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:[EMAIL PROTECTED] > -----Original Message----- > From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] > Sent: Thursday, August 11, 2005 10:41 AM > To: [EMAIL PROTECTED]; 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 partys 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 > >>>[email protected] > >>>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 > >[email protected] > >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 [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
