asterisk card,voip phone,gsm cell phone

How to build an IP PBX with Asterisk Card?

2009-04-05T08:04:00.000-07:00

How to build an IP PBX with Asterisk Card?

This article demonstrates how to build your own ip pbx in about an hour . Provided that the instructions herein are followed carefully, you too should be able to set up your very own ip pbx system and all for the cost of the target hardware of your choice.
1) . What is a PBX ?
A PBX (Private Branch Exchange) is a switch station for telephone systems. It consists mainly of several branches of telephone systems and it switches connections to and from them, thereby linking phone lines.
Companies use a PBX for connecting all the internal phones to an external line. This way, they can lease only one line and have many people using it together, with each one having a phone at the desk with different number. Inside a PBX, you only need to dial three-digit numbers to make a call to another phone in the network. We often refer to this number as an extension.
Most people are familiar with PBXs because they've used them in an office environment, but what if you wanted to use one in small office or your house? Enter Asterisk , which is free and offers all this functionality right out of the box.
Asterisk is a software implementation of a hardware PBX and can run on a variety of hardware platforms. The features and benefits of owning an Asterisk PBX are numerous, and seemingly only limited by the imagination of the person who sets up and uses an Asterisk PBX.
2). Build your Asterisk ip pbx
There are many ways of setting up an Asterisk ip PBX , with some being easier than others. The easiest amanner in which to set up a fully functional Asterisk ip PBX is to download and install aprecompiled/tuned distribution such as Trixbox, Asterisk NOW, etc. Just take your pick of those. These are great if you don't care to learn about how your PBX operates or need to do a mindless "click and drool" install. The problem with precompiled asterisk distributions is, while they are simple, they are not as flexible and extensible as a home-brewed Asterisk installation from the source code. When you install one of these "distros", it is quite often that important decisions are already made for you, and you must work within the framework of the distribution to keep the box "happy". When you roll your own Asterisk from the sources, you get more say in what goes into the build and how it is customized/extended at the expense of a slightly more complicated planning and installation process. However, it's not as hard as you might think and it's quite worth it in the end to "roll your own".
In this article I will cover the steps needed to install Asterisk for the Ubuntu Linux operating system. This article assumes you know how to do a minimal command line only base system install of Ubuntu Linux. You certainly don't want to run a GUI on your Asterisk box as it would consume unnecessary resources.

fxo card,FXO Module

2009-03-13T01:28:00.000-07:00

fxo card,FXO Module
The FXO module allows the Asterisk card X400P card to terminate analog telephone lines (POTS). Because of the modular design, a user can activate additional ports at any time with more FXS or FXO daughter cards. The FXO module passes all the call features any standard analog telephone line will support.

Asterisk card :FXS module

2009-03-13T01:26:00.000-07:00

Asterisk card :FXS module
FOR ASTERISK

fxs card FXS Module
The FXS module allows the Asterisk card X400P card to terminate analog telephones.Because of the modular design, a user can activate additional ports at any time with more FXS module daughter cards. The FXS module passes all the call features any standard analog telephone will support.
Note: To use the FXS module, you must provide 12V DC power to the Asterisk card X400P, either from the computer's power supply.

te110p - Wildcard TE110P ISDN PRI pci e1 card

2009-03-01T04:04:00.001-08:00

te110p - Wildcard TE110P ISDN PRI pci e1 card

The Wildcard te110p brings a high-performance, cost-effective, and flexible single span togglable T1/PRI or E1/PRI interface to the Digium line-up of telephony interface devices.
The TE110P is a single span, selectable E1 or T1 card that supports all the functionality of our quad T1/E1 card. That is you can select the E1 or T1 by a jumper.

This card supports both voice and data modes on its single span. For example, the card can support 12 channels dedicated to voice and 12 to data while passing all traffic through to the Asterisk Open Source PBX, which reliably routes the channels to their designated locations. This eliminates the need for an external router.

The te110p supports industry standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC, and Frame Relay data modes. The board drives both line-side and trunk-side interfaces, including call features.

By utilizing our TDMoE (TDM over Ethernet) technology, an exclusive Digium process, one can easily connect multiple PCs equipped with the TE110P and achieve voice quality on par with single PBX implementations. Scalability for this product is derived from adding multiple TE110Ps to each individual PC. Add addition cards as you need them for your expanding applications.

sip phone

2009-02-18T07:10:00.000-08:00

sip phone

sip phone is a voip phone that supports SIP protocol, which is one of the voip protocol.
sip is the most popular voip protocol in the world, comparing with h.323 , iax,mgcp , sip is more and more popular.
there are many sip phone manufacturer in the world, however, welcome to consider ChinaRoby ,which is a professional sip phone manufacturer.

SIP has the following characteristics:
Transport-independent, because SIP can be used with UDP, TCP, ATM & so on. Text-based, allowing for humans to read SIP messages

you may also see more informations from the website of chinaroby about sip protocol and sip phone.

VoIP links

2009-02-15T17:03:00.001-08:00

VoIP links

Asterisk Consultant.com Bringing Freelance Consultants and Customers together. Lots of VoIP information, about the hardware or software.
AsteriskGuru.com Forums and technology Support. Everything you need to get started with and maintain Asterisk PBX solution,click to join in.
Asterisk Indian Community Forum Asterisk Forums, Howtos, Tips...etc ,from India.
All about Voip in dutch What is voip? Why voip? How ? Everything about voip (in dutch). Welcome to visit the website.
Asterisk.org.ph A Philippine Based Asterisk Community, Forums for Howtos, Tips, Troubleshooting, Selling, Buying, etc.

more voip informations>>

voip phone supplier
voip provider
ip phone provider
fxo gateway

we updated the website about the oem voip phone page

2009-02-15T17:00:00.000-08:00

we are supplying with oem voip phone services, we updated the website .
If you need such service,welcome to visited my website to check.

how to customize my voip phone

2009-02-13T16:48:00.000-08:00

ChinaRoby.com is supplying with ip phone oem service and voip gateway oem services.
The OEM services including the items list below,
1) print some stickers or labels and put them on the ip phone or gateway
2) Print customer's logo on the ip phone or voip gateway by silk screen printing
3) customize gift box
4) re-design pcb of the ip phone or voip gateway
5) show the company name on the lcd of the ip phone
6)show the company name or logo in the firmware
7) other firmware special requirement.

IP Phone OEM

Printing client's logo/brand on ip phone
LCD display client's brand
Putting client's logo into firmware of ip phone
Customizing language
Printing text on the panel/keys/buttons in client's language
Customizing package of ip phone
Printing labels
Asterisk card OEM
Printing client's logo/brand on the Asterisk card
Customizing package of asterisk card
Printing labels
VoIP gateway OEM
Printing client's logo/brand on gateway
Putting client's logo into firmware of voip gateway
Customizing language
Printing text on the panel/keys/buttons in client's language
Customizing package of ip phone
Printing labels

what is the chipset of voip phone

2009-02-12T23:51:00.001-08:00

what is the chipset of the voip phones ?

1) PA1688
Pa1688 is the old voip phone chipset, people use them to produce voip phone several years ago,however, the chipset supplier stop supplying with the chipset anymore.
2) AR1688
AR1688 is the replacement of the PA1688 , however, it supports SIP only.
3) Infineon
It is the most popular voip phone chipset, people use them to produce many voip phones, it supports SIP + iax or H.323+SIP, you may select the protocols from them . It is with 2 RJ45 ports, with a Router built in , it supports 2 SIP accounts and 1 IAX2 account at the same time.
4) CM5000
CM5000 voip phones are also popular, however, it is expensive and I donot think it is a good choice.

What is a PBX ?

2009-02-12T23:47:00.000-08:00

What is a PBX ?
A PBX (Private Branch Exchange) is a switch station for telephone systems. It consists mainly of several branches of telephone systems and it switches connections to and from them, thereby linking phone lines.

Companies use a PBX for connecting all the internal phones to an external line. This way, they can lease only one line and have many people using it together, with each one having a phone at the desk with different number. Inside a PBX, you only need to dial three-digit numbers to make a call to another phone in the network. We often refer to this number as an extension.
Most people are familiar with PBXs because they've used them in an office environment, but what if you wanted to use one in small office or your house? Enter Asterisk , which is free and offers all this functionality right out of the box. Asterisk is a software implementation of a hardware PBX and can run on a variety of hardware platforms. The features and benefits of owning an Asterisk PBX are numerous, and seemingly only limited by the imagination of the person who sets up and uses an Asterisk PBX

ip pbx

What Are Some disadvantages of VoIP?

2009-02-12T05:31:00.000-08:00

What Are Some disadvantages of VoIP?

If you're considering replacing your traditional telephone service with Internet Voice, there are some possible differences:Some Internet Voice services don't work during power outages and the service provider may not offer backup power; It may be difficult for some Internet Voice services to seamlessly connect with the 911 dispatch center or identify the location of Internet Voice 911 callers; orThey may or may not offer white page listings.

ip phone manufacturer

List of SIP software

2009-02-11T16:28:00.001-08:00

List of SIP software
Free software and open-source licenseAsterisk - an open source SIP/IAX PBX The Open Source SIP PBX (also available in commercially supported form Pingtel) Axon PBX - Free SIP PBX (Windows) YATE - a professional open source SIP, H.323,IAX PBX and client. It works as a SIP-H.323 translator. (supported commercially by Null Team)
SIP communications products
Proprietary licenseIntTalk Enterprise VoIP solutions — softphone, SIP MCU and more. CommuniGate Pro IP Communications Platform - Carrier Class Internet Communications Platform Objectworld UC Server — Unified Communications for Windows. MessengerSDK — SIP based voice and video telephony SDK. 3CX Phone System for Windows — SIP server. Brekeke PBX and Brekeke SIP Server IVR Technologies, Inc. Talking SIP 3.0 — an integrated application, media and real-time billing server Vonexus — Microsoft-based IP Telephony Solution. Microsoft Office Live Communications Server Creacode NicIVR Advanced SIP IVR and application development platform Pactolus SIPware solutions — carrier-grade calling card, broadband telephony, conferencing, and custom applications Rostrvm Switchless — standards based call centre architecture using SIP. pbxnSIP SIP-based PBX for Windows Dialexia Dial-Office — SIP-based IP-PBX for both Windows and Linux. 3Com SIP PBX — Enterprise IP Telephony applications suite using SIP. Xeepe Hybrid SIP/ISDN Phone System for SOHO and SMB.
SIP servers
Free software and open-source licenseYATE Yet Another Telephony Engine - An H.323, SIP, IAX server and client for both Windows and Linux made for Enterprise environment. Mobicents - The Open Source JSLEE and SIP Server SIProxd SIP proxy - a proxy/masquerading daemon for the SIP protocol SIPX, for Linux. Asterisk (PBX) SIP IAX/MGCP/H323 and more OpenSER - GPL SIP Server focused on security and scalability OpenSBC - Reference implementation of Session Border Controller (including simple proxy and registrar) using the MPL licensed SIP Express Router (SER) - GPL SIP Server of iptel.org, focused on security and scalability YXA is SIP software written in Erlang CallWeaver - OpenSource telephony server

voip phone provider
t1 card
fxo ata
voip gateway supplier
voip gateway
sip ata

Proprietary licenseAvaya Communication Manager SIP Enablement Services BEA Systems WebLogic SIP ServerBrekeke SIP Server is available for Windows, Linux, Mac OS X and Solaris. It is free for personal and educational use. Creacode NicSC SIP Session Border Controller] High performance SIP Call Controller, Registrar and NAT Traversal solution. Eyeball SIP Proxy Server IVR Technologies, Inc. Talking SIP 3.0 - An integrated application, media and real-time billing server Nortel SIP Multimedia Communication Server 5200 Pactolus RapidFLEX Service Delivery Platform - Highly scalable SIP application server, media server, and service creation environment Paradial RealSIP Server Cisco SIP Proxy Server Pingtel Call Manager (based on SIP Foundry open source) Ubiquity SIP Application Server 3Com VCX IP Telephony Module: Back-to-Back User Agent SIP PBX Oracle Multimedia communication Engine (M2CE) Interactive Intelligence SIP Proxy, specialising in Load Balancing and Fault Tolerance CommuniGate Pro - universal communication server - SIP Proxy/Registrar, NAT Traversal/SBC, IVR, Conference, IM/Presence via SIMPLE/XMPP, E-Mail and GroupWare. Communology SIP Clients, specialising in mobile Clients and SIP testing Squire Technologies SIP H323 SBC Class 4 softswitch
SIP-capable firewallsSecure Computing's SnapGear firewall includes the SIProxd SIP proxy Intertex SIP Transparent Routers, Firewalls and ADSL modems - For broadband deployments and the SOHO market Ingate SIP Transparent Enterprise Firewalls and SIParators - Enabling global SIP communication on the enterprise LAN Secure Computing's Sidewinder 7 firewall includes a SIP proxy Balabit IT Security Zorp Professional firewall includes SIP proxy BorderWare SIPassure firewall translates TLS and SRTP
SIP clients
Free softwareEkiga, formerly known as GnomeMeeting, GPL KCall, using Qt libraries KPhone, using Qt libraries, GPL YATE - An H.323, SIP, IAX server and client for both Windows and GNU/Linux, license GPL with an embedded browser. OpenWengo - A SIP based softphone and IM client using the Qt libraries, available under the GPL PhoneGaim, based on Gaim. PJSUA, light SIP stack in C (designed for embedded system) with command line interface, GPL TudoMais, Java/Flash-based for Windows. Zap!' is cross platform, based on XULRunner, licensed under the MPL SIP Communicator, the Java VoIP and Instant Messaging client, runs on Windows, Linux and Mac OS X, LGPL Linphone, with a core/UI separation, the GUI is using GTK+ libraries MiniSIP, with a core/UI separation and encryption, alpha version for Nokia 770 Twinkle, using Qt libraries XMeeting is Mac OS X SIP/H.323 client, based on OpenH323 libraries, licensed under the MPL wxCommunicator, based on SIPXtapi and wxWidgets libraries, GPL
Proprietary freewareMicrosoft Windows Messenger (not to be confused with MSN Messenger or Windows Live Messenger) SJphone runs on Windows, Mac OS X, Linux and PocketPC. Tivi runs on Windows NCH Swift Sound Express Talk, GSM, uLaw, ALaw and PCM codecs 3CX Phone for Windows - a free SIP client for Windows. Mindspring, for Windows. Voice, chatting, and file-sharing capabilities. RadiusCat for Windows, SIP RADIUS AAA Billing Software CounterPath X-Lite for Windows, Mac OS X and Linux The Cornfed SIP User Agent for Linux by Cornfed Systems is available with Command-Line Interface (CLI) and Gnome GUI clients. Gizmo Project for Windows, Mac OS X and Linux, primarily used for the SIPphone network SightSpeed for Windows and Mac OS X. G.711, iLBC, Speex and H.263 codecs Phoner for Windows, G.711, iLBC, G.726, GSM and Speex codecs PhonerLite for Windows, G.711, iLBC, G.726, GSM and Speex codecs fring enables users to make mVoIP (mobile VoIP) calls with any SIP provider even from non-SIP enabled phones
Commercial proprietary softwareAGEphone runs on Windows, PocketPC and Windows Mobile. Hampton Software's Articulation runs on Palm OS 5 handhelds including Clie and Treo. is-phone integrated softphone, Session Initiation Protocol (SIP) based, PBX independent Movial PC, Symbian and Windows CE Clients Pingtel, Java-based for Windows. VeriCall Edge for voice/video over IP calling runs on embedded Linux. Virbiage Cubix for Windows, Inter-Asterisk eXchange Protocol, G.711, iLBC, G.729, GSM and Speex codecs CounterPath X-Pro for Windows SightSpeed for Windows and Apple OSX G.711, iLBC, ADPCM and Speex codecs
SIP test tools
Proprietary licenseSFTF: SIP Forum User Agent Test Framework (open source; written by the SIP Forum). Codenomicon SIP Test Tool: Commercial SIP robustness and security test tool (next generation PROTOS SIP test tool [see open source test tools]) Mu Security: Commercial SIP-VoIP, RTSP-IPTV Triple Play security analysis & testing platform (automate existing 3rd party testing and scripts like Nessus and PROTOS Bandwidth.com offer a tool to test the SIP ports on your network SIPient Systems - SIPFlow: SIPFlow captures SIP traffic and draws callflows (ladder diagrams) for each transaction.
Free software and open-source licenseSIPp - an Open Source SIP test tool (functional and performance tests). SIPsak - a command line tool which can send simple requests to a SIP server (Open Source). PROTOS c07-SIP test suite: PROTOS c07-SIP robustness and security test suite (initial results of running the test suite are documented in US/CERT advisory). pjSIP-perf - an Open Source SIP transaction/call performance measurement tool. Asteroid - is a SIP Denial of Service testing tool. See MITRE CVE-2006-5444 and CVE-2006-5445
SIP protocol stacks
HelloSoft Ltd. Voip protocol for wlan, cellular & gateway, including SIP Stack, Voice Media Processing, Jitter Buffering, Call Control Manager. VeriCall Edge integrates its own SIP stack with all of the required media processing, packet handling and call control functionality needed to develop voice/video over IP end-user devices. amSIP SIP/audio/video SDK based on oSIP/eXoSIP2/oRTP/mediastreamer2 for softphone/server developments. Proprietary licenseRADVISION SIP Developer Suite for developing carrier-grade IMS Compliant SIP Client and Server applications on any platform. The Cornfed SIP User Agent System Developer's Kit (SDK) for Linux by Cornfed Systems is available for integrating fully integrated softphones into OEM product designs. Ulticom Signalware? SIP is a carrier-grade SIP stack designed for operator core networks SIP .NET: SIP client API for .NET Framework and .NET Compact Framework Ageet Corporation offers the small and fast microSIP stack
Free software and open-source licenseOpen SIP Stack - Free Software Open source C++ SIP stack released under MPL 1.0 license. Designed for scalability and reliability. GNU oSIP - Free Software Open source (also available under commercial license) C SIP stack released under the LGPL. Small footprint. Support Linux/Win32/MacOSX/WM5.0/VxWorks. Designed for a basic library in order to develop SIP client & gateway. eXoSIP - Free Software Open source (also available under commercial license) C SIP stack based on oSIP and released under the GPL. Support Linux/Win32/MacOSX/WM5.0/VxWorks. Designed for a basic library in order to develop SIP client. ReSIProcate: Free Software SIP stack Sofia-SIP - a Free Software SIP user agent library licensed under the LGPL. It can be utilized conditionally in the IMS network. JAIN-SIP JAVA SIP Stack - Public domain implementation of RFC 3261 and extensions. Reference Implementation for JSR 32 javax.SIP package. PJSIP: Open Source SIP stack in C. Tiny footprint, ultra portable, and very fast. Yass - Yet Another SIP stack and is the YATE SIP stack written in C++. It can be used under Windows or Linux in a single or multithread environment for state-full software like clients, servers and proxies. It contains a transaction engine and of course a SIP not very efficient but safe parser.

What Are Some Advantages of VoIP?

2009-02-11T16:26:00.000-08:00

What Are Some Advantages of VoIP?
Because Internet Voice is digital, it may offer features and services that are not available with a traditional phone. If you have a broadband internet connection, you need not maintain and pay the additional cost for a line just to make telephone calls.With many Internet Voice plans you can talk for as long as you want with any person in the world (the requirement is that the other person has an Internet connection). You can also talk with many people at the same time without any additional cost.
What Are Some disadvantages of VoIP?
If you're considering replacing your traditional telephone service with Internet Voice, there are some possible differences:Some Internet Voice services don't work during power outages and the service provider may not offer backup power; It may be difficult for some Internet Voice services to seamlessly connect with the 911 dispatch center or identify the location of Internet Voice 911 callers; orThey may or may not offer white page listings.

FXO vs FXS

2009-02-11T00:39:00.000-08:00

FXO vs FXS

FXO
In telecommunications, a Foreign Exchange Office, or FXO, is a telephone signaling interface that receives POTS, or "plain old telephone service". It generates the on-hook and off-hook indicators used to signal a loop closure at the FXS's end of the circuit. Analog telephone handsets, fax machines and (analogue) modems are FXO devices, though the term is rarely used except in connection with Foreign exchange service (FX).FXO interfaces are also available for computers and networking equipment, to allow these to interact directly with POTS systems. These are commonly found in devices acting as gateways between Voice over Internet Protocol (VoIP) systems and the public switched telephone network (PSTN).

FXS
In telephony, a Foreign eXchange Station, or FXS, is a telephone interface which supplies battery power, provides dialtone, and generates ringing voltage. A device that connects to such an interface contains an Foreign exchange office (FXO) interface and could be a standard analog telephone or a private branch exchange (PBX) to receive telephone service.Any telephone exchange is an example of an FXS, as is the telephone jack on the wall, though the term is rarely applied except in connection with foreign exchange service.An FXS interface utilizes an FXO protocol to detect when the terminating device (telephone) goes on-hook or off-hook, and can send and receive voice signals.

from http://www.chinaroby.com

What is VoIP ?

2009-02-11T00:38:00.000-08:00

What is VoIP ?
VoIP allows you to make calls using an IP network, over a data network like the Internet. VoIP converts the voice signal from your telephone into a digital signal that travels over the internet then converts it back at the other end so you can speak to anyone with a regular phone number. When placing a VoIP call using a phone with an adapter, you'll hear a dial tone and dial just as you always have. VoIP may also allow you to make a call directly from a computer using a conventional telephone or a microphone.
What Kind of VoIP Equipment Do I Need?
A broadband (high speed Internet) connection is required. This can be through a cable modem, or high speed services such as DSL or a local area network. You can hook up an inexpensive microphone to your computer and send your voice through a cable modem or connect a phone directly to a telephone adaptor.
Asterisk Hardware
Asterisk is an open source IP PABX that runs on the Linux operating system. It is an extremely powerful product capable of the most advanced PABX functions including voice-mail, conference calls, trunking, hunt groups, and much more. Being IP based, users connecting to Asterisk can be geographically distributed, yet still connected to the same business grade PABX. In order to connect telephones directly to an Asterisk box, or to connect Asterisk to the PSTN, interconnect cards are required.
VoIP Phone
VoIP phone looks and feels like standard desk-phones, but connect to the Internet (either via an Ethernet connection or WiFi) rather than the phone line socket in wall. They are independent from your computer, meaning you do not have to have your computer on to make and receive calls. A wide range of IP phones are available from the ultra-basic phones, to advanced phones with multi-line support, high quality speaker-phone, power over Ethernet, and more!
VoIP gateway ATA
VoIP gateway ATA allows you to use any standard analogue telephone as an IP phone. Simply plug your telephone (cordless phone if you like) into the RJ-11 port on the ATA, connect your ATA to the Internet by plugging it into a spare Ethernet socket on your router, and your old telephone is transformed into an Internet telephone. Once set up, they are independent from your computer, meaning you do not have to have your computer on to make and receive calls. Many analogue telephone adapters also have a built in router, providing some advantages over the standard ATAs. These are lited in the router category.
How Can I Place a VoIP Call?
Depending on the service, one way to place a VoIP call is to pick up your phone and dial the number, using an adaptor that connects to your existing high-speed Internet connection. The call goes through your local telephone company to a VoIP provider. The phone call goes over the Internet to the called party's local telephone company for the completion of the call. Another way is to utilize a microphone headset plugged into your computer. The number is placed using the keyboard and is routed through your cable modem.
If I have Internet Voice service, who can I call?
Depending upon your service, you might be limited only to other subscribers to the service, or you may be able to call any phone number, anywhere in the world. The call can be made to a local number, a mobile phone, to a long distance number, or an international number. You may even utilize the service to speak with more than one person at a time. The person you are calling does not need any special equipment, just a phone.
What Are Some Advantages of VoIP?
Because Internet Voice is digital, it may offer features and services that are not available with a traditional phone. If you have a broadband internet connection, you need not maintain and pay the additional cost for a line just to make telephone calls.With many Internet Voice plans you can talk for as long as you want with any person in the world (the requirement is that the other person has an Internet connection). You can also talk with many people at the same time without any additional cost.
What Are Some disadvantages of VoIP?
If you're considering replacing your traditional telephone service with Internet Voice, there are some possible differences:Some Internet Voice services don't work during power outages and the service provider may not offer backup power; It may be difficult for some Internet Voice services to seamlessly connect with the 911 dispatch center or identify the location of Internet Voice 911 callers; orThey may or may not offer white page listings.
Does my Computer Have to be Turned on?
Not if you are making calls with a phone and adaptor or special VoIP phone , but your broadband Internet connection needs to be active. You can also use your computer while talking on the phone.Can I Take My Phone Adapter with me When I Travel?
You may be able to use your VoIP service wherever you travel as long as you have a high speed Internet connection available. In that case it would work the same as from your home or business.
What hardware is required For VoIP?
Here you will find a brief overview of various options for your VoIP service.
For more information only the actual products please check our VoIP Hardware Catalog for a detailed description of the most common VoIP hardware.
How Do I Know If I have a VoIP phone Call?
It will ring like any other call.
ADSL Modem/Routers with VoIP
"All-in-one" combo units are becoming very popular amongst home and small business users. They combine an ADSL modem (some are ADSL2 and ADSL2+ compatible), router, and built in voice over IP telephone adapter in the one unit. Some units also have multi-port Ethernet switches, 802.11g wirless access points, and advanced firewall and VPN functions. A single unit covering multiple needs means less hardware collecting dust, and fewer used power points! Additionally, a combination unit ensures that you will not have any NAT or firewall issues as the router inherently 'knows' about the VoIP traffic. These units automatically prioritise traffic to ensure call quality is never compromised by other Internet activity. Plain routers (without the ADSL modem component) are also available.
PC Headsets
These headsets can be used with any voice over IP application on your computer. In fact, they can be used for any application that requires a microphone and/or speakers! We have sourced only the most comfortable headsets that are suitable for use with Internet telephony. Some of our headsets plug into a spare USB port and act as a separate audio device for your computer, whilst others simply plug into your sound card, allowing them to be used on any operating system without requiring drivers to be installed. We have a selection of both stereo and mono headsets (which some people find more natural for telephone conversations).PC Handsets, Speakerphones & Telephone Adapters
'Telephone-like' handsets connect to your computer via the sound card or a USB port and work together with a softphone to provide you with a complete VoIP experience. We also have USB based hands-free speaker-phones, and telephone adapters which are similar to ATAs, but connect to your computer rather than your router. All of these devices either work without any special software or come with free software. Many work on MacOS and Linux systems as well as Windows.
Business Grade Media GatewaysMedia gateways are suitable for businesses requiring a number of voice over IP lines - either as an independent option or hooked up to an existing PABX. These gateways do the job of multiple ATAs in the one unit as well as additional routing functions. The Octtel series of gateways have a built in phonebook manager allowing you to manage a group of devices without any external SIP server. This phonebook feature can also be used in conjunction with a SIP proxy if desired.
Gateways with FXO ports are not yet A-ticked and may only be used for internal testing purposes.Routers with VoIP
These products are 'all-in-one' devices that combine a router and a voice over IP adapter in one. By using a single device for your router and VoIP hardware, not only do you have one less piece of hardware to deal with (and one less powerpoint that is occupied), you are guaranteed that you will not have any NAT or firewall issues as the router inherently 'knows' about the VoIP traffic. Additionally, these units automatically prioritise traffic to ensure call quality is never compromised by other Internet activity. Routers with built in ADSL modems are also available.Is there a difference between making a Local Call and a Long Distance Call?
Some VoIP providers offer their services for free, normally only for calls to other subscribers to the service. Your VoIP provider may permit you to select an area code different from the area in which you live. This means you may not incur long distance charges if you call a number in your area code regardless of geography. It also means that people who call you may incur long distance charges depending on their area code and service.Some VoIP providers charge for a long distance call to a number outside your calling area, similar to existing, traditional wireline telephone service. Other VoIP providers permit you to call anywhere at a flat rate for a fixed number of minutes.
from www.chinaroby.com

ISDN BRI and ISDN PRI Services

2009-02-11T00:36:00.000-08:00

ISDN BRI and ISDN PRI Services

There are two main services associated with ISDN – Basic Rate Interface (BRI) and Primary Rate Interface (PRI). Both services consist of multiple channels over which data can be sent (known as B channels) and also include a signaling channel (the D channel). The D channel is used for control and signaling purposes, such as setting and tearing down ISDN call. Referred to as “out-of-band” signaling, this method ensures that other ISDN calls do not interfere with existing connections, that bandwidth on the B channels is reserved for data only, and ultimately results in quicker call setup and teardown.

Basic Rate Interface (BRI) ISDNBasic Rate Interface ISDN is made up of two 64 Kbps B channels that are used for sending and receiving data in full duplex, and one 16K D channel for signaling. In total, an ISDN BRI interface provides 144K of bandwidth (64+64+16). ISDN BRI is often referred to as 2B+D. Many Cisco router models include built-in BRI interfaces, but they can also be added using modular WAN interface cards.

Primary Rate Interface (PRI) ISDNFor companies with higher bandwidth requirements, ISDN Primary Rate Interface (PRI) service exists. In North America, PRI service consists of 23 64 Kbps B channels, and one 64Kbps D channel, for a total possible bandwidth of 1.544Mbps (T1 equivalent).In Europe, PRI service consists of 30 B channels and 1 D channel, for a total bandwidth of 2.048Mbps (E1 equivalent). ISDN PRI interfaces are typically implemented as modular WAN interfaces on Cisco routers, although some models do include built-in PRI ports.

what is auto provision for a voip phone

2009-02-11T00:30:00.000-08:00

there is a very good feature called auto provisioning for a voip phone.
With this feature, you can get the latest firmware or confirguration file from a FTP or http server.
For example, you are a voip provider, and you have 1000 users, each user is using an ip phone , now you have a new firmware , how to upgrade all your ip phones ?
Of course you cannot upgrade them one by one , it wasts too much time, the best way is that you build a FTP server , then put the firmwares into the FTP server, then, all the ip phones can get the latest firmware when you boot the ip phones.
with which , you can update all the 1000 ip phones easily.
see the details from http://www.chinaroby.com/english/download/AutoProvisioning.pdf

how to know the bands of my gsm cell phone

2009-02-09T00:02:00.000-08:00

There are many gsm cell phone manufacturers in China, however, most of them are producing gsm cell phones based on MTK chipset.
If the cell phone is based on MTK chipset, then, you can know find the bands supported from a code.
It is *#3646633# .
press the keys list above , then, you can find the bands that the cell phone can work.
You can select the correct bands in your country.
You may also select auto , if so , the mobile phone can search the bands by itself.

gsm cell phone - dual sim

2009-02-08T17:10:00.001-08:00

Chinese gsm cell phones are full of the world.
they usually supports 4 bands ( gsm 850/900/1800/1900MHz) and they are unlocked, so , they can work all over the world.
Chinese gsm cell phones can usually support 2 sim cards,that is, they are with dual sim slots, you can insert 2 sim cards into the same mobile phones.
there are 2 type of dual sim,
one is called dual sim dual standby , the other is called dual sim single standby.
that is dual sim dual standby ?
dual sim dual standby is that , a mobile phone is with dual sim and dual sim can work at the same time.
dual sim single standby is that , a mobile phone with with dual sim , however, there are only one sim can work at the same time. you can switch from one to the other one.
And you may use forwarding function to forward a call from one to the other.
welcome to select chinese unlock gsm cell phone.

the security of voip

2009-02-08T01:35:00.000-08:00

what is voip ?
VoIP is that using the Internet Protocol network for voice transmission, which represents the internet protocol IP.
Through Internet , you can send e-mail, instant messaging, as well as tens of thousands of web pages transmitted to the PC or phone. Some people say it is the traditional telecommunications killer, some people say it is a revolutionary factor in international affairs. All in all a lot flatter. However, perhaps the use of this service in your time, perhaps you have a hacker to steal personal information even destroy your network. All affect the data network attacks are likely to affect the VoIP network, such as viruses, spam, illegal intrusion, DoS, hijacking of calls, eavesdropping, sniffing and other data. The only difference is that we are more willing to take some measures to protect other networks. For VoIP, rarely have any specific measures. In fact, only if we take some protective measures, the technology can be real success. Explore the following 25 kinds of ways to protect VoIP:
1, limiting all the VoIP data can only transfer to a VLAN on the Cisco recommends that voice and data were divided into VLAN, this will help to deal with in order of priority, voice and data. VLAN division also contributed to defense costs of fraud, DoS attacks, eavesdropping, hijacking communications. VLAN of the user's computer division to enable the formation of an effective closure of the circle, it will not allow any other computer access to their equipment, thus avoiding the computer attacks, VoIP network will be quite safe; even under attack, it will be lost to a minimum.
2, monitor and track the VoIP network communication mode Monitoring tools and intrusion detection systems can help users identify those VoIP network intrusion attempts. VoIP Log detailed observations can help find some irregular things, such as the inexplicable or the international telephone companies or organizations not linked to the basic international calls, multiple login attempts to crack the password, such as voice exploded.
3, the protection of VoIP server Efficiency measures must be taken to protect the security of the server in order to protect ourselves from internal or external intruders using sniffer technology to intercept the data. Because the VoIP telephone has a fixed IP address and MAC address, so an attacker which is easy to sneak into. Recommend that users limit the IP and MAC address, simply visit the VoIP system does not allow the super user interface, and SIP gateways prior to the establishment of another firewall, this will to some extent limit the intrusion of the network system.
4, the use of multiple encryption Only send encrypted data packets is not enough, we must all phone signal encryption. Of voice encryption will prevent the interception of voice were inserted into the user session. In this regard, SRTP protocol to encrypt-to-end communications, TLS to encrypt the entire communication process. Should be adopted at the gateway, network, host level to provide strong protection to support encrypted voice transmission.
5, the establishment of VoIP network redundancy mechanism Need to be ever ready may be a virus, DoS attacks, they may lead to paralysis of network systems. Construction to set up multi-layer nodes, gateways, servers, power and network call router system and with more than one Internet provider. Recurrent network system for each test to ensure that its work well, when the main service network at a standstill, a standby facility to quickly take over the work.
6, will be placed behind a firewall device The establishment of the separation of the firewall, so that the border through the VLAN communication is limited to the available agreements. In case the client is infected, this will prevent the spread of viruses, Trojans, proliferation to the server. The establishment of separation behind a firewall, system security strategy for the maintenance will become simple. When necessary, the firewall must be properly configured in order to open or close some ports.
7, regularly updated patch VoIP network security, will depend on the underlying operating system are also dependent on the operation of their applications. Maintain the operating system and VoIP application software patch to update the procedures for the defense or infectious malicious code is very important.
8, will be separated from the internal network and the Internet Telephone management systems and network systems at Internet direct access to outside is a good choice, voice services and other server under the domain phase separation and restrict their access.
9, will softphone phone (softphone) to minimize the use of VoIP soft phone terminal computer vulnerable to hacker attacks, even if it is located at the company behind a firewall, because such things are with the ordinary PC, VoIP software and a pair of headphones for use with. Moreover, the soft phone terminal does not separate voice and data, therefore, vulnerable to virus and worm attacks.

why I buy an asterisk card ?

2009-02-08T01:22:00.000-08:00

asterisk is an open source ip pbx software. You may download it free from their official website.
If you buy an asterisk card , you can build your own ip pbx solution in your office or home.
asterisk card is not must , you can test the asterisk pbx without the card.
however , since it is not expensive, why not buy one ?

about the voip market of the world

2009-02-06T19:32:00.000-08:00

voip business grows slow because of the economy crisis.
And it is not esay to make money from the voip recently.
However, voip asterisk business is better and better, it must be the best feild in VoIP ,because people would like to build an ip pbx based on asterisk for their own use.
Chinese asterisk cards are always low cost, so, I think chinese asterisk cards must full of the world in future.

unlock gsm cell phone

2009-01-30T22:28:00.000-08:00

unlock gsm cell phone ChinaRoby is supplying with unlock gsm cell phoneHere are the features,welcome to visit the website for details.

3.0" Large Screen
It is a Touch Screen PDA,has a 3.0 inch super sized LCD touch screen,which is very revolutionary and sensitive.

Dual SIM
You can insert 2 SIM cards simultaneously. Both can work at the same time.

Camera
2.0 Megapixel Camera with Auto Focus, which includes a video recorder and digital voice recorder.

MultimediaMusic player:
Support MP3 player,with mp3,midi,wav format.
Movie player:
support 3GP/MP4 movie player with full-screen ,support Fast Forward / suspension.Voice recorder,
Video recorder.
Games built in + downloadable

Ring Tone64 polyphonic,
MP3,
6 individual speakers

Network
GSM850/900/1800/1900, Unlocked to all networks.
Working for any SIM card &any Tele company,it is a unlock gsm cell phone.Simply place your sim card in and its activated.

Internet
WAP/ GPRS (provided from your country Telecom) You can browsed on internet by this phone.Plug and play in Windows XP, no drivers or software needed

Bluetooth2.0

high Speed Transfer

Memory
Built in 256MB , Expandable to 2GB MaxPersonal Assistant Calendar, To Do list, Alarm, World Clock,Calculator,Currency Converter,Health,Pedometer

Language
English(by default), French,Spanish,Arabic,Thai,Vietnamese,Russian
SMS
SMS/MMS/GPRS Function included,WAP/GPRS/MMS/SMS Enable
Standby Time :
Up to 150 hours
Talk Time : Up to 180~300 mins

Security Setup
SIM Lock , phone lock, Keypad Lock,Black List, Filter List
Phone book Memory phonebook support 300 cards record;
support caller ringing / caller video player

the unlock gsm cell phone can work all over the world.

网络电话

2009-01-30T21:12:00.000-08:00

支持IAX2,SIP 与大多数系统和设备兼容
网络电话
兼容Asterisk
DHCP,
带2 RJ45接口,支持路由功能
大屏幕LCD液晶显示
单/双以太网接口
内置H.323代理服务器
快速启动和H.245隧道
支持H.245和Q.931keypad带外DTMF传送方式
支持呼叫转接
呼叫转移
三方通话等高级功能.
使用Q.931协议发送和接收主叫号码(Caller ID)
支持通过IP地址呼叫，可直接呼叫语音网关呼叫。
使用关守时，支持E.164电话号码方式呼叫。
可在局域网内自动搜索关守
支持ITSP发行的预付费卡呼叫PSTN普通话机（如net2phone;eTalk）
支持PPPoE协议（ADSL，Cable Modem接入时使用）
符合ITU-T标准信令音（如拨号音，忙音等）和DTMF音生成与检测。
支持基于E.164编码的拨号规则
可储存112组快速拨号号码
可记录40个未接来电，已接来电和已拨电话的号码
可储存16条语音信息;
听筒音量、扬声器音量可独立调节
可通过标准网页浏览器（如IE6.0）和标准Telnet方式设置话机参数
可通过FTP方式升级程序
支持 G.723.15.3k/6.3k G.729, G.711A-Law、μ- Law语音编解码算法
动态语音检测；舒适噪声生成; 动态语音缓冲技术
符合G.165 16ms回声抵消
voip电话

ip pbx 与 Asterisk

2009-01-28T02:16:00.000-08:00

ip pbx

1)什么是IP PBX? 什么是Asterisk？
IP PBX顾名思义,是建立在VoIP上的PBX,IP PBX的最终目的是要替换现有的PBX系统。与传统的PBX不同,IP PBX开放能力和拓展能力都大幅度增强，创造出大量增值业务，真正满足用户需求。在商业推广中，各个厂家根据自身的技术架构，开发出很多功能各异的IP PBX，有在原有PBX上插一块以太网卡，就叫做IP PBX的，有TDM和IP两个核心的IP PBX，也有纯IP核心的，基于Intel服务器架构或者嵌入式的IP PBX，而Asterisk的初衷就是做这种纯IP核心的IP PBX。纯IP核心的IP PBX，可以有几种形式，最基本的就是一台服务器，装上Asterisk软件，加上自身开发的应用，这种IP PBX必须使用IP终端，如网络电话、小网关、中继网关、软电话等；比较流行的形式是一台服务器插各种板卡，这些板卡带s口、o口、E1/T1口，用来连接电话机、局端放出的模拟或者数字线路，IP终端和普通终端都可以接入。Asterisk是一个IP PBX的开源代码。IP PBX的开源代码很多，但Asterisk是影响力最大的一个。既然是开源的,就可以在Asterisk上进行修改，可以根据客户的需要修改很多内容，比如界面、数据库、数据结构、协议栈等，让它变成客户想要的产品，或者利用这个过程提供服务。不同于GATEKEEPER或者SIP Server，Asterisk要做一个IP PBX。

2）Asterisk能做什么？Asterisk(英文意思是“星号”),最初的设想是,任何和电话系统有关的功能都能实现。但是实际上，并不是所有和电话有关的功能靠Asterisk实现都是最好的，这个要根据自身的需求，在Asterisk上定制开发，才会真正达到客户所要的目标。Asterisk最擅长的就是做一个IP PBX，当然有人用来做VoiceMail服务器、IVR服务器、会议服务器、中继网关、SIP SERVER、发卡系统等等。另外，Asterisk也有H323的支持能力，但是客户不能把它的H323当作GK，而只能作为一个支持H323的端点设备。Asterisk是动态的，它不断推出新的版本，比如T38的支持能力，可能在不久的将来，就有版本完全实现。

3）如何学习Asterisk？在学习Asterisk之前，客户必须了解互联网和通信网两方面的知识。客户必须知道一个普通电话是如何打通的，也必须知道一个VOIP网络是如何架构的。Asterisk本身就是一个开源的代码，具有很多分支，每个人看到它的角度都可能是不完全一致的，因此最好的学习方法就是根据客户的需求自己摸索。

4）目前用Asterisk开发的产品都有哪些？商用情况如何？目前市场上流行的IP PBX系统基本上都是从Asterisk修改而来.人们通过修改Asterisk的代码,满足不同的需求,从而生产出功能各异,各具特点的IP PBX.比如Jajah，Voipdiscount等等，都是基于Asterisk开发的。Asterisk的SIP协议栈的兼容性应该经过大量实践的检验，是相对成熟的。无论是在国外还是在国内,这样的IP PBX是很多的。

2009-01-28T02:09:00.000-08:00

网络电话

支持IAX2,SIP 与大多数系统和设备兼容
大屏幕LCD液晶显示
支持Asterisk
带路由功能
快速启动和H.245隧道
支持H.245和Q.931keypad带外DTMF传送方式
使用Q.931协议发送和接收主叫号码(Caller ID)
内置H.323代理服务器
单/双以太网接口
使用关守时，支持E.164电话号码方式呼叫。
可在局域网内自动搜索关守
支持ITSP发行的预付费卡呼叫PSTN普通话机（如net2phone;eTalk）
支持PPPoE协议（ADSL，Cable Modem接入时使用）
可通过标准网页浏览器（如IE6.0）和标准Telnet方式设置话机参数
支持通过IP地址呼叫，可直接呼叫语音网关呼叫。
支持 G.723.15.3k/6.3k G.729, G.711A-Law、μ- Law语音编解码算法
可通过FTP方式升级程序
支持DHCP自动分配IP地址等参数
符合ITU-T标准信令音（如拨号音，忙音等）和DTMF音生成与检测。
符合G.165 16ms回声抵消
动态语音检测；舒适噪声生成; 动态语音缓冲技术
支持基于E.164编码的拨号规则
可储存16条语音信息; 听筒音量、扬声器音量可独立调节
可储存112组快速拨号号码
可记录40个未接来电，已接来电和已拨电话的号码
宽带电话 voip phone

usb phone is for sale

2009-01-28T01:59:00.000-08:00

usb phone
skype phone
usb handset

USB phone with LCD or without LCD display
CE, FCC certified.
Caller ID function of VoIP calls .
Acoustically isolated speaker and microphone to eliminate echo.
No external power required
Customized hand phone casing designs available
Complies with skype
PC-to-PC and PC-to-phone operation Driver built-in
USB1.1 compatible, no need sound card.

Package:

Skype Phone *1
Color: White/Black
Installation CD (Driver and Manual )

System Requirements

PC running Windows 98/ 2000/ XP, 128MB RAM, 105MB disk space available for installation, and Internet Connection (any broadband: cable, DSL,etc.) 10MB or more hard disk spacek
P2 with 400 MHz CPU or better with USB interface .

180mm*120mm*35mm (L × W × H)

G.W. : 0.13
N.W. : 0.078 kg

ip phone with PoE

2009-01-28T01:42:00.000-08:00

ip phone
low cost voip phone
sip phone

Key features:
SIP and IAX2 support.
or SIP and H.323 support.
Support two sip accounts running at the same time.
with Router.
DHCP server and client.
Support PPPoE, (for ADSL, xDSL).
Support major G7.xxx Codec.
VAD,CNG.G.165 compliant 16ms
echo cancellation.
E.164 dial plan and customized dial rules.
Support hotline.
Call ID displayDND(Do Not Disturb),
Black List,Limit List Phone book,
maximum 100 entries.
POE (optional)
Call Transfer,
Call Forward,
3-way conference calls

--------
We are looking for distributors for our VoIP products, welcome to resell them !

How to build and customize an IP PBX with Asterisk ?

2008-12-11T19:04:00.000-08:00

How to build and customize an IP PBX with Asterisk ?
This article demonstrates how 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.

What is a PBX ? why set one up?
A PBX (Private Branch Exchange) is a switch station for telephone systems. It consists mainly of several branches of telephone systems and it switches connections to and from them, thereby linking phone lines.

Companies use a PBX for connecting all their internal phones to an external line. This way, they can lease only one line and have many people using it, with each one having a phone at the desk with different number. The number is not in the same format as a phone number though, as it depends on the internal numbering. Inside a PBX, you only need to dial three-digit or four-digit numbers to make a call to another phone in the network. We often refer to this number as an extension.

Most people are familiar with PBXs because they've used them in an office environment, but what if you wanted to use one in small office or your house? Enter Asterisk.

The beauty of Asterisk (see also Asterisk.org) is that it is free and offers all this functionality right out of the box. Asterisk is a software implementation of a hardware PBX and can run on a variety of hardware platforms. The features and benefits of owning an Asterisk PBX are numerous, and seemingly only limited by the imagination of the person who sets up and uses an Asterisk PBX. (Click to see the features of Asterisk ).

Prepare for setting up your PBX
There are many ways of setting up an Asterisk PBX, with some being easier than others. The easiest manner in which to set up a fully functional Asterisk PBX is to download and install aprecompiled/tuned distribution such as Trixbox, PBX in a flash, Asterisk NOW, etc. Just take your pick of those. These are great if you don't care to learn about how your PBX operates or need to do a mindless "click and drool" install. The problem with precompiled asterisk distributions is, while they are simple, they are not as flexible and extensible as a home-brewed Asterisk installation from the source code. When you install one of these "distros", it is quite often that important decisions are already made for you, and you must work within the framework of the distribution to keep the box "happy". When you roll your own Asterisk from the sources, you get more say in what goes into the build and how it is customized/extended at the expense of a slightly more complicated planning and installation process. However, it's not as hard as you might think and it's quite worth it in the end to "roll your own".

Asterisk installed from the source code enables you to pick your choice of underlying OS (Debian, Ubuntu, Fedora, Mandrake, SuSE, etc) and also fine tune the installation for your OS environment and your hardware for best possible performance. In this article I will cover the steps needed to install Asterisk for the Ubuntu Linux operating system. This article assumes you know how to do a minimal command line only base system install of Ubuntu Linux. You certainly don't want to run a GUI on your Asterisk box as it would consume unnecessary resources.

Step 1: prepare a minimal Ubuntu machine, wiht a command line only install

Step 2: download source packages
After you've prepared your minimal Ubuntu machine, you will need to download a few source packages from Asterisk. Please perform the following steps on your command line as the root user. To set your root password, log in as your regular user account you created during installation and type the following commands:

sudo passwd root

when asked for your password enter your regular USER password, hit ENTER, followed by the desired root password and then hit ENTER, confirm the desired password once more and then you're done.

Logged in as the root user we'll download the needed packages by running the following script in your terminal screen. If you are familiar with Putty, you can also do this remotely via a windows desktop or SSH into the box as root from any other Linux/Mac machine.

NOTE: we will be using Asterisk 1.4 NOT 1.6 as 1.6 is the unstable release.

Source packages needed :

asterisk-1.4.21.2
zaptel-1.4.11
libpri-1.4.5
asterisk-addons-1.4.7
asterisk-gui

I have written a script file to simplify the process on an Ubuntu 8.04 system. Copy and paste the following code into a new file in a terminal window and save the file as "astsetup.sh". Then when you save the file, make it executable by running this command: chmod +x astsetup.sh.

——> BEGIN COPYING CODE TO NEW FILE
######## Prepare Build Environment

apt-get install build-essential speex libspeex-dev gsm-utils \
libgsm1-dev libgsmme-dev libgsm1 libgsm-tools \
libncurses5-dev ncurses-term linux-headers-2.6.24-19-generic \
libncurses5-dbg doxygen subversion

######## Download Asterisk Source Packages

wget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.21.2.tar.gz
wget http://downloads.digium.com/pub/zaptel/releases/zaptel-1.4.11.tar.gz
wget http://downloads.digium.com/pub/libpri/releases/libpri-1.4.5.tar.gz
wget http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.4.7.tar.gz
svn co http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui

######## Extract all packages for building

tar -zxvf asterisk-1.4.21.2.tar.gz && tar -zxvf zaptel-1.4.11.tar.gz \
&& tar -zxvf asterisk-addons-1.4.7.tar.gz && tar -zxvf libpri-1.4.5.tar.gz

######## Begin Building and Installing Source Packages

cd libpri-1.4.5
make
make install
cd ../zaptel-1.4.11
make clean
./configure
make menuselect
make install
make config
cd ../asterisk-1.4.21.2
make clean
./configure
make menuselect
make install
make samples
make progdocs
cd ../asterisk-addons-1.4.7
make clean
./configure
make menuselect
make install
make samples
cd ../asterisk-gui
./configure
make
make install
make samples
cd /root
mkdir packages
mv *.gz /root/packages/.
echo INSTALLATION COMPLETE PLEASE CONFIGURE WEB GUI

———-> END COPYING CODE TO NEW FILE

One final step remains in order to bring the GUI up and running. We must edit two files:

/etc/asterisk/http.conf and
etc/asterisk/manager.conf.

/etc/asterisk/http.conf:

[general]
enabled=yes
enablestatic=yes ; without this, you can only send AMI commands, not display
; html content
bindaddr=0.0.0.0 ; address you want the Asterisk HTTP server to respond on
bindport=9000 ; port you want the Asterisk HTTP server to respond on
prefix=asterisk ; will form part of the URI, similar to a directory name

/etc/asterisk/manager.conf:

[general]
enabled=yes ; you may already have AMI enabled if you are using it for other things
webenabled=yes ; this enables the interaction between the Asterisk web server and AMI
[admin] ; This is the manager username. You can name the user whatever you want
secret = password
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config

———–

That's it! Restart your PBX server and point your browser to:

http://ip-of-pbx:9000/asterisk/static/config/cfgbasic.html (where IP of PBX is the IP address of your PBX box)

You should then see the web GUI and should be able to logon and start creating extensions (users).

If you don't see the GUI after restarting your PBX, login as root and just start Asterisk by typing asterisk on the command line. You can also set it to start at system boot by adding it to your /etc/rc.local startup script. Also, if you have an account with a VOIP service provider, you can setup your PBX to be the "gateway" to your office/home with ease.

Asterisk is very flexible, scalable, and limitless in its scope of use, so there is no way I can tell you everything that can be done with this amazing piece of software. It's use is not just limited to communication and phone management and there are thousands of hacks that can be applied to Asterisk. Some of them are:

Home Automation using X10 addressable switches
Remote home/office alarm/lock control
Remote environmental/audio monitoring
Conference Rooms
E-mail reader (text to speech access to any email account)
Remote Dictation
Voice Recognition
Ability to run system commands by phone - Remote Control
Phones and Extension Types
Asterisk supports any standards based SIP or IAX compatible voip phone. Softphones and VOIP ATAs (Analog Telephone Adaptors) are also another way to interface with your Asterisk system. With an ATA you can turn your existing analog corded or cordless phones into VOIP phones.

Voip Service Providers/PSTN
Asterisk can connect with the world either directly with a voip service provider that allows a "byod" (Bring Your Own Device) plan, or directly with the PSTN (Public Switched Telephone Network) via an Asterisk card.

cheap asterisk card

2008-12-11T18:58:00.000-08:00

Cheap asterisk card

What is a PBX ?

Why you need setup an ip pbx based on my cheap asterisk card ?

There are many asterisk card vendors, however, my asterisk card is cheap but good .

A PBX (Private Branch Exchange) is a switch station for telephone systems. It consists mainly of several branches of telephone systems and it switches connections to and from them, thereby linking phone lines.
Companies use a PBX for connecting all their internal phones to an external line. This way, they can lease only one line and have many people using it, with each one having a phone at the desk with different number. The number is not in the same format as a phone number though, as it depends on the internal numbering. Inside a PBX, you only need to dial three-digit or four-digit numbers to make a call to another phone in the network. We often refer to this number as an extension.
Most people are familiar with PBXs because they've used them in an office environment, but what if you wanted to use one in small office or your house? Enter Asterisk.
The beauty of Asterisk (see also Asterisk.org) is that it is free and offers all this functionality right out of the box. Asterisk is a software implementation of a hardware PBX and can run on a variety of hardware platforms. The features and benefits of owning an Asterisk PBX are numerous, and seemingly only limited by the imagination of the person who sets up and uses an Asterisk PBX.

Cheap Asterisk card with 1 E1 port

2008-12-05T03:17:00.000-08:00

VoIP Asterisk PCI card RX-1E1
supports both telephony and data modes. RX-1E1 has one E1/PRI
interfaces supporting PRA signaling, HDLC, PPP and Frame Relay
data modes, and any combination thereof. With the VoIP Asterisk card RX-1E1, you can
easily build your IP-PBX or Carrier system .

Interfaces:

-1 X E1 interfaces

-PCI 5.0V interface

Features:

One PRI port 30 channels

Hardware DTMF detection

Conference Bridge

PCM bus connectors (PCM30, PCM64, PCM128)

4 dual-color LEDs (layer 1 state indicators)

Active channel switching (across multiple cards over the external PCM bus)

PRI ISDN protocol stack

Suitable for 3.3 volts and 5.0 volts 32 bit PCI 2.2 slots

Applications:

ISDN PRI PBX

ISDN least cost router

Voice over IP PRI gateways

VoIP integration of ISDN equipment

PBX to PBX trunking

IVR

Data Modes:

Cisco HDLC

HDLC

PPP

Multilink PPP

Frame Relay

Voice Modes:
-EuroISDN

-AT&R 4ESS

-Lucent 5E

-National ISDN2

IP Phone supports SIP and IAX2,welcome OEM

2008-11-23T18:29:00.000-08:00

IP Phone supports SIP and IAX2,welcome OEM !!

[img]http://www.chinaroby.com/english/images/PY-60_voip%20phone.jpg[/img]

->IP Phone with 2 RJ45 ports,
->supports SIP,IAX2 and H.323
->with Router built-in
->welcome OEM

[img]http://www.chinaroby.com/english/images/PY-90s_voip%20phone.jpg[/img]
See also http://www.chinaroby.com

Mr. Gery

Roby Technology Co., Limited

Email: info@chinaroby.com
Web : www.chinaroby.com
MSN : gery668@hotmail.com
Skype ID: gery668
Tel : +86-755-83843088

VoIP Asterisk card RX-4E1 - VoIP Asterisk PBX - Trixbox card

2008-11-20T19:27:00.000-08:00

VoIP Asterisk card RX-4E1 is a Quad-Span E1 half-length PCI card which supports both telephony and data modes. RX-4E1 has four E1/PRI interfaces supporting PRA signaling, HDLC, PPP and Frame Relay data modes, and any combination thereof. With the VoIP Asterisk card RX-4E1, you can easily build your IP-PBX or Carrier system with handling 124 channels/calls. Interfaces:-4 X E1 interfaces-PCI 5.0V interface Features:

1. Be used for VoIP/Call Center/PBX

2. Provide four E1/PRI interface for to 124 channels/calls

3. Supports standard telephony and data protocols. (Including RBS and PRI protocol families for voice and PPP, Cisco HDLC and Frame Relay data modes).

4. 100% compatible VoIP Asterisk PBX all features.

5. Support SS7 (through chan_ss7)

Applications:

1. PBX/IVR Services

2. VoIP Services

3. VoIP Gateways

4. FAX Server, FAX On Demand

5. Voice Mail Services

6. Call Centers

7. Prepaid Calling Card System

8. Callback Service

cheap voip asterisk card

VoIP Asterisk PBX -- PCI Asterisk card with 1 FXO modules

2008-11-20T19:24:00.000-08:00

VoIP Asterisk PBX -- PCI Asterisk card with 1 FXO modules

-end PBX/Voicemail platform. Scalable and Effective SOHO Solution,

Target Applications

-Small Office Home Office (SOHO) applications

-Gateway Termination to Analog Telephones

-Add Inexpensive Analog Phones to Existing PBXs

-Wireless Point-to-Point Applications between Asterisk Servers

Services and Features

-Caller ID and Call Waiting Caller ID

-ADSI Telephones -PCI Half-length Slot

-RJ-11C Connector

Module:

Motherboard: TDM400P

FXO module: 4,3,2,1,0 pcs (optional)

FXS module: 0,1,2,3,4 pcs (optional)

Cheap Asterisk card

VoIP Asterisk PBX -- PCI Asterisk card with 3 FXO modules

2008-11-20T19:21:00.000-08:00

VoIP Asterisk PBX -- PCI Asterisk card with 4 FXO modules

The wildcard tdm400p is a half-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. It’s a 4 Port PCI Card for Asterisk, TrixBox and other Open Source Telephony projects with 4 included FXO modules. These cards are fully compatible with all analog Digium and other similar analog cards and modules with no changes to the drivers. FXO modules are used to plug existing analog telephone lines into your phone system.With the Wildcard TDM400P hardware, Open Source Asterisk PBX software, and a standard PC, users can create a Small Office Home Office (SOHO) telephony environment which includes all the sophisticated features of a high
-end PBX/Voicemail platform. Scalable and Effective SOHO Solution,
Target Applications
-Small Office Home Office (SOHO) applications
-Gateway Termination to Analog Telephones
-Add Inexpensive Analog Phones to Existing PBXs
-Wireless Point-to-Point Applications between Asterisk Servers Services and Features
-Caller ID and Call Waiting Caller ID -ADSI Telephones
-PCI Half-length Slot -RJ-11C Connector
Module:
Motherboard: TDM400P
FXO module: 4,3,2,1,0 pcs (optional)
FXS module: 0,1,2,3,4 pcs (optional)
Cheap Asterisk card

VoIP Asterisk pbx -- PCI Asterisk card with 4 FXO modules

2008-11-20T19:16:00.000-08:00

VoIP Asterisk PBX -- PCI Asterisk card with 4 FXO modules

-end PBX/Voicemail platform. Scalable and Effective SOHO Solution,

Target Applications

-Small Office Home Office (SOHO) applications

-Gateway Termination to Analog Telephones

-Add Inexpensive Analog Phones to Existing PBXs

-Wireless Point-to-Point Applications between Asterisk Servers Services and Features

-Caller ID and Call Waiting Caller ID -ADSI Telephones

-PCI Half-length Slot -RJ-11C Connector

Module:

Motherboard: TDM400P

FXO module: 4,3,2,1,0 pcs (optional)

FXS module: 0,1,2,3,4 pcs (optional)

Cheap Asterisk card

VoIP Asterisk - wildcard tdm400p

2008-11-20T19:02:00.000-08:00

wildcard tdm400p

Model :TDM400P
FOR VoIP ASTERISK/Trixbox

The wildcard tdm400p is a half-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. It’s a 4 Port PCI Card for Asterisk, TrixBox and other Open Source Telephony projects with 4 included FXO modules. These cards are fully compatible with all analog Digium and other similar analog cards and modules with no changes to the drivers. FXO modules are used to plug existing analog telephone lines into your phone system.With the Wildcard TDM400P hardware, Open Source Asterisk PBX software, and a standard PC, users can create a Small Office Home Office (SOHO) telephony environment which includes all the sophisticated features of a high-end PBX/Voicemail platform. Scalable and Effective SOHO Solution, Target Applications -Small Office Home Office (SOHO)

applications

-Gateway Termination to Analog Telephones

-Add Inexpensive Analog Phones to Existing PBXs

-Wireless Point-to-Point Applications between Asterisk Servers Services and Features

-Caller ID and Call Waiting Caller ID

-ADSI Telephones

-PCI Half-length Slot

-RJ-11C Connector

Module:Motherboard: TDM400P

FXO module: 4,3,2,1,0 pcs (optional)

FXS module: 0,1,2,3,4 pcs (optional)

Cheap asterisk card