> Extensions.conf > [mainmenu] > exten => 501,1,Answer > exten => 501,n,Wait(2) > exten => 501,n,Playback(velkommen_abacus) > exten => 501,n,Set(Loop=0) > exten => 501,n,While($[${Loop} < 3]) > exten => 501,n,Background(tast123vent_) > exten => 501,n,WaitExten(5) > exten => 501,n,Set(Loop=$[${Loop}+1]) > exten => 501,n,(LoopEnd),EndWhile
This should be: exten => 501,n(LoopEnd),EndWhile I don't understand, i do have the same thing you wrote above. > Connected to Asterisk 1.6.2.6 currently running on asterisk (pid = 2467) > Verbosity is at least 3 > == Using SIP RTP TOS bits 184 > == Using SIP RTP CoS mark 5 > -- Executing [...@phones:1] Answer("SIP/301-00000248", "") in new stack > -- Executing [...@phones:2] Wait("SIP/301-00000248", "2") in new stack > -- Executing [...@phones:3] Playback("SIP/301-00000248", > "velkommen_abacus") in new stack > -- <SIP/301-00000248> Playing 'velkommen_abacus.slin' (language 'en') > -- Executing [...@phones:4] Set("SIP/301-00000248", "Loop=0") in new stack > -- Executing [...@phones:5] While("SIP/301-00000248", "1") in new stack > -- Executing [...@phones:6] BackGround("SIP/301-00000248", > "tast123vent_") in new stack > -- <SIP/301-00000248> Playing 'tast123vent_.slin' (language 'en') > -- Executing [...@phones:7] WaitExten("SIP/301-00000248", "5") in new > stack > -- Timeout on SIP/301-00000248, continuing... > -- Executing [...@phones:8] Set("SIP/301-00000248", "Loop=1") in new stack > [Jun 18 10:38:16] WARNING[1692]: pbx.c:3680 pbx_extension_helper: No > application '' for extension (phones, 501, 9) You put '(LoopEnd)' in the place for the application. Hence empty application with 'LoopEnd' as its input. > == Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-00000248' -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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