[Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit - EXPLAINED

[Alex Robar]

I'm still curious as to WHY it's getting a new IP everytime an incoming POTS
call comes in. If I were you, I'd be asking Bellsouth why this happens
instead of getting a static IP. A static IP may not even solve your issue
too. If the problem is that a POTS call disconnects the modem and causes
PPPoE authentication to re-occur, then you'll still see a VoIP call
disconnect when this happens, even if the same IP is received when the DSL
connection is re-established.

Alex

On 5/9/06, Hadar Pedhazur <hadar at unorthodox.com> wrote:
>
> 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!
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



--
Alex Robar
alex.robar at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060509/e306c5c3/attachment.htm