that you'll add yourself as a watcher or comment on the issue so that
once somebody gets around to looking at it, you'll be notified and can
assist.
Jeff Peeler
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
in the featuremap work anymore, include blindxfer or
automon.
Any ideas what may be the problem?
Have you set the parkedcallreparking, parkedcalltransfers, and other
associated options?
--
Jeff Peeler
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us
On Sun, Apr 26, 2009 at 1:28 PM, jonas kellens jonas.kell...@telenet.bewrote:
part of extensions.conf:
*exten = 11,1,Answer()*
*exten = 11,n,NoOp(CallerID : ${CALLERID(all)})*
*exten = 11,n,Playback(/tmp/welkom-tcs.alaw)*
*exten = 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)*
*;
is from OPAL and from what I remember totally incompatible
with h323 plus.
I just tested on F10 and everything compiles fine.
-
Jeff Peeler
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
remains down, but we will have it back up as
soon
as we can. Thank you for your patience.
Mark Michelson
Bugs.digium.com is now back online and has been updated. If there are
any problems found you should be able to report them under the Mantis
project.
--
Jeff Peeler
Digium, Inc. | Software
On Tue, Nov 18, 2008 at 08:08:51AM -0500, Jerry Geis wrote:
Say in 1.4 there was a meetme command that would show active meetme
conferences.
What is that same command in 1.6?
I looked at core show help - didnt see it.
I looked at UPGRADE.txt and didnt see it.
The 1.4 command was simply
- Lucas Alvarez [EMAIL PROTECTED] wrote:
Hi, I'm having a problem conecting my asterisk 1.4.21 with zaptel
1.4.11
to a Panasonic PBX. I'm using dynamic features to send hook flash to
the
zap channels to make a call transfer to the pbx without tying a
channel.
When I call from
- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Oct 01, 2008 at 01:39:29PM -0500, Jeff Peeler wrote:
Nope, that's the best you can do without restarting Asterisk. Is
requiring two restarts reproducible? I'd really like to see console
output with verbosity and debug set to 4
- Jeff LaCoursiere [EMAIL PROTECTED] wrote:
One of my clients today had a POTS line with a bad punch, and no
dialtone.
I used zap destroy channel x remotely to keep it from being used to
send
outbound calls, which worked fine. Line repunched, ready again to
use,
but how do I
- Elliot Murdock [EMAIL PROTECTED] wrote:
Thanks Tzafrir,
This is what I get:
module unload chan_zap.so
-- Unregistered channel -2
-- Unregistered channel 1
...
-- Unregistered channel 122
-- Unregistered channel 123
-- Unregistered channel 124
CLI module
On Fri, 2008-06-27 at 11:36 -0300, Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.13 as our SIP server and several SIP
clients using ZRTP support (with Zfone module in Windows and libzrtp in
Linux).
People say that it's necessary to use an Asterisk patch in order tu
support
- Rilawich Ango [EMAIL PROTECTED] wrote:
Segmentation fault occurs after executing the following cmd.
Dial(SIP/[EMAIL PROTECTED]|35|Ttr)
Is it a bug and how to fix it?
Below is the core dump message converted by gdb.
#0 0x068be1ad in realtime_peer (newpeername=0x1b37844
On Mon, 2008-03-24 at 16:42 +0700, kitti jaisong wrote:
Hi all,
I have install G729 license to asterisk 1.4.18 and use distro
debian etch 4r2. It'snot complete. when i use ldd commant to show the
library it show
/usr/lib/aserisk/modules/codec_g729a.so: /lib/libc.so.6: version
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