It is reasonable 'n' is not usable as priority number. How can asterisk
know the second priority if all other priority have 'n' as priority number?
In a relational database there is no 'sequential read'.
In other words, you need to assign the priority to all entries.
Leandro
Il giorno
Hello,
I was wondering if there is a tool that can create the realtime database
structure for latest Asterisk version or a web resource/file containing
the sql scripts. Hope I haven't missed obvious things, I had no luck
searching on the web, in the wiki I found few pages with bits of sql or
Carlos Chavez писал 02.08.2012 20:24:
originate
SIP/protel-out/0445540881644 application playback tt-monkeys
And if you
route it via local context? For example,
originate
local/outgoing/0445540881644 application playback tt-monkeys
where
outgoing is your context which knows how to route
If you check the contrib/realtime direco
2012/8/3 Daniel-Constantin Mierla mico...@gmail.com
Hello,
I was wondering if there is a tool that can create the realtime database
structure for latest Asterisk version or a web resource/file containing the
sql scripts. Hope I haven't missed obvious
If you check the contrib/realtime/mysql directory in the source tree,
you'll find scripts for almost all the tables.
Leandro
2012/8/3 Daniel-Constantin Mierla mico...@gmail.com
Hello,
I was wondering if there is a tool that can create the realtime database
structure for latest Asterisk
It means ... Asterisk don't make any IVR at realtime. It just fire
Mysql/Odbc query and get *app and appdata.*
On Fri, Aug 3, 2012 at 11:50 AM, Leandro Dardini ldard...@gmail.com wrote:
It is reasonable 'n' is not usable as priority number. How can asterisk
know the second priority if
Hi Ikka,
I'm using asterisk 10.0.0 10.5.2 10.6.0 10.6.1 and they all leave
empty files.
Hope somebody can help.
Regads
Thorben G. Jensen
2012/8/3 Ikka Vertika (Mitra Kreasindo) ikka.vert...@mitrakreasindo.com
HI,
** **
What version is your asterisk ? I’m using 10.2, 10.4. 10.6,
On Friday 03 August 2012, virendra bhati wrote:
Hi Team,
I want to used *'n*' as priority in asterisk realtime but asterisk don't
support n as next priority
I am using Asterisk 1.4.41
Well, what you are wanting would be mathematically impossible!
A simple text file is read
Hi
I've made a call to our elastix server and the call was redirected to the
voicemail, which the user should leave a message. Intead recording the call
the service returned a message like Sorry, but the user's mailbox can't
accept more messages. I'm a little lost in the configs here, what
Leandro
I have to disagree reasonable designers would have done a better job with
this one. But we developers are not always so reasonable. The issue is many
developers when pushing to put features in they don't put on their
designers hat and think out side the box first.Heaven knows I have
Thorben Jensen wrote:
From: Thorben Jensen i...@thorben.dk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 3, 2012 4:15:58 AM
Subject: Re: [asterisk-users] MixMonitor creating file on non-bridged calls
with option b
Hi
Hi Jonathan,
If I set the MixMonitor option on a queue, it will not create an zero
length file if the call is not bridged, and I just assumed it would be the
case with option b.
I have set the fileformat to raw, but if I set it to wav it will create a
64 byte file.
Off course I can
I am kissing every inch of land where each one of the asterisk's developer
is putting his feet. In the last 10 years I have worked thanks to the
availability of the asterisk code. Most of my income was possible just
thanks to asterisk, so I am pretty biased when trying to evaluate if the
asterisk
I have read an earlier posts in this forum that said it was a bug in asterisk
10.4 and there's a patch to fix it. So it already reported as bug.
But i use asterisk 10.61 and that bug is still there, unfixed.
We upgrade to the newest version for the new feature that we can use, and also
Not to bash on the developer who did this I get that we don't always think out side the box all the timeYou can bash others all you want for not thinking outside the box, but where is your effort to think outside the box yourself?. All you have to do, (that's what I did, and took me like 4 hours)
Ikka.vertika wrote:
From: Ikka.vertika ikka.vert...@mitrakreasindo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 3, 2012 9:20:03 AM
Subject: Re: [asterisk-users] MixMonitor creating file on non-bridged calls
with option
Jonathan Rose wrote:
Thorben Jensen i...@thorben.dk wrote:
Hi Jonathan,
If I set the MixMonitor option on a queue, it will not create an
zero
length file if the call is not bridged, and I just assumed it would
be the case with option b.
I have set the fileformat to raw, but
On Friday 03 August 2012, C. Savinovich wrote:
Not to bash on the developer who did this I get that we don't always
think out side the box all the time
You can bash others all you want for not thinking outside the box, but
where is your effort to think outside the box yourself?. All you
Well, it gets even stranger
I've installed version 10.2.1, instead of 10.7.1, and copied the configuration
files from another identical server that is running 10.2.1 is working
correctly.
I STILL can't get voicemail to play back. I can hear the password prompts
Theses are, what I think
AJ, You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle?Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] Asterisk
On Fri, Aug 3, 2012 at 9:35 AM, C. Savinovich
c.savinov...@itntelecom.comwrote:
AJ,
You don't use 'n's in your dialplan?, you number it yourself? man,
what if you have a 300 line dialplan and then you decide to insert a new
line in the middle?
Some might say that you should never do
The structured way of thinking (that cursed philosophy where you write 100
lines of code to avoid a goto) says you should have your contexts small
enough to not need ns.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
I was looking over Queue and I don't think there is actually an option
for Queue that will automatically start a MixMonitor. I see a few options
involving mixmonitor (x and X), but they appear to be more about allowing
the parties involved with the call to start MixMonitor through dialplan
On Friday 03 Aug 2012, C. Savinovich wrote:
You don't use 'n's in your dialplan?, you number it yourself?
man, what if you have a 300 line dialplan and then you decide to
insert a new line in the middle?
If you ever used BASIC you'd remember the trick is to increment line
numbers
MixMonitor creates the file before it starts recording. The b option
simply waits until the bridge event to take audio frames and add them to
the stream. I don't really see why this is a problem. It isn't like you
are going to run out of space for zero length files, and more to the point
if
No, numbers have to be in sequence.
Leandro
I am typing from my mobile phone...
Il giorno 03/ago/2012 20:28, Raj Mathur (राज माथुर) r...@linux-delhi.org
ha scritto:
On Friday 03 Aug 2012, C. Savinovich wrote:
You don't use 'n's in your dialplan?, you number it yourself?
man, what if
Using n with labels is what most people do. A dialplan isn't javascript, you
don't need two hundred 3 line functions.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
Sent: Friday, August 03,
Thorben Jensen wrote:
I was looking over Queue and I don't think there is actually an
option for Queue that will automatically start a MixMonitor. I see a
few options
involving mixmonitor (x and X), but they appear to be more about
allowing
the parties involved with the call to start
Basic?... no man, I am kid!Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as
extension's next priority
From: "Raj Mathur (राज माथुर)" r...@linux-delhi.org
Date: Fri, August 03, 2012
Looking at my voicemail.conf I note this snippet:
; Maximum number of messages per folder. If not specified, a default value
; (100) is used. Maximum value for this option is .
;maxmsg=100
So in my case max messages is .
Assuming you are storing your message in files instead of
Starting around twenty minutes ago, we have begin having some network
difficulties affecting community services. The issue is being
investigated and will be resolved as promptly as possible.
These technical issues appear to be causing an outage to at least the
following services:
On 8/3/2012 3:24 PM, Asterisk Development Team wrote:
Starting around twenty minutes ago, we have begin having some network
difficulties affecting community services. The issue is being
investigated and will be resolved as promptly as possible.
These technical issues appear to be causing an
On Saturday, August 4th, 2012, the Asterisk community services
listed below will be undergoing maintenance (software upgrades and
updates). The services will be shut down at approximately 9:00 PM CDT
(3:00 AM August 5th UTC), and will return no later than 10:00 PM
CDT. We apologize in advance for
Un-top-posting...
On Fri, 3 Aug 2012, Luis H. Forchesatto wrote:
I've made a call to our elastix server and the call was redirected to
the voicemail, which the user should leave a message. Intead recording
the call the service returned a message like Sorry, but the user's
mailbox can't
I am looking for ways to detect if there is some person talking on the
other side of the line and trigger some events based on that.. is there any
possible way through which this could be done in asterisk ?
Thanks,
Sathiish
--
_
Look for AMD (Answering machine detection).
On Fri, 2012-08-03 at 14:42 -0700, sathiish kumar wrote:
I am looking for ways to detect if there is some person talking on the
other side of the line and trigger some events based on that.. is
there any possible way through which this could
What the minimum Server Specifications do I need to run
200 concurrent channels at the time with .WAV recording (MixMonitor)?
It will be connected via VOIP sip account.
Codec will be ulaw.
Which UK dedicated server provider do you recommend and how much bandwidth
do I need?
Thanks
--
Solving my own issue: The order of linking was incorrect.
Change the Makefile as follows to solve this issue.
diff --git a/Makefile b/Makefile
index dec892d..90d79df 100644
--- a/Makefile
+++ b/Makefile
@@ -73,7 +73,7 @@ all: banner $(AST_INC_CHECK) $(AST_VER_CHECK)
@echo
$(NAME).so
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