[asterisk-users] About voip gateway

[Kevin Larsen]

> I want to create a voip service, I do not know much about it, but 
> the first thing I want to know if more than one client can make a 
> call at the same time through internet to the PSTN, and what gateway
> should I use for this.
 
I think the first recommendation any of us will have is to research all 
you can as there are a lot of mistakes to be made in the telephony world 
and some of them can be expensive and/or dangerous. The kinds of questions 
you are asking are not bad to ask, but they do place you squarely at a 
beginner level.

It is hard to answer your questions without having further information. 
What are you trying to accomplish with this system? Do you need to carry 
more than one call? What types of phone service are available where this 
will be installed?

For example, a single POTS line will allow you one call in or out of the 
PSTN. This is not a limitation of Asterisk, this is a limitation on how 
POTS lines work. A PRI style connection (E1 or T1 depending on location) 
will allow many more (over 20 calls at once). A SIP trunk is only limited 
by the number of lines your trunking provider allows and the bandwidth of 
your internet connection. The gateway you would want to use will depend 
entirely on what type of connection to the PSTN you are using. A lot of 
manufacturers make hardware compatible with Asterisk for physical 
connections to the PSTN and a SIP trunk just requires an internet 
connection of sufficiently high bandwidth, low latency and a reasonably 
stable path to the SIP provider.

Without knowing more about what you are aiming to do, it is hard for 
anyone to give you any specific help. You were earlier asked for a 
specific example of what you wish to accomplish. Please provide that and 
you will get more people responding.
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