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

Hadar Pedhazur hadar at unorthodox.com
Mon May 8 14:36:28 MST 2006


Alex Robar wrote:
> I'll lean this way too. I had a DSL line from Bell Canada in Kingston, 
> Ontario, and an incoming call on that line to the POTS phones would 
> cause VoIP traffic to become completely unintelligble. The VoIP call 
> would have to be re-established to fix things. A quick call to Bell had 
> a technican out to check the lines, and put a fix in place for me.

I was afraid of doing that, unless I specifically explain that it's a 
VoIP thing, because otherwise, if the tech asks "what was interrupted", 
I won't be able to show anything else...

Thanks for the suggestion!

> Alex Robar
> 
> On 5/8/06, *Jerry Jones* <jjones at danrj.com <mailto:jjones at danrj.com>> wrote:
> 
>     I would guess either the DSL itself is bad or perhaps the dsl Modem.
>     perhaps calling Bellsouth would be helpful? Does other Internet
>     traffic get interrupted also?
> 
> 
>     On May 8, 2006, at 1:42 PM, 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!



More information about the asterisk-users mailing list