[Help] Setting up an Asterisk PBX to replace the current one


#1

Hi all,

I’m pretty new to the world of telecommunications, and I’m actually just a summer student doing a internship at a large company, but I have been assigned the project of setting up an Asterisk system to test out VoIP, and maybe even to implement as the major phone system for the company.

Now since I’m new to this, I have run into quite a few problems, and hopefully some of you out there can assist me. I’ll start by explaining everything below, and asking questions as I go along.

Firstly, I have installed Asterisk on a older Dell box (not the Poweredge’s, or whichever ones have been having problems), it’s a Pentium 3, 1.33 Ghz I believe. Asterisk is installed on top of the Ubuntu operating system. This took a while as there is little to no support to be found when setting up and configuring Asterisk on Ubuntu (even Ubuntu’s own website was a little vague). We purchased a Wildcard TE410P as we have a T1 line in the office, and this is what we are intending to use.

After getting it installed and setup, me and a few co-workers starting testing with the Diax softphone, and that didn’t work to well, so we moved from IAX over to SIP and began using the X-lite softphone. The only problem we ran into with this softphone was we had to make sure that all computers were using the same duplexing, or the sound quality would come out horribly.

Now here is where most of my questions start to come in.

First, while we do have MusicOnHold working with mpg123, the quality seems pretty… bad. I have stripped out all of the ID3 tags, but it still sounds awful. It sounds like there is a popping noise in the background, and the sound cuts out for a split second every couple of seconds. Any suggestions on how to fix this? Also whenever I shut down Asterisk, I get the ‘Yuck’ error, claiming that there is a broken pipe. I’m sure this has to do with MusicOnHold. Any fix to this?

I’ve been trying to start setting this system up so we can handle external calls. I’ve been attempting to dial outside of the office, to other employees cellphones, but I have had no luck. Firstly, can anyone provide a good example of a dialplan for it? As well, this may be my problem, but since we have our own T1 line, and already pay for the line, do we also have to pay for a VoIP provider? I have been under the assumption that since it is our own line, and since we already pay the bills to have the line, that we would not need to pay for another VoIP provider as we would be our own provider. If it’s the case that we do not need to pay for a VoIP provider, how do I go about dialing into the system from the outside? I still have not figured that part out yet either.

Huh, I thought I had more questions than that, but I guess that’s it. So hopefully one (or maybe multiple!) of you out there can help me out.

Cheers,
Moody


#2

Well I figured out one part at least. By paying for a phone T1 line we don’t have to pay for a VoIP provider. Hoorah!


#3

Hi!

I’m pretty much in the same situation as you, though my company wants to become a voip service provider and i’m the one who’ supposed to solve the technical part :unamused: :wink:
check out enum for the topic above :wink:


#4

goto the wiki www.voip-info.org they have alot of good information there for you in regards to dail plans and configuring your iax and sip channels. if you post a little more detail on what it is your trying to do i’ll help you out some more. but first search the wiki .