Greetings, I'm trying to continue to do some processing after a TIMEOUT (absolute). In my dialplan below, when a call comes in to [default], I call macro-phonenum and pass it a timeout of 20 seconds. macro- phonenum sets TIMEOUT(absolute), then loops saying the phone number that was called (in MACRO_EXTEN). When the timeout expires I want to call my macro-hangup (so it can say "goodbye" or whatever). But the system is just hanging up. The dialplan and log output is below. Any info is appreciated. This is on version 1.6.0.5.
[macro-answer-and-join] exten => s,1,NoOp() exten => s,n,Answer() exten => s,n,Wait(4) exten => s,n,SendDTMF(1) exten => s,n,Wait(1) exten => s,n,SendDTMF(1) exten => s,n,MacroExit [macro-hangup] exten => s,1,NoOp() exten => s,n,Playback(goodbye) exten => s,n,Hangup() ; exten => T,1,NoOp() exten => T,n,Playback(goodbye) exten => T,n,Hangup() [macro-phonenum] exten => s,1,NoOp() exten => s,n,Macro(answer-and-join) exten => s,n,Set(TIMEOUT(absolute)=${ARG1}) exten => s,n,Set(i=1000) exten => s,n,While($[${i} >= 1]) exten => s,n,SayDigits(${MACRO_EXTEN}) exten => s,n,Wait(5) exten => s,n,Set(i=$[${i} - 1]) exten => s,n,EndWhile() exten => s,n,MacroExit ; exten => T,1,NoOp() exten => T,n,Macro(hangup) exten => T,n,MacroExit [default] exten => _X.,1,NoOp() exten => _X.,n,Macro(phonenum,20) exten => _X.,n,Macro(hangup) ; exten => T,1,NoOp() exten => T,n,Macro(hangup) The log when the timeout occurs: <snip> (I'm in macro-phonenum) -- <SIP/70.124.61.17-082a69a8> Playing 'digits/5.ulaw' (language 'en') -- <SIP/70.124.61.17-082a69a8> Playing 'digits/1.ulaw' (language 'en') -- <SIP/70.124.61.17-082a69a8> Playing 'digits/2.ulaw' (language 'en') -- <SIP/70.124.61.17-082a69a8> Playing 'digits/1.ulaw' (language 'en') -- <SIP/70.124.61.17-082a69a8> Playing 'digits/2.ulaw' (language 'en') -- Executing [...@macro-phonenum:7] Wait("SIP/ 70.124.61.17-082a69a8", "5") in new stack == Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/ 70.124.61.17-082a69a8' in macro 'phonenum' == Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/ 70.124.61.17-082a69a8' Scheduling destruction of SIP dialog 'D8FE9724-1DD1-11B2-9F1A- a4ef9db84...@192.168.1.98' in 32000 ms (Method: ACK) set_destination: Parsing <sip:70.124.61.17:5060> for address/port to send to set_destination: set destination to 70.124.61.17, port 5060 Reliably Transmitting (NAT) to 70.124.61.17:5060: BYE sip:70.124.61.17:5060 SIP/2.0 <snip> Cheers, - Brendan Brendan Sterne QA Lead, Callvine -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users