Displaying 20 results from an estimated 1000 matches similar to: "Trouble getting feature codes to work"
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
2010 Jan 21
0
Feature codes not detected
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 =>
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
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
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 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
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
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?
2012 Nov 05
6
err: Could not request certificate when I run "puppet device"
1. I get the following error when I run “puppet device’
err: Could not request certificate: Could not write
/var/opt/lib/pe-puppet/devices/certname/ssl/private_keys/certname.pem to
privatekeydir: Permission denied -
/var/opt/lib/pe-puppet/devices/certname/ssl/private_keys/certname.pem
Any thought?
Thanks,
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2003 Mar 06
3
Some questions.....
Hi all,
I need some help, advice or whatever you can explain to me because I haven''t
got a clear idea about how to do the following assembly, and I''d be very
grateful if I got some help from an expert like you.
I''m trying to build a system which represents the following:
I''ve got a hosts unit (host1, host 2, ...) which have IP in the network 192
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
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
2009 Sep 16
3
Music on Hold
Hi,
I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
1.6.1.4. The call goes on hold, MOH is started, and then stops right away.
Here are the files both of type .raw:
Tsunami*CLI> moh show files
Class: default
File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2
File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1
These files
2006 Apr 05
3
queue issue
Hi,
I have several queues configured at my call center for different support levels.
Today, something weird happened:
- A client called queue 1 and was answered by an agent
- The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf
- The user transferred the client to another Queue, by using the second channel and the XFer key of her
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
2006 Mar 17
7
problems with emailing voicemail
Hi,
I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05)
running on RedHat 9.0. Everything's been great but a couple of days ago, we
all stopped receiving emails of our voicemail. There's been no changes to
our configuration
I bet I'm expereiencing a Linux problem rather than an Asterisk problem, but
because I know only as much Linux as required to get