[asterisk-users] Why does it take several seconds to interpret DTMF-input ?

[Jonas Kellens]

Hello,

I notice that it takes 4 to 6 seconds between someone pressing a cipher 
and Asterisk continuing inside the dialplan. How come ???

Taken from verbose logfile :

(attempt 1)
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin '1' received on 
SIP/SipAgenT01-00001eb0
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin ignored '1' on 
SIP/SipAgenT01-00001eb0
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF end '1' received on 
SIP/SipAgenT01-00001eb0, duration 180 ms
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF end passthrough '1' on 
SIP/SipAgenT01-00001eb0

[Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30]   == CDR 
updated on SIP/SipAgenT01-00001eb0
[Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30]     -- 
Executing [1 at pbx-routing:1] Set("SIP/SipAgenT01-00001eb0", "choice=1") 
in new stack
[Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30]     -- 
Executing [1 at pbx-routing:2] System("SIP/SipAgenT01-00001eb0", "echo 
"'418','IVR','1','','SipAgenT01-00001eb0','$(date +%s)'" >> 
/var/log/asterisk/loggingAST/SipAgenT01-00001eb0.csv") in new stack

(attempt 2)
[Jun 11 15:30:21] DTMF[18780] channel.c: DTMF begin '8' received on 
SIP/SipAgenT01-00001ec1
[Jun 11 15:30:21] DTMF[18780] channel.c: DTMF begin ignored '8' on 
SIP/SipAgenT01-00001ec1
[Jun 11 15:30:21] DTMF[18780] channel.c: DTMF end '8' received on 
SIP/SipAgenT01-00001ec1, duration 160 ms
[Jun 11 15:30:21] DTMF[18780] channel.c: DTMF end passthrough '8' on 
SIP/SipAgenT01-00001ec1

[Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27]   == CDR 
updated on SIP/SipAgenT01-00001ec1
[Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27]     -- 
Executing [8 at pbx-routing:1] Set("SIP/SipAgenT01-00001ec1", "choice=8") 
in new stack
[Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27]     -- 
Executing [8 at pbx-routing:2] System("SIP/SipAgenT01-00001ec1", "echo 
"'418','IVR','8','','SipAgenT01-00001ec1','$(date +%s)'" >> 
/var/log/asterisk/loggingAST/SipAgenT01-00001ec1.csv") in new stack



Why doesn't Asterisk continue immediately inside the dialplan after 
having received the DTMF-input ?


Kind regards,

Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130611/72cee402/attachment.htm>