Hello Michael,
Thanks a lot for your immediate help. After applying patch MixMonitor
started works normally,
I can understand that this can be Happen in asterisk 10.4 but as a stable
and Long support version 1.8.12.0 this should not happen. I got same error
in both version.
Anyways this patch
Am 24.05.12 23:46, schrieb bilal ghayyad:
Thanks for all for the help and kindly reply.
One last point that will help me alot:
I am thinking to have 4 Servers running Asterisk and 2 Servers to be for
database. The load to be distributed on the 4 Asterisk Servers with ability
to be
Hi all,
I am running the same Asterisk 1.4.21.2 with the same configuration on all the
servers in the region.
I got this function called func_devstate which I use to control the lights of
the Polycom phones.
This module works well for all the Asterisk servers except this one.
To get it
Hello,
I need to return to the original call leg that I wanted to transfer the
call to. in case the destination IVR has put me in a rather long queue.
Please suggest a way I can hang up the atxfer leg and return to the first
call leg.
The hangup parameter in dial app using '*' key works only
You can unload the features module maybe
On Wed, May 23, 2012 at 9:18 PM, Eduardo Pimenta
e...@akivasoftware.com.brwrote:
Hi Guys,
is there any way to disable all Asterisk Features? We are having false
dtmf detections and randon calls being put on-hold and suspect that dtmf
features is
On Thursday 24 May 2012, Jayesh Labade wrote:
Hello All,
I have installaed asterisk 10.4 in my machine. Now suddenly MixMonitor
application starts generating 44 Bytes of Recording file.
Is this new tye of Bug? Help me..
44 bytes is very interesting, as this is the canonical length of a .wav
Hi John;
For 20,000 users: Is it better to use Asterisk realtime configuration or to use
conf files?
I readed the below link but did not understand which GUI that works with
asterisk realtime?
http://www.freepbx.org/trac/wiki/AsteriskRealtime
Regards
Bilal
My question is:
- Original Message -
From: Jayesh Labade jayesh.lab...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 25, 2012 2:09:58 AM
Subject: Re: [asterisk-users] Asterisk MixMonitor starts recording 44
bytes file
We'd like you all to help us welcome Rusty Newton to Digium's Asterisk
development and community support team! Rusty has been with Digium for
over five years, starting in the Technical Support department and then
moving to a sales position where he assisted customers with Asterisk and
Switchvox
Hello ISDN Users.
I am hit by some frustrations because my Server has only two PCI-X slots
and my Eicon Diva 4BRI-8M, which should work fine with Asterisk, is only
PCI 2.0 standard and does not fit into the PCI-X slot.
Currently I use two AVM Fritz! cards, but while my Xeon 604 2000MHz had
a
Hi again,
does someone use the USB-Stick Huawei K3765-HV with Asterisk?
I have an 4-Port TT-Hub (Cypress Chip) and 4 USB-Sticks with 4 different
GSM Provider (Germany, France, Turkey, and Iran)
Currently I use the Vodafone Easybox 803 A together with an ISDN
connection to my Asterisk
Swift() is an asterisk wrapper around the text-to-speech engine cepstral. Looks
like this is a dev issue - I'll start a new thread on the dev mailing list.
Justin Killen
Senior Programmer / Analyst
All American Asphalt
951-736-7600 x 2060
On 05/25/2012 11:10 AM, Michelle Konzack wrote:
I am hit by some frustrations because my Server has only two PCI-X slots
and my Eicon Diva 4BRI-8M, which should work fine with Asterisk, is only
PCI 2.0 standard and does not fit into the PCI-X slot.
This does not make sense; PCI 2.0 cards
Hi
Recently our asterisk system stopped beign recognized by URA in others
telephones exchanges. What's the troubleshoot steps for this issue?
--
Att.*
***
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com
On 5/25/2012 3:18 AM, Lee, John (Sydney) said:
-- Executing [*1223*1**1900@incoming:78] Set(SIP/1900-08ee1da8,
DEVSTATE(Custom:cfalw1900)=INUSE) in new stack
I use
'Set(DEVICE_STATE(Custom:var)=BUSY)'
in my 1.4 dialplans to set device state.
mark
--
Asterisk Realtime is better for administration.
Performance, IMHO is the same issue. I'm not lucky to made large
implementations to test these.
Regards,
--
Ing CIP. Alejandro Celi Mariátegui
a...@linux.org.pe
http://cipher.pe/web/nuestra-experiencia.html
El vie, 25-05-2012 a las 07:06
Realtime is probably better for administration, but do you want to throw a
layer of complication into such a large undertaking? I wouldn’t want 20,000
people screaming at me because MYSQL crapped out.
From: asterisk-users-boun...@lists.digium.com
Howdy,
Since the subject is Viteiy Setup, I don't think this is off topic.
My big problem with Vitelity is getting my server to register for
incoming calls. I can make outgoing calls just fine. My server says
it is registered with Vitelity, but no calls come in. Every attempt
to call the
Is your IAX2 peer registered?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralph Green
Sent: Friday, May 25, 2012 4:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
If your server says it is registered, that could be part of the problem.
Vitelity doesn't use trunk registration, only IP authentication. You should
not be using a registration string in your trunk definition. I don't know
if it will hurt but it won't help.
It sounds like you might have only 1
I am using asterisk for voice mail. During DTMF collection Asterisk
stop sending any RTP Packets. The gap between two consecutive packets
are 4 seconds, which is huge enough to screw up the jitter buffer. When
ever asterisk stops to receive DTMF, the RTP stream is cut and we loose
audio.
I
On 05/25/2012 04:30 PM, Dave George wrote:
I am using asterisk for voice mail. During DTMF collection Asterisk
stop sending any RTP Packets. The gap between two consecutive packets
are 4 seconds, which is huge enough to screw up the jitter buffer. When
ever asterisk stops to receive DTMF, the
Hi Kevin,
I have two asterisk boxes with the same issues.
Box 1: asterisk ver 1.4.21.2
Box 2: Asterisk 1.8.7.1
setup:
CDMA Phone CDMA Media Gateway WCM sip Asterisk voice mail
The calls are SIP Based. DTMF collection is when the user is entering a
password for voice mail access or voucher to
Hi;
In Voicemail.conf
If am am using
format=h263|gsm ,and i want to store only audio , then it is not storing.In log
it shows that video is deposite less then 5 second. If i want to store video
and audio both then it will store properly.
If am using
format=gsm|h263 ,then
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