Hi Group,
is it possible to read the DTMF tones from a caller while he is in a meetme
conference?
I would like to read the pressed key sequence and call a command like
MeetMeAdmin or System Commands.
I'm using Asterisk 1.8.7.
Thanks for help
Daniel
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Hi Group,
is in MeetMe any option to identify the own number (from the view of a caller)?
I would like to write an option to set on the telephone an request for voice,
if the room is muted. That request should display on our Conference Control
Website and an Admin should unmute this person.
Hello nice group,
having a Problem with CDRs.
If i change the context with Goto() Asterisk write the new exten in dst cdr
field.
How can i keep the old entry? Any ideas makes me very happy.
Thanks for helping me.
Daniel
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...@lists.digium.com] On Behalf Of Daniel Knoll
Sent: Friday, March 30, 2012 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] keep dst cdr record if context change
Hello nice group,
having a Problem with CDRs.
If i change the context with Goto() Asterisk
[
Exten = s,1,playback(vm-goodbye)
Exten = s,n,hangup()
And you get two CDR records, 1 with default and 1 with foo?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Knoll
Sent: Friday, March 30, 2012 2
Looks nice, was also my first idea, but field dst is read only. I can't
overwrite this and get an error like this
ERROR[2474]: cdr.c:345 ast_cdr_setvar: Attempt to set the 'dst' read-only
variable!.
Am 30.03.2012 um 22:00 schrieb Warren Selby:
On Fri, Mar 30, 2012 at 2:21 PM, Danny Nicholas
I found a solution that works fine for me
Set(var1=${SHELL(shellcommand)})
Bye Daniel
Von: Daniel Knoll dan...@danielknoll.de
Datum: 16. April 2011 13:13:28 MESZ
An: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Betreff: write system command
Hey Guys,
In which Version of Asterisk is EventFilter: in manager.conf working?
Higher than 1.6.2.10 or from the 1.8.0 Version?
Thank for your answer
Daniel
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um 17:15 schrieb Godson Gera:
On Thu, Dec 9, 2010 at 1:29 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hey Guys,
for debugging i need to read the Events from AMI. But i have a lot of
unwanted RTCPSent Events.
How can i filter this Events in Asterisk 1.6.2.x Version of Asterisk?
You
Hey Guys,
for debugging i need to read the Events from AMI. But i have a lot of unwanted
RTCPSent Events.
How can i filter this Events in Asterisk 1.6.2.x Version of Asterisk?
Thanks a lot for your answers
Daniel
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Hi @ all,
what is the best way to to use features like MeetmeCount without leaving the
conference.
I use Meetme(,X) and MEETME_EXIT_CONTEXT=context, but the problem is that the
caller leave the Conference :(
Is it possible to press a key, without this obstacle?
Thanx for your answers
Daniel
Has anyone a solution for me
- with Meetme(,Ms)asterisk plays conf-invalid if a room not exist
- with Meetme(123,Ms) asterisk plays not conf-invalid if the room not exist
and asterisk hangup
I am happy about any proposal.
Thanks
Daniel
--
Hello,
is it possible to check more than one condition for GOTOIF in the dialplan?
Or is the normal way to cascade the diaplan each GOTOIF?
The Background is that I would like to check more than 2 values from a
Variable, and then route the call based on the value.
Thanks for your help.
Daniel
Hi Kai-Uwe,
thank you for your answer. but it doesn't work.
i use this dialplan.
exten = 22,n,Answer()
exten = 22,n,NoCDR()
exten = 22,n,WaitExten(2)
exten = 22,n,Set(CHANNEL(musicclass)=music)
exten = 22,n,Set(CHANNEL(language)=de)
exten = 22,n,Read(roomid,conf-getconfno,6,1)
exten =
Hi Paul,
i set Answer() .. just Cut the first, my fault.
is that the normal case, to treat errors like wrong conference Room?
Daniel
Am 07.09.2010 um 15:01 schrieb Paul Belanger:
On Tue, Sep 7, 2010 at 3:11 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hi Kai-Uwe,
thank you for your answer
Hi Group,
i have a MeetMe Question.
I use MeetMe(,Ms) in the Dialplan and if a Conference Room does't exist
Asterisk play (conf-invalid.slin)
If i use MeetMe(${room},Ms) (value from DTMF Read) and the Conference Room
doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk Hangup
Hello Everybody,
does anyone knows an opensource stresstest client for the IAX protocol, like
sipp?
Thanx for your answer.
Daniel
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New to Asterisk?
Hi Group,
short question. is it possible to use
#include asterisk/alaw.h instead of #include asterisk/ulaw.h
in app_meetme.c or is ulaw required in meetme?
thanx for the answer.
Daniel
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Hi Group,
actual i have a transcode problem and i have no idea to solve this. All my wav
files are alaw encoded and i allow only alaw codec.
But sometimes the WriteFormat is slin and if i recall the same number the
WriteFormat is alaw for the Channel.
Why the channel has sometimes slin and
Hi all,
is it possible to send a Variable to another System via IAX Protocoll by using
AMI / Orginate
Like this:
Action: Originate
Channel: IAX2/user1:passw...@192.168.1.2/6...@default
Application: Meetme
Data: 111,q
Variable: var1=111
and the Remote System knows the Variable var1 ?
In my Test
,MeetMe(${CALLERID(num)},q)
Maybe it is helpful for all others.
Daniel
Am 12.07.2010 um 18:58 schrieb Daniel Knoll:
Hi all,
is it possible to send a Variable to another System via IAX Protocoll by
using AMI / Orginate
Like this:
Action: Originate
Channel: IAX2/user1:passw...@192.168.1.2/6
Dear Group,
after a compile Asterisk 1.6.2.6, i have strange issue.
1.) i get a recording message in Log, but i don't set the Option r
2.) if the room number has a entry pin, the caller get a voice Message to left
a Name, i'd never set this Option.
How can i disabled to play vm-rec-name / this
Is it possible to join 2 meetme conferences (each on different server)
together, that if i load balance the callers, they can see altogether
something like a inter system communikation ?
Thanx for your help.
Daniel
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Hello wise Group,
if i plan to load balance incoming SIP Calls for MeetMe Conference to 2 or more
Server, i think it is a Problem, because each Server opened his own MeetMe
Room/Channel. Is it possible to made some interconnect the dahdi or MeetMe
Channels over many Servers? (Like PHP, it can
ok, thanx for your answer.
Daniel
Am 20.06.2010 um 19:17 schrieb Tilghman Lesher:
On Saturday 19 June 2010 10:47:07 Daniel Knoll wrote:
Hello Group,
what does the Compiler Option mean LOTS_OF_SPANS ?
The description is: More than 32 DAHDI spans
Does this mean, more than 32 DAHDI Channels
Hello Group,
what does the Compiler Option mean LOTS_OF_SPANS ?
The description is: More than 32 DAHDI spans
Does this mean, more than 32 DAHDI Channels ?
Thanx for help.
Daniel
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Hi all,
i got a lot of this messages if only one caller is in a meetme
conference and it playing a MusicOnHold Sound. If a second Caller
entry the Conference the messages ended.
DEBUG[11794] channel.c: Internal timing is disabled
(option_internal_timing=0 chan-timingfd=61
What does this message
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Daniel Knoll
Liberdastr. 9
12047 Berlin
fon +49 (0)179 20
Hi Guys,
sometimes if one caller or many callers are in a meetme Room and a new one join
the room,
then he or another caller into the same room where kickt from the room.
It's very strange for me and in logs (full) I can't see anything. is it
possible to log more from meetme.c ?
can anyone
:
try using confbridge in lastest asterisk version
On Sat, Jun 12, 2010 at 11:30 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hi Guys,
sometimes if one caller or many callers are in a meetme Room and a new one
join the room,
then he or another caller into the same room where kickt from
. there are nothing in logfiles with sip
debug on :(
Has anyone the same problem and a solution for me?
Thanx for all.
Daniel Knoll
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: Originate
Channel: Local/1122
Application: Meetme
Data: 1234,q,
Can anyone reproduce this ?
Thanx for your help.
Daniel Knoll--
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/
On May 16, 2010, at 4:16 AM, Daniel Knoll wrote:
Hello User-List,
is it possible to play a sound file directly to a caller channel?
Like this AMI command
Action: Originate
Channel: SIP/20-1d41
Application: Playback
Data: /path/to/audio/file
I get an Error Message. My
Hello User-List,
is it possible to play a sound file directly to a caller channel?
Like this AMI command
Action: Originate
Channel: SIP/20-1d41
Application: Playback
Data: /path/to/audio/file
I get an Error Message. My intension is to play a sound file to a caller and
the other callers
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