Displaying 20 results from an estimated 3000 matches similar to: "Emulating attended transfer through the dialplan"
2009 Apr 14
4
Ignoring time spent waiting in queue in CDR
Hello,
I'm working on an Asterisk configuration for a call center, and they
bill based on the time spent talking to an agent, but not for any time
spent waiting in a queue. The CDR information contains the entire
duration of the call as billable seconds, including time spent waiting
in the queue. I would like the billable seconds to only include the
time spent actually talking to an agent.
2014 Aug 27
1
features.conf and mixmonitor stop and start
Hello,
I have a recording started in the dialplan with the MixMonitor application.
I want to be able to stop it during a call and maybe restart it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is running?
[featuremap]
automixmon =>
2005 Jul 27
1
Attended transfer not working (atxfer)
While on conversation with another party, I dial the atxfer key
sequence. Asterisk says "Transfer" then gives you a dial tone, while put
the other party on hold music. I dial the transferee number and talk
with the transferee, then I hang up and the other party must be
connected with the transferee.
But this doesn't work the transferee hears a beep. -- Playing 'beep'
2010 Mar 29
0
MixMonitor and StopMixMonitor
Hello list,
how does StopMixMonitor know which 'monitoring channel' to stop when
there are multiple conversations that are being monitored/recorded ??
I want to use StopMixMonitor in a macro, called from within
applicationmap (features.conf).
Jonas.
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2009 Jun 15
1
Opinion on Attended transfer in features.conf
Hi,
In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind
Transfer when transferer hangs up before callee answers :
- in Blind Transfer, caller (ie transferee) is hearing Ringing tone when
callee's phone is ringing
- in Attended Transfer, caller (ie transferee) is hearing Music On Hold when
callee's phone is ringing
- in Attended Transfer, if callee don't answer
2018 Apr 13
2
Disable blind and attended transfer during call
Hi
Is there a way to disable blind and attended transfer during a call.
I am trying this configuration but unfortunately with no luck:
- in features.conf
[applicationmap]
disabletransfer => 9*9,self,GoSub(disabletransfer,s,1)
- in extensions.conf
[incoming]
exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer)
exten => 99,n,Dial(Sip/alice,120,tT)
exten => 99,n,Hangup()
2009 Jul 20
0
No subject
I got this notion
monitor-format = wav49
wav49 presents much louder than regular wav and gsm in my experience
--
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, January 22, 2010 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
2006 Dec 05
4
Attended Transfer
Dear List,
I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable
attended transfer feature. but i just can't do it work. I've already
set "atxfer = *" (and many other combinations) and all extensions on
extensions.conf have the t and T option. But when I'm going to test,
it doesn't work. Is there any other file that i have to configure in
order to
2010 Jan 20
1
Setting MixMonitor options from Queue
Hello,
We are recording our calls to queues by putting the appropriate options in
our "queue.conf". This is all working properly.
We would now like to set the MixMonitor option to adjust the caller volume
(which is very quiet). With the regular MixMonitor application, we would
just add the "v4" option to make it much louder. I don't see a way to set
this option when
2006 Jun 11
3
JIAX status
HI,
Anyone knows the current status of JIAXclient?
I tried to recompile the sources available in sourceforge but
they reference a old java package that I was not able to find.
I tried to e-mail the author but seems that his account is no longer valid.
I in need of a java IAX client that could be loaded as an applet. I know
that
is a lot of viable SIP alternatives, but due to NAT/Firewall
2009 May 29
1
Attended transfer and dialplan
Hi,
How can you add specific statements into Asterisk dialplan (extension.ael,
...) for attented transfers ?
I can see Asterisk sending Transfer or Masquerade events through AMI (in
1.6.1) but I could use an external program to catch those events but I would
prefer to use dialplan instead.
Any idea ?
Regards
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2014 Jul 16
0
Function transfer RFC 5589
Hello,
I have the following scenario:
1. VoIP Gateway G400 connected to PSTN
2. Asterisk server 1 (working as IVR)
3. Asterisk server 2 (working as ACD, with several agents connected)
I have incoming calls coming from PSTN through the VoIP Gateway to Asterisk
server 1 (IVR). When the IVR ends working with the call, transfers it to
the Asterisk server 2 (ACD).
In Asterisk server 1
2009 Jul 22
2
Waiting for a call to complete with AMI Originate
Hello,
I'm using an AMI Originate command to send a fax. The fax is sent by
a script, and I'd like my script to send the fax, wait until it has
succeeded or failed, then exit with an appropriate error code (it is
driven by a mail system, so the exit code will tell the mail system
whether to retry the fax later).
The script works great if the fax succeeds, or if the line is busy or
2008 Jan 18
3
Circular links and backups
Hello,
I ran into an interesting problem earlier today. I have a Unix
machine I maintain in a largely Windows shop. They use Windows Backup
for their backups, and so I created a readonly share of the entire
filesystem with one user, "backup", who is an admin user. This lets
them back up the entire Unix machine by attaching to the "backup"
share, but nothing can be changed.
2005 Mar 26
1
Transferred calls CDRs
Hello!
I have been doing some tests with call transfers and I have been looking at
the CDRs that Asterisk generates.
Scenario 1:
A calls B
B answers and does a blind transfer to C (using # key)
C answers and talks with A
Scenario 2:
A calls B
B answers and does an attended transfer do C (using the phone's transfer
key)
C answers, B hangs up, and C talks with A
For scenario 1, the CDR shows
2006 Jan 05
1
Re: Has anyone tried using flash() in features.conf (applicationmap) - RESOLVED
Problem resolved.
This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as opposed to the dialplan scripts that I have seen that require tranferring the call, hanging up, and waiting for a call back. That was too confusing for my wife. Now all she has to do is pres *3 and it is done. No transfers. No hanging up. No dial back.
extensions.conf
[context]
exten =>
2013 Mar 07
2
Recording with MixMonitor and AGI
Hi,
I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan:
[macro-ccdev2-rec]
exten => s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten => _X.,1,NoOp(Will send call to ${EXTEN})
exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-rec]
exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})
2009 Jul 16
1
Stop recording on SIP attended transfer
Hello,
We have an application where operators will sometimes take an incoming
call from a queue, then contact an outside line, do a consultation,
and finally do a SIP attended transfer to join the two parties
together. We'd like to record the incoming caller's conversation with
the operator and the attended part of the outgoing call, but not the
unattended part, after the transfer has
2023 May 30
0
Can't stop Mixmonitor
Hi all
Using asterisk 16.25
I was trying to stop Mixmonitor using features. The code is executed but I
realized that I was executing StopMixmonitor from another channel so I opted to
use AMI.
When I call MixMonitor I store the channel name in a var and then I use
StopMixmonitor from AMI sending the stored channel name as parameter.
What I've seen is that the app returns failure and going
2008 Nov 22
5
CDR Desgin
I've taken the liberty of starting a new thread to discuss the design
of the Asterisk CDR mechanism. The discussion has been kindly
initiated by murf putting together a proposal:
http://svn.digium.com/svn/asterisk/team/murf/RFCs.
After reading the proposal I still don't think it's the right way to
go. To my mind adding more channel variables increases the complexity
in a situation