IP PBX

Brekeke :Hosted IP Telephony Platform - Multi-Tenant SIP IP-PBX

Brekeke PBX Multi-Tenant Edition is designed to be a Hosted IP-PBX Platform for Service Providers. The Multi-Tenant Edition provides easy setup and maintenance for hosting multiple clients under one server. This Multi-Tenant Edition provides the most flexibility and profitability for your IP telephony service.

Supporting the industry standard, Session Initiation Protocol (SIP), Brekeke PBX delivers a scalable and reliable VoIP phone platform. Brekeke PBX supports Microsoft Windows and Red Hat Linux.

More on  Brekeke :Hosted IP Telephony Platform

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

How to build and customize your own PBX with Asterisk:

  Here is a step by step guide tutoral of how to build and customize your PBX with Asterisk.

This article demonstrates how easy it is to roll your own PBX in about an hour or two. Provided that the instructions herein are followed carefully, you too should be able to set up your very own switchboard/PBX system and all for the cost of the target hardware of your choice.

PBX Tutorial

TrixBox Guide Tutorial (TrixBox Without Tears)

trixbox®, with a lowercase ‘t’, is an IP-PBX software solution designed for small and medium-sized businesses. trixbox comes in two flavors: the open-source community edition and a hybrid-hosted, commercially-proven solution.

trixbox CE and Pro are supported by a world-class technical support team that has been built over the last four years and processes over 2,000 calls and 10,000 emails per month. Our team is here to help.

trixbox Pro

Resell a reliable and easy-to-manage telephony solution to your SMB customers

Here is an very good Asterik/trixbox tutorial.

The following is the table of contents of this Asterisk/Trixbox User Guide:

1  Introduction. 8

1.1  The Components. 8

1.1.1  The IP PBX. 8

1.1.2  Phones. 9

1.1.3  SIP Gateway. 9

1.1.4  Home Network. 9

1.1.5  VOIP Service Providers. 9

2  Is VOIP for You?. 10

2.1  What is it going to cost ?. 10

2.2  What will the Quality of the phone calls be?. 10

3  Installation. 11

3.1  Change default Settings. 12

3.1.1  To get Help. 12

3.1.2  Change Linux Password. 13

3.1.3  Change IP Address (set IP address to Static) 13

3.1.4  Set Time Zone. 14

3.2  Connect to AMP (or FreePBX)  from Web Browser 15

3.2.1  Log in to Asterisk Management Portal (AMP) 15

or FreePBX Administration. 15

3.3  General Settings. 18

3.3.1 General Settings for AAH 2.7 or Earlier 18

3.3.2 General Settings for AAH 2.8 or Later 19

3.3.3  Dial Command Options. 19

3.4  Extensions. 22

3.5  Follow Me (AAH 2.8 with freePBX) 24

3.6  Ring Groups. 25

3.6.1  Now it’s a good time to set up your softphone. 26

4  Incoming Calls Handling. 27

4.1  Incoming Calls (Prior to AAH 2.8 with FreePBX) 27

4.2  Incoming Calls (AAH 2.8 with FreePBX) 27

4.2.1  Time Conditionds. 28

5  Trunks And Routes. 29

5.1  What is a Dial Pattern?. 29

5.2  What is a Trunk?. 29

5.3  Let’s Create The Trunks. 30

5.3.1  Pennytel 30

5.3.2  Oztell (SIP) 32

5.3.3  Oztell (IAX) 33

5.3.4  Astratel 34

5.4  Create Outbound Routing. 37

5.4.1  What is an Outbound Route?. 37

5.4.2  How does it work?. 37

5.4.3  International 38

5.4.4  Domestic. 38

5.4.5  MobileAust 39

5.4.6  Oztellonly. 39

5.4.7  Astratelonly. 40

5.5  ENUM.. 41

5.5.1  Setting up ENUM Trunk. 41

5.5.2  Setting up ENUM Outbound Route. 42

5.6  Inbound Route (DID Routes A@H version 1.x) 44

5.6.1  Inbound Route (AAH version 2.7 with AMP and earlier) 44

5.6.2  Inbound Route (AAH version 2.8 with freePBX) 44

6  Digital Receptionist 46

6.1  Setting Up Digital Receptionist 46

6.2  Customising Individual Extension. 50

7  Get Under The Bonnet 51

7.1  Editing The .conf Files. 52

7.1.1  iax.conf 52

7.1.2  Indications.conf 52

7.1.3  enum.conf 52

7.1.4  extensions.conf 52

7.1.5  extensions_custom.conf 53

7.1.6  sip.conf 53

7.2  Port Forwarding – Routers. 54

7.3  QOS – Routers. 54

8  Check your System.. 56

8.1  System Status. 56

8.2  Asterisk Info. 57

9  Interfacing Asterisk to PSTN.. 58

9.1  Digium Wildcard X100P FXO PCI Card. 58

9.1.1  zapata.conf 58

9.1.2  zaptel.conf 59

9.1.3  modules.conf (modprobe.conf for AAH 2.x) 59

9.2  Digium TDM400P FXO/FXS Card. 59

9.2.1  zapata-auto.conf 59

9.2.2  modules.conf (modprobe.conf for AAH 2.x) 60

9.3  Rebuilding Zaptel Driver 61

9.4  Sipura SPA3000 as a PSTN Interface. 62

9.4.1  Log in to SPA3000. 62

9.4.2  Change the settings. 62

9.4.3  Add SIP Trunk. 64

9.4.4  SPA3000 as an outbound PSTN Trunk. 65

10  PSTN to VOIP Gateway. 66

11  DISA. 67

11.1  DISA prior to AAH version 2.8 using AMP.. 67

11.2  DISA – AAH version 2.8 using FreePBX. 68

12  Setting Up a Soft Phone. 69

Profile Tab. 69

Audio & Video Tab. 69

Network Tab. 70

STUN Tab. 70

Call Forward. 71

13  Flash Operator Panel (FOP) 72

13.1  Setting Up. 73

13.1.2  Setting the Admin Password. 73

13.1.3  Hang-up a Call 73

13.1.4  Transfer a Call 73

13.1.5  Initiate a Call 73

13.1.6  Barge in or Create a Conference. 73

13.1.7  Alternative Flash Operator Panel. 73

14  Call parking and transfer. 74

14.1  Call Transfer - Managed. 74

14.1.1  How is it done?. 74

14.2  Call Transfer – Blind. 75

14.3  Put a Call On Hold. 75

14.4  Call Pickup. 75

14.5  Filter Your Incoming Calls - Only Accept Known Calls. 76

15  MeetMe – Teleconference. 78

15.1   meetme.conf 78

15.2   meetme_additional.conf 78

15.3   extensions.conf 78

16  Voicemail Email Notification. 80

16.1  Installing sendmail 80

16.2  /etc/hosts. 81

17  Fax through AAH (fax to e-mail) 83

17.1  Installing Fax in AAH version 2.3/2.4. 83

17.2  Installing Fax in AAH version 2.8 with FreePBX. 86

18  Weather Forecast 88

18.1  Configure Weather Report – On Demand. 88

18.1.1  Create Extension codes. 89

18.2  Configure Weather Report – Background Method. 90

18.2.1  Create Extension codes. 90

18.3  Now put it in your Digital Receptionist 91

19  Remote Management 92

19.1  httpd.conf 92

20  How to set up a Remote Extension. 93

20.1  Create a new extension. 93

20.2  IAX.Conf 95

20.3  Sip_Nat.Conf 95

20.4  The correct softphone for IAX. 95

20.4.1  First the general options. 96

20.4.2  Accounts options. 96

21  Tools. 97

21.1  Webmin. 97

21.2  Putty. 97

21.3  WinSCP.. 97

22  STUN Servers. 98

23  Dialing through MS Outlook®.. 98

23.1  Download AstTapi 98

23.2  Install AstTapi 99

23.3  Configure Outlook Address Book. 99

24  Speed Dial 101

24.1  The Dumb-Me method. 101

24.2  Asterisk@Home’s method. 102

25  How to use Window Messenger® 5.x. 103

26  How to interconnect 2 boxes. 105

26.1  Method 1 - With the Peer Asterisk box as Extensions. 105

26.2  Method 2 - In a Peer/User Arrangement 106

27  Backup and restore. 108

27.1  Creating a Backup using the console. 108

27.2  Restoring a Backup using the console. 108

27.3  Maintaining multiple backups. 109

28  Customised Voice (prior to AAH 2.8 with freePBX) 110

28.1  Customised Voice Prompts. 110

28.2  Installing Other Languages. 111

29  My Asterisk® IP PBX Network. 112

30  Publications and References. 113

30.1  The future of Telephony by O’Reilly Publishing. 113

30.2  Voice over IP – Per call bandwidth consumption. 113

30.3  Other Asterisk@Home Forums, Tutorials and Wikis. 113

31  USB Phone Support 114

32  Bugs Reports. 115

32.1  Backup and Restore. 115

32.2  Conflict between ZAP Card and FAX Module. 115

32.3  Editing email notification message. 115

32.4  Endless Loop during Directory search (v2.0) 115

32.5  Flash Operator Panel (Bug Detected in version 1.3) 116

32.6  Max Channels Bug. 116

32.6.1  Corrected in version 2.8. 116

32.7  Ring Group Hunt Syntax Bug (v2.2 and v2.5) 117

32.7.1  Corrected in version 2.8. 117

32.8  Securing Asterisk@Home console Alt+F9. 117

32.9  Sound Directory Permission (v2.6) 117

32.10  WakeUp Calls (V2.2) 118

32.10.1  Corrected in version 2.4. 118

33  Touble Shooting. 119

33.1  Asterisk Feature Codes not working. 119

33.2  Asterisk Drops Calls after a few seconds. 119

33.3  Pennytel Asterisk Problem.. 119

33.4  Unable to receive Incoming Calls. 119

Appendix A. 120

A.1  Asterisk Feature Codes. 120

A.2  Asterisk CLI commands. 120

A.2.1  General commands. 120

A.2.2  AGI Commands. 121

A.2.3  Database Handling. 121

A.2.4  IAX Channel Commands. 122

A.2.5  SIP Channel commands. 122

A.2.6  Server management 122

A.3  Asterisk Special Extensions. 123

A.4  Asterisk Common Variables. 123

A.5  Codec (Coder Decoder)

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

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

 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 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