Le 22/10/2012 04:27, Binan AL Halabi a écrit :
Hello,
It means that one of clients, is using 'silence suppression' mechanism
which sends audio frames that do not contain any samples.
Asterisk complains about silence supression and appears these warnings
on CLI.
If the client turn off the
2012/10/22 Binan AL Halabi binanalhal...@yahoo.com
Hi,
You are using b flag in monitor command. This means don't begin recording
untill call is bridged.
So what you get if you delete this flag ?
If I dont use the b flag then I get two separate files just like in the
case when B waits till
Hello all,
My name is Danilo and I have a problem with the ISDN. I hope I have the
wrong section. =P
I have a CS1000 Nortel central with release 5.50. This central is
attached to an Asterisk server with Sangoma PRI ISDN.
I need to read the headers of ISDN and comes running from Nortel to
Just add noload=cdr_csv.so to modules.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson
Sent: Friday, October 19, 2012 5:09 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Dear All,
I have successfully setup Asterisk realtime. Now I can create extensions
dynamically. But when I put this command on cli mode
sip show peers
it returns no result.
can any one guide me to fix this problem.
Thanks--
On Mon, 2012-10-22 at 19:17 +0500, Control Oye wrote:
Dear All,
I have successfully setup Asterisk realtime. Now I can create
extensions dynamically. But when I put this command on cli mode
sip show peers
it returns no result.
can any one guide me to fix this problem.
Thanks
On 22 Oct 2012, at 15:21, Ishfaq Malik wrote:
On Mon, 2012-10-22 at 19:17 +0500, Control Oye wrote:
I have successfully setup Asterisk realtime. Now I can create
extensions dynamically. But when I put this command on cli mode
sip show peers
it returns no result.
can any one guide me to
Grzegorz Pycia wrote:
Hi
I have some problem with monitor application when call i transferred
in
attended mode and the transfer occurs before call is answered.
Here is how it looks:
A calls B(let's assume ${UNIQUEUEID}=1)
exten = _,1,NoOp
seme =
I'm using latest 1.8, althought I did check and this behaviour is the same
since 1.6.2.11. I will file a bug report about it in 1.8.17.0.
Auto Mixing would not bother me, i will check the Mix monitor.
Regards.
22 paź 2012 17:22, Jonathan Rose jr...@digium.com napisał(a):
Grzegorz Pycia wrote:
In general there is no guaarantee as which call will connect; each queue is
independent AFAIK.
Lenz- big fan :) And I'm sure this topic is of interest to you...
I'll admit, I had a feeling that it's random would be the response to my
original question. I remember reading the app_queue code a
Just add noload=cdr_csv.so to modules.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson
Sent: Friday, October 19, 2012 5:09 PM
To: asterisk-users@lists.digium.com
Subject:
Il 22/10/2012 18:44, Alex Forster ha scritto:
*DEVELOPERS* - If I took a crack at fixing this issue, what general tips do
you have for me to make it most likely that my solution can be integrated
into HEAD? I believe I can justify spending some time at work to deal with
this, but not without at
If I am using asterisk (server) and then asterisk on client (sound port)
and I want to get the best MP3 sound I can get - how can I do that with
ulaw codec
and wav file conversion.
I used gst-launch to convert my MP3 to WAV (16K and mono) then playing
over ulaw
to the other client. I know
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).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Your best bet in 1.4.X is going to be to use a LAME/SOX combination to
convert your MP3's to wav/ulaw files.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, October 22, 2012 12:10 PM
A customer has asked us to provide that feature. I know there are a few
methods and products out there, but I haven't paid attention in a while.
It is for about 300 users, and we'll consider open as well as paid-for
products. We would prefer to pay for supported products as the cost will
be
Unless I missed something, there isn't anything out there that is as cheap
or reliable as human translation in this case. If I did miss it, I know
somebody will correct me.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos
On Mon, Oct 22, 2012 at 2:16 PM, Carlos Alvarez car...@televolve.comwrote:
A customer has asked us to provide that feature. I know there are a few
methods and products out there, but I haven't paid attention in a while.
It is for about 300 users, and we'll consider open as well as paid-for
On Mon, Oct 22, 2012 at 12:40 PM, Christopher Harrington ch...@acsdi.comwrote:
All automated solutions -- paid or free -- are terrible. The technology
simply does not exist at this point at a level that is acceptable to most
customers. If quality is paramount, you are better off doing the
Carlos
I have tried several solutions and non of them have been worth the money. I
have worked with transcription companies and they are the best but they are
expensive. If you do find something that works let the groups know as there
are a few of us out here that are looking for that holy
On Mon, 22 Oct 2012 12:47:51 -0700
Carlos Alvarez car...@televolve.com wrote:
In-house transcriptions are definitely out of the question, but any
experience with outsourced solutions would be useful. As far as I can tell
the current service is automated, and as awful as Google Voice, yet
We have a customer with a dozen phones and they want nearly all of them to
ring.Unfortunately this causes a firestorm of call presence notifications
that overwhelm something on their network. Any existing calls get gappy audio
for a few milliseconds when a new call comes in and when
On Mon, Oct 22, 2012 at 3:05 PM, Lefteris Zafiris zaf@gmail.com wrote:
If you are able to find a reliable way of chopping speech samples in
segments no bigger
than 20 seconds based on silence detection, so words wont be cut in half,
you might come
up with something very similar to Google
You could do a simple PHP/Perl script to query hints and ring only the
not-in-use phones. Or more simply that that do a ChanIsAvail() against the
list and ring the returned array. If I do
ChanIsAvail(line1/line2/line3/line4/line5) and 1 and 3 are in use, it
returns an array with 2/4/5 and I can
Check the notifyringing option in sip.conf
-Original Message-
From: Chris Owen ow...@hubris.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 22 Oct 2012 15:17:27
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk
On 22/10/2012 at 16:02 -0400, Bryant Zimmerman wrote:
Carlos
I have tried several solutions and non of them have been worth the
money. I have worked with transcription companies and they are the
best but they are expensive. If you do find something that works let
the groups know as there
On Mon, Oct 22, 2012 at 2:43 PM, Nickolay V. Shmyrev
nshmy...@nexiwave.comwrote:
There is no holy grail yet, speech technology deployment requires a
close cooperation between the speech technology provider and the users.
It's not plug and play but after some joint efforts automated
On Oct 22, 2012, at 4:25 PM, Danny Nicholas da...@debsinc.com wrote:
You could do a simple PHP/Perl script to query hints and ring only the
not-in-use phones. Or more simply that that do a ChanIsAvail() against the
list and ring the returned array. If I do
On Oct 22, 2012, at 4:11 PM, isr...@gmail.com wrote:
Check the notifyringing option in sip.conf
Interesting. Looks like exactly what I want other than it looks like it is a
global only setting? I'll play with it tonight but any idea if this is still
global only?
Chris
--
On Oct 22, 2012, at 4:11 PM, isr...@gmail.com wrote:
Check the notifyringing option in sip.conf
Looks like this really doesn't do what I had hoped:
;notifyringing = no ; Control whether subscriptions already INUSE
get sent
; RINGING when another
I'm running Asterisk 10.7.0 with three sip trunks to my call
termination provider. For the most part everything works great.
However, at apparently random times and usually about 20 mins or so
into the call, the outbound audio stream dies. The call stays
connected and the inbound audio works fine.
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