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.
<br><br>Alex<br><br><div><span class="gmail_quote">On 5/9/06, <b class="gmail_sendername">Hadar Pedhazur</b> <<a href="mailto:hadar@unorthodox.com">hadar@unorthodox.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Replying to my own post (and my most recent follow-up). I have now<br>confirmed 100% that the DSL modem gets a _new_ IP address every time his<br>"real" phone gets answered, or hung up! This (of course) disrupts the
<br>audio coming from to him, since the sending machine (Asterisk in my<br>case), no longer has the correct IP address to send to him.<br><br>I lowered his registration from the default 1 hour to 1 minute, so after<br>we're disconnected, I can see that he's re-registering with a new IP
<br>address, each and every time :-(.<br><br>I told him to call Bellsouth and ask about a Static IP address, but I<br>don't know if they offer it, or how much they charge.<br><br>While this one isn't "solved", it's at least "explained".
<br><br>Thanks to everyone who responded!<br><br>Hadar Pedhazur wrote:<br>> I haven't seen anything this strange, and it's 100% reproducible. I'm<br>> hoping that there are some clever ideas out there for what to look for,
<br>> since I can test to my heart's desire on this one...<br>><br>> My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has<br>> a regular POTS line connected on the same line. He has the appropriate
<br>> filters on every jack that has a phone connected to it, and he even<br>> replaced one or two of them (when I thought that was the problem).<br>><br>> I sent him a HandyTone GS-486 (HT), configured to connect back to my
<br>> Asterisk server. He only has a single computer in his apartment, so it's<br>> connected into the HT, and the HT is connected into the DSL modem.<br>><br>> He can make and receive calls on the HT, and the quality is excellent.
<br>> If he's speaking via the HT (meaning a VoIP-only call) and the "real"<br>> phone rings, everything continues fine (temporarily). If the real phone<br>> is answered, either by a person, or by the answering machine (which is
<br>> in another room, connected to a filter on another jack), then the audio<br>> on the Asterisk conversation becomes _one way_. My father can be heard<br>> _perfectly_ by the remote side of the conversation, but he can hear
<br>> nothing. When the POTS line is hung up, then both sides of the VoIP call<br>> go dead (audio-wise). Of course, he can now redial a VoIP call, and both<br>> sides work perfectly...<br>><br>> At first, I couldn't imagine that it was anything other than a bad
<br>> filter, but other than replacing the filter (which didn't help), nothing<br>> else stops working. He can continue to use the Internet connection on<br>> his PC just fine, and I can continue to hear him speak over the VoIP
<br>> connection with no problems either, so the Internet connection has not<br>> been lost.<br>><br>> I have to admit to being completely clueless as to what to even look<br>> for, so _any_ advice as to things to test for would be appreciated. As I
<br>> said at the top, I can reproduce this 100% of the time, so I can easily<br>> setup any debugging environment in advance, and trigger the problem at<br>> will, etc.<br>><br>> Thanks in advance!<br>_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Alex Robar<br><a href="mailto:alex.robar@gmail.com">alex.robar@gmail.com</a>