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5

The best is to block all the ports (iptables default to DROP) and open only what you need (probably port 5060 at least for your asterisk server). If you need to administer asterisk from outside, port 80 needs to be open if you administer it from a web browser or port 22 if you use ssh. Update: For Asterisk, I don't think you need other ports open and even ...


3

Yes, there is, a PCI fax modem. or for laptops, a USB fax modem Then any good old "Caller ID" software will do what you're looking for.


1

A pbx is short for Private Brance Exchange. This means, it acts like a phone system you see in most medium to larger companies. You can transfer calls between extensions, provide voice mail, conference multiple lines together, even connect two different PBX's. These are private though, and do not connect to the public telephone service unless you either ...


1

You will have to link the two PBX's. You will not be able to just plug in another PBX and have it work. You will also need a switch for the other phones and power adapters if the switch isn't POE.


1

SPA942s should automatically re-register based on the option in the web interface under Admin -> Advanced -> Ext(1-4) -> Proxy and Registration -> Register Expires (in seconds). A successive registration will overwrite a previous one. As long as the phone is set to send a registration within the interval of time that the PBX will hold the registration, it ...


1

I solved this by doing the following: Download the CSV format from Newer Elastix by going to the Batch Extension tab. Download the CSV file from your old Elastix by going to the Batch Extension tab. If you diff both files you will find that there is an extra header by the name of "Tech" in the Newer Elastix CSV and the value is "Sip" (which you need to ...


1

Ideally you want to know the scope of the phone network but as your network admin is on leave it's time for some trial and error 1st IP 10.22.0.253 subnetmask 255.255.255.0 This will allow you to access the 10.22.0.* range 2nd IP 10.22.0.253 subnetmask 255.255.0.0 This will allow you to access the 10.22.*.* range Test on both to see if ...


1

That depends how you're going to provide connectivity to your Asterisk box. If you plan to use your single analog phone line then you will be limited to its capabilities. For example, you can't do caller ID on the Asterisk box if your telco doesn't provide it. If you're willing to get a SIP trunk (from a provider such as voip.ms or sipstation.com) you ...


1

It turns out that Database named as asterisk was needed along with asteriskcdrdb but wasn't mentioned in the tutorial. Creating the database and all tables in it fixed the issue. Cause: Some SQL scripts in the updates enforce usage of asterisk database, even if you override the database in initial settings. To resolve previous errors (which, BTW, also ...


1

You can buy an analog card like this one from Digium: http://www.digium.com/en/products/telephony-cards/analog/4-port (I assume you have a standard analog line. You can also buy ISDN cards). Install the card on your PC then you can use the PSTN AND your SIP provider as required. you can setup a dialling plan to cater for both. eg dial 9 prefix for ...



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