Hi, i have looked on wikipedia about SIP servers and have a very very rough idea of what it does… i will read more later today so i dont expect all my newbie questions to be answered. A appreciate nothing is more anyoning that a newbie how hasnt bothered reading!!
I would like to rig my tech support up with VOIP (no analog lines at the moment). As i understand it SIP servers will redirect calls, voice mail etc for me. I take it i would setup users addresses on the server such as sip:chris@mydomain.com and people would be able to call me on them from their sip enabled hardware?
As this is very new to me i am quite happy to download and install the server on my linux box, however on a scale of 1-10 how hard is it to get the server up and running? I am resonably-ish competent with linux administration, so i am more concerned about the application complexity. Is the server admin done via scripts, or by http or similair?
Are there any tutorials about that would be of use for setting up for a similair scenario?
Finally, say i do want to rig up to analog phone lines, do i need any special hardware or will a standard modem do?
SIP hardware (phones or softphones) can call either extensions or by name. Depends on the hardware. You’d have to check to see what they support. It might just be extensions, and then your SIP hardware would just call a number to reach your users.
The basic Asterisk application is setup exclusively with .conf files. There are “versions” like Asterisk @ home floating around that can do some changes through a web interface, but the web interface is still just writing .conf files. Scales of 1-10 are not easily assigned. If you’re good with.conf files, then it’s a 1, if you’re not, it’s an 11 or 12.
I’ve not seen any really useful tutorials. This is definately a “search and study” project. The Wiki and the Actual code itself are your best documents. Having said that though the Asterisk @ Home project is probably the most completely documented.
If you want to hook into analog lines, you should look into the Digium boards made to do exactly that. Look for TDM boards with FXO cards.
thanks for the prompt reply! config files are ok with me, dependent on docs! Either way ill take your sugestion and get digging!
A quicky regarding the dial by extension or dial by name - are there any standards which indicate these capabilities. I presume dial by name is better? For instance, on some phones i see implementations of “E.164 Number and DNS (ENUM, RFC2916)”, are there any which correspond to dial by name?
Next step for me is to get a couple of desk VoIP phones in - any recommendations? Im looking to spend ~$120 per phone for test purposes?
Standards for anything in VOIP are still being created. Some are largely centered around “de-facto” standards. (What is implemented the most.) IAX is one of those. There’s lots and lots of usage, but it’s not a real “standard” protocol.
Go with what’s easiest at first, and expand.
As far as equipment goes, try some of the softphones first. They’re free.
SJphone and X-Lite are the two that most people seem to mention.
You’ll quickly want to switch to a hardphone. Softphones suffer from poor sound quality and noticable delay. About all they’re good for is testing and checking voicemail.