Hi all,
(I am testing on Asterisk 11.7.0~dfsg-1ubuntu1)
I am using the following format to append to a logfile, according to the
documentation http://www.voip-info.org/wiki/view/Asterisk+func+FILE:
same = n,Set(FILE(/tmp/mylog.txt,,,a)=my-log-message)
But this does not append a newline. So I
Hi,
Let's say I do:
Set(data=xxx,yyy)
Gosub(my-sub,s,1(${data}))
My subroutine will only receive xxx for ARG1. How can I pass a literal
with a comma to a single argument in a subroutine?
(The point is: when calling the subroutine I do not know if the variable
has a comma or not.)
Thanks,
Dear all,
i'm using Chanspy to dynamically play a sound file on a specific channel.
It works the caller and the callee can hear the file playing during
their conversation.
However, i'm also using Mixmonitor to record the call. The thing is, in
the resulting wav i can of course
hear the
Quick and drity:
1) Meetme has to be configured to record the media stream.
2) You have to install a streaming server. Maybe ffmpeg could do the job:
https://trac.ffmpeg.org/wiki/StreamingGuide
3) Then your website should be able to get the stream from the streaming
server.
You should be able
Tonight several community services will have intermittent
availability due to maintenance. This maintenance will begin at
approximately 9:00 PM CST[1] and should last no longer than three
hours, ending around 12:00 AM CST.
The affected services are:
* JIRA issue tracker (issues.asterisk.org)
*
One thing that concerns me is that this post is from 2009, even though the
newest version of the app on Sourceforge is up to date. I have a customer who
has been using a conference server that I built for him using app_konference
for several years now and he routinely runs conferences with
El Lunes, 8 de diciembre, 2014 12:51:42, Matthew Jordan
mjor...@digium.com escribió:
On Fri, Dec 5, 2014 at 3:23 PM, Rodrigo Montiel
guevara2...@yahoo.com.ar wrote:
Hi masters,
I’m not an expert on this my friends, but I’m trying to understand which the
expected behaviour is
On Monday, December 08, 2014 02:21:16 PM Matthew Jordan wrote:
Hey everyone!
This is a friendly reminder that Asterisk 12 will be entering security fix
only mode soon. As a Standard release of Asterisk, Asterisk 12 received one
year of maintenance fixes, and will receive one year of security
On Tuesday 09 Dec 2014, Daniel Gonzalez wrote:
Hi,
Let's say I do:
Set(data=xxx,yyy)
Gosub(my-sub,s,1(${data}))
My subroutine will only receive xxx for ARG1. How can I pass a literal
with a comma to a single argument in a subroutine?
(The point is: when calling the subroutine I do
hi,
i upgraded few asterisk systems from last asterisk 1.8 to 11 and i see
in graph that cpu usage is ~50% higher
any ideas? configuration, modules, .. is the same
--
---
Marek Cervenka
===
--
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Tuesday, December 09, 2014 7:01 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Playing audio to bridged channels using
Hi Everyone.
I was referred here by malcolmd of the Asterisk forums. What follows is a copy
of this question: http://forums.asterisk.org/viewtopic.php?f=1t=92007?
I've recently upgraded from Asterisk 11 to Asterisk 13.
Most of it went smoothly thanks to the documentation detailing how to
I want to create a voip service, I do not know much about it, but the first
thing I want to know if more than one client can make a call at the same
time through internet to the PSTN, and what gateway should I use for this.
2014-12-08 13:07 GMT-08:00 Steve Edwards asterisk@sedwards.com:
On
On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont
p.beaum...@hatsoffsoftware.co.uk wrote:
Hi Everyone.
I was referred here by malcolmd of the Asterisk forums. What follows is
a copy of this question:
http://forums.asterisk.org/viewtopic.php?f=1t=92007
I've recently upgraded from
Thanks Richard. This is exactly the answer I was looking for.
I'm now assuming that Asterisk 11 was using it's equivalent bridge_simple but
I was getting confused because the only bridge module I saw in modules.conf was
bridge_softmix. When I upgraded to Asterisk13 that would have been the
I want to create a voip service, I do not know much about it, but
the first thing I want to know if more than one client can make a
call at the same time through internet to the PSTN, and what gateway
should I use for this.
I think the first recommendation any of us will have is to
On Tue, Dec 9, 2014 at 2:58 PM, Patrick Beaumont
p.beaum...@hatsoffsoftware.co.uk wrote:
Thanks Richard. This is exactly the answer I was looking for.
I'm now assuming that Asterisk 11 was using it's equivalent
bridge_simple but I was getting confused because the only bridge module I
saw
Hi,
In my office have setuped the Elastix machine and i have a static IP
(external IP given by ISP), now the issue is that whenerve call from
outside sip extensions which is register to the sip server , am not able
hear audio from both side.
both callee and caller cant hear audio.
please help me
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