[asterisk-users] How to hang-up a FXO call without answering it?

[Frank Tarczynski]

I'm using Asterisk 13.4.0 and DAHDI 2.10.2.  I've got a FXO line that I 
use for in and outgoing PSTN calls.  Unfortunately I'm getting a lot of 
spam calls on the number.

I had the extension configured to forward incoming calls to 2 SIP 
extensions or go to voicemail.  But now I'm getting loads of junk 
voicemail messages, so I removed the voicemail command:

[from-pstn]
exten => s,1,Wait(1)
exten => s,2,Set(WHO=${CALLERID(num)})
exten => s,3,Verbose(CALLERID is ${CALLERID(num)})
exten => s,4,Verbose(Time is ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten => s,5,Dial(SIP/1000&SIP/1100,30)
;exten => s,6,Voicemail(1000,u)
exten => s,6,Hangup()

Now incoming calls will cause the SIP extensions to ring for 30 seconds, 
but then the FXO line isn't disconnected.  The [from-pstn] context seems 
to keep looping on the Dial() command:

[Sep 19 11:16:59]     -- Starting simple switch on 'DAHDI/4-1'
[Sep 19 11:17:00]     -- Executing [s at from-pstn:1] Wait("DAHDI/4-1", 
"1") in new stack
[Sep 19 11:17:01]     -- Executing [s at from-pstn:2] Set("DAHDI/4-1", 
"WHO=919961XXXX") in new stack
[Sep 19 11:17:01]     -- Executing [s at from-pstn:3] Verbose("DAHDI/4-1", 
"CALLERID is 919961XXXX") in new stack
[Sep 19 11:17:01] CALLERID is 919961XXXX
[Sep 19 11:17:01]     -- Executing [s at from-pstn:4] Verbose("DAHDI/4-1", 
"Time is 20150919-111701") in new stack
[Sep 19 11:17:01] Time is 20150919-111701
[Sep 19 11:17:01]     -- Executing [s at from-pstn:5] Dial("DAHDI/4-1", 
"SIP/1000&SIP/1100,30") in new stack
[Sep 19 11:17:01]   == Using SIP RTP TOS bits 184
[Sep 19 11:17:01]   == Using SIP RTP CoS mark 5
[Sep 19 11:17:01]   == Using SIP RTP TOS bits 184
[Sep 19 11:17:01]   == Using SIP RTP CoS mark 5
[Sep 19 11:17:01]     -- Called SIP/1000
[Sep 19 11:17:01]     -- Called SIP/1100
[Sep 19 11:17:01]     -- SIP/1000-00000095 is ringing
[Sep 19 11:17:01]     -- SIP/1100-00000096 is ringing
[Sep 19 11:17:31]     -- Nobody picked up in 30000 ms
[Sep 19 11:17:31]     -- Executing [s at from-pstn:6] Hangup("DAHDI/4-1", 
"") in new stack
[Sep 19 11:17:31]   == Spawn extension (from-pstn, s, 6) exited non-zero 
on 'DAHDI/4-1'
[Sep 19 11:17:31]     -- Hanging up on 'DAHDI/4-1'
[Sep 19 11:17:31]     -- Hungup 'DAHDI/4-1'
[Sep 19 11:17:35]     -- Starting simple switch on 'DAHDI/4-1'
[2015-09-19 11:17:39.1] ERROR[27434][C-00000079]: callerid.c:567 
callerid_feed: No start bit found in fsk data.
[2015-09-19 11:17:39.1] WARNING[27434][C-00000079]: chan_dahdi.c:1374 
my_get_callerid: Failed to decode CallerID
[2015-09-19 11:17:39.1] WARNING[27434][C-00000079]: sig_analog.c:2569 
__analog_ss_thread: CallerID returned with error on channel 'DAHDI/4-1'
[Sep 19 11:17:39]     -- Executing [s at from-pstn:1] Wait("DAHDI/4-1", 
"1") in new stack
[Sep 19 11:17:40]     -- Executing [s at from-pstn:2] Set("DAHDI/4-1", 
"WHO=") in new stack
[Sep 19 11:17:40]     -- Executing [s at from-pstn:3] Verbose("DAHDI/4-1", 
"CALLERID is ") in new stack
[Sep 19 11:17:40] CALLERID is
[Sep 19 11:17:40]     -- Executing [s at from-pstn:4] Verbose("DAHDI/4-1", 
"Time is 20150919-111740") in new stack
[Sep 19 11:17:40] Time is 20150919-111740
[Sep 19 11:17:40]     -- Executing [s at from-pstn:5] Dial("DAHDI/4-1", 
"SIP/1000&SIP/1100,30") in new stack
[Sep 19 11:17:40]   == Using SIP RTP TOS bits 184
[Sep 19 11:17:40]   == Using SIP RTP CoS mark 5
[Sep 19 11:17:40]   == Using SIP RTP TOS bits 184
[Sep 19 11:17:40]   == Using SIP RTP CoS mark 5
[Sep 19 11:17:40]     -- Called SIP/1000
[Sep 19 11:17:40]     -- Called SIP/1100
[Sep 19 11:17:40]     -- SIP/1000-00000097 is ringing
[Sep 19 11:17:40]     -- SIP/1100-00000098 is ringing
[Sep 19 11:17:49]   == Spawn extension (from-pstn, s, 5) exited non-zero 
on 'DAHDI/4-1'
[Sep 19 11:17:49]     -- Hanging up on 'DAHDI/4-1'
[Sep 19 11:17:49]     -- Hungup 'DAHDI/4-1'

The caller just hears the line ring and ring and the SIP extensions are 
dialed over and over until the caller hangs-up.

Is there anyway to force a hang-up or disconnection of the incoming call 
if the SIP extensions don't answer?

I'd like to do this without actually answering the call if at all possible.

Frank