asterisk users - May 2004 - New TDM04B 4-port FXO card problems

Just installed the new 4-port FXO card and moved two pstn lines from
existing x100p cards to ports on the FXO card. All zapata.conf entries
that were functional on the x100p's were copied to the new FXO channels
(including callprogress=no).

Observations thus far:
1. asterisk will spontanously decide a pstn call has arrived, and ring
   the sip phone designated in the dailplan. Verified callprogress=no
   is in place, and monitoring the two pstn lines with an external
   analog phone (with line lamps) indicates no one was on the phone
   and no ringing actually occurred.
2. Incoming CallerID seems to be slightly less reliable on the FXO 
   compared to the x100p. (Eg, there "seems" to be more cases of the
   callerid showing up as "asterisk".)
3. The echo issues that have been so well documented with the x100p's
   seem to be identical on the new FXO card. 
4. Incoming pstn calls that either go to IVR menues or VM do not properly
   sense disconnect supervision. Again, monitoring the pstn line via the
   LEDs on an analog phone "does" indicate approximately .5 second of
   no-battery (disconnect). After the disconnect, * does not release the
   pstn line which then causes dial tone from the Central Office to be
   recorded in the VM, etc. The CLI does not indicate a hangup until
   _after_ the sip phone hangs up. MAJOR Problem, and reported to support.

Additional testing is being conducted. Anyone seeing these same problems?

Rich

>
> Just installed the new 4-port FXO card and moved two pstn lines from
> existing x100p cards to ports on the FXO card. All zapata.conf entries
> that were functional on the x100p's were copied to the new FXO channels
> (including callprogress=no).
>
> Observations thus far:
> 1. asterisk will spontanously decide a pstn call has arrived, and ring
>    the sip phone designated in the dailplan. Verified callprogress=no
>    is in place, and monitoring the two pstn lines with an external
>    analog phone (with line lamps) indicates no one was on the phone
>    and no ringing actually occurred.
I saw this a few times today on my X100P.. Problem in zaptel code perhaps..
> 2. Incoming CallerID seems to be slightly less reliable on the FXO
>    compared to the x100p. (Eg, there "seems" to be more cases of
the
>    callerid showing up as "asterisk".)
Yes I see errors more here too..

> 3. The echo issues that have been so well documented with the x100p's
>    seem to be identical on the new FXO card.
> 4. Incoming pstn calls that either go to IVR menues or VM do not properly
>    sense disconnect supervision. Again, monitoring the pstn line via the
>    LEDs on an analog phone "does" indicate approximately .5
second of
>    no-battery (disconnect). After the disconnect, * does not release the
>    pstn line which then causes dial tone from the Central Office to be
>    recorded in the VM, etc. The CLI does not indicate a hangup until
>    _after_ the sip phone hangs up. MAJOR Problem, and reported to support.
>
I worked with Mark today and this problem is now fixed in the CVS.
> Additional testing is being conducted. Anyone seeing these same problems?
>
> Rich
>
Scott

On Sat, 2004-05-01 at 23:28, Scott Weis wrote:
> >
> > 4. Incoming pstn calls that either go to IVR menues or VM do not
properly
> >    sense disconnect supervision. Again, monitoring the pstn line via
the
> >    LEDs on an analog phone "does" indicate approximately .5
second of
> >    no-battery (disconnect). After the disconnect, * does not release
the
> >    pstn line which then causes dial tone from the Central Office to be
> >    recorded in the VM, etc. The CLI does not indicate a hangup until
> >    _after_ the sip phone hangs up. MAJOR Problem, and reported to
support.
> >
> 
> I worked with Mark today and this problem is now fixed in the CVS.
I just downloaded the most recent CVS, compiled, removed modules, and
reloaded.

Problem still persists.

Here is the bug ID I posted: Add comments I guess

http://bugs.digium.com/bug_view_page.php?bug_id=0001522
----- Original Message ----- 
From: "Roger Gulbranson" <roger@gulbranson.com>
To: <asterisk-users@lists.digium.com>
Cc: "Roger Gulbranson" <roger@gulbranson.com>
Sent: Saturday, May 01, 2004 11:57 PM
Subject: Re: [Asterisk-Users] New TDM04B 4-port FXO card problems

> On Sat, 2004-05-01 at 23:28, Scott Weis wrote:
> > >
>
> > > 4. Incoming pstn calls that either go to IVR menues or VM do not
properly
> > >    sense disconnect supervision. Again, monitoring the pstn line
via
the
> > >    LEDs on an analog phone "does" indicate
approximately .5 second of
> > >    no-battery (disconnect). After the disconnect, * does not
release
the
> > >    pstn line which then causes dial tone from the Central Office
to be
> > >    recorded in the VM, etc. The CLI does not indicate a hangup
until
> > >    _after_ the sip phone hangs up. MAJOR Problem, and reported to
support.
> > >
> >
> > I worked with Mark today and this problem is now fixed in the CVS.
>
> I just downloaded the most recent CVS, compiled, removed modules, and
> reloaded.
>
> Problem still persists.
>
>
>
> _______________________________________________
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>

On Sat, 2004-05-01 at 23:57, Roger Gulbranson wrote:
> On Sat, 2004-05-01 at 23:28, Scott Weis wrote:
> > >
> 
> > > 4. Incoming pstn calls that either go to IVR menues or VM do not
properly
> > >    sense disconnect supervision. Again, monitoring the pstn line
via the
> > >    LEDs on an analog phone "does" indicate
approximately .5 second of
> > >    no-battery (disconnect). After the disconnect, * does not
release the
> > >    pstn line which then causes dial tone from the Central Office
to be
> > >    recorded in the VM, etc. The CLI does not indicate a hangup
until
> > >    _after_ the sip phone hangs up. MAJOR Problem, and reported to
support.
> > >
> > 
> > I worked with Mark today and this problem is now fixed in the CVS.
> 
> I just downloaded the most recent CVS, compiled, removed modules, and
> reloaded.
> 
> Problem still persists.
I see Mark just posted a new fix.  Problem now resolved (at least for
me).  Thanks.