Displaying 20 results from an estimated 10000 matches similar to: "the correct way to setup a transfer with REFER in SIP"
2009 Mar 12
4
log to cdr each dialpan action, not only one record for each call
Hi to all.
What can i do if a customer needs to log in the CDR all the dialpan
actions related to a call?
I mean, not only the lastapp e the lastdata but all the dialpan actions!
I know that the actual CDR system store one record for each call (and
for billing purposes this can be correct) but in some cases the
approach needed is something similar to the queue_log.
I know that exists ResetCDR
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 =>
2009 Jun 10
1
problem with transfer application (REFER)
I'm experiencing some problem using the transfer()
application,expecially when a call in received from a queue.
I'm using Asterisk 1.4.22.1
This is my scenario:
; this is the piece of code in extensions.conf that place the call in
the queue when 1111 is called
exten => 1111,1,Answer
exten => 1111,n,Queue(2000|t)
;this is the piece of code that calls the user test when 2222 is
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 Mar 08
2
Fwd: add a new queue strategy: SBR
Hi., do you think that sbr policy in queue strategy will be useful?
Bye
---------- Forwarded message ----------
From: nik600 <nik600 at gmail.com>
Date: Sat, 7 Mar 2009 15:21:14 +0100
Subject: add a new queue strategy: SBR
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Hi to all isn't there any plan to add the Skills Based Routing
strategy in
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
2016 Jun 30
2
how to join 2 channels using AGI/AMI
this is the point, and the strange thing:
DTMF is set to rfc2833, but is working both on incoming and outgoing calls,
it is not working only on calls generated with the Originate AMI command,
or with the queue member that point to Local dialplan, as you suggested
2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>:
> Looking at your logs it looks like you may need to
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 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
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 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
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
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.
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
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 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]
2008 Nov 18
4
busy-level / busy-limit Asterisk 1.4.22
Hi to all
the busy-level / busy-limit setting in sip.conf is available for
Asterisk 1.4.22 ?
This is a piece of my sip.conf:
[202]
type=friend
secret=202
host=dynamic ; This device registers with us
username=202 ; Username to use when calling this device before registration
limitonpeers = yes
call-limit = 2
busy-level = 1
The directive busy-level is ignored....
I've also tried
2016 Jun 30
2
how to join 2 channels using AGI/AMI
thanks John
yeah, your approach is much siple, i've tried it but i'm not able do detect
DTMF tones.
it seems that on calls that i receive DTMF tones are handled correctly, but
on calls generated from Asterisk to the world when the called side sends
some DTMF digits they are not detected:
-- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in
new
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 Jan 15
1
problem with PlayDTMF: no error but no tone
Hi to all
i'm using PlayDTMF with AJAM, after the authentication, i make a
request like this:
host:8088/asterisk/mxml?action=PlayDTMF&Channel=SIP/200-sdadsadkioah&Digit=1
the result is:
<ajax-response>
<response type='object' id='unknown'><generic response='Success'
message='DTMF successfully queued' /></response>