Hello Mathew,
Thank you for the reply.
I will open an issue and send debug information.
Can you explain more about the workaround? A reference to the documentation
would be fine.
Thanks again,
Yaron.
On Sun, Oct 26, 2014 at 10:46 PM, Matthew Jordan mjor...@digium.com wrote:
On Sun, Oct
2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com:
Am 25.10.2014 00:09, schrieb Olivier:
Hello,
I need to play some musiconhold content starting at a random duration
from the start.
Thanks to mode=custom option and either madplay or mpg123 programs, I
could successfully get
Hello,
Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when
a call has been hung up because the SIP rtptimeout has been reached?
Thank you,
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
Am 27.10.2014 08:54, schrieb Olivier:
2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com:
Am 25.10.2014 00:09, schrieb Olivier:
Hello,
I need to play some musiconhold content starting at a random duration
from the start.
Thanks to mode=custom option and either madplay or
Am 26.10.2014 00:43, schrieb lee:
Hi,
how can I make asterisk do something when an outgoing call is picked up?
The background is that I would like to record incoming and outgoing
phone calls. In order to do this, I need to play an announcement
telling the person calling or being called
Hi all,
what should i do if i want to know how long asterisk server take a time for
registration 1 client on server side?
especially just for voip server authentication, when we have to registered
username and password in sip.conf and extensions.conf files? it's not how
long registration for one
On Mon, Oct 27, 2014 at 1:20 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hello Mathew,
Thank you for the reply.
I will open an issue and send debug information.
Can you explain more about the workaround? A reference to the
documentation would be fine.
Sure - really, what you are
On Sun, Oct 26, 2014 at 10:42 PM, Todd R. tjrl...@live.com wrote:
Does anyone know how to set the music on hold class with the Manager
Interface in 1.8?
Here is what I am using but I end up just getting no music when I put this
in place, when I remove it the default is back.
The classes I
Thanks Matt.
I tried that already, no luck.
Still, I get blank nothingness instead of MOH. I will try again just to be sure
I didn't miss something.
I have also tried surrounding musicclass with CHANNEL() but that didn't
work and didn't seem right anyhow since it already knows it's a channel
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Mudgett
Sent: Friday, October 24, 2014 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk
On Mon, Oct 27, 2014 at 10:56 AM, Murthy Gandikota mgandik...@nts.net
wrote:
--
Thanks, Richard. How do I get manager events such as VarSetEvent (
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_VarSet)
using ARI?
Events are provided
I have released an updated AppKonference that compiles with Asterisk 13. You
can download the latest code from source forge:
sourceforge.net/projects/appkonference
That said Asterisk 13 doesn’t get that much attention because I use Asterisk
1.4 + some hacks. Here’s a link to my Asterisk 1.4
The reason the dial plan can never be deprecated is because Asterisk wouldn’t
be Asterisk without the dial plan. Sure, you could re-engineer Asterisk so that
it would be “better for a small select group of users at the expense of the
majority of community that use the product as designed for
On Mon, Oct 27, 2014 at 2:04 PM, Paul Albrecht palbre...@glccom.com wrote:
The reason the dial plan can never be deprecated is because Asterisk wouldn’t
be Asterisk without the dial plan. Sure, you could re-engineer Asterisk so
that it would be “better for a small select group of users at
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
Jordan
Sent: Monday, October 27, 2014 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
On Mon, Oct 27, 2014 at 2:40 PM, Murthy Gandikota mgandik...@nts.net
wrote:
I am unable to detect the Manager_Setvar event using ARI.
Can you please let me know, in ARI lingo, the curl or javascript code to
detect the AMI Manager_Setvar event for myvar in the following dialplan:
Howdy,
I'm trying to get my feet wet with pjsip using the conversion script
mentioned on the Wiki on this page:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
I'm using the copy of the script that's included with Asterisk 13
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
Jordan
Sent: Monday, October 27, 2014 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
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