Displaying 20 results from an estimated 700 matches similar to: "3-way conference call"
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
2013 Feb 20
1
Meetme and MEETME_EXIT_CONTEXT
Hello,
using Asterisk 1.8.12.2
I am having trouble with exiting the conference room by entering a
single digit.
option X of the Meetme()-application should do this.
I have following in extensions.conf :
/exten => _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)//
//exten => _1000X,n,MeetMe(${CONFNO},dMX)//
//
//
//[dynamic-nway-invite]//
//exten => 0,1,NoOp(confno =
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]
2007 Feb 02
1
WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)
Hi All,
I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ;
when I dial ,there have this warning:
-- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack
Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting
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
2011 Jun 02
1
Three-way conference in Asterisk
Hi
How to set a threeway conference in asterisk only for VOIP (I am
using only SIP channel).
Thanks
Nikhil
2003 Aug 06
1
S4 methods bug in naming of slots (PR#3665)
Hello,
I am using R 1.7.1 on a Redhat Linux machine, version 7.3.
The following works fine:
setClass("ok", representation(
"A" = "matrix",
"Cmatrix" = "matrix"))
new("ok",
"A" = diag(4),
"Cmatrix" = diag(4))
But the following doesn't work:
setClass("notok", representation(
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
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
2010 Jul 30
1
asterisk-users Digest, Vol 72, Issue 82
thanks for your reply but i think ${BRIDGEPEER} will work only when both
channels are connected. i want to get channel-id before dialing so that i
can dial using that channel id.
> ${BRIDGEPEER} is probably a good way to do what you want.. if Channel
> A calls Channel B, and you want Channel A to "get" the channelID of
> Channel B, as long as the two channels are bridged,
2008 Dec 02
0
How to get both channel ids from diaplan ?
Hi,
I think this have been talked over several times but I couldn't find any
answer.
Sorry for asking.
I want from dialplan, to transfer a callee to a context-extension-priority
that would play a given fax file to callee (callee is supposed to be a fax
number).
I can get caller's channel id (with built-in CHANNEL variable).
I found BRIDGEPEER but its value remains unset (see bellow)
2009 Jul 24
3
Goto from a feature macro is not working?
Hello,
I'm trying to implement multi-party calls according to these
instructions:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
They are almost working, except that the Goto at the end of
[dynamic-nway-start] doesn't seem to work. When I turn verbosity up a
bit, I get something like this in my error log:
== Channel 'SIP/SWG-0085a180' jumping out of macro
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
2007 Jun 29
0
nway call
I'm using asterisk 1.4.5 , on Centos. My kernel version is 2.6.9-55.ELsmp
I've configured the nway call. I made entries in extension.conf, feature.conf, as per required.
I'm trying to make a 3-way conference with the 1 user myself ( using asterisk), and two others are PSTN line users.
I'm making a first call , then putting that person on hold by pressing **( as per feature.conf
2006 Apr 05
0
What does this error mean "app.c: Huh....? no dial for indications?"
Hi,
What does the following error mean:
Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications?
Here is the 'full' log around the error:
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to
agent '3002', on 'Local/510@default-6b6c,1'
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002
Apr 5 12:38:24 VERBOSE[22755]
2004 May 23
1
*** Asterisk Sunday News: Conferences on the phone and IRL - "in real life"
Here in Sweden, it's supposed to be springtime. A wonderful time of the year,
with sunny skies and wonderful weather. Almost summer. Today, it's not.
It's winter all over again with rain and only 3 degrees celsius outside.
Better to stay inside and write a weekly Asterisk newsletter :-)
This week's topics:
-------------------
* Looking beyond Asterisk 1.0/1.1 - what's up?
*
2011 Apr 11
0
update CDR fields after Queue
Dears;
I have been faced with a problem that I am not sure about how can I solve
it...
I my scenario there is a variable which will be ready just after the callee
had hanged up and the caller, which coming throw a Queue.
But the CDR fields are logged into DB just after the Queue application. so
the '*userfield' *field will remained Blank.
is there any way to suspend CDR write INTO DB