asterisk users - May 2006 - PSTN Incoming call on real line disrupts VoIP call over DSL circuit

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!

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!
> _______________________________________________
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>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

Well, I have no idea how DSL lines are connected in the US but what happens 
to a normal Internet connection when the phone is being picked up ?
Test scenario could be that your Dad is listening to an Internet radio 
station or other audio stream and then being called....
BTW, how are the "real" phones and the answering machine being
connected ?
Is the HT in front of them in the POTS line ?

--Juergen


>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

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!