Asterix - how to organize a mini-PBX?

Question

Asterisk

Good time of the day!
In office for the sake of experiment I want to organize automatic telephone exchange mini. Asterix, I would like more, there are many reasons for this.
There is 1 analog number, I need to spread it to 3-4 phones (ip), I need a call waiting, and a greeting with a choice of department, and of course, if the number is busy, then everyone is busy (and not so that they shout into the phone "Vasya hang up pipe, otherwise an urgent call"), there is a server (not very big, but it will do).
I need to find out what boards are needed to convert analog to ip, what ip phones would you recommend, on which platform should asterix be installed? How to do it? And then it is planned to expand analog lines.

Answer 1

Asterisk is installed on linux, debian or ubuntu is fine. The functionality that you describe is implemented without problems. Analogue conversion requires FXS/FXO boards or gateways.
Good boards are produced by Digium, for example , here you can choose a complete set by bus type (PCI / PCI-Express) and the number of FXO (external) and FXS (internal) analog lines. In your case, if you do not use a fax, then AEX801E will do, it can then be expanded using modules for 1 or 4 FXS / FXO ports.
From IP phones - Cisco Linksys SPA5xx (good, but expensive), Siemens Gigaset with a radio handset, Fanvil, Grandstream and others are quite cheap, but with quality - as lucky.

Answer 2

The most important thing you MUST DO FIRST is turn off international calls on all lines and set an acceptable limit of outgoing calls per day (based on the amount of money).
You must understand that:
1) Telephony makes a specialist through losing money during the training phase
2) It is not uncommon to lose hundreds of thousands of dollars in one night
3) There are criminal organizations in the world that engage in telephone fraud by hacking into popular telephone systems.
As for asterisk now, it is possible to inject your own shell into it by picking up the password of any user to voice mail , which consist of 4 digits and more often they are "0000".
Then authorization takes place in / recordings (User Panel), and through an oversight of FreePBX programmers, it is possible, using the asterisk SHELL functions, to upload your file to the web server visibility area. Usually this is a file/command manager through which the asterisk configs are read and all sip passwords are found out, as well as the diaplan is read and once it is used.
If you want, I will provide consultations - free of charge - in a personal. I have been doing this for more than two years, my tools, my achievements, I provide hosting specifically for asterisk with technical support.

Answer 3

You can also use Linksys SPA3102 as IP <-> Analogue gateways, and if analog phones are connected, then on 2 lines, for example, Linksys PAP2T.

Answer 4

There may be problems with an analog external number. If possible, it is better to negotiate with the provider for sip, or change the provider.
It is not necessary to take boards and ip phones. You can also take gateways with FXS / FXO ports - the price will be cheaper than 3 times. And if you want exactly an ip phone, then spa-303 is worth its money, in our office there are such I did not use it myself, but dlink ip phones are not recommended.
PS The downside of the boards is that they are more difficult to use in virtualization - you have to throw the device into the VM. Gateways are easier.
PPS And still I recommend to be defined at once with faxes. To be or not to be. Faxes have strict requirements, especially if they go through G.711

Answer 5

Use VoIP gateway with port fxo.
Raise ATC on Asterisk. I recommend trying Kerio Operator first , it unfolds in 5 minutes, everything is set up easily, for understanding everything is simple. Next, connect a VoIP gateway with the fxo port as an interface and route incoming and outgoing calls. In the same operator, this is done very simply by drag & drop, no problem. Incoming calls are routed to an IVR extension. IVR will need to be thought out for your goals and ordered from professional announcers, for your 4 departments it will cost 2000 ± 500 - a penny in fact. Operator eats for 70 users so many resources , it spins on the host like a virtual machine. but Kerio is paid, there are free solutions like FreePBX . Good luck!

Answer 6

take a ready-made integrated distribution kit asterisknow / freepbx , configure it on a virtual machine with 512 memory and do not suffer