[sf-lug] Anyone using Asterisk with VoIP on Linux?

[Mark Weisler]

On Thursday 10 July 2008 14:08:05 John F. Strazzarino wrote:
> A friend is setting up a VoIP system for his office.  It is a small
> operation, he just wants to get 4 phones working.  He does have a Trandnet
> router with the IP phone plugged into it and he gets a dialtone, but that's
> it.  His Linux server has a myriad of error messages (of course, none of
> which I have captured!) which would probably explain why the setup is not
> working.  
> Does anyone have any experience/knowledge of how to do this?
Yup.
>  
> I'm reading 'Asterisk for Dummies' like mad, so that would explain my
> knowledge level. 
> Also, do all phones in a VoIP system need to match (same model and
> features) or can you just use 'any old' IP phone?
Nope.
You can't use just any old phone.
A basic dichotomy is is analog versus digital.
If you use analog phones you must have equipment to convert the analog signal 
to digital. This can take the form of a card in the PC like the...
http://www.digium.com/en/products/analog/tdm410.php
or an external 'telephone adapter' like the ...
http://www.digium.com/en/products/analog/s101i.php (You plug in an analog 
phone on one port and the other is an Ethernet connection to your PC running 
Asterisk.)
It can make sense to use analog phones in certain situations especially where 
you don't want to be responsible for people having to learn a lot of new 
technology if they are not so inclined. (Some businesses for example.)



On the digital side of things, you can use IP telephones like the Polycom ...
http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip.html
or you can use a headset from someone like Polycom in combination with 
a 'softphone' - an application on your computer that looks sort of like a 
mobile phone interface and provides dialling and other interaction with your 
phone and Asterisk.




You also have to decide on the method of transmitting voice from your site to 
the called party. You have two realistic choices:
* IP
* The public switched telephone network (PSTN).

Increasingly we are just using IP to leave the premises as the call quality 
using just IP is generally just fine. A while back (months or a year or two) 
it was wise to have a card in the PC that would convert the signal to 
something the PSTN could understand and the call would leave via PSTN 
protocols. That is, you would use an analog line like a plain old home 
telephone line or you would use a T-1 set up for PSTN protocols and calls 
would leave and enter your site via the PSTN.

But the economics and quality of IP mostly favor IP now. (There are situations 
where you have both IP and PSTN and one backs up the other.)

Good luck. The Asterisk world is large and moving fast.


> 
> Thanks
>  
> John
>  
> P.S. Feel free to contact me 'off list' if you feel that the discussion is
> not appropriate to this list. 



-- 
Mark Weisler 
PGP: 0x68E462B6  http://pgp.mit.edu:11371/ 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part.
URL: <http://linuxmafia.com/pipermail/sf-lug/attachments/20080710/5df59d12/attachment.pgp>