Displaying 20 results from an estimated 4000 matches similar to: "Redirecting a call to another extension using asterisk java"
2010 Jul 27
2
urgent:how to transfer a call using asterisk FAGI
Hi,
I have xlite registered with a user. Now i dial an extension say 1500 which
has the dial plan as follows.
exten==>1500,1,AGI("localhost//hello.agi"
So when i dial extenstion 1500 the script hello.agi is invoked which in turn
plays a welcome message. I now want to transfer the call now to operator.
How can i achieve this???Please help me in this regard as this is very
urgent.
2010 Jul 27
2
How to transfer a call to operator using FAGI asterisk
Hi,
I have xlite client registered with a user. Now i dial an extension say 1500
which
has the dial plan as follows.
exten==>1500,1,AGI("localhost//hello.agi")
So when i dial extenstion 1500 the script hello.agi is invoked which in turn
plays a welcome message. I now want to transfer the call now to operator.
How can i achieve this???Please help me in this regard
Thanks &
2010 Aug 04
2
How to record a file and play some other file at the same time
Hi,
I have an xlite registered with asterisk server. When i dial a number AGI is
invoked. and in this we are running to threads one to record files and one
to play files. So i dialed the extension and i started recording and playing
at the same time. On the xlite i m getting an indication when recording my
voice and at the same time i could see playing the other file too. But in
the directory
2010 Jul 29
2
How to record and playback at the same time
Hi,
we are using Asterisk to record and playback. Both services are working well
independently but it seems we can't start playback of a file while we are
still recording it, even if the file is already in the hard disk.
Is it possible to playback while recording the same audio file? Or is there
a way to enable it?
Regards,
Jahnavi.
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2010 Aug 23
1
How to do barging using asterisk server.
Hi All,
I have this requirement. I have an xlite client registered with asterisk
server. And from this when i dial an extension say xxx it invokes an AGI
script which gives me a series of instructions like "Welcome to this IVR
system. Press 1 to trade 2 to sell....and so on". I want to stop this and
press 1 or talk even before the prompt finishes. How to achieve this. I was
told that
2008 Mar 18
2
call screening feature
Hi,
I have our software with SIP running on it.I configured asterisk server as
proxy. How do I implement the call screening features(incoming and outgoing)
using asterisk server.Please suggest me how to proceed on this.
Thanks & Regards,
Jahnavi.
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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 =>
2010 Jul 26
2
No audio using xlite
Hi,
I installed asterisk server in my linux box. I configured a user 1000 using
xlite and registered with asterisk server in the same linux box. I
configured one more user 1001 in other box and this user also got registered
with asterisk. But i am facing two issues here.
1. When a call is made from 1001 to 1000 i could see an incoming call
blinking but no audio flow is observed.
2. When i made a
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
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
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
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
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
2020 Feb 05
1
Hangup hook to put back a call into a queue
hi,
I hope someone can help me:-)
we’ve got a freepbx server. there are 2 special extensions (2001, 2002).
if someone calls this extensions (or a call is forwarded to these
extensions) and these extension hangup (not the caller party), then we’d
like to put the calls back into a queue (1000) and wouldn’t like to hangup.
I read your description about hangup hooks:
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
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'
2009 Jan 25
2
Zaptel transfer using any button or code, but not flash hook
Hi List;
I need to do a call transfer using analoge phone connected to fxs, but I do not need this to be done using flash hook, let it to be using the # or * or any code, but how I can configure that this code is for transfer? Also, I do not need the flast hook to be used for trasfer as it cause usually a confusion to distinguish between the hangup and the call transfer.
Any advise?
Regards
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
2007 Apr 10
0
Dialplan help - MeetMe (or ChannelRedirect) 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