[Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Juan J. Sierralta P.]

On Sat, 2004-04-17 at 13:12, Olle E. Johansson wrote:
> Chris Orme wrote:
> 
> >>>exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> 
> Isn't the 'r' forcing a 'ringing' signal from start, regardless
> of what the device you are calling are signalling. If you are calling
> a SIP device, that device might return 'busy' and that's propably
> why you first hear 'ringing' and then a 'busy' signal.
> 
> I would like app_dial gurus to explain the 'r' option a bit
> more so we can document it better.

	I have a similar problem when my SIP devices dial outside via a Zaptel
trunk interface. If I don´t use "r" my 7960/ATAs/Polycoms work just
fine, but Granstream IP Phone and ATA-286 don´t get ringback tone.
	I believe the problem is that Grandstream doesn´t support Session
Progress which is a major drawback because when you´re calling to some
cell phone which has voicemail it is usual that you get the voice prompt
of the voicemail has a progress tone in order to just bill after the
beep which is when the line is answered, with Grandstream if some user
calls a cell phone (with voicemail) he/she gets the line answered and
just hears silence the problem is that silence is being billed :(

-- 
Juanjo sin .sig