[code]
res_ldap.conf
[_general]
;
; Specify one of either host and port OR url. URL is preferred, as you can
; use more options.
;host=192.168.1.1 ; LDAP host
host=lync-demo.local ; LDAP host
port=389
url=ldap://ad.lync-demo.local:389
protocol=3
Hello,
I have a problem.
One every couple of months my asterisk system crashes with a segmentation fault.
kernel: asterisk[20527]: segfault at 0808 rip 2aaac952d8f2 rsp
40edb910 error 4
(This is in /var/log/messages)
If I look at the same timestamp in the warning log
Hello all,
I have the following challenge: I have to add a variable to the
destination channel with the following conditions:
1) It has to be set in the dialplan, in runtime.
2) The source channel can't have the same variable has the destination.
I had two ideas so far, but they seem
Check us out at: http://digium.com http://asterisk.org
Arjan Kroon wrote:
Hello,
I have a problem.
One every couple of months my asterisk system crashes with a
segmentation fault.
Normally if you are getting a segfault, that's a good reason to file a bug
report. However...
I use the
On Sun, Oct 28, 2012 at 2:48 PM, giuseppe...@gmail.com wrote:
Hello guys,
I would like to use asterisk with a html sip web client.
What asterisk version or particular question are required?
If you're starting without any pre-existing configuration, it would be
smart to use the current
Jakob Hirsch wrote:
Hello everyone!
Hola,
We use Asterisk for various services like voicemail. Our SIP clients
usually use rtp events (rfc2833) for DTMF, which works just fine and
independent from the codec (g711 vs. g726 etc.).
Now we noticed there are some SIP clients that announce
Dmitry Melekhov wrote:
19.10.2012 08:40, Dmitry Melekhov пишет:
Hello!
Hola,
I'm trying to use psi+ to conect to asterisk using chan_motif and vise
versa.
Connection looks good, but no sound.
As I see there is some traffic (22.229 is my desktop with psi)
08:38:37.463506 IP
Alexandre Rodrigues wrote:
Hello all,
Hola,
I have the following challenge: I have to add a variable to the
destination channel with the following conditions:
1) It has to be set in the dialplan, in runtime.
2) The source channel can't have the same variable has the destination.
In our sales queue, we have wrapup time set to 15 seconds. When the
phones are really busy, the operators would like the ability to bypass
that 15 second wait and grab the next call in the queue. Is that
possible? How to accomplish?
--
Mitch
--
As I read the queues.conf.sample file I would say no since you would have to
set the value to 0 and reload the queue. If you state your asterisk version
and whether you're using realtime, someone might offer a solution.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I don't think you can. But you could set it to a lower value like 3 seconds
and give your operators a feature key to pause themselves in the queue if
they need extra work time.
- Logs
On Oct 29, 2012 12:15 PM, Mitch Claborn mitch...@claborn.net wrote:
In our sales queue, we have wrapup time set
Asterisk 1.8
Not currently using realtime.
Mitch
On 10/29/2012 12:19 PM, Danny Nicholas wrote:
As I read the queues.conf.sample file I would say no since you would have to
set the value to 0 and reload the queue. If you state your asterisk version
and whether you're using realtime, someone
Since you're not using realtime, your best bet is probably going to be to
give your operators a menu to increase/decrease their queue penalties and
set your wrapup time low like the other poster suggested.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
JR Richardson wrote:
My bad. I sent Igor to the boneyard to fetch 1.6.0.28 and it appears to me
that by commenting out lines 309-312 and doing a fresh make you eliminate
the extra files (or make them empty).
Appriciate the suggestion but commenting out 309-312 refused to compile:
cdr_csv.c
Thanks Joshua for the quick reply.
Both of these may seem complicated to you because they don't expose a
single option that just does exactly what you want.
That's exactly what I was thinking. I know that asterisk provides
tools to implement a lot of different things and because of
this I was
Hello,
I am asking the user to enter his mobile phone followed by # using Read().
From time to time the Read() application disconnects the user while he is
typing his number, though there is a 15 seconds timeout, and even if I type
the number very fast it still may happen to me.
*same =
On Tue, 30 Oct 2012, Thomas Thomas wrote:
I am asking the user to enter his mobile phone followed by # using
Read(). From time to time the Read() application disconnects the user
while he is typing his number, though there is a 15 seconds timeout, and
even if I type the number very fast it
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