Ignore me I figured it out. The dangers of copy and paste. After looking through the code line by line I noticed the 'b' parameter to monitor(). Fine to use before the dial command but shouldnt be used when a call is in progress.
Gareth Blades wrote: > I have got call recording working on our 1.4.30 asterisk box together > with a recording pause ability and being able to play different audio to > each party at the start and end of the pause. This all works perfectly > but one wish is to have the audio files have a beep or something in them > so when you listen later you can tell where the audio was paused. > > So I changed things around so that instead of pausing and unpausing the > recording was stopped and then started again with a new file. An AGI > script would then join and mix the files together. The problem that I am > having is that I can stop the recording but when I ran the monitor > command again the recording didnt start. > So I decided to start with a simpler test and just have recording off by > default and then use #2 to start it but that doesnt work either. I > wondered if it was something to do with the native bridging so set > canreinvite=no on the test handsets I am using but no change. > > Any ideas? > > > > features.conf > [applicationmap] > pauseMonitor => #1,peer/callee,Macro,recpause,monitor-disabled > startMonitor => #2,peer/callee,Macro,recstart > unpauseMonitor => #3,peer/callee,Macro,recunpause,monitor-enabled > > > extensions.conf > [macro-recpause] > exten => s,1,Playback(disabled) > exten => s,n,PauseMonitor > > [macro-recunpause] > exten => s,1,Playback(enabled) > exten => s,n,UnpauseMonitor > > [macro-recstart] > exten => s,1,Set(FNAME=callrec_${MACRO_EXTEN}_${UNIQUEID}_GWTEST_${EPOCH}) > exten => s,n,Monitor(wav,${FNAME},b) > > [internal] > exten => 100,1,Dial(SIP/100,20) > exten => 110,1,Answer > exten => > 110,n,Set(DYNAMIC_FEATURES=pauseMonitor#unpauseMonitor#startMonitor) > exten => 110,n,Set(FNAME=callrec_${EXTEN}_${UNIQUEID}_GWTEST_${EPOCH}) > ;exten => 110,n,Monitor(wav,${FNAME},b) > exten => 110,n,Dial(SIP/110,20) > exten => 110,n,Hangup > > > log :- > -- Executing [...@internal:1] Answer("SIP/100-00000004", "") in new > stack > -- Executing [...@internal:2] Set("SIP/100-00000004", > "DYNAMIC_FEATURES=pauseMonitor#unpauseMonitor#testfeature#startMonitor") > in new stack > -- Executing [...@internal:3] Set("SIP/100-00000004", > "FNAME=callrec_110_1272534191.4_GWTEST_1272534191") in new stack > -- Executing [...@internal:4] Dial("SIP/100-00000004", > "SIP/110|20") in new stack > -- Called 110 > -- SIP/110-00000005 is ringing > -- SIP/110-00000005 answered SIP/100-00000004 > -- Packet2Packet bridging SIP/100-00000004 and SIP/110-00000005 > -- Packet2Packet bridging SIP/100-00000004 and SIP/110-00000005 > -- Executing [...@macro-recstart:1] Set("SIP/100-00000004", > "FNAME=callrec_110_1272534191.4_GWTEST_1272534203") in new stack > -- Executing [...@macro-recstart:2] Monitor("SIP/100-00000004", > "wav|callrec_110_1272534191.4_GWTEST_1272534203|b") in new stack > -- Packet2Packet bridging SIP/100-00000004 and SIP/110-00000005 > == Spawn extension (internal, 110, 5) exited non-zero on > 'SIP/100-00000004' > -- _____________________________________________________________________ -- 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