hello everyone,
After I make a search, I could only find the
Is there any body know where should I start? is there any basic?
Please give me some suggestion.
Are you talking about a general FAQ on a web site or a FAQ for this Usenet group?
If a FAQ for this Usenet group ther eare procedures in place within Usenet to follow.
Why not try miniSipServer? It is very easy to establish your own VOIP network with miniSipServer.
Its manual document gives out a simple demo voip network. You may be benefit from it.
miniSipServer can be downloaded from
Enjoy SIP!
By the way, I also suggest you to visit Voip-info website. Lots of VOIP informations can be found in this website. I think you can almost get everything about VOIP from this website.
Its URL is
Very good advise. I can find nearly all the products information over there. But I can't find some thing such as how to open a call shop, how to design the network, how to use FXO gateway. That will be what I like to write down.
Here is what I wrote in these days. Anybody give me some advises, or help me to revise it? I am very busy and tired to front so many clients to ask similar question. So I write it down in my blogger, to help others.
VOIP FAQ
There are many device IP phone, ATA Analogue Telephone adapter. Those devices are belong to terminal. There are also many device on the server side, such as Softswitch, Billing, IPPBX, E1 gateway, etc.
The kernel device is the softswitch or the IPPBX.
it likes what you do for your PC connected to internet. There are some internet connection method:
Cable modem, ADSL, Static IP, DHCP, WiFi, Dialup.
For Cable modem, ADSL, the users always are asked for input user name and password. They always use ppp.o.e protocol to login into internet. So if your voip terminal support PPP.O.E, you could input it there.
Diagram here
For Static IP access, you will get the ip address, gateway, netmask, dns from your internet provider. you should input it into your voip terminal. For DHCP access, it is very easy. Just make sure the terminal WiFi
Dialup is a narrow band internat access. It could only be used on IP phone or 1 port ATA.
Every terminal's regist information is nearly same. Whatever band it is, whatever VoIP protocol it is using, to make a basic phone call, you must have the following information: server IP or domain name, port, account, password. This information is related to the softswitch which is owned by the service provider. If you don't have it, please contact your service provider.
Before make a call, one thing you must know. The dialplan. The dialplan is a text which used to tell the softswitch how to route calls. Normally, you don't need to config it in your terminal. but you need to know how to dial. such as localcall, long distance call, international call. It depends on the service provider's setting. It is always different to the land line.
For enterprise use, they need more than make a cheap call. such as call transfer, call waiting, voice mail, etc. Especial, the enterprise needs a computer attendant to play welcome information. For these feature, it is belong to the server side. So if you want to use these service, please make sure your service provider provide it, or your server has open this feature. For the terminal, what you do is click some items. The most famous server is Asterisk, a free open source software.
For carrier users, when the end user can make call, it is just begin to provide service. You have to use the billing system to make a correct and real time bill. To develop some agent, you need the billing interface for different user. To open a callshop, you need a callshop billing for the callshop's clients.
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