Displaying 20 results from an estimated 3000 matches similar to: "atxfer attended transfer feature"
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
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 Mar 25
49
atxfer
Hi list,
This wll be my first post, so I want to thank all the developers for the
great product they have created.
Now, the question,
I have installed asterisk 1.05 on debian sarge (binary package)
with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100)
This all works fine, exept for som echo on the ISDN channel, but I'll
replace the I4L card with an AVM-C4 card next
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 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]
2005 Mar 03
2
Attended Transfer (ATXFER) with CVS asterisk r 1_
Hi,
I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I
would like to use the atxfer function but is not included in the stable
asterisk.
Is there a way to include it in my version of asterisk: I did no used the
last cvs because I can't compile the chan_capi .in it. :(
Bye
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'
2005 Jun 14
2
Features.conf for secretary function
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer => *0
blindxfer => #0
I completly restart asterik, and not just make a RELOAD. But during a
call, when I press # it runs a blind transfer and if I press * I am
disconnected.
I am using the CVS version of * get as explain here
2008 Oct 23
1
Atxfer Command
Hi,
We are testing new Asterisk 1.6.0.1 because we would like to use the
Attended Transfer feature and we are trying to use the new action Atxfer
developed for AMI.
As far as we know, it is suposed to be in this release as it can be read
in Digium's changelog
/New command: Atxfer. See doc/manager_1_1.txt for more details or manager show command Atxfer from the CLI/
But, when we try to
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 Mar 15
1
blind xfer works atxfer doesn't...help!
Hi all
I am having problems with atxfer
if I do the extact same thing with blind xfer it works fine
when I hit press #2 (defined in conf for atxfer) i get "transfer"
I dial the number I want and i get the following on the console
-- Playing 'pbx-transfer' (language 'en')
-- Executing Dial("Local/18005558355@jesnjer-f97a,2", "/18005558355")
2005 Jan 12
0
Attended transfer problem with Atxfer
Hi everyone,
I'm trying the new atxfer functionality. All seems to work fine at the
beginning, but there is no audio between the party at the end of the
transfer. Plus, after that, even normal calls won't work well (they
can't hangup!).
I'm using the Asterisk CVS from 2005-01-10 with Asterisk-OH323.
Here is my dialplan:
[default]
exten => h,1,NoOp(bye)
exten =>
2018 Apr 13
2
Disable blind and attended transfer during call
Hi
Is there a way to disable blind and attended transfer during a call.
I am trying this configuration but unfortunately with no luck:
- in features.conf
[applicationmap]
disabletransfer => 9*9,self,GoSub(disabletransfer,s,1)
- in extensions.conf
[incoming]
exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer)
exten => 99,n,Dial(Sip/alice,120,tT)
exten => 99,n,Hangup()
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
2006 Nov 15
2
some questions about atxfer usage
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an extension:
- The extension rings during 15 seconds and the call returns to the
"transferer". Is there any possibility to recover the call before the
timeout of 15 seconds expires?
I mean, I would like
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 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
2015 Jan 27
1
Inline transfer
Hello,
while most of the physical phones have keys to handle attended and blind
transfer, most soft phones have no support for it. Asterisk offers a
"featuremap" to assign a key to blindxfer and atxfer and they work fine if
the call is still in the same starting context, but if the call has moved
in another context, then the new call will be started from such context
with unpredictable
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 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: