Hello,
I need to play some musiconhold content starting at a random duration
from the start.
Thanks to mode=custom option and either madplay or mpg123 programs, I
could successfully get what I was after on a Debian Wheezy system.
Now I realized sox version on my target system (Debian Squeeze) ca
On Fri, Oct 24, 2014 at 1:47 PM, sean darcy wrote:
> On 10/24/2014 02:21 PM, Jeffrey Ollie wrote:
>>
>> restorecon -rv /etc/asterisk
>
> I'd never have guessed.
Yeah, if you "mv" the data instead of "cp" the data from one place to
the other, the SElinux labels don't get updated. I like SElinux,
On Fri, Oct 24, 2014 at 1:19 PM, Murthy Gandikota
wrote:
>
> In
> https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann
> els
>
> it is stated:
>
> channel-dump.js in action
>
> Here's sample output from channel-dump.js. When it first connects there
> are no channels in Asteris
On 10/24/2014 02:21 PM, Jeffrey Ollie wrote:
restorecon -rv /etc/asterisk
I'd never have guessed.
Thanks. I owe you a beer. At least one.
sean
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New to
Depending on how the data was copied from one install to the other,
you may be running into SELinux issues. Try running:
restorecon -rv /etc/asterisk
and see if that helps.
On Fri, Oct 24, 2014 at 11:56 AM, sean darcy wrote:
> On 10/23/2014 01:19 PM, sean darcy wrote:
>>
>> On 10/23/2014 11:2
In
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann
els
it is stated:
channel-dump.js in action
Here's sample output from channel-dump.js. When it first connects there
are no channels in Asterisk - (sad) - but afterwards a PJSIP channel
from Alice enters into extension 1
On 10/23/2014 01:19 PM, sean darcy wrote:
On 10/23/2014 11:26 AM, sean darcy wrote:
Running 11.13.1 on Fedora.
This is a new install, but a copy of a previous - working -install.
module load chan_sip
Unable to load module chan_sip
Command 'module load chan_sip' failed.
SIP channel loading...
[
On Fri, Oct 24, 2014 at 10:09 AM, Paul Albrecht wrote:
>
> When Matt says deprecating the dial plan would be difficult and would take a
> long time it seems to me he’s being evasive and misleading. He doesn’t say
> it’s never going to happen and he doesn’t share whatever he thinks the
> Asterisk v
On Fri, 24 Oct 2014, Marco Carvalho wrote:
I set up a new server for Asterisk with 11 cert 6 on it. I am migrating
from a previous server. I have replicated all the configurations,
modules and setup that I know of. However, when I tested an outbound
call, it didn’t work. Checking the asterisk
Hi
I'm using asterisk 1.8 but I'm sure this applies to other versions.
If someone puts a call divert on a handset such as a Snom phone I get this
type of SIP message on receipt of an inbound call:
Got SIP response 302 "Moved Temporarily" back from xxx.xxx.xxx.xxx:x
Which then triggers a loc
On Oct 23, 2014, at 1:58 PM, Kevin Larsen
wrote:
> > From: Paul Albrecht
>
> > Seems like now is as good a time as any to raise these issues, in
> > fact, sooner is better than later because once developers start down
> > a path it’s very difficult to get them change their minds no matter
On 10/23/2014 05:00 PM, Matthew Jordan wrote:
On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton
mailto:dfullertaster...@shorelinecontainer.com>> wrote:
Hello all,
I'm setting up a couple of test boxes and I'm running into a
problem. What I need help with is determining whether I'm
Hi there,
I am running
Asterisk 11.9.0
WANPIPE Release: 7.0.10
DAHDI Version: 2.9.0 Echo Canceller: HWEC
libpri version: 1.4.12
When I start the ConfBridge application I get the following warning:
[2014-10-24 14:36:21] WARNING[29177][C-6934]: config_options.c:790
uint_handler_fn: Attempted t
Hello,
I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a
previous server. I have replicated all the configurations, modules and setup
that I know of. However, when I tested an outbound call, it didn’t work.
Checking the asterisk message log yielded nothing. Any idea
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