[Asterisk-Users] Supervised transfer problem with BudgetTone
Nicolas Schmerber
nicolas.schmerber at wanadoo.fr
Thu Aug 11 07:40:56 MST 2005
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:611 at voiceserver.thevoipconnection.com
>
>
>
>
>>-----Original Message-----
>>From: Nicolas Schmerber [mailto:nicolas.schmerber at wanadoo.fr]
>>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
>>
>>steve at daviesfam.org 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 at 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 at 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.
More information about the asterisk-users
mailing list