Displaying 20 results from an estimated 1000 matches similar to: "Attended Transfer with REFER"
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
2007 Oct 01
1
Odd one way RTP on SIP to SIP calls
Hi everyone,
I'm having an odd problem with one way RTP on SIP to SIP calls.
I have two SIP servers, one is an Asterisk and the remote SIP server
is a Nortel SIP server.
When a call comes to the Nortel server through the PSTN and is routed
to the Asterisk, audio is fine. Two way RTP and no problems. When a
SIP client registered on the Nortel server calls the Asterisk, the
Asterisk
2015 Jul 15
2
how to return a transfered call to the transferrer?
Hi all
Any of you guys could point me in the right direction?
I need to make that a blind transfer to return to the transferrer when the transferee does not answer.
Scenario:
. Miss Jane Doe, our front desk attendant, picks up an external call to
Mr. Smith;
. Miss Doe flashes, dial Mr. Smith's extension and then hangup;
. Mr Smith's phone rings until timeout;
. At this point, how
2007 Jun 18
1
atxfer attended transfer feature
I would like to know if atxfer is supported somehow
because there seems to be little documentation for
this feature. I know most people expect a good SIP/IAX
phone to do the job but I think it's nice to be able
to do attended trasnfers with a simple ATA-connected
analog phone. I have Asterisk 1.2/Freepbx and
features.conf has a line regarding atxfer and I set it
to *2 (Default). While # works
2010 Oct 29
1
Asterisk 1.8 and character sets and AMI
Hi,
Just tried upgrading to 1.8 and ran into two problem immediately;
1. Caller-ID behavior is different -- now when I set the caller-id
name to something with special characters (?, for example), the SIP
INVITE now has %C3%96 instead of the ? character. I've tried doing
Set(CALLERID(name-charset)=utf8) as well as iso8859-1, but it's always
the same behavior.
2. My AMI scripts have
2006 Dec 05
4
Attended Transfer
Dear List,
I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable
attended transfer feature. but i just can't do it work. I've already
set "atxfer = *" (and many other combinations) and all extensions on
extensions.conf have the t and T option. But when I'm going to test,
it doesn't work. Is there any other file that i have to configure in
order to
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'
2010 Sep 17
1
Attended Transfer does not release channels
Hi all,
i have the following setup
PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk
1.6.2.9 -> SIP -> agent
Does work quit fine - then agent does have the abibility to transfer a call
to a third party - the agent can initiate the transfer over a web interface
- it does generate a asterisk manager atxfer request...
So agent does initiate transfer - call
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]
2003 Sep 18
1
CDR of calls transferred via IAX[2]
Let's say i have a network of * boxes connected via IAX, one of them is a
"switch", one or more are the "gateways".
- An IAX[2] "customer" register himself on the "switch" (and gets an
accountcode for te purpose of cdr)
- The customer places a call to the "switch", the switch does some magic and
decides which "gateway" the call
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
2011 May 31
3
AMI buffering event output?
Hi,
I'm seeing weird behavior with AMI where no events are output until
some input is detected (can be an empty line), at which time all the
buffered output is spewed out at once.
I am maintaining multiple Asterisk installations, and with one
installation I have run into a weird buffering problem with AMI.
The version is 1.6.1.11 in this particular case, which I am running at
multiple
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 Nov 23
1
1.6.1.10 Music On Hold
Hello.
I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold
functionality has changed (or is bugged?).
I have Aastra 6757i and Aastra 6731i phones, and now when i press the
MusicOnHold button / change lines on the phone, MOH no longer starts. It did
this in v 1.6.0.9.
The invites received are exactly the same, only 1.6.1.10 doesn't ever start
MOH.
Is there some
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
2005 Jun 06
2
Features.conf - atxfer
I am trying out the new atxfer feature from CVS-HEAD. I set atxfer
equal to *7 and it seems to work OK. I am having a problem getting it
to work the way a receptionist would want. If an extension calls me, I
hit *7 and I hear the voice say "transfer". I dial another extension.
If the newly dialed extension goes to voicemail, I can't figure out how
to get the original call
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 Jan 16
2
CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.
Hello,
When I bridge an incoming and outgoing call (attempting to simulate
call-forwarding) I'm only getting one CDR -- that of the outgoing call.
A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone
on PSTN) and bridges the call.
The only CDR created is from B to C. I have even tried using Answer() and
ForkCDR() to get two CDRs, but to no avail.
I am starting to
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")