Simon P. Ditner wrote:
The "S0" means end of input, so what you've effectively done is
require 3 or more digits of input before routing the call to Asterisk
by changing it to "xx.". If you set that to <S0:s>, it will try to
reach extension 's' at the default SIP proxy, or if you entered
<S0:123> it would try '123'
On the "PSTN Line" screen there is an "Answer Delay" under the section
"FXO Timer Values (sec)". The default value is something like 16
seconds. North American analog ringing is 2s on, 4s off, 2s on, 4s
off, etc.., and in my tests I found that it consistently captured
caller ID down as low as 5 seconds in that field. You should then only
be hearing 2-3 rings when calling in before asterisk picks up.
Thanks! Both suggestions worked exactly as you predicted.
On to next example and I have a new problem.
[incoming]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Background(vm-enter-num-to-call)
exten => 1,1,Playback(digits/1)
exten => 1,2,Goto(incoming,s,1)
exten => 2,1,Playback(digits/2)
exten => 2,2,Goto(incoming,s,1)
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(incoming,s,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup
This plan works if I press any digit while "vm-enter-num-to-call" is
playing; but the book mentions a timeout which is not happening. As
soon as "vm-enter-num-to-call" finishes; the call is disconnected.
== Auto fallthrough, channel 'SIP/pstn-079ba000' status is 'UNKNOWN'
I never hear the "vm-goodbye" message.
---------------------------------------------------------------------
To unsubscribe, e-mail: [email protected]
For additional commands, e-mail: [email protected]
