It makes me a little leary to try the Trixbox if I will only be "lucky"
to get cell phone quality on a cable line. Voice quality is important,
especially when we call long-distance and many of our suppliers
and customers don’t speak perfect English to begin with.
Good afternoon, Al.
I have really limited experience with Asterisk … could barely spell Asterisk a couple of weeks ago … but since then, with a LOT of help from the fine people in this forum, I have built up a fairly feature-rich though very small PBX for my consulting partnership. I did it by building Asterisk and its drivers from sources, on a general-purpose Fedora Core 5 box, and not by installing one of what I understand are canned Asterisk solutions like Asterisk@Home and Trixbox (correct me if I’m wrong about the “canned solutions” part).
I strongly urge you not to give up on Asterisk just yet.
My system is a 2.5-GHz Celeron with 1GB RAM, and connects to the Internet via a low-speed DSL with 1.5Mbps download speed and 192Kbps upload speed. We have a Digium TDM-400P board with one FXS port and three FXO ports. Two of the FXO ports connect to telco wall jacks. I have the PBX set up so that if someone calls in via the PSTN, and selects the extension number for one of us, and the person being called isn’t here, the PBX can automatically redirect the call out the other PSTN port to a cell phone number, so that the call then goes in one PSTN line, through Asterisk, out the other PSTN line, and to the cell phone.
My partners live 40+ miles from here in two different directions, one with DSL and the other with Comcast cable, and the one with DSL connects to the PBX via X-Lite 3.0 softphone (my other partner isn’t connected yet… we need to give him a boot in the ass and get him to set up and configure X-Lite). Within the office here, there is one analog phone connected to an FXS port, and two VoipVoice USB phones connected to PCs running X-Lite 3.0.
So far we are pretty happy with the results. The audio quality is good except when I’m downloading large files, or sometimes when I’m on Skype, then the low-speed DSL into here tends to get loaded down and people calling in from the outside over SIP tend to get not-so-good results (internal phones and connections to the telco still work well, though).
I have no idea what might happen if and when I start loading it up with more PSTN connections or start handling more simultaneous SIP / IAX2 calls. We have no immediate plans to do that so what we have now works very well for our needs.
Now, it could be that my expectations just aren’t very high. I’m tickled pink that it even works at all and was as easy as it was to set up. So, it could be that what’s perfectly fine for us will be only marginally acceptable to your environment.
If you would like to try our system, to see how it sounds to you, you are welcome to call me at 603-437-1811 ext 21, or else set up X-Lite 3.0 on your machine and send me an email at firstname.lastname@example.org (I’ll reply with my “real” email address), and I’ll send you the information you need to connect to us via SIP. That might give you some useful feedback regarding how the performance and audio quality might be in your application.
I do warn you that our Automated Attendant messages (“Press 1 for … press 2 for… etc.”) don’t sound very good. I recorded them myself using a seriously sub-optimal recording configuration, back during my first few days of figuring out how to set this thing up. I know what to do to re-record them and make them sound better, just haven’t gotten around to it yet.