Displaying 20 results from an estimated 900 matches similar to: "Three-way conference in Asterisk"
2007 Mar 15
1
asterisk n-way call problem
Hi,
i am using the n-way-call dialplan solution found on voip-info. i have
added its entry in applicationmap of features.conf file. the problem
is......its not working. to activate the n-way call i dial *0 but nothing
happens. i have played around with dtmf and codec settings but no success.
the extensions and sip configuration is below if you want to have a look. I
dont have any clue why its not
2007 Apr 23
1
problem with 3-way conferenicing
Hi,
I am trying to achieve 3-way conferencing taking hint from wiki link
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
Here is the scenario:
1. user "ua1" calls user "ca1"
2. "ua1" then presses the feature code "*0" to redirect "ca1" to
conference room 300
3. "ua1" then dials the user "33"
4. user
2008 Aug 20
1
3-way conference call
Hi,
I am trying to achieve 3-way conferencing taking hint from wiki link
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
Here is the scenario:
1. user "user1" calls user "user2"
2. "user1" then presses the feature code "*0" to redirect "user2" to
conference room 300
3. "user1" then dials the user "user3"
4.
2015 Dec 22
2
asterisk 13 n-way call problem
Hello!
I need to use n-way call as it described here:
http://habrahabr.ru/sandbox/52259/
It is in russian, but dial plan is quite clear.
It works in asterisk 11:
-- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer)
priority 1
-- Executing [0 at fromtransfer:1] NoOp("OOH323/7272-6385", "") in new
stack
-- Executing [0 at fromtransfer:1]
2009 Jul 26
0
MeetMe time doesn't show up in CDRs?
Hello,
I'm working on some dialplan rules to pull multiple users into a
conference call. I have some fairly straightforward rules which start
up a new MeetMe conference, allow escape with the * key to invite more
users, then use a features.conf sequence to bring the new user into
the conference with ChannelRedirect.
The problem I'm running into is the time in the MeetMe conference
2006 Mar 09
2
Bizarre problems with two Samba in the same workgroup
I had to setup a new Samba server as a PDC in my job (Samba 3.0) to replace
and existing Samba server PDC (Samba 2.2), the old server was running ldap
as backend, the new is running tdbsam. I have a lot of bizarre problems when
I put both in the same workgroup with other machines. I will give you a list
of them:
1) The new server is configured as local master browser and domain master
browser,
2016 Sep 19
2
Ast 13.11.2 : bridgepeer variable empty ?
Hello
I can confirm that the variable DIALEDPEERNAME contains the information
that I would expect in the variable BRIDGEPEER.
But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of
Asterisk version 13 ?!
So if this is not the intention, then yes this is probably a bug and
should be reported.
Kind regards.
Jonas.
On 18-09-16 19:58, Ludovic Gasc wrote:
> Hi,
>
>
2016 Sep 17
2
Ast 13.11.2 : bridgepeer variable empty ?
Hello
a call goes out and is answered :
[Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c:
SIP/myprovider-0000010b is making progress passing it to
SIP/mysippeer-00000108
[Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c:
SIP/myprovider-0000010b answered SIP/mysippeer-00000108
[Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel
SIP/myprovider-0000010b joined
2011 Apr 08
2
Call Recording using MixMonitor - close, but would like some more words of wisdom.
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the MixMonApp =>
*1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I
cannot get it to
2006 Feb 21
3
Send flash through zap channel
Hi everyone,
our setup includes a NEC PBX connected to our asterisk via bri lines.
The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door.
So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so.
Setup: asterisk
2018 Aug 08
2
Queue breaks Dynamic_Features on Attended Transfer
On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo <dan at keshercommunications.com>
wrote:
> > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp.
> > AgentA answers and is able to use that feature code.
> > If AgentA performs an attended transfer of a call from a queue to
> AgentB, the
> > feature code no longer works.
> >
> > It only
2014 Feb 20
2
Variables are empty after Redirecting a channel
Guys,
I am using
Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on
a x86_64 running Linux on 2013-01-18 19:52:25 UTC
How can I set variable in one context and then Redirect a channel to
another context and use variable there? The code below doesn't work, so
I've got empty VAR1 in context_2
[context_1]
exten => s,1,SET(__VAR1=VALUE1)
exten =>
2006 May 10
2
REPOST: features.conf *1 Call Recording
Hi all. I posted this earlier but never got any advice that helped. If
anyone knows how to get this going, I'd appreciate some advice.
I am attempting to setup Asterisk to allow me to press *1 while in a
call to use automon to record the call but have had absolutely no
success. Is there a trick to this?
In extensions.conf
[globals]
DYNAMIC_FEATURES=>automon
[default]
exten =>
2018 Aug 08
2
Queue breaks Dynamic_Features on Attended Transfer
Hi,
I think I've identified an issue and just want to check before completing a bug report.
Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code.
If AgentA performs an attended transfer of a call from a queue to AgentB, the feature code no longer works.
Cases that do work are as follows...
Calls using both Queue() and
2010 Jun 26
2
Detecting hook flash in asterisk
Hello,
Is it possible to detect a hook flash in asterisk. I want to be able to
perform some functions an hook flash.
I have the following entry in features.conf which executes a Macro on
detecting key press '**'.
[applicationmap]
test => **,caller,Macro,testflash
Is it possible to do this action on hook flash?
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2006 Jun 21
1
getting zap peer of sip channel
I'm wanting to capture the zap channel that a sip channel has connected to.
I came across the ${BRIDGEPEER} variable documented on the wiki, and if
I show channel SIP/<channel> when a call is connected I can see
BRIDGEPEER as one of the channel variables.
However ${BRIDGEPEER} is not set when I want it: I run a macro when the
call has been connected.
Does anyone have a hint on how
2012 Jan 23
1
ConfBridge details
running Asterisk 1.8.9.0-rc2, what are the ways to interface with
ConfBridge ?
I see the CLI command 'confbridge' documented for asterisk 10, but i
dont see how to interface with confbridge on 1.8
What I'm trying to do is keep track of conferences that are used.
I tried something like the below, but not only does Confbridge not
return, but i'd need something that erases the
2007 Jun 29
2
features.conf / DTMF / automon hell
I have been trying for a very long time to get asterisk to detect and
utilize dtmf tones from my sip clients within my dial scripts. I have
set automon=>#9 in my features.conf, I have Dial(....,gWw) in my dial
scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in
my extension. I can see the dtmf tones on the wire as SIP INFO
packets. Using the Read() app I have verified that * is
2007 May 31
2
applicationmap on features
I want to be able to send a prerecorded message to the person I am
calling. I know that you can use the application map to do this. Just
to test I enabled the testfeature example that is in the features.conf
file. When I hit #9 during a call the other user does not hear the
monkeys, they only hear a series of beeps. I have tried with different
soundfiles and they all give the same problem.
2010 Apr 29
1
Starting call recording using a dynamic feature to call a macro
I have got call recording working on our 1.4.30 asterisk box together
with a recording pause ability and being able to play different audio to
each party at the start and end of the pause. This all works perfectly
but one wish is to have the audio files have a beep or something in them
so when you listen later you can tell where the audio was paused.
So I changed things around so that instead