Jag och mitt skal – Introduktion till terminaler och shells

Denna video är ju rolig på sitt vis, många tar den säkert dock på allvar, och tror verkligen att terminaler och skal i *nix (Unix, GNU/Linux, BSD, jag kommer i fortsättningen skriva *nix) är något svårt och komplicerat att använda sig utav. Men så är det inte, terminalen, och skalet, är det Sanna sättet att använda en dator på, och är inget som är svårt, farligt, eller långsamt, tvärtom.
Continue reading “Jag och mitt skal – Introduktion till terminaler och shells”

How to set up a SIP trunk in the Asterisk PBX

In my previous article we configured Asterisk with some SIP-devices, and created a basic dialplan so that they could dial eachother. We also created two additional extensions for test purposes. This time I will show you how to configure a SIP trunk, and add extensions in the dialplan so that the telephones can dial out through the trunk. And if you also have a telephone number (DID) associated with the trunk, for others to be able to dial your phones, through your Asterisk PBX.

What is a SIP trunk?

A SIP trunk is often defined using many buzz- and marketing words throughout the web, but, what it basically is, is a two-way connection to a VOIP-provider, that routes the calls you send to it, out on the PSTN for you, and charges you for the calls you make. If you also have a DID (Direct Inward Dialing) number at the provider, calls made to you are forwarded to your Asterisk PBX, then you switch the calls as you see fit. Through a trunk, many calls can be sent, the limit is only your bandwidth and computer resources at the machine where your Asterisk runs, unless your VOIP-provider, or you for that matter, limit the number of calls in some way (by configuring the PBX at either end of the trunk), that are allowed to go through it. Continue reading “How to set up a SIP trunk in the Asterisk PBX”