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yes, u are quite right, you can find this feature
in almost every pbx now.
We are also wondering whether, presently some one
is implementing this feature or not, if no body is doing that, we
can
start on that
Surajee
----- Original Message -----
Sent: Wednesday, June 04, 2003 3:36
AM
Subject: RE: [Asterisk-Users] Call
Transfer Problem
so,
What should the call initiator do if s/he wants to transfer the call initiated
by himself/herself, by using flash keypad or what else ?
I
can see such application can be used in some big office, where the BOSS always
asks the secretary to make the call, once the call is connected, then the
secretary can trasfer the call to the BOSS. in order to let the BOSS talk on
the phone. am I right ??
Please let me know once the feature is
implemented.
George Lin
U get the following output when u execute the
"show application Dial" command in the Asterisk prompt,
-= Info about application 'Dial' =-
[Synopsis]:
Place an call and connect
to the current channel
[Description]:
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):
Requests
one or more channels and places specified outgoing calls on
them.
As soon as a channel answers, the Dial
app will answer the originating
channel (if it needs to be
answered) and will bridge a call with the channel
which first answered.
All other calls placed by the Dial app will be hunp up
f a timeout is not
specified, the Dial application will wait indefinitely
until
either one of the called channels answers, the user hangs up, or
all
channels return busy or error. In general, the dialler
will return 0 if it
was unable to place the
call, or the timeout expired. However, if all
channels were
busy, and there exists an extension with priority n+101 (where
n is the
priority of the dialler instance), then it
will be the next
executed extension (this allows you to
setup different behavior on busy from
no-answer).
This
application returns -1 if the originating channel hangs up, or if
the
call is bridged and either of the parties in the bridge
terminate the call.
The option string may contain zero or more of the
following characters:
't' -- allow the
called user transfer the calling user
'T'
-- to allow the calling user to transfer the
call.
'r' -- indicate ringing to the
calling party, pass no audio until
answered.
'm' -- provide hold music to the
calling party until answered.
'd' --
data-quality (modem) call (minimum delay).
'c' -- clear-channel data call (PRI-PRI
only).
'H' -- allow caller to hang up by
hitting *.
'C' -- reset call detail record
for this call.
'P[(x)]' -- privacy mode,
using 'x' as database if provided.
In addition to transferring the
call, a call may be parked and then picked
up by another user.
The optionnal URL will be sent to the called party if the channel
supports
it.
Surajee
----- Original Message -----
Sent: Monday, June 02, 2003 1:11
PM
Subject: FW: [Asterisk-Users] Call
Transfer Problem
Hi,
Which
document describes the Dial
with “T” option ? Could you let me know or email it to
me.
Thanks,
George
Lin
-----Original
Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Surajee
Ratnayake
Sent: Sunday,
June 01, 2003 9:10 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Call
Transfer Problem
hi
All,
We are working
on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING
aspects of Asterisk.
We were able to
do one type of call transfering, ie, the called person can transfer the
original call to another person.
but we were
unable to do the other, that is, call initiator him/her self couldn't
transfer the call. Eventhough the documentation for Dial
application intructs to use "T" to achieve that.
and we learnt
that it has not been implemented yet in Asterisk. Is this true?
Is some one
workin on this issue? if the answer is NO, we can give a try to implement
it, with a help of u all , ofcourse :-)
(cos, we
are quite new to asterisk-only 1 month of experience, but amazed of its
great performance)
Thank you very
much,
Surajee
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