HI,
well, that was what I expected in my posting yesterday. For me, your wiring
looks strange. Here in Germany, we have "spiltters" connected to the
incoming line which have two outputs: A high pass filter output for the DSL
signal and a low pass output with DC pass-through for the POTS signal. the
DSL output is being connected to the DSL-modem and the POTS output will
feed your internal POTS wiring.
Therefore, there is _NO_ filter needed on each POTS outlet, because there
is nothing to be filtered out on your internal line anymore.
Seen from my German wiring knowlegde, your cabling is wrong and causes the
interruptions on the DSL service.
Don`t you have something like a "spiltter" available ? It should be the
_ONLY_ filter on your incoming line and then the DSL-modem and the POTS
phone should be connected to it ...
--Jürgen
Replying to my own post (and my most recent follow-up). I have now
confirmed 100% that the DSL modem gets a _new_ IP address every time his
"real" phone gets answered, or hung up! This (of course) disrupts the
audio coming from to him, since the sending machine (Asterisk in my case),
no longer has the correct IP address to send to him.
I lowered his registration from the default 1 hour to 1 minute, so after
we're disconnected, I can see that he's re-registering with a new IP
address, each and every time :-(.
I told him to call Bellsouth and ask about a Static IP address, but I
don't know if they offer it, or how much they charge.
While this one isn't "solved", it's at least "explained".
Thanks to everyone who responded!
Hadar Pedhazur wrote:
I haven't seen anything this strange, and it's 100% reproducible. I'm
hoping that there are some clever ideas out there for what to look for,
since I can test to my heart's desire on this one...
My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has
a regular POTS line connected on the same line. He has the appropriate
filters on every jack that has a phone connected to it, and he even
replaced one or two of them (when I thought that was the problem).
I sent him a HandyTone GS-486 (HT), configured to connect back to my
Asterisk server. He only has a single computer in his apartment, so it's
connected into the HT, and the HT is connected into the DSL modem.
He can make and receive calls on the HT, and the quality is excellent. If
he's speaking via the HT (meaning a VoIP-only call) and the "real" phone
rings, everything continues fine (temporarily). If the real phone is
answered, either by a person, or by the answering machine (which is in
another room, connected to a filter on another jack), then the audio on
the Asterisk conversation becomes _one way_. My father can be heard
_perfectly_ by the remote side of the conversation, but he can hear
nothing. When the POTS line is hung up, then both sides of the VoIP call
go dead (audio-wise). Of course, he can now redial a VoIP call, and both
sides work perfectly...
At first, I couldn't imagine that it was anything other than a bad
filter, but other than replacing the filter (which didn't help), nothing
else stops working. He can continue to use the Internet connection on his
PC just fine, and I can continue to hear him speak over the VoIP
connection with no problems either, so the Internet connection has not
been lost.
I have to admit to being completely clueless as to what to even look for,
so _any_ advice as to things to test for would be appreciated. As I said
at the top, I can reproduce this 100% of the time, so I can easily setup
any debugging environment in advance, and trigger the problem at will, etc.
Thanks in advance!
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