The MWI detection is done using fsk modem detection within chan_zap itself.
(It does not support neon MWI detection.) The driver plays no real part in the
detection.
Doug Bailey
- Original Message -
From: "Jim Duda" <[EMAIL PROTECTED]>
To: asterisk-users@lists
If you want to flush your disk cache to see how much memory is being eaten
cache pages, try this:
echo 3 >/proc/sys/vm/drop_caches
- "ast erisk" <[EMAIL PROTECTED]> wrote:
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -
Shaun wrote:
>I've been trying to get the polycom 550 phones to show a idle display bitmap
>but have not been successful. Anybody have any experience with this? The
>manual gives instructions
>(http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soun
essage waiting status of the FXO port in question. This may encoded
in the caller ID indicator or may be contained in its own message spill.
This is output as a NOTICE logging message.
Regards,
Doug Bailey
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-- Bandwidth and Colocation Provided by
- "sean darcy" wrote:
> Doug Bailey wrote:
> > - "sean darcy" wrote:
> >
> >> pstn incoming on a TDM400P, sometimes i* won't answer, going into
> >> a loop like this:
> >>
> >> -- Starting simple switch on
- "sean darcy" wrote:
> OK. Calmer now. If fact a 410 would have the same problem.
>
> I'll make the fix on our machines. Should I file a bug, or does the
> 169154 commit already fix it?
>
> sean
>
>
The issues has been corrected in trunk and the 1.6.1 branch.
Sicne we have addressed
- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> wrote:
> Tell your box to not expect Caller*ID information. You set that with
>
> usercallerid=no in /etc/asterisk/zapata.conf
>
> Since you are using the Asterisk Appliance you would have to contact
> Digium for support.
>
> Sydney Web H
stdexten macro. In that macro is a Dial statement
with a timeout of 20. You would have to adjust that timeout manually and save
it off (Run the save_config script)
One caveat is that the AA50 is not supported when you manually modify the dial
plan. The changes you make are at your own risk.
- D
Have you tried to contact Digium Support about this question?
They are very good at answering AA50 configuration questions.
Regards,
Doug Bailey
- "Fidel Garcia" <[EMAIL PROTECTED]> wrote:
> I have a Digium Appliance AA50 configure with 8 lines and two dial
> plans
When you run configure, you need to spec the "host" parameter for the
architecture
and environment you will be running under.
For example, a 1.4 distro being built for a blackfin running uclinux would run
./configure host=bfin-uclinux
This will imply that the compiler being used to build the
- "Mike" wrote:
> Folks,
>
> I have recently installed Asterisk 1.6.1.1. I have two PSTN lines
> connected to a TDM400 and two VoIP lines using SIP. I have a CISCO
> 7940
> using SIP as my desk phone. Calling any of the four lines should
> ring
> the desk phone. This works fine, except
Ring Pulse Alert Signalling.)
In any case, this is set by defining "mwisendtype" in chan_dahdi.
The default for this is fsk spills.
It can be set to "nofsk" if you want to disable the fsk spills.
The line reversal is set by specifying
mwisendtype=lrev
Regards,
Doug Baile
- "Barry Miller" wrote:
> On Wed, Sep 02, 2009 at 09:44:05AM -0500, Doug Bailey wrote:
> >
> > - "Barry Miller" wrote:
> >
> > > Hi,
> > >
> > > Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840
>
xdetect and
> transfer are turned off.
>
If the fsk spill that you hear is at the beginning of the call, it is due
to the transmission of CID information. (This is usually transmitted
between the first and second rings.)
You should be able to set usecallerid=no in chan_dahdi.conf to di
- "Barry Miller" wrote:
> On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote:
> > > > - "Barry Miller" wrote:
> > > >
> > > > > Hi,
> > > > >
> > > > > Using 1.4.26.1 & DAHDI 2
- "Doug Bailey" wrote:
> - "Barry Miller" wrote:
>
> > On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote:
> > > > > - "Barry Miller" wrote:
> > > > >
> > > > > > Hi,
> > &
- "Doug Bailey" wrote:
> - "Doug Bailey" wrote:
>
> > - "Barry Miller" wrote:
> >
> > > On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote:
> > > > > > - "Barry Miller" wrot
with error on channel 'DAHDI/1-1'
>
The error you are seeing ("No start bit found in fsk data") indicates that the
fsk processing code cannot lock onto the fsk spill. You may want to adjust the
gain applied to the incoming signal while it processes cid.
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