Hi there Wiley! On Sat, 14 Aug 2004 14:43:05 -0700, Wiley E. Siler <[EMAIL PROTECTED]> wrote: > My office build is the same as yours. 15 or so extensions, low traffic > 100MB network, and a desire to have a phone system that uses VoIP. I > have my system working as a PBX just like you would. I use two TDM400s > for my 8 POTS lines and Polycom IP 500 phones at the desktop. I also > tested with the Grandstream phones you suggested. SO, we have the same > system requirements so here are the answers as I have found them for my > implementation....
Thanks for your e-mail!!! Your setup and your envoironment are really encouraging, since they are very similar to what I have in mind (except for the quantity of POTS lines - we won't use that many). > Voice quality on the SIP based phones has a lot to do with the codec you > use. The lowest compression codec is uLaw and that is what I use since > we have tons of bandwidth to spare. Also, my HP switch has COS (class > of service which is like QOS) so I can prioritize the packets coming > from my phones over the standard network traffic. Even without this > switching feature turned on, performance was great. The phones > themselves play another role in the quality. Grandstreams are pretty > good and I have only used mine for testing so I will not disparage them. > However, the old saying stands. You get what you pay for. Raising your > phone budget from $85 to more like $150-250 will get you a phone with > more features and greater expandability in my IHO. However, you can > still do great things with the cheaper Grandstream phones and still have > a system that works very well. IT is all up to what you can spend and > what you need. Google the archive by putting "site:lists.digium.com" in > front of your search string (no quotes though). You should see some > good info on phones. Well, I'm not really looking for a lot of phone features, just the basics (transfers, call retrieval, etc.). And voice quality is not what I am most worried about, but the delays on the conversation. However, on your mail, you say that latency is, in most cases, unnoticeable, and those are great news to me, as I feel more comfortable to suggest our office to buy ip-phones and use them, knowing they will serve us well. > Latency is gonna be there on any network. However, on my network (which > is just like yours) the latency is very very low. We are talking > 20-40ms tops and it is completely unnoticeable when using the phone. > The only problem I have had at all has been with occasional echo. It > does not happen often and it usually takes about 5 seconds for the * box > to train up and remove it. Most of this seems to originate in the fact > that I am using POTS lines. The solution that uses a T1 PRI has better > features and I think it has less echo potential. However, that would > not work for me since my T1 provider wanted to make me pay 6 grand to > switch to a PRI from my standard data T1 with POTS. Just some food for > thought... I'll most likely use a BRI. Do you think this will help to avoid echo? > I have been a VoIP user for about 1 month after spending another > researching what when where how... So, we know I am not an expert... > but as a fellow user and new VoIP initiate, I can tell you that Asterisk > is a phenomenal product for SMB level offices like yours and mine. When > I compared it to a PBX system of comparable power, expandability, and > feature set, Asterisk won easily since the only real cost I have had was > for my phones. I have my system in place for around 3000 dollars and it > is competitive with all the 10K dollar solutions the vendors threw at me > plus it has an undeniable advantage in upgrade path. All upgrades to > the system are free and the sky is the limit to what you can build using > the framework that all the * gurus have built into this system. Not to > mention the fact that if anything ever goes wrong with the server, I can > have a new one in place in under and hour. Try that with a PBX when > some proprietary part goes belly up. You could wait days potentially. > My $.02. Hope this helps. That's also what I hope it will happen here! If we want to expand, we don't want to end up with a closed-system that won't handle more extensions or phone lines. And since things are converging, and things like FWD, Vonage and others are helping ppl to communicate, the use of a voip based system would certainly help us more to communicate with our clients and with ourselves. Yours, Francis _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
