal directory.

    * Simple Deployment
      By simply plugging the AastraLink Pro into an existing LAN (Local Area Network) it’s ready to start communicating without the need of an installation technician. The AastraLink Pro will auto discover and configure any new Aastra SIP phones connected to the system making the need for a service technician to perform moves, adds and changes a thing of the past.

    * Networking, Teleworking and Scalability
      The AastraLink Pro grows as your business expands. It can host up to 50 extensions when SIP trunking is used, and up to 9 AastraLink Pro systems can be linked together via IP to create a WAN linked multi site office environment. Remote tele-working from a home office is as simple as taking an auto discovered AastraLink SIP phone and plugging it into your UPnP enabled home network.

    * Affordable Solution integrates with Aastra SIP phones
      With no hidden application license or user costs, or the need of an installation technician, AastraLink Pro is a cost effective and powerful small business communication system that tightly integrates with the following Aastra SIP
      phones and expansion modules: 51i, 53i, 55i, 57i, 57i CT, 536M, and 560M, and the newly released 9143, 9480i and 9480i CT.

More info on Aastra PBX Phone system

VOIP IP PBX Grandstream

Written by admin on Wednesday, November 5th, 2008 in IP PBX.

VOIP IP PBX Grandstream

Grandstream’s IP-PBX product segment consists of the GXE5024 and GXE5028.  The GXE502x appliance is a powerful all-in-one voice + video + fax + data communication solution for the small to medium sized business, especially companies with sub-30 seats per location.  The GXE502x takes modern business communication systems to a heightened level of innovation, quality, reliability, ease of deployment and affordability.

Designed from ground up to support distributed IP communications, intelligent unified messaging, advanced application integration and popular PBX features, the GXE502x product family also optimally integrates legacy PSTN trunk and telephone interfaces for fail-safe hybrid communication needs in all circumstances including power or network loss.

Features & Benefits

Key Features

    * Integrated high performance data router with advanced voice/video QoS support
    * Integrated legacy PSTN trunks, analog phone/FAX ports & unlimited SIP trunk options
    * Integrated session border controller (SBC) for NAT/firewall traversal and secure telecommuting
    * Integrated conference bridges that allow any combination of IP or PSTN calls using any codecs(built-in transcoding)
    * Unified messaging including voicemail-to-email, fax-to-email and video-to-email (pending)
    * Power and network failure survivability and recovery; Integrated PoE (802.3af)
    * Support true & local emergency call routing in all circumstances
    * Automated detection and provisioning of IP phones, video phones, ATA and other endpoints for easy deployment
    * Rich PBX features such as presence, shared line appearance, call park & pickup, call queue, ACD, intercom & paging, ring group, customizable auto attendant & IVR, personal music-on-hold, branch office system peering
    * Hardware accelerated encryption engine to ensure strongest security protection using SRTP and TLS
    * Personal Web portal to manage individual phone/call setting, personal greeting, new or saved voice/fax/video messages for each extension user
    * Flexible dial plan, call routing and call recording (pending)

More Details on VOIP IP PBX GXE5024 and GXE5028

 

ShoreTel IP PBX

Written by admin on Tuesday, November 4th, 2008 in IP PBX.

 The ShoreTel IP phone system is a completely integrated system that scales seamlessly from 1 to 10,000 users including PBX, voice mail, and automated attendant functions.

The ShoreTel system is built from the ground up and designed to be the easiest to use, easiest to manage, full-featured IP phone system on the market today. Its distributed architecture is ideal for multi-site companies that span multiple locations because the ShoreTel IP phone system appears and behaves as a single, unified system.

Distributed Architecture

ShoreTel’s distributed architecture is ideal for multi-site companies that span multiple locations because the ShoreTel IP phone system appears and behaves as a single, unified system.

ShoreWare Director

With ShoreWare Director, your entire IP phone system can be managed from a single browser-based interface. Instead of multiple PBX, voicemail and automated attendant systems, ShoreWare Director unifies all systems into a single interface.

Unified Messaging

The ShoreTel IP phone system provides a complete messaging solution from voicemail to automated attendant, along with desktop tools designed to increase employee productivity.

More IP PBX Information
 

Linksys SPA9000 IP PBX

Written by admin on Tuesday, November 4th, 2008 in IP PBX.

Linksys SPA9000 :IP Telephony System
Full Featured IP PBX System for the Small Business and Home Office

* IP PBX system with high-end features comparable to traditional large business voice services
* Supports 16 Linksys IP Phones per SPA9000
* Powerful self-configuration capabilities enabled with Linksys IP Phones
* Works with most Internet Telephone Service Providers

The SPA9000 marries the rich feature set of high-end PBX telephone systems with the convenience and cost advantages of Voice over IP. It has common voice system features such as an auto-attendant, shared line appearances, three way call conferencing, intercom, music on hold, call-forwarding and much more. The SPA9000 opens up access to the benefits of VoIP, including low cost long distance service, telephone number portability, and one network for both voice and data.

The SPA9000 is so easy to configure that a fully working system can be set up in minutes. New telephones are automatically detected and registered when they are connected to the SPA9000. The SPA9000 has an integrated web server that allow features to be configured using a web browser. The web server has multiple levels of password protected access to user and service level features. Service level settings may be locked by the Internet Telephone Service Provider to ensure they are not inadvertently corrupted. The Internet Telephone Service Provider also can remotely update the software and settings through a secure encrypted connection.

With its integrated router, the SPA9000 can be either connected directly to the internet connection or to another router on your network. The SPA9000 has separate WAN and LAN Ethernet ports. The WAN connection can be connect through DHCP or a fixed IP address. The LAN port can assign IP addresses to IP Phones and computers using NAT and DHCP.

While the SPA9000 will work with any SIP compatible IP Phone, it is the ideal host for Linksys IP Phones, such as the SPA901, SPA921, SPA922, SPA941, and SPA942. Powerful configuration capabilities enable the SPA9000 to support a greater set of advanced features with these IP Phones, such as shared line appearances, hunt groups, call transfer, call parking lot, and group paging. With its two FXS ports, the SPA9000 can support traditional analog devices such as telephones, answering machines, FAX machines, and media adapters. The SPA9000 supports 16 Linksys 900 series IP Phones.

More Details on SPA9000 IP PBX

How does an IP PBX work?

Written by admin on Monday, October 27th, 2008 in IP PBX.

How does an IP PBX work?

Here is a very good article which explains how an IP PBX work. also what is hosted IP-PBX,
what is the difference between tradition PBX and IP PBX.
One thing is important you should know about Hosted IP PBX :
Hosted IP PBX requires lower latency and greater attention to QoS controls. Small business Internet connections over asymmetric broadband connections such cable modems or DSL are rarely capable of providing acceptable service for more than a few users.

D-Link’s Small Business VoIP Kit DVX-4010

Written by admin on Monday, October 27th, 2008 in IP PBX.

D-Link’s Small Business VoIP Kit DVX-4010

Enable IP Telephony throughout your office with D-Link’s Expandable IP Telephony Kit (DVX-4010). This kit includes one SIP IP PBX (DVX-1000), one Analog Trunk Gateway (DIV- 140), and ten Business IP Phones (DPH-140S ) to allow you to create a comprehensive IP communications network.



D-Link IP PBX

Microsoft Response Point

Written by admin on Monday, October 27th, 2008 in IP PBX.

Microsoft Response Point

Microsoft Response Point is an advanced software-based telephone system developed by Microsoft. Response Point, a PBX system targeting small businesses with less than 50 employees,was launched in March 2007, with systems available on the market in the fourth quarter of that year. Response Point is VoIP-based, and uses SIP as its’ signaling and call setup protocol.Response Point supports voicemail and multi-party calling in addition to two party VoIP calls.Response Point features innovative voice recognition technology to manage calls and voice mail. Voicemail messages can, optionally, be sent to e-mail where they can be retrieved and archived. Response Point voice dialing can works with the Response Point phone directory which is currently limited to 1100 contacts per user. Contacts may be imported from the Windows Address Book or Microsoft Outlook.Response Point automatically detects gateways and phones connected to the network.

Quanta’s Syspine was the first OEM product on the market. D-Link followed shortly afterward with their VoiceCenter product. Aastra became the third OEM entry with a system, Aastralink RP, introduced in July 2008 coinciding with the delivery of Response Point Service Pack 1.

Announcements in September 2008 have added additional manufacturers providing new gateway options (a T1/E1/PRI gateway and a 4-port FXS gateway) and new phone options (full-duplex conference room phones).

Service Pack 1

MS Response Point Service Pack 1 (RP SP1) was released in July 2008, This update is freely available for download at http://www.microsoft.com/responsepoint with the following features:

    * Click-to-Call Dialing (the ability to double-click or right-click on a co-worker’s extension or an imported Outlook contact, and automatically dial the phone in that fashion)
    * SIP trunking support (SIP trunking allows an PBX owner to replace traditional PSTN lines with PSTN connectivity via a SIP trunking service provider (ITSP). This new support allows RP to use a supported SIP Trunking provider if there is an internet connection provided to the RP base unit)
    * Direct-inward-dial (DID) support when used with a SIP Trunking provider (ITSP)
    * Enhanced call logging functionality in the RP Administrator program
    * Several call handling and performance improvements
    * Updates and enhancements to the RP Assistant and RP Administrator programs

Source: Wikipedia, the free encyclopedia

IP PBX (Zultys)

Written by admin on Monday, October 27th, 2008 in IP PBX.

IP PBX (Zultys)
Overview

The MX30 is a powerful system that enables multimedia communications for all workers in a small office. By integrating the functions of many devices into a compact box, the MX30 simplifies the VoIP network of any deployment. As well as being a comprehensive solution, the MX30 is easy to install, use, and maintain.

The MX30 is designed specifically to connect Internet Telephony service providers (ITSPs) using SIP. This allows customers to realize the full benefits of VoIP and save money because of the superior rates offered by ITSPs.

The system combines the functions of an IP PBX, Internet gateway, network server, and application server. By using standard protocols, it is interoperable with phones, gateways, and devices from other manufacturers.

With the MX30, workers are accessible on a central communication system from any location with an Internet connection. Direct connectivity among all users is easily achieved, regardless of their location within a building, campus, city, country, or region. Users of the system can log in from the office, from home, while traveling, or from a different site and still have full access to all system features.

All users, regardless of role within the organization, get a high quality voice system and access to productivity tools that increase their ability to get more done in less time. The system offers powerful applications such as presence, instant messaging, unified messaging, call handling, screen pops, and detailed call logs. These tools can operate on a single system or among multiple MX systems of varying sizes for a centrally managed platform that painlessly scales to support large enterprises.

Key Features
 

    * Complete PBX functionality with integrated voice mail
    * Based on open standards: SIP, Linux, VoiceXML, TAPI, SQL
    * Supports 30 users in a single box
    * Expandable to support 10,000 users in multiple locations
    * Gateway to ITSPs, PSTN, and Internet
    * Fax termination, origination, and storage
    * Instant messaging, presence, and chat
    * Unified messaging
    * Call detail reports
    * Archiving for regulatory compliance
    * Multiple automated attendants
    * Multiple ACDs, hunt groups, and operator groups
    * Advanced call center features
    * Flexible paging groups
    * Remote users over the Internet
    * Integration with external CRM applications
    * Multiple languages and worldwide support

IP Telephony
The MX30 supports true end to end IP telephony for users among different sites, remote locations, and temporary work stations over the WAN and Internet. Users are no longer bound to any physical phone or port. They can travel from one location to another and still be directly connected to the company’s communication system.

The MX30 can interoperate with any SIP compliant phone. Zultys has its own range of IP phones that are fully compatible with the MX30. These include desktop phones as well as the soft phone included in the user software (MXIE). You can provision and maintain IP phones directly from the administration software, which allows for management of all IP phones within the office.

The MX30 can connect to one or more ITSPs (Internet telephony service providers). This permits the use of the Internet for all voice calls without the need to connect to the PSTN. Alternatively, the MX30 provides you with the flexibility to route some calls to the PSTN and other calls to the ITSP.

Backup Telephony Interfaces
The system has a single slot for an optional telephony interface module to support backup PSTN connectivity in the case where calls are not routed to an ITSP.

The ISDN BRA modules provide one or two full-duplex S/T circuits that support ETSI and Japanese ISDN. The analog FXO modules provide two or four two-wire circuits which support detection of caller ID. The 2-port analog module also provides one connection for a phone that can be automatically connected to the first FXO circuit on the PSTN in the event of a power failure.

Music on Hold
The MX30 can play MoH to callers.This is played from a WAV file on the hard disc and can be music or an announcement describing the company’s products or services.

Dial Plan and Least Cost Routing
The system has a flexible dial plan that allows you to specify the routing of calls based on dialed patterns and available bandwidth. Your organization can ensure least cost routing for all calls, regardless of the user or location.

For each dial pattern on the dial plan, you specify the source of the call, the transformed pattern, the primary route, and alternative routes. You can create multiple dial patterns to handle internal calls and external calls. These calls can be routed over the LAN, the WAN, the Internet at an external gateway, or a line interface to the PSTN.

Call Accounts and Restrictions
Call accounting allows you to force specified users to enter an account code (contract code) before making certain types of calls. You can obtain call detail records based on the account code and therefore portion expenses among the various accounts. This is useful for organizations that bill their clients for calls made on behalf of the client.

You can restrict calls to specific destinations based on the user’s profile or where the user’s phone is located. The restriction can force the user to enter his or her password prior to making the call. The system can allow calls from certain phones without authentication, allow phones in other locations to require authentication, and require users wherever they are located to authenticate prior to making a call.

Data Networking
The MX30 has two Ethernet ports – one connects to the service provider and one connects inside the office. A comprehensive firewall is between these two ports.

The WAN port connects directly behind the broadband IAD (integrated access device) and provides termination of traffic and address translation. The translation comprises NAT (network address translation) for Internet traffic and ALG (application layer gateway) for SIP traffic. The port can have a fixed IP address, an IP address received from the ISP using DHCP, or use PPPoE to obtain all information. Default and static routes can be provisioned to control the proper routing of voice and data traffic.

The LAN port connects directly to an Ethernet switch. Over this port, the MX30 can provide servers for DHCP, TFTP, and NTP. You can independently disable any of these if you provide them externally to the MX30.

The system can terminate multiple simultaneous VPN sessions with remote IP phones or other MX30 or MX250 systems. This allows users at branch offices or home offices to securely access all functions of the MX30 and the corporate network without the need for external equipment.

MXIE—MX Interface for End Users
MXIE (pronounced “mixee”) is the PC client interface for users on the MX30. It enhances a user’s ability to be more efficient in communicating with other users and external callers. With MXIE, users have address books, detailed call logs, call handling rules, message delivery rules, voice mail, faxes, instant messages, chat, and presence. MXIE can be used as a stand-alone application with a built in soft phone, or in conjunction with a desktop phone. There is only one login required for access from any location, regardless of the user’s role as an operator, a call center agent, or an individual.

MXIE runs under Windows, Mac OS/X, and Linux. Not all features of MXIE are available on all platforms.

Management of Users
Adding or deleting users on the MX30 can be done in seconds. Multiple users can be imported by uploading an existing database. The MX30 supports multiple user profiles to set rights and privileges on the system. These rights include password settings, voice mail access, MXIE access, long distance dialing, and paging authority.

Accessibility of Users
The MX30 can be configured to handle calls in the most efficient manner possible. Users can have eight contact points where they can receive calls. A contact point can be an IP desktop phone or soft phone. An incoming call can alert all contact points simultaneously, with each contact point being located anywhere on the network. Users can create custom call handling rules that can route their calls to any destination based on any combination of date, time, day of week, caller ID, presence, and location of log in.

Auto Attendant (AA)
The MX30 can accommodate multiple AAs to service different applications. Each AA can be assigned a direct external number (DID) and an internal extension number. An AA can be scheduled to run different scripts at different times of the day and on different days of the week. Special scripts can be scheduled to run only for specific days and times. You can also schedule times when operators answer all calls.

The scripting is accomplished with a graphical user interface and creates a standard VoiceXML program. You can create a script to prompt a user for input and then provide details on different products or services.

The MX30 has integrated text to speech capability in multiple languages. You can easily create an announcement for callers and integrate it as part of the AA script.

Voice Mail (VM)
You can select how VM storage is divided among the users by setting the limits on the user profiles. Limitations that you can set for a user profile include the total number of received messages, the duration of each message, and the total disc space allowed for storage of all messages on the MX30.

Users can access, save, delete, and forward VM by using a phone (internal or external to the MX30) or by using MXIE. Users can save their voice messages without taking up storage on the MX30 by dragging the messages from MXIE into folders on the local PC. Voice messages are saved as standard WAV files and forwarded to others outside the system through standard means of file sharing such as email and network directories.

Fax Origination and Termination
The MX30 can send and receive faxes, eliminating the need for a paper fax machine. To send a fax, the user prints a document from the PC to the MX fax printer driver. The MX30 sends the fax as soon as possible. The MX30 terminates incoming faxes, converts them to a TIF file, and makes them available to users through MXIE. The user can access and manipulate them in the same way as voice messages.

Operators
You can define multiple groups of operators on the MX30. Operators within a group can be assigned different priority levels to accommodate skills based routing and back up shifts. A user can belong to multiple operator groups and the MX30 can distribute operator calls to the user based on his or her priority within the group.

Operators use their MXIE login as the console, and thus there is no need for special equipment. Using MXIE, they have access to the complete directory of users and can use the 10-key operation on a PC keyboard to transfer calls. Any user of the system who is already familiar with MXIE can immediately become functional as an operator. Without a requirement for a physical console or special software training, an organization can dynamically assign operators at any time of the day. Operators, at all priority levels, can be located anywhere and can log in at any time to provide uninterrupted coverage of incoming calls.

Automatic Call Distribution (ACD) and Hunt Groups
The MX30 can provide multiple ACD and hunt groups for either the informal or formal call center. You can configure ACD or hunt groups for different services within the business. Each group can be assigned a direct inward dial number (DID) in addition to an internal extension. You can assign a user to be an agent for one or more groups, and agents within a group can be assigned different priority levels for call distribution. A user who belongs to one or more groups can still make and receive individual calls.

For the formal inbound call center, the advanced ACD features on the MX30 provide call queues, real time supervisory monitoring, queue manipulation, group and agent statistics, queue overflow handling, and playing of promotional messages to callers in queue.

Unified Messaging
The MX30 supports unified messaging for delivery of voice, fax, and other notifications to the email client of choice. Each user can set up email notification of these messages with or without the message content attached. He or she can set up rules for email delivery based on media type, date and time, source, or age of the unchecked message.

Integration with External Applications
Integration with an external CRM or IVR package can be achieved through access with SIP, TAPI, HTTPS, and VoiceXML. The caller’s information can be passed to the CRM package to invoke a screen pop with immediate presentation of account information to the agent.

Call Detail Records (CDR)
The MX30 provides comprehensive CDR for reconciliation of billing and tracking of system usage. Predefined reports include usage reports for automated attendant, users, groups, emergency services, trunk lines, and dial plan routes. Activity can be filtered by user, extension, location, or group. For custom reports, the system integrates with external applications by providing read access to its MySQL database. Administrators can use Crystal Reports or any other third party reporting tool to generate the most appropriate call detail reports for the organization.

Paging Groups
The MX30 supports paging over the speakers of IP phones. You can configure multiple paging groups, and assign a user as a member of one or more paging groups. A paging group can comprise users from any location, thus users are logically divided rather than physically divided into zones. For example, a customer support agent can hear the announcements for sales and technical support teams, while a technical support agent can hear announcements only for the technical support team. Users can receive specific announcements from any location within the enterprise, even over the WAN and Internet. Paging authority can be restricted by password authentication and by assignment of paging profiles.

Encryption
The MX30 supports AES encryption to fully secure speech traffic between the system and internal or external callers. When enabled, encryption provides security for calls that occur between a user and voice mail or auto attendant. Encryption can also be enabled for all traffic between two MX30 systems on a WAN.

Archiving
The MX30 allows for the archiving of all voice mail, instant messages, and chat sessions to an external server. This archiving can meet regulatory requirements such as those imposed by the HIPAA act and the SEC.

Codecs and Voice Compression
You can specify the type of codec available for transmitting voice during calls between the MX30 and other IP devices. You can select between G.711 µ-law, G.711 A-law, G.729A, G.729AB, or any combination. The MX30 will automatically negotiate the codec that is available. When you have multiple MX30s and MX250s communicating within a group, you can specify which codec to use between sites to ensure the most efficient use of bandwidth within a WAN.

Locations and Emergency Dialing
Locations are created to display the correct time on a telephone when it is in a different time zone from the system. They are also used to ensure that when a remote user dials an emergency phone number, the call is routed to the local public service access point (PSAP). A location can be based on an IP address or selected by the user. The dial plan on the MX30 routes emergency calls properly when it is connected to the PSTN. If the emergency call is routed to the ITSP, the ITSP must ensure proper handling of the emergency call. All operators are alerted of the user’s name and location when an emergency call is made.

System Administration
The MX30 is managed from a single graphical user interface that runs on a PC under Windows. The PC can be located anywhere in your network. Different users can have different administration rights. All provisioning is done with this interface, which simplifies learning and increases productivity.

Administrators have real time views into the system, including call detail reports, current sessions, active registrations, status of SIP messaging, and status of telephony circuits. Critical events are sent to a Syslog server so administrators can receive a page or call if there are problems. All system configuration, voice mail messages, faxes, and call detail reports can be periodically backed up. You can schedule complete or partial back up to a network directory or FTP server.

System Capacities
When you purchase the MX30, it is equipped with all the hardware necessary to support 30 users and 120 SIP registrations. You purchase what you need initially, and subsequently expand the functionality and capacity by purchasing software licenses for the system. These licenses can be added at any time from any location without having to power down the system.

Multiple MX30 and MX250 systems can be networked using MXgroups to provide capacities to 10,000 users with a transparent database of users. Users at one site can communicate with users at the same or another site with equal capabilities. Users who travel among the sites can log into any MX and continue to make and receive calls as if they were at their normal location.

Global Features
Zultys sells and supports its products worldwide, allowing the MX30 to be readily deployed in one or more countries. The system supports worldwide telephony protocols so it can connect directly to the local PSTN. You can install any one of a number of language packs available for the voice prompts. The language for the MXIE user interface can be selected dynamically by the user.

IP PBX (Zultys) detaled information

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