The integrators normally use a Fluke meter to measure the resistance/impedance on the line.
If you know some old Bell telephone line installer/troubleshooter, he will be able to tell you exactly which Fluke test meter they use to monitor the line impedance and levels. Stephan Monette. On 2011-10-14, at 8:53 AM, John Van Ostrand wrote: > I was trying to get away from using a PC for several reasons, I'm not ready > to trust a voip dids entirely and this solution is geared towards offices > with about 4 line (+ or -) so PRI is out. > > How does one know which level to choose when adjusting audio? > > ----- Original Message ----- >> Hey John, >> >> I used several FXO ATAs in the past and they all had different issues >> like >> DTMF, echo, static,... I tested Mediatrix, Quintum and they both had >> issues that customers didn't like. >> >> A lot of our Asterisk PBX integrators seems to prefer using a PC with >> Asterisk and Sangoma Analog FXO boards. They always use the hardware >> base >> echo canceller with the Sangoma cards. >> >> Because of the analog phone lines, you always need to tweak and adjust >> audio levels on each line. You will find that you need to re-adjust >> the >> line levels 2-3 times a year to get ride of some issues depending on >> the >> weather! >> >> Sangoma has really good support on this matter. >> >> Some PBX integrators got tired of tweaking FXO ports and refuse to >> sell >> systems with FXO lines! They only sell VoIP or PRI-T1 services with >> their >> PBX. With the rates of PRI-T1 being sub $500.00 (we have a few >> customers >> that purchased PRI-T1 form Primus at sub $500/month!), many >> integrators >> pushes the customer to get rid of analog lines (and got rid of the >> headaches too!). >> >> I would be curious to hear who has perfect results with FXO lines on >> Asterisk with no re-tuning on all their installation (not just one >> install)! >> >> Good luck. >> >> -- >> Stephan Monette >> >> Business Development Director >> Primus Business Services. >> Tel.: 613-656-2980 >> >> >> >> >> On 11-10-11 4:02 PM, "John Van Ostrand" <[email protected]> wrote: >> >>> Hi, >>> >>> We are struggling with a Cisco (nee Sipura) 8800. It's not capturing >>> DMTF >>> reliably on inbound calls to our automated attendant (asterisk) and >>> it >>> happens only with certain callers and never with other callers. The >>> first >>> one, two or three digits will be missing so a caller pressing 5102 >>> could >>> get option 1, 0 or 2 instead of my extension. It also is more >>> reliable >>> when callers wait for silence between prompts before dialling. >>> Flipping >>> between AVT and inband doesn't change much. We gained better >>> reliability >>> by reducing the gain on the outgoing audio. When AVT is configured, >>> we >>> can see missing SIP event packets and the 8800 doesn't log the digit >>> either so it's clear that it's just unable to recognize the tones. >>> >>> Is the 8800 reliable? What is a good solid ATA? What about a >>> Mediatrix >>> 1204? Any other suggestions? We need a 4 port FXO ATA. >>> >>> Thanks, >>> >>> -- >>> John Van Ostrand >>> CTO, co-CEO >>> Net Direct Inc. >>> 564 Weber St. N. 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