Displaying 20 results from an estimated 10000 matches similar to: "Bug in attended transfer or as expected?"
2005 Jan 18
14
Attended call transfer
Hi All,
Does any one know if attended call transfer has been added into the STABLE
release of asterisk yet? Potentially using a mix of phones would create
confusion in a user base, any ideas on attended transfer or how to achieve
this / mods to dial plan etc would be greatly appreciated.
I have been on an almost vertical learning curve with Asterisk and Linux for
6 months this is just
2005 Apr 25
5
UK (english) sound files
Hi all,
After many complaints (including car manufacturers saying the american
prompts are unexceptable, EEEK) I started on a quest for real "English"
asterisk prompts.
The only one I have found is here >>
http://www.g7ltt.com/VoIP/vmfiles.html
<http://www.g7ltt.com/VoIP/vmfiles.html>
And no nothing else on the WIKI looked helpful (e.g. only American voice
actors etc)
2006 Feb 17
3
MixMonitor and command
Has anyone had any success using the MixMonitor() plus "command" as
nothing I have tried works.
I am using 1.2.1 I did google the archive but couldn't see any mention
of anyone using this. What I am hoping to do is run a macro on hangup,
current method I am using seems to miss some calls 5% of calls fail to
mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.
2005 May 25
15
PHP/AGI Problem
Hi
I am currently developing a IVR application using
PHP/AGI. I am using the PHPAGI class at
http://phpagi.sourceforge.net/ to handle the
commuication with my *.
The application basically asks a caller to enter in
some information which is then processed and a answer
is read back out to them. I want the application to
loop back to the beginning after giving the answer so
they can try another
2004 Jul 14
8
Directed Call Pickup
In the list I found some messages that *8 doesn't work so well. Is there
any possibility to create a extention that you can call, and if you are
fast enough, pick up a number? (Also if you are outside your callgroup)
like
pseudo code:
exten => 888, 1, EnterPhoneNumber()
exten => 888, 2, EnterPass()
exten => 888, 3, TransferCallToThisPhone()
exten => 888, 103, Invalid()
2013 Feb 04
1
CallerID external call after Attended Transfer
Hello,
using Asterisk 1.8.12.2
case :
I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer
to my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other
side.
Receptionist transfers the call and I am connected to my colleague ( B )
My question is about the CallerID that the
2006 Jan 18
1
Attended transfer reconnect when goes to voicemail?
Hi
Running bristuffed 0.3.0-PRE-1f which is 1.2.1.
Using *2 in features.conf for attended transfer. Works well if someone
answers.
But the following sequence causes issue:
1. Receptionist takes call.
2. *2 then 123 to transfer to extension 123.
3. 123 is busy or does not answer so receptionist hears 123 voicemail
4. How can receptionist reconnect to calling user - could wait for voicemail to
2004 Sep 29
7
Credit Card machines / interop
Hi all,
One of the areas I am trying to research before I can confidently start
deploying Asterisk is "Credit Card Machines". (PDQ / Streamline machines
/ any similar)
I'm talking about the credit card swipe boxes at point of sale desks. I
believe they dial out to the specific bank provider everytime a card is
swiped.
My question is:
- Does anyone have any experience using
2006 Mar 12
7
stop monitor on transfer
Guys.
This idea has been banging my headfor days now and I feel the need to share
with you.
Imagine this scenario: all calls come in thru a receptionist, asterisk
records all incoming calls, the receptionist's work is to transfer the calls
to internal people but some of them are bosses and you know how bosses are,
they don't want their calls to be recorded, so, I have been trying to
2006 Apr 14
22
attended transfer issue
Hi!
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!
At this point the attended transfer should go into a blind transfer. The
phone of Person B should still be ringing and the
2006 Feb 08
4
Fedora Core 3 or Fedora Core 4? yum update ornot?
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Rich Adamson
> Sent: 08 February 2006 08:41
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum
update
> ornot?
>
> However, if you expose the box to
2005 Feb 23
6
List tips for new subscribers
*spews coffee over keyboard*
- FUNNIEST - THREAD - EVER -
Also one of the most insightful.
Teddy, your gmail invite is on the way.
2008 Feb 27
3
Attended transfers through a GUI
Greetings list,
I've been playing around this afternoon with Flash Operator Panel, trying to
get it to do attended transfers. I am running the latest version.
Has anyone managed to get this working reliably, and if so, would you mind
sharing how you did it please?
Alternatively, are there any other GUIs (free or commercial) that reliably
support attended transfers?
I'm trying to
2005 May 10
1
Redirect to an application on other asterisk server
Hello,
I'm a newbie in connection several asterisk servers with each others.
I've got the following situation.
I've got 9 asterisk servers (asterisk00 till asterisk08).
When I call to asterisk08 then I want to redirect an application which
runs on asterisk00.
But how can I redirect in an application on asterisk08 to an application
on asterisk00?
Or isn't this possible?
2006 Feb 22
2
Asterisk hints
Hi All
Does anyone know how the hints in asterisk works? How does a SIP phone
interact with the hints? I am having a problem with certain phone
models that do not set the hints correctly when I list the hints with a
'show hints'.
Thanks
Garth
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
2006 Jan 06
3
Announcing a call transfer
With our current pbx system, a call comes in from the PSTN to the
receptionist. She then hits flash, which puts the caller on hold, calls
my extension, says "so and so is on the phone for you", I say "ok put
him through", she hangs up and I am connected to the caller.
With asterisk@home I can it # then the extension to transfer to and it
will ring there. But is there a
2004 Oct 04
1
Macro's and Var Scope's
Hi,
I am having difficulty getting my record phone call dial-plan script
working. I have tried the example record call scripts but they start
recording before they are actually connected to an end point, e.g. you
get ringing or announcements being recorded.
It seems to me that these are bugs with the Dial() command:
1) Using M(x) in a dial command does not allow argument to be passed.
Using
2004 Sep 09
12
SNOM 200 can't conference.
Hello,
Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone.
Thanks
-Matt
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2006 Jan 24
4
which gui for asterisk on web
Hi there,
I want to use asterisk for sip comminication with max 1000 users
Which gui shuld i use for adding users and managing asterisk?
I tried AMPortal, it added extensions to mysql but asterisk did not find
users i added
? installed asterisk 1.2.2 on FC4
Toygun