Displaying 20 results from an estimated 5000 matches similar to: "How to get _out_ of an attended transfer?"
2008 Feb 27
0
Attended transfers and orginal caller ID
Greetings list,
Have there been any further developments recently regarding presenting the
original caller's caller ID to SIP devices after an attended transfer? I've
googled around on the topic, but most of the threads I've found (some from
this very list) are all dated back in mid-2006 and I wondered if there have
been developments on the issue.
To recap, the desired behaviour
2006 Oct 12
1
Attended transfer hanging PRI channel
The attendant attempts an attended call transfer (all phones are IP501).
The attendant pushes "hold", "transfer", dials the extension and
announces the call. When the attendant pushes "transfer" the second
time, the original call is lost.
The reason this is a big problem is that the PRI channel for the call
remains busy. Subsequent inbound calls on that
2002 Jul 24
2
ssh-keygen listing fingerprints little unclear
Since ssh-keygen is not listing the _types_ of keys I have in my file,
wouldn't it be a good idea to make the -t switch filtering out the
selected type of key when doing a listing with -l?
i.e. in this case I see both rsa1, rsa, and dss keys:
$ ssh-keygen -l -f ~/.ssh/known_hosts
1024 a9:4f:0b:b6:33:d7:d0:ad:6a:11:b4:57:25:7e:1e:f8 fluff.x42.com
1024
2005 Aug 02
0
Problem with attended transfers...
We have two Asterisk servers running CVS-HEAD (06/02/05 and 06/28/05).
Most of our calls are either incoming or outgoing to external (PSTN
or non-Asterisk) numbers, and only our internal users can initiate the
transfer. Only half of the attended transfers work. It goes like
this:
1)Extension 8123 calls number 19876543210
2)During the call, extension 8123 dials *2 to do an attended
(non-blind)
2009 Sep 21
0
Asterisk 1.6 dynamic agents
I've downloaded and installed Trixbox 2.8 (asterisk 1.6) ..I encounter 2
problems for dynamic agents login and logout -
1. When agent from sip phone dials *11 , he is prmpted to enter extension
number first - but if he feeds the extension number, asterisk doenst allow
him to login but if he enters the AGENT ID - he is given instant access and
then he can enter queuenumber* to enter the
2007 Feb 07
0
Connection problem w/ Attended Transfer
Hi all,
I'm new posting here, though not to perusing. I'm having an issue
with attended transfer and was wondering if anyone had heard of the
problem/had any suggestions... Apologies in advance if this post is
excessively newb-oid.
- An incoming call C is passed to A, a POTS telephone connected via a
Handytone 286 ATA.
- A presses atxfer key, then dials B, a Win XP laptop running
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
2009 Oct 26
1
Cancel attended transfer
Hi folks,
I have a simple question regarding attended transfers. I have some
queues where agents take calls and I have configured attended transfers
between queues. That is, the agent dials the attended transfer extension
that routes it to the aproppiate transfer queue where the second agent
answers and they both talk for a while. Finally the transferrer leaves
the call with *, connecting
2017 Apr 04
2
Hiding partitions at boot time
Hi,
My problem is I have a particular system which has Linux (LFS) and
Windows XP on one drive and Windows 10 on a second drive.
I use extlinux to boot my systems and all three systems boot as
expected, however in use I have found unsatisfactory interactions
between the Windows XP and the Windows 10 systems.
My question is:
Is it possible on booting one of the Windows systems to hide
2007 May 14
0
How is Context Determined when Transferring a Call?
When trasferring a call, how is the context determined?
When using a zap device, and the DTMF code for blind or attended transfer is
entered, does the tranfer originate at the context the zap device is set to
be in, or does it originate from where the outside call being transferred
originated in, or the context the current call is in?
I ask because I am seeing strange behavior when trying to
2009 Jun 15
1
Opinion on Attended transfer in features.conf
Hi,
In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind
Transfer when transferer hangs up before callee answers :
- in Blind Transfer, caller (ie transferee) is hearing Ringing tone when
callee's phone is ringing
- in Attended Transfer, caller (ie transferee) is hearing Music On Hold when
callee's phone is ringing
- in Attended Transfer, if callee don't answer
2005 Feb 16
0
Attended xfer
Does anyone know if the attended transfer in CVS head works with
app_queue (and more importantly, chan_agent ?)
This is the only thing stopping me from deploying the attended transfer
patches.
Cheers,
Ben
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mark
Benson
Sent: 16 February 2005 14:44
To: Asterisk
2018 Aug 08
2
Queue breaks Dynamic_Features on Attended Transfer
Hi,
I think I've identified an issue and just want to check before completing a bug report.
Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code.
If AgentA performs an attended transfer of a call from a queue to AgentB, the feature code no longer works.
Cases that do work are as follows...
Calls using both Queue() and
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
2009 Jul 16
1
Stop recording on SIP attended transfer
Hello,
We have an application where operators will sometimes take an incoming
call from a queue, then contact an outside line, do a consultation,
and finally do a SIP attended transfer to join the two parties
together. We'd like to record the incoming caller's conversation with
the operator and the attended part of the outgoing call, but not the
unattended part, after the transfer has
2005 May 11
0
Re: samba Digest, Vol 29, Issue 14
Este correo no es de yanier
----- Original Message -----
From: <samba-request@lists.samba.org>
To: <samba@lists.samba.org>
Sent: Wednesday, May 11, 2005 1:18 AM
Subject: samba Digest, Vol 29, Issue 14
> Send samba mailing list submissions to
> samba@lists.samba.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>
2009 Jul 27
0
Emulating attended transfer through the dialplan
Hello,
I'd like to implement something similar to an attended transfer, but
with a little more control (I'd like to be able to use MixMonitor and
StopMixMonitor to control the call recording, set the account code,
etc. I'm on Asterisk 1.4.26.
All of the ways I have seen to do this are complicated plans using
MeetMe and applicationmap features, and playing with those over the
2009 Sep 05
0
Remote attended transfer
Hi,
I'm having problems with sip remote attended transfer using 2 asterisk
boxes (same version, latest 1.4.X). Whenever I transfer from a call
from box A to a call on box B, one call leg of the transferring phone
is not disconnected (the one that is normally dropped by server side,
phone disconnects the other one). The same situation works perfectly
with local attended transfer.
Is anyone
2013 Sep 16
0
Transfer rights for attended transfers
Recently I asked a question about possibly unwanted calls due to extended transfer rights after
attended transfers using DTMF sequences
(http://lists.digium.com/pipermail/asterisk-users/2013-September/280536.html). Obviously,
transferring with SIP INVITEs (hold + transfer keys) is not immediately affected by the this,
but it is not always possible to enforce this.
Meanwhile I have changed the
2015 Jan 30
0
Remote Attended Transfer
Hello,
I'm trying to find more information about this Remote Attended Transfers,
as is explained in
https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers
for Asterisk 12 using pjsip stack
Was Remote Attended Transfer implemented in previous versions of Asterisk
(versions without PJSIP, Asterisk 11 and previous)?
Where can I find configuration examples to do it work