Hello, I’m very new to Arterisk and have following question
It is possible to build a system that will automatically call multiple users at the same time and each of them will hear a shot message that already been created and save in the system?
If it possible
should I use a land lines to call users or the VoIP? which way is cheaper?
What hardware that I need?
What is the maximum number of simultaneous calls that asterisk support?
Thank you
Yes, it is possible. You can create a record in your system and implement a pager/intercom to execute auto"magically" at some hour and every call phone will auto-answer and hear the messages -only apply to IP-Phones-
Depends the price and the plan from your provider, in my country is cheaper a VoIP carrier than land line or E1.
A server for run asterisk, if you will use E1 lines or analog lines a TDM card -openvox,digium or sangoma-, IP-Phones. This just for start.
Depends on your server features, per example: CPU Dual Core 1.8ghz, 1GB DDR2 667 Mhz, can support 40 simultaneous calls
Thank you for your reply that help me a lot 
I have some more questions regard the simultaneous calls, could you please help me out.
1 is the number of simultaneous calls depended as well on the bandwidth and static ip?
2 if I use the same machine but one machine connect to T1 another to normal DSL, is the number of simultaneous calls will be different? or it only depended on the machine power?
3 If I want to make 40 simultaneous calls do I need to ahve 40 different static IP? or I only need one?
4 in the other hand, If I want to use land line to make 40 simultaneous calls do I need 40 different land line number or I only need one?
Thanks again
Yes, but even depends on the CPU performance. When you generate a call this call use a codec, generally G711, but you can define your preferred codec like G729, this codec is more used when you use a Internet route to make your call. Every codec has a bandwidth consumption so if you have a internet dedicated of 2Mb you can generate 63 simultaneous call.
This is by: (2048 - 3% ) / 31.2, the 3% is the headers more or less that you consume in every call, and the 31.2 is the consumption for a call with g729 codec. Obviously you need a SIP provider.
When we talk about T1, here the maximums calls you can generate are the maximums channels in your T1 for the DSL applies the above answer. The CPU help to translate between codecs, so the performance will decreased if you use differents codecs in your call.
No, only one dynamic or static IP, but one sip provider.
Yes, you will need 40 differents land lines, for that number is better a E1,T1 or sip provider.
Thanks for your replay again
sorry I’m totally new to this so please bear with me
so basically I have 2 option now T1 or Sip provider + dsl
for SIP provider + dsl
1 I need to pay the sip provider per minute to make a call
2 the simultaneous calls will be depended on Hardware and Bandwidth
for T1
1 I need to pay for T1 service, I don’t need to pay per minute call
2 simultaneous calls will depended on number of channels in T1 typically around 23 channels
is that correct? and it is hard to switch between the 2 methods?
Thanks again
Nop, asterisk can support both in the same server.
Generally you will pay a monthly rent and pay for call not for minute, -again i my country is cheaper than E1-
Yes, and yes. If you pay for a codec g729 you can place more calls.
[quote]for T1
1 I need to pay for T1 service, I don’t need to pay per minute call
2 simultaneous calls will depended on number of channels in T1 typically around 23 channels[/quote]
Yes.
You can have both in the same server even add land line to your server, so you can create dialplan to generetae outgoings calls from your preferred trunk.
The CPU make the translations between codecs so you can call with g711 -maybe from one public phone- and talk with sip g729.
Thank you so much navaismo thanks for clear thing up for me
Thanks again
You are welcome.