What you say...John Novack (jnov...@stromberg-carlson.org):
>
> Carlos Alvarez wrote:
> >
> >
> >On Tue, Mar 5, 2013 at 2:32 PM, Hose ><mailto:hose+aster...@bluemaggottowel.com>> wrote:
> >
> >We have an asterisk frontend terminating all o
them, even when running
ping tests for hours during heavy call volume periods. The loads on the
machines are minimal - never seen the load go above .10 during normal
operation. But it does seem like something between them is making them
drop calls.
hose
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/g1 or dahdi/G1...
> regards,
> yves
>
> Am 05.03.2013 07:31, schrieb Hose:
> >Hello,
> >
> >If I put two spans' worth of channels, say 1-23 from span 1 and 25-47 in
> >span 2, in one group, but only span 2 was showing OK and the other was
> >down
di/g1/(number) would just jump to channel 25?
Testing seems to bear this out, but I'm not positive about it.
hose
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What you say...Richard Mudgett (rmudg...@digium.com):
> > What you say...Richard Mudgett (rmudg...@digium.com):
> >
> > > > I've always used dahdi-genconf to just create the
> > > > dahdi-channels.conf
> > > > and since our PRI is fairly simple (just dump all the channels
> > > > into
> > > > one
rmine if it even has any effect.
>
> Richard
Ah, ok. Is it possible that this file isn't even being used then? The
chan_dahdi.conf is similarly terse and doesn't have an include = line:
[channels]
context = g
(dahdi/g1), but it seems to work? :) It's completely
confused me as to why this actually works.
hose
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What you say...David Backeberg (dbackeb...@gmail.com):
> On Fri, Jun 24, 2011 at 4:55 PM, Hose
> wrote:
> > Can anyone recommend some kind of virtual t.38 fax software? I'd like
> > to test/debug some of the t.38 stuff, but it'd be much easier if I had a
> &g
faxes from.
There doesn't seem to be much out there, and the stuff that's out there
is kind of expensive for me just to be using for testing.
hose
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N
t seem to be any other error on the console when this
happens, I'm at a bit of a loss how to diagnose it further.
Suggestions?
hose
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of silence would be more acceptable to me than
setting maxsilence to 2 seconds and having an undue pause in the left vm
causing a hangup, or setting maxsilence to 0 seconds and having a
situation where the PRI or whatever gets wedged, and the voicemail just
keeps recording for... a long time.
hose
What you say...Hose (hose+aster...@bluemaggottowel.com):
> Because some users have requested transfer beep confirmations I've
> switched our phones over to using the asterisk transfer feature instead
> of the built in transfer functions of the phones. While testing it was
> wo
http://forums.digium.com/viewtopic.php?f=1&t=77154
hose
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h
e
IAX trunk. This is asterisk 1.6.2.16.1. Did switch get removed and I
missed it? It doesn't seem to be in the applications anywhere.
hose
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New to As
eject = yes is
on in sip.conf). Anyway, the logs don't show anything more useful
either. Is there something obvious I'm missing? Cranking up verbosity
on the console doesn't seem to do anything.
hose
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is on the
list, then routes accordingly. Does anyone have any suggestions as to
how to approach that, or if they have a entirely different way in mind?
hose
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n hold is going when
there is no Answer command. Incidentally, in 1.6.1.x the Answer appeared to be
explicit after dumping a call into a queue, which is how I came across
this issue after upgrading to 1.6.2.11.
hose
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What you say...Hose (hose+aster...@bluemaggottowel.com):
> I can't seem to locate any documentation on what this does. I tested it
> out with a simple static conference room:
>
> exten => conference,1,MeetMe(,1aMqw)
>
> and a static room defined in meetme.conf
hold when they enter.
hose
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> Tilghman & Teryl
> with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
> and Harry, BB, & George (dogs)
Is there a way to determine which changes have been rolled into a
release, or should we just assume that anything committed since the r
ssed so far in
the 1.6.0 branch?
hose
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What you say...Tilghman Lesher (tilgh...@mail.jeffandtilghman.com):
> On Thursday 09 July 2009 14:13:28 Hose wrote:
> > I understand that standalone macros have been deprecated in 1.6 for
> > gosub routines. I've been working on converting them all but was
> > wonder
I understand that standalone macros have been deprecated in 1.6 for
gosub routines. I've been working on converting them all but was
wondering about dial macros - it doesn't look like there's a replacement
yet to call a gosub routine from the dial command. Or am I looking at
th
What you say...Dave Fullerton (dfullertaster...@shorelinecontainer.com):
> Kevin P. Fleming wrote:
> > Hose wrote:
> >
> >> I have a feeling that the issue is between transcoding of ulaw to g.722
> >> and it's too loud during the transcoding - anyway to a
e call?
I have a feeling that the issue is between transcoding of ulaw to g.722
and it's too loud during the transcoding - anyway to adjust the levels?
Thanks!
hose
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asteris
What you say...Allan Oepping (al...@pacificwebworks.com):
> I'm not sure if this posting will go to the correct thread or not, as I
> am subscribing to make this post, and don't have a message to reply to.
> Hose, if this does not end up in the thread can you post in
card is no longer generating timer interrupts
properly or possibly 2) chan_dahdi is losing track of the timer events?
hose
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What you say...Hose (hose+aster...@bluemaggottowel.com):
> Hi,
>
> I'm getting the following error from an asterisk 1.6.0.9 installation:
>
> [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error:
> Asked to delete sched id -1???
> [May 20 06:07:18] ERROR[
happens; didn't happen
with 1.4 and chan_zap.
Anyone run into the issue before?
hose
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