The VoIP Connection a écrit :
Section 4.3.7.2 from the Bugetone Manual:
The user can transfer an active call to a third party with announcement.
The user presses the “flash” button and hears a dial tone, then dial the 3rd
party’s phone number followed by pressing send button. If the call is
answered, press “flash” to complete the transfer operation, if the call is
not
answered, pressing “flash” button to resume the original call.
Notes:
• If attended Transfer fails, the BudgeTone phone will ring the user to
remind that
another party is still on the call, the user can then pick up the call using
handset
or speaker.
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-----Original Message-----
From: Nicolas Schmerber [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 11, 2005 5:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Supervised transfer problem
with BudgetTone
[EMAIL PROTECTED] a écrit :
On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
All the features I need work just not one : the supervised call
transfers. I know there are a lot of posts about that, but
none gave
me the correct answer (unless I missed it).
Hi,
You'll need to switch to the CVS-HEAD version of Asterisk in
order to
have supervised transfers.
Steve
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
When looking at a recent firmware changelog of Grandstream ,
it says BT should support supervised transfer, so shouldnt it work ?
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Tried this manipulation a few minutes ago :
A calls B , B pushes flash button ( A is waiting with a mp3 played)
B calls C pressing Send ;
C answers
B presses flash button again ;
C is so on hold (with a mp3 played)
B hangs up
But A and C arent in connect ; the phoneof B rings ( to tell someone is
in wait : A)
So it seems to fail
What should i put in grandstream config for the next item :
/Enable Call Features: Y/ N ?
//Disable Call-Waiting: Y/N ?
//Send DTMF: / in-audio / via RTP (RFC2833) / via SIP INFO
/Send Flash Event: Y / N ? /
Any others Ideas ?.
Thx
Nicolas S.
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
