happen
with 1.4 and chan_zap.
Anyone run into the issue before?
hose
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
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[18517]: chan_dahdi.c:10515
timer interrupts
properly or possibly 2) chan_dahdi is losing track of the timer events?
hose
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
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 in there?
I am getting
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
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 adjust the levels?
There was a flaw
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
this wrong?
hose
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
wondering about dial macros - it doesn't
branch?
hose
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
, 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 release date
in the ChangeLog has been implemented?
hose
.
hose
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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:
conf = 123456,22,1
Users can get
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
--
_
-- Bandwidth and Colocation
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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
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
--
_
-- Bandwidth and Colocation Provided by http
/viewtopic.php?f=1t=77154
hose
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk
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
working fine, but I
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
on the console when this
happens, I'm at a bit of a loss how to diagnose it further.
Suggestions?
hose
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
What you say...David Backeberg (dbackeb...@gmail.com):
On Fri, Jun 24, 2011 at 4:55 PM, Hose hose+aster...@bluemaggottowel.com
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
software
), but it seems to work? :) It's completely
confused me as to why this actually works.
hose
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
= national
signalling = pri_cpe
rxgain = 1.0
txgain = 1.0
group=1
echocancel=yes
channel = 1-23
jbenable=no
callprogress=yes
musiconhold=default
usecallerid=yes
hose
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
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
group) it works
/(number) would just jump to channel 25?
Testing seems to bear this out, but I'm not positive about it.
hose
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
...
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 / showing a RED alarm, would asterisk automatically skip over
trying to use channels
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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
What you say...John Novack (jnov...@stromberg-carlson.org):
Carlos Alvarez wrote:
On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.com
mailto:hose+aster...@bluemaggottowel.com wrote:
We have an asterisk frontend terminating all our SIP phones
27 matches
Mail list logo
