I had the exact same problem, removing the hardware echo fix the problem but
this is not a solution for a production system. I'm now using another brand
of hardware.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Webster,
Andrew
Envoyé : 23 janvie
Same result, more FXO interfaces.
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Al
Envoyé : 17 janvier 2007 00:39
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] TDM404B VS TDM2401B
Hi List,
any good comparison between TDM404B and TDM2401B
i'm not very happy
Hi,
I'm having hard time to emulate agencallbacklogin. Agent can logon
and receive call without any problem using addqueuemember. The problem comes
when I try to evaluate their performance using queuemetrics. Here is an
exemple of my log script:
;Agent Login
exten => _60XXX,1,Macro(
Which version are you using? There was a problem in 1.2.12.1 with the page
application. Update to 1.2.13.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steven
Ringwald
Envoyé : 15 novembre 2006 13:45
À : Asterisk Users Mailing List - Non-Commerci
Hi,
I’m sure
some else has been facing this problem. I want to record all the call coming in
my queue. I want this format: MMDD-HHMMSS-AgentID-CallerId - UniqueID. I’m
using the monitor feature inside the queue.conf. I can’t use the
agents.conf monitor features because I’m using dyn
Hi,
I just registed my digium licence and I received the successful
message after the registration process. However, When I load Asterisk, I’m
getting this error :
[format_g729.so] => (Raw G729 data)
== Registered file format g729, extension(s) g729
[codec_g729a.so]Oct 2 01:2
I had the exact same
issue with the TDM2400E, choppy voice on the SIP side. I never called Digium Support,
I simply removed the hardware echo can and everything was fine. I finally
decided to buy a Sangoma A200.
David
De :
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]igium.com
You could take a WRTSL54gs, install openwrt then openser
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve Kennedy
Envoyé : 24 septembre 2006 08:47
À : asterisk-users@lists.digium.com
Objet : Re: [asterisk-users] DSL router with integrated SIP
I had a similar problem
and the problem was the hardware echo can who was defect. Try removing the echo
can hardware echo can and test the line.
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Robson Ribeiro
Envoyé : 21 septembre
2006 12:28
À :
asterisk
You are right,
You can change the
default volume play with thoses parameter :
Those seem to be dB
value.
David
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Damon Estep
Envoyé : 19 septembre
2006 11:24
À : Asterisk Users Mailing List - Non-Comm
A bug has been opened for
this :
http://bugs.digium.com/view.php?id=7982
You should had this
comment to the bugs
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de marvin horst
Envoyé : 19 septembre 2006
09:18
À :
asterisk-users@lists.digium.com
Objet
Mike,
If you could send the answer here it would be greate. I would like to add
this features to my Polycom phones too.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Anthony
Rodgers
Envoyé : 18 septembre 2006 19:09
À : Asterisk Users Mailing Lis
Are you having this problem with an analog line or PRI ?
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Tobias Wolf
Envoyé : 18 septembre 2006 11:41
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Dial and Time
You must had the latest information in your sip.cfg. Compare the template
from version 1.6.7 and the one you are using.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Louis-David
Mitterrand
Envoyé : 13 septembre 2006 05:55
À : asterisk-users@lists
Hi,
I would first of all know
which frequencies Polycom HD Voice use 16 kHz? Also, now that Polycom as
released a phone that support “16Khz” sound and that more device
will probably support this in the near future, is there any plan to support higher
frequencies in Asterisk.
Hi Chris,
I'm would like to get more information about this problem. Why the
phone isn't registering properly with the new firmware and what did you mean
by hard coding the sip settings?
Thx
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
I would recommend you to
call Unlimitel as they have a very good support. Or just send a copy of your
post to : [EMAIL PROTECTED]
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Mike
Envoyé : 7 septembre 2006
11:32
À : 'Asterisk Users Mailing
List - Non-
: Re: [asterisk-users]
Asterisk 1.2.11 and # key
But the behaviour should
like: if pressed once the # should be transmitted. If pressed ## (fast) the it
should be blind transfer. Isn't it?
On 9/5/06, David Gagnon <[EMAIL PROTECTED]>
wrote:
That why, when you dial one # th
Look at queuemetrics.
Pretty good.
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Technical Support
Envoyé : 4 septembre
2006 16:59
À :
asterisk-users@lists.digium.com
Objet : [asterisk-users] Call
center reports
Can someone point me to call ce
dxfer
=> ##" line in my features.conf
On 9/5/06, David Gagnon
<[EMAIL PROTECTED]> wrote:
Are you sure this is not because of the dynamic
features in features.conf ?
By default, # is defined for the transfer
feature.
David
De : [EMAIL PROTECTED]
[mailto:[EMAIL PRO
- Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits
David Gagnon wrote:
> Ronald,
>
> Like someone already told you, you should explain more clearly the
> way you try to transfer, we need more details on the procedure, using
which
> button on
Are you sure this is not
because of the dynamic features in features.conf ?
By default, # is defined
for the transfer feature.
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Michael Strelnikov
Envoyé : 4 septembre 2006
09:53
À :
asterisk-users@lists.di
Ronald,
Like someone already told you, you should explain more clearly the
way you try to transfer, we need more details on the procedure, using which
button on which phone. We need every detail to help you. This as nothing to
do with the way the dial plan is loaded, this is totally false.
Ronald,
You seem to be a little bit angry about VoIP. If so, I could give
you my old Nortel system. Does this would make you happy?
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 2 septembre 2006 04:20
À : Asteri
Take a look at the citel handset gateway. The SIP one. It's an ATA for 24
nortel/meridian phones
http://www.citel.com/products/handset_gateways/
david
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Andre
Courchesne - Consultant
Envoyé : 29 août 2006 11
You could try : relaxdtmf=yes in zapata.conf
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve Edwards
Envoyé : 28 août 2006 21:49
À : Asterisk Users Mailing List
Objet : [asterisk-users] Can I increase DTMF sensitivity?
I have a client that i
Look over there : http://bugs.digium.com/view.php?id=6953
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de J. Oquendo
Envoyé : 24 août 2006 13:54
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Call Parking Ring
I don't think you can use the template of another brand with your Fanvil.
You must configure the phone manually. First time I ear about FAnvil IP
phone so I cannot help you
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL PROTECTED]
Envoyé : 2
8:50 À : Asterisk Users Mailing List - Non-Commercial
Discussion Objet : Re: [asterisk-users] Hint extension issue - bug?
I'm using asterisk 1.2.10
David Gagnon wrote:
>Are you having this problem with the trunk?
>
>
>
>-Message d'origine-
>De : [EMAIL PROTECTED]
D]
[mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez
Envoyé : 23 août 2006 08:50
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Hint extension issue - bug?
I'm using asterisk 1.2.10
David Gagnon wrote:
>Are you having this problem with the trunk?
&
Brad,
It works with friend. I'm using this config since 1 year. I dunno why it
didn't work for Andrew.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Watkins,
Bradley
Envoyé : 23 août 2006 08:48
À : Asterisk Users Mailing List - Non-Commercial Di
Are you having this problem with the trunk?
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez
Envoyé : 22 août 2006 18:23
À : Asterisk Developers Mailing List; Asterisk Users Mailing List -
Non-Commercial Discussion
Objet : [asterisk-users]
Most of the time, the
sample provided by Polycom is the same than the other version. So you don’t
need to use the new xml. If they add something, it’s written in the
release (pdf) so you just need to add what they say.
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] D
I'm using a Windows software call mp3gain. It can normalize directory.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Dennis P.
Clark
Envoyé : 22 août 2006 15:35
À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Objet
PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de David
Gagnon
Envoyé : 20 août 2006 12:48
À : 'Asterisk Users Mailing
List - Non-Commercial Discussion'
Objet : [asterisk-users]
Metermaid - Parking Slot
Hi there,
I was using the old metermaid patch with Asterisk 1.2.9
Hi there,
I was using the old
metermaid patch with Asterisk 1.2.9.1. I just updated to the current trunk and
the hint for my parking slot doesn’t work anymore. I’m always
getting State:Unavailable. Is there anything I need to change of this features
is broken ?
[EMAIL PR
For the ATA, I would
recommend Mediatrix. They have some enterprise grade FXS external gateway with
echo can.
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Lito Lampitoc
Envoyé : 20 août 2006 00:11
À : Asterisk
Users Mailing List - Non-Commercial Discussion
Obj
CTED] De la part de Russell
Bryant
Envoyé : 19 août 2006 00:17
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] SLA Doc
On Fri, 2006-08-18 at 19:17 -0400, David Gagnon wrote:
> I just saw there is a branch in the SVN that support SLA. The latest
> trun
Hi,
Take a look at ramfs (http://plume.bxlug.be/articles/7). All you
need then is to create a link (ln -s) in /var/lib/asterisk/sound to then
ramdrive you created using ramfs.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Nitin Gupta
Hi,
I just saw there is a branch in the SVN that support SLA. The latest trunk also seems to have some kind of SLA support. Is there any doc about how to setup a shared
line or any docs concerning this feature?
Thx
David
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