Displaying 20 results from an estimated 2000 matches similar to: "Transferring Unanswered Calls"
2005 May 30
4
R: R: R: AT-320 + supervised transfer
I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes.
So i can only update asterisk with CVS and try atxfer.
Thanks for all
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill
Inviato: luned? 30 maggio 2005 18.40
A:
2009 Jun 16
1
Unable to use # as feature key prefix
Hi folks,
I was using the following featuremap:
blindxfer => *1
disconnect => *9
atxfer => *2
parkcall => *7
automixmon => *0
and everything worked.
Then the need arouse to use some features like automixmon during
a conference, but MeetMet has the * key bound to the
(admin) menu. Thus, in order to enable features like automon and
transfers even during a conference, I
2005 Jan 25
2
New native assisted transfer (atxfer) usage info required
Hi, I would like to use the new atxfer (native assisted transfer, see
mantis item #3241) , but I've partially been able to
make it work.
I can receive a call and then having the caller hear MOH while talking
with another extension (the one I want to transfer to), but then I can't
make the caller and the trasferred talk hanging up or pressing any key
combination I'm aware of.
My
2009 Feb 09
1
Transfer Asterisk 1.6 Telephone IP
Hi List.
I have a small problem in using the transfer key transfer of IP Phone in
Asterisk 1.6, I think I spend some detail in the configuration but can not
find.
What happens is, when I do a transfer using the Transfer button, the
phone, does not play the music on hold, which is waiting on the phone is
silent, and I have the same settings on a 1.4 server, and the music plays
correctly when
2004 Aug 03
6
features.conf
Is features.conf included in the cvs as of 8-1-04? I have updated, but am
not seeing it?
2005 Oct 17
1
Call transfer - atxfer
Hi,
I try to set up attended transfer in my Asterisk Box . My
features.conf look like this:
[general]
parkext => 100
parkpos => 1-5
context => parkedcalls
parkingtime => 100
transferdigittimeout => 3l
courtesytone = beep
xfersound = beep
xferfailsound = invalid
featuredigittimeout = 500
;adsipark = yes
pickupexten = *8
[featuremap]
atxfer => *2
blindxfer => #
disconnect
2005 Mar 07
2
Call transfer questions
Dear all
I am trying to work out how make call trasfer work the way I want is
I am the called party I want to transfer a call so I press # and enter the
ext but then it disconnects me
this is a blind transfer
how do I make it so its not a blind transfer so i can talk to the person
before i transfer the call...and go backl to the orig caller if the
transfered to ext doesnt answer....
also can
2005 Jul 26
1
Supervised transfer over SIP to outside POTS lines
Hello all,
I am trying to complete my dial plan and have come up with an
interesting situation. My configuration is set up with 12 xlite SIP
clients on SUSE linux workstation. They are calling out via 10 analog
lines, TE110P->rhino 24 fxo.
It all works and dials out great ... but ... this unit was brought in to
handle the "global" office. So the help desk support on the Suse
2005 Jul 20
1
getting problem in Picking up the parked call
Hi all.
I am trying following scenerio for call park & pickup.
voice is flowing established between B & C, after call-pickup (
instead of A & B ).
can anyone please clarify why it is happening like this, ( or ) do i
need some more configuration for park&pickup ?
A
B
2005 Jul 01
1
Attended transfer works for caller, not for callee
Hi,
I have been trying to enable attended transfer for callee. When the
callee pressed *2, DTMF tone was heard by the caller. But when the
caller pressed *2, attended transfer started. It's strange.
I used two SIP phones. My Asterisk version is "Asterisk CVS-HEAD built
by root@router on a i686 running Linux on 2005-06-27 06:07:18".
In features.conf, I have:
[featuremap]
2006 Mar 16
1
Attended call transfer with GXP-2000
Can someone explain me attended transfer with Grandstream GXP-2000?
Hitting TRNF button, I get:
Dial number (BLIND) or
Select line (ATTENDED)
What's the exact meaning of 'Select line'?
Thanks
Mimmus
2007 May 25
1
Problem with call parking
I am trying to test the call parking, but It doesn't fonction :(these are my config files.extensions.conf:include=>parkedcallsexten => 4000,1,Dial(SIP/4000,60,tT)exten => 4001,1,Dial(SIP/4001,60,tT)exten => 4002,1,Dial(SIP/4002,60,tT)In features.conf:[general]parkext => 700 parkpos => 701-720 context => parkedcalls [featuremap]blindxfer => # disconnect => *0
2005 Jul 04
3
Call Transfer using SIP clients
Hello all,
First of all, let me apologize about the length of this message, but I suppose
it was necessary to include the details.
I've spent quite some time already trying to get the call transfer function to
work on my Asterisk installation. Let me first describe the general situation
of the setup I am using, so you might be able to pinpoint the cause of the
problem.
I'm currently
2008 Apr 03
1
Hearing "transfer" during call
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word "transfer", I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:
sip.conf:
2013 May 17
0
Temporarily features (transfer) off during Read
Hello all.
Dialing with tT options and function Read (to prompt number) has a
trouble for me.
Can I temporarily features off during Read?
features.conf:
[featuremap]
blindxfer => ## ; Blind transfer (default is #)
atxfer => ** ; Attended transfer
I try:
exten => s,n,Set(LOCAL(tmp_atxfer)=${FEATUREMAP(atxfer)})
exten =>
2005 Sep 27
1
blindxfer & atxfer not working?
I'm wondering whether there's a problem with the blindxfer and atxfer commands.
I was using Asterisk STABLE and pressing the # key to transfer calls
worked fine, except of course when you called up FedEx and they asked
"Enter the number of packages, followed by the Pound key".
I found on the wiki
(http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf)
that
2005 May 31
2
R: R: R: R: AT-320 + supervised transfer
Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ?
This is my Dial()
exten => 605,1,Dial(${GIORDANO NAT},60,Ttr)
I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press
2006 Mar 01
9
Asterisk transfer conflict
I have a problem with my Asterisk system.
When I use my phone to call my office mailbox I have to end my password with
#. (The office do not use Asterisk)
" # " is also used as a transfer button on my asterisk, so when I press it I
hear my Asterisk trying to transfer the call.
Is there any way to change the transfer button or remove it ?
Fredrik
2005 Mar 15
2
Asterisk retains DTMF Control Even whenan External IVR System is dialed
Eric Wrote:
-----------
The trick is not to use options you don't understand. "show application
dial" will show you what the t and T options are for.
Most people use the transfer feature of their phone, rather than using
the T/t hack on the Dial line.
Sounds like you are using CVS-HEAD and so will have to configure stuff
in /etc/asterisk/features.conf.
/Snip/
Eric,
Thanks for
2007 Oct 14
1
Problem: features (from features.conf) not available if call was originated by manager API or call file
Hello asterisk-users,
I setup my asterisk to support several features like
automon,blindxfer,atxfer,parkcall etc. by using features.conf and the
global variable
DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in
extension.conf. Every Dial() command in my diaplan has the appropriate
parameters out of {tTkWwW}.
For calls from my SIP phones everything works fine. Pressing #1 will