( bounced again, one of these days i'll get used to this new email program)


just some ideas to try

a. the 616 ports will not provide cpc  (or will any kxt or kxtd as 
anyone that every hooked up an answering machine will know ;-)

b. the "battery" of the 616 port is about 30 vdc, the sip might be 
expecting 48 vdc, you might try a common 9v battery and 300 ohm resistor 
iin series with the 616 port so the sip sees about 40v on its port.  the 
sip has lots of settings, it might be selectable for 24v or 48v  "co 
battery".  the 9v batt is just a way of testing and won't last long at 
30 ma.

c. is the 616 the pots only version or really a 616-10 that supports apits?

d. listen to the port with a butt set, i remember the orig 616 ports had 
audio on them when the port was not in use.  in fact, it can have bgm on 
it, which would get the sip real confused. and i think the 616 had the 
built-in "greensleeves" moh chip too.  i think there is a switch for 
internal/external moh.  make sure bgm is off on the port (code 1 8 
sticks in my mind as the feature code) 
e. is the 616 port sending stutter dial tone?

f. if the sip is looking for a quiet port when not in use, there is a 
trick with a bridge and diode to make it quiet.  wire a small bridge 
rectifier (~) leads in series with the port, wire a diode to the bridge 
(+) -->|-- (-).  what happens-  the bridge and diode has a 2v drop that 
will absorb the ac signal when not in use, but once the port is siezed, 
the current passing thru the bridge/diode will now pass the ac (audio).  
0 Dbm is less that 1v peak.

g.  while testing, put a swtch in series with the port.  switch off, set 
up the call (get an error yet?), after a few seconds, flip on the switch 
(get an error msg?)  it could be a timing problem.  part 98 calls for 2 
second silence between line sieze and "dial tone".  the 616 presents 
tone immediately.  (at least it is precise dialtone ;-)
-larry

keep us informed!




Roland H. Alden wrote:

>I have an old KXT 616 and a brand new Sipura 3000. The 3000 has an FXO
>port and works fine when plugged into Verizon but when it tries to make
>a call out through a station port on the KXT it (I think) tries to seize
>the line and then reports a "AUD:Stop PSTN Tone" event. 
>
>This event appears to be only documented on Google by people like me
>having the same class of problem, and no answers as well.
>
>Some people think this has something to do with the Sipura falsely
>detecting disconnect supervision. 
>
>In the Sipura a lot about "PSTN Disconnect Detection" is configurable.
>For example these are the default parameters:
>
>Disconnect tone: [EMAIL PROTECTED],[EMAIL PROTECTED];4(.25/.25/1+2)
>Detect polarity reversal: yes
>Detect CPC: yes
>Min CPC Duration: 0.2
>Detect Polarity Reversal: yes
>
>I tried the obvious and set Detect CPC to no but that did not fix the
>problem. I know that the KXT has certain peculiarities in this area and
>thought I'd ask here before I start a serious program of trail and
>error.
>  
>


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