[Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

The VoIP Connection a écrit :

>Nicolas,
>
>Just did some quick testing and the instructions are incorrect.  You need to
>press "transfer" to complete the transfer instead of the second "flash".
>This actually makes more sense.
>
>Attended and regular transfer both work perfectly with the following
>settings:
>
>Enable Call Features: "Yes"
>Disable call Waiting: "No"
>Send Flash event: "No"
>
>DTMF should be whatever * is set to, but in-band won't work properly if your
>codec is anything other than U-Law.
>
>By the way, the newest firmware also makes the long overdue conference
>feature work properly.
>
>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 10:41 AM
>>To: asterisk-biz at thevoipconnection.com; Asterisk Users 
>>Mailing List - Non-Commercial Discussion
>>Subject: Re: [Asterisk-Users] Supervised transfer problem 
>>with BudgetTone
>>
>>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.
>>
>>    
>>
>
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>
>
>  
>
Thank you all, now it works
The last method (grandsteam manual but with transfer key instead) was 
the right

Thanks

Nicolas