Displaying 20 results from an estimated 2000 matches similar to: "Feature codes not detected"
2010 Jan 22
2
Trouble getting feature codes to work
Hi,
I'm having trouble getting feature codes to work in Asterisk 1.4.21.2.
Features.conf contians this:
blindxfer=##
atxfer=*2
automon=*1
disconnect=**
I'm really most interested in getting disconnect to work so that I hear
"Goodbye" when I press ** during a call connected this way in my dial plan:
exten => 1,n,Dial(SIP/14168724765 at 6135551212-sw1|120|gtT)
exten =>
2010 Jan 21
2
Caller hang up not detected
Hi,
I'm having trouble getting Dial to exit when the caller hangs up in Asterisk
1.4.21.2.
I use a POTS line to call into the DiD given to me by VOIP service
provider. When the call comes in, I have the VOIP provider send it to
another POTS line. All this works fine however when the caller (me) hangs
up, the Dial command does not exit. The callee stays connected (and my
billing
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
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
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 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
2008 Mar 05
2
Transferring Unanswered Calls
Hi list,
I'm wondering if it's possible to transfer a call that is still ringing??? I
Have some Grandstream GXP-2000 and with the TRNF button it's impossible. So,
I've configured some keys to transfer the calls like this:
[featuremap]
blindxfer => #2 ; Blind transfer (default is #)
disconnect => *0 ; Disconnect (default is *)
;automon => *1
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 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]
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
2007 Aug 29
0
call pickup problem
i have TB instaled and i cant get call pickup when another phone rings
i tried ** , *8 , *8# , **+ext but nothing seems to be ok.on extention menu
i put call pickup=1 and call group=1 but nothing look at my
features.conf;
; Sample Parking configuration
;
[general]
; do not manually enter parkinglot config information, use the parkinglot
module
;
; the parking_additional.inc file is
2006 Apr 27
0
How can conference room can call out?
I have a group of local users. They want to participate on a conference
call to a PSTN line in USA.
To connect they need to enter a code there as well.
I want to use a local conference room, where LAN users and local users
can call in. The conference room should be connected to the conference
abroad.
features.conf has:
[featuremap]
blindxfer => #1 ; Blind transfer
;disconnect
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
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
2012 Oct 25
0
Asterisk 1.8 not playing parking slot announcement to parker
Just upgraded to 1.8, we use the multi lot parking feature by dialling *4.
We are not getting the parking slot announcement being played to the person
who parks the call, so it's impossible to tell which slot they've gone
into. Could someone check our config?
On Debian Squeeze using packages from
http://packages.asterisk.org/debsqueeze main (Asterisk
1.8.11.1-1digium1~squeeze)
2005 Jul 27
0
[PLEASE RESPOND] Supervised transfer over SIP to outside POTS lines
PLEASE RESPOND IF THERE'S A SOLUTION
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
2006 Jan 20
1
applicationmap
Hi -
I'm trying to implement the applicationmap stuff in features.conf, and I
can't seem to get it to work. I'm testing it out on 1.2.2 with Polycom
IP500s and Snom190s.
My features.conf looks like this:
[general]
parkext => 700
parkpos => 701-720
context => parkedcalls
parkingtime => 240
transferdigittimeout => 2
;courtesytone = beep