[Asterisk-Users] Line Noise HELP!
Andre Normandin
anorman at superdata.com
Tue Apr 12 03:56:11 MST 2005
rusty*CLI> show version
Asterisk CVS-HEAD-03/26/05-17:05:44 built by root at rusty.superdata.com on a
i686 running Linux
rusty*CLI>
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Rich
Adamson
Sent: Monday, April 11, 2005 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Line Noise HELP!
And, what asterisk version are you running?
------------------------
> Ooops, sorry folks.. A correction..
>
> I don't have digium X100 cards, I have Digit Networks X100P clone cards..
Don't know if it
matters, but wanted to get the facts straight :-)
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Andre Normandin
> Sent: Monday, April 11, 2005 5:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Line Noise HELP!
>
> Hi,
>
> I'm having very similiar problems.. However, I'm running a development
version, and it
happens on both SIP phones, and on Analog
> phones connected via Sipura SPA-2000's (I have 2 different SPA2000's,
and 4 analog lines..
Seems to happen on all of them as well)..
>
> The problem seems to be EXACTLY as described.
>
> THe call seems fine at first, then within minutes the call degrades to
the point that
neither end can hear each other.. First, the volume
> seems to lower, and then static, breaking up, etc..
>
> I have both DIGIUM X100 cards for my pots lines (3 of them), and
BROADVOICE for outgoing
calls. It seems to happen no matter if I'm
> on an analog line (I.E. someone called me), or if it was me that
initiated the call
(BROADVOICE outbound).
>
> I do have a 'remote' SIPURA SPA2000 located at a friends house in a
different state -- he
is an extension on my pbx so he can call me, and
> he can call his friends locally (He just moved away) via my POTS or
BROADVOICE line.. He
experiences the same problems as I described
> above, unless he calls me directly at my 'internal' extension, or I
call him at his
'internal' extension.. I.E. If it doesn't touch POTS or
> BROADVOICE, the problem doesn't seem to occur..??
>
> The other interesting thing that has happened of recent development is
that some people
are complaining that they are hearing the
> 'electronic beep' sound as if the call is being recorded, but I am not
recording the call.
This has occured with my friend as well as incoming
> and outgoing POTS/BROADVOICE calls.
>
> If anyone has an idea, I'd love to hear it.. The problem is driving me
(and others who
talk to me) crazy!!!
>
> - Andre
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Damian Funnell
> Sent: Monday, April 11, 2005 3:08 PM
> To: junk at irqx.com; Asterisk Users Mailing List - Non-Commercial
Discussion
> Subject: Re: [Asterisk-Users] Line Noise HELP!
>
> Hi mate,
>
> Interesting - you're using a different version of Asterisk than I
am, but your problem
sounds identical. We thought it was the SIP
> phones that we were using as well, but then it started occurring
on the analogue
phones as well.
>
> Post again when you've tried a new phone, will you? Let us know
if the problem goes
away.
>
> Cheers,
> Damian.
>
> Paul wrote:
>
> @page Section1 {size: 8.5in 11.0in; margin: 1.0in 77.95pt
1.0in 77.95pt; }
P.MsoNormal { FONT-SIZE:
> 12pt; MARGIN: 0in 0in 0pt; COLOR: #000066; FONT-FAMILY: "Times
New Roman" }
LI.MsoNormal {
> FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; COLOR: #000066;
FONT-FAMILY: "Times New
Roman" }
> DIV.MsoNormal { FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; COLOR:
#000066; FONT-FAMILY:
"Times
> New Roman" } A:link { COLOR: blue; TEXT-DECORATION:
underline } SPAN.MsoHyperlink
{ COLOR:
> blue; TEXT-DECORATION: underline } A:visited { COLOR: blue;
TEXT-DECORATION:
underline }
> SPAN.MsoHyperlinkFollowed { COLOR: blue; TEXT-DECORATION:
underline }
P.MsoPlainText {
> FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: "Courier
New" } LI.MsoPlainText
{
> FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: "Courier
New" }
DIV.MsoPlainText {
> FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: "Courier
New" } PRE {
FONT-SIZE: 10pt;
> MARGIN: 0in 0in 0pt; COLOR: #000066; FONT-FAMILY: "Courier
New" }
SPAN.EmailStyle18 {
> COLOR: navy; FONT-FAMILY: Arial; mso-style-type: personal }
DIV.Section1 { page:
Section1 }
>
> Damian,
>
> pbx*CLI> show version
>
> Asterisk CVS-HEAD-03/23/05-00:44:07 built by root at pbx on a
i586 running Linux
>
> There is my version info. Someone on the list has suggested
that its my SIPura
phone. It could very well be the phone, but it
> just seems unlikely that the conversation would be perfectly
clear for some time
before the static starts. I tried it today and was
> able to go approximately two minutes before it started. The
FXO card is a generic
x100P. Im going to try to get another IP
> phone and test it to see if its the phone. Let me know if you
come up with any
ideas.
>
> Paul
>
> ________________________________________
>
> From: Damian Funnell [mailto:damian.funnell at fff.co.nz]
>
> Sent: Monday, April 11, 2005 12:49
>
> To: junk at irqx.com; Asterisk Users Mailing List -
Non-Commercial Discussion
>
> Subject: Re: [Asterisk-Users] Line Noise HELP!
>
> Hi Paul, there was a thread yesterday in regards to a few of
us experiencing a
very similar problem - a problem that (if the same
> for all of us), doesn't seem to have been properly diagnosed
yet.
>
> One thing that appeared to be common to all of us was the
version of Asterisk that
we are running (1.0.6), is this the version
> that you are running? What FXO card are you using?
>
> Does the noise that you are complaining about occur on every
call and, if so,
always after exactly a minute, or is it more
> random?
>
> Starting to wonder if there isn't a problem with Stable,
interested to hear what
version you're running.
>
> Damian.
>
> Paul wrote:
>
> I recently hooked up my sipura IP phone and set it up as an
SIP device to
>
> connect to asterisk. I am able to dial a number on the SIP
phone, connect to
>
> an external number via the PSTN connected to asterisk and
begin the
>
> conversation. At first, the audio quality is PERFECT, in both
directions. I
>
> can hear the person clearly and they claim to hear me like im
on a regular
>
> POTS line. After approximately 1 minute, the quality turns
horrible and the
>
> person can no longer hear me, but I can faintly here them.
There is a lot of
>
> static on the line, it almost sounds like an electronic device
is
>
> interfering with it. I thought maybe it was a wireless phone
or router, so I
>
> disconnected all those and put my cell phone in the other
room. Still no
>
> change. Anyone have any ideas, this is really getting to be a
problem.
>
> Paul
>
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list