POTS Lines are analog, and a key system is the best way to deal with analog lines since you are simply switching the voice path with no analog to digital to analog conversion in the process.
VoIP is digital, so starting with a digital signal is always the better way to go. Consider the conversion that must take place on a POTS interface Caller speaks into phone - analog signal Call reaches Telco CO - converted to digital for switching on the PSTN Call routed to your POTS line - converted back to analog - no need on ISDN Call answered by * - back to digital - no need on ISDN Call router to your hard phone - back to analog 5 A/D D/A conversions, with lossy compression along the way (VoIP) Also, digital service provides rock solid and predictable signaling for call starts, ends, CID, etc. POTS lines provide shady analog signaling the is hard to interpret because of line quality variations. So unless you can get really clean POTS lines you are far better off with ISDN/PRI, and if you cant get that a key system with an analog voice path will sound better, always. I realize this does not answer your question and solve the problem, it is just an attempt to explain why you are not getting a better answer. I would assume MOST * deployments will be in environments where nothing but T1/E1 would ever be considered, leaving the SOHO user in the same position they are in now, spend more or live with the limitations. With all that being said, start by checking your POTS lines (or asking the Telco to) for excessive resistance, impedance and noise. The fact that you are not having good experiences with a variety of hardware where others have been able to get good (acceptable) results indicates there may be a line issue. Damon (new to * but not new to Telecom and VoIP) -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 03, 2004 8:14 AM To: Asterisk Group Subject: [Asterisk-Users] An anniversary and a lament for FXOs This week marks one year since I first setup an Asterisk server in the hopes of transitioning my home office to a total VoIP system. The process has been an incredible learning experience. I've tried numerous IP hard phones, eventually settling upon the Polycom IP600 as my choice. I've also used multiple ATAs including all the Sipura products. Using Asterisk has been a challenge, a thrill and (when its working) a joy. However, the one thing that I am not satisfied with is the performance of the FXO interfaces that bring in my PSTN lines. I've tried X100p cards but found them horribly unreliable. I presently use Sipura SPA-3000s but they're only marginally better. How is it that my Panasonic 4 line SOHO phone system (KX-TG4000B) can have four stable, reliable FXOs with no echo at all in a device with a total cost of <$500? It seems to me that there ought to be hardware available that behaves just as well, but bridges the PSTN to the SIP/IAX domain? I've read a lot on the list about how difficult designing FXOs can be, but that flies in the face of the fact that every small multi-line phone system has them...and without expection those behave better than the devices I've been able to try with Asterisk. The Sipura SPA-3000 has several settings to adjust for line impedance and inductive/capacitive line loading....lots of settings, but it provides nowhere near the basic performance of one of the lines on the Panasonic KSU. It's simply mind boggling. So, while I've posted with respect to FXOs previously, I must ask again....what FXO interface device can anyone recommend from real experience? Michael P.S. - I even investigated switching my lines to ISDN to get around the need for FXOs, but SBC won't do it where I live. -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 _______________________________________________ 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 _______________________________________________ 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