Displaying 20 results from an estimated 20000 matches similar to: "Indication of transfer on display"
2011 Apr 25
1
Transfer beep w/ Polycom phone
Hi all.
When a user transfers a call by pressing the "transfer" soft button on their
phone, I'd like it to "beep" at them when the transfer is complete. I've got
it turned on in features.conf:
xfersound = beep ; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
However, it seems that
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
2007 Jun 04
0
no ringing tone making attended transfer whith an IAX client
Hi
I have configured attended transfer in features.conf like this
[general]
parkext => 70 ; What ext. to dial to park
parkpos => 00-99 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
parkingtime => 300 ; Number of seconds a call can
be parked for
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
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:
2009 Jun 10
0
Problem with attended transfers
I need attended transfers, but I do not have time to talk to another
extension and see if they accept the transfer, my features.conf is:
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
parkingtime => 220 ; Number of
2011 Mar 31
1
Transfer feature dialing out after one digit
Because some users have requested transfer beep confirmations I've
switched our phones over to using the asterisk transfer feature instead
of the built in transfer functions of the phones. While testing it was
working fine, but I changed something in features.conf and suddenly any
time I hit transfer (*2), I can only enter one digit before asterisk
immediately tries to dial that extension.
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
2007 Oct 03
4
IAXy and hook flash transfer
In features.conf, I have uncommented the transfer features under feature
map, but I still cannot transfer using a POTS phone on an IAXy adapter.
I think I am missing something here.... Any help is appreciated.
Here is features.conf:
;
; Sample Parking configuration
;
[general]
parkext => 700 ; What extension to dial to park
parkpos => 701-720 ;
2009 Jan 22
1
Zap connection problem
Greetings all,
I'm trying to connect to an AT&T teleconference, but the
call is never marked as ANSWERED by asterisk and therefore won't bridge and
continue. The only work-around I've come up with so far is to dial like
this:
Exten => 744,1,Dial(Zap/g1,,p)
The "private" mode keeps the line open without trying to do a bridge, but
requires the
2006 Feb 23
0
Features set in the features.conf stopped working after upgrade.
Hi,
I recently moved all of my conf files over to a new Asterisk 1.2.4
server and every works except the features enabled in features.conf. Was
there a syntax chnage in 1.2.4? Or is there something else... Here is my
features.conf:
********************
[general]
parkext => 880 ; What ext. to dial to park
parkpos => 881-890 ; What extensions to park calls on
context
2006 Mar 19
0
Transfer to specific park number
Hi
I'd like to allow users to transfer a call to a specific park number. This
way, the receptionist can tranfer a call park for ext 100 at park number
7100 etc...
It seems like this should be fairly simple using the Park(ext) app but it
doesn't work for me. No matter what I extension I use, the system just
picks the next available park number. I've simplified my dialplan for
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
2009 Dec 14
0
pickupexten on chan_dahdi
Hi,
I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here my settings:
chan_dahdi.conf
[trunkgroups]
[channels]
context=default
switchtype=national
facilityenable=yes
rxwink=300 ; Atlas seems to use long (250ms) winks
; where the ring cadence is changed *after* the callerid spill.
usecallerid=yes
hidecallerid=no
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)
2017 Feb 16
2
Beep on Attended Transfer
Hi,
During an attended transfer using the SIP phone feature buttons, I'm getting a few complaints from recipients that they can't tell when the call they are receiving has been transferred.
Is there any way (even if it's complicated) to generate a beep tone to the recipient of the transferred call when the transfer is completed?
I know you can do this with DTMF codes but they want to
2010 May 12
1
Voicemail() app not available?
Hi all,
I have a demo machine I'm running up on Lenny - it has the packaged
Asterisk version installed (1.4.21.2+stuff).
I'm trying to add an extension to leave a voicemail message, just with
Voicemail(1234), which I've done before (on 1.2 at least), but it's
saying "no application 'Voicemail' ".
"module show like voi" shows
2005 Jul 25
1
"Cannot native bridge" on licensed G729
Hi folks,
In an effort to save bandwidth (our 7905s run over a WAN) we've
switched from ulaw to g729a. We purchased 4 licenses from Digium (4
SIP clients, low call volume), and they seem to have been accepted:
[codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator)
== G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e
== Found license
2017 Jun 12
2
OT: Explain where mailing list bouncing comes from ?
Ditto; a Gmail issue?
Andrew
On 12 June 2017 at 16:00, Marcelo Terres <mhterres at gmail.com> wrote:
> It is happening the same with me.
>
> Regards,
> Marcelo H. Terres <mhterres at gmail.com>
> IM: mhterres at jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
2011 Apr 26
0
play audio file to destination SIP channel on attended call transfer
Hello List,
Please help with the following problem,
I have a situation, where I need to play an audio announcement to the caller SIP channel once an attended transfer is successful. The attended transfer is done from client. I can see a transfer event in AMI. I am not using 'T/t' option in dial() command. The transfer is completely on client side using SIP signaling.
1. A calls B
2. B