Displaying 20 results from an estimated 200 matches similar to: "Dialplan help - MeetMe (or ChannelRedirect) and call monitoring"
2007 Apr 10
1
Dialplan help - MeetMe and call monitoring
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio channel but not speak (it's a monitoring application for a callcenter)
The person in charge of monitoring cannot use ChanSpy or
2007 Apr 17
4
Using meetme like call
hi all, I have a little question about meetme in Asterisk.
One of my client ask me that all call can, if is necessary, become
conference for 3-4 user during conversation.
I think that are 2 way for make this:
1- all call (instead if the users are only 2) are conference
2- using n-way call
(http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO)
I decide to implement the first way because
2011 Jun 02
0
ChannelRedirect
Hello,
I am implementing a small ACD system on Asterisk 1.6.2.17.2 I need help with ChannelRedirect. I have a caller connected to an agent. The agent may request additional help by consulting another department. I can't use manual process with blind or directed transfer as the agent have many different numbers to dial. The message with the proper dial number is coming from the host. I got
2007 May 16
1
MeetMe and ChannelRedirect
Hi,
i'm trying to implement the following scenario:
- A user calls number 700
- Asterisk then dials to extensions 100, 200, 300, 400 and 500
- And then bridges all calls to a conference room
I tried to use MeetMe and ChannelRedirect, but seems that after
channel redirect nothing more is executed. So, this seem to work for the
caller and first called, but the others
2007 Apr 01
1
Asterisk 1.2 and res_perl - lock that leads to weird behaviour
Hi guys,
as I wrote in a previous thread I was experiencing dropped audio
(apparently randomly) and SIP + IAX peers getting REACHABLE /
UNREACHABLE without reason, servers were in the same LAN.
Investingating deeply in the problem I also noticed that 'show channels'
command on the CLI, sometimes were returning strange results, for
example it wasn0t showing some channels I was sure
2010 Jun 17
1
applicationmap and ChannelRedirect
Hi,
I'm struggling with a feature in my home phone setup. I have several
phones using both SIP and SCCP. What I try to do is to create a dynamic
feature that works similar to the blindxfer feature built into Asterisk.
What I want is the possibility for the called part to push a number
sequence (for example *#) to redirect the callee to a fixed extension or
(for example *123#) to redirect the
2007 Mar 31
4
Sponsored development - Monodirectional audio handling
Hi Guys,
we're needing a special implementation on Asterisk
Our intention is to contribute the development and share back the code
to Asterisk community
Here is what we need:
- An option to Asterisk Dial command which, if used, when calls is
answered gives monodirectional audio
(Caller should hear the called party but not vice-versa)
- A DTMF sequence (maybe handled in features.conf) for
2007 Jan 26
0
Asterisk dropping audio
Hi all,
I have a problem with Asterisk dropping audio.
While in call, audio gets dropped for a while (from 5 to 60 secs, and
obviously users often hangup, this means that I'm not sure the audio is
always coming back after 60 secs), in the meantime the call remains up
and no SIP signalation is generated.
It happens randomly so it's very difficult to debug.
I cannot see common
2007 Mar 23
3
SIP/IAX peers UNREACHABLE and audio loss
Hi all,
I'm having a problem with some Asterisk servers interconnected with
each other using IAX (I also tried with SIP without solving the problem)
Sometimes, with apparently no reason, some peers become UNREACHABLE
(I have qualify=yes in iax.conf) and REACHABLE again as soon as
another qualify test is made.
Our users are also complaining about audio loss during their calls,
apparently
2007 Jul 16
0
Dial and option G
Hi all, I use the G option in my dials for redirect both parties in the
conference.
There is a way for auto-include in a conference other parties that first
two without using AGI?
I try with:
[from-internal]
exten => 9999,1,Dial(IAX2/DIP02/9999||G(fromiax^9999^1)
[fromiax]
exten => 9999,1,MeetMe(9999,qdxAa)
exten => 9999,2,MeetMe(9999,qdx)
exten =>
2007 Aug 15
1
CDR billsec greater than duration
Hi all,
I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1
Doing a select in the CDR table I noticed there are some calls with
billsec greater than duration, duration is always 0 in those calls.
How can this happens ? Am I missing something ?
Tnx in advance
Regards
Edoardo Serra
WeBRainstorm S.r.l.
2009 Jun 05
5
How run AsyncAGI commands in background
Hi all,
I have an external application commanding asterisk by AMI and AsyncAGI. I
also have a dialplan like this:
; AsyncAGI extensions
exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN});
exten => _8.,n,AGI(agi:async);
exten => _8.,n,Hangup();
; Meetme extensions
exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT});
exten =>
2015 Jan 02
1
Help in building R with minGW
Dear R users,
I would need some help in building R using minGW in windows 8.1. After
using the command *configure *(./configure --enable-R-shlib
--with-readline=no --with-x=no), I use the command *make. *This results in
the following error:
[...]
make[2]: Leaving directory `/home/Edoardo/r-3.1.2/src/nmath'
make[2]: Entering directory `/home/Edoardo/r-3.1.2/src/unix'
make[3]: Entering
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 =>
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
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
2007 May 06
2
Call waiting tone when calling a busy station?
Hello,
When dialling a SIP phone which is already in a call the caller hears a
"regular" ringing tone and does not know that the called party is engaged in
another call. Is there a supported way inside SIP to tell the calling party to
play a stuttered ringing tone?
Thanks! __Yehavi:
2013 Jan 02
3
Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and ChannelRedirect
3. The local extension looks like something this:
exten => _X.,1,Dial(SIP/${EXTEN},60,?)
exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..)
I have looked through all arguments of Dial
2010 Jul 28
1
Redirecting a call to another extension using asterisk java
Hi,
My problem is as follows.
I registered an xlite client and dialed 1500 extension. In the
extensions.conf i set as follows.
exten=>1500,1,AGI(localhost//
hello.agi.
This hello.agi when connected plays a greeting message. Once this is
connected from the script i want to transfer the call to another extension
say 1600. How do i achieve this. I tried using ChannelRedirect but it didnt
work.