Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got:
-- Executing [301@from-test:1] Dial("SIP/300-00000045", "SIP/301,60,rjtTg") in new stack -- Called SIP/301 -- SIP/301-00000046 is ringing -- SIP/301-00000046 answered SIP/300-00000045 -- Auto fallthrough, channel 'SIP/300-00000045' status is 'ANSWER' -- Executing [h@from-test:1] Goto("SIP/300-00000045", "play,s,1") in new stack -- Goto (play,s,1) -- Executing [s@play:1] NoOp("SIP/300-00000045", "play") in new stack -- Executing [s@play:2] SayDigits("SIP/300-00000045", "123579") in new stack [Feb 21 10:35:00] WARNING[31945]: file.c:833 ast_readaudio_callback: Failed to write frame -- <SIP/300-00000045> Playing 'digits/1.ulaw' (language 'en') == Spawn extension (play, s, 2) exited non-zero on 'SIP/300-00000045' This is my dialplan: [from-test] exten => _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten => h,1,Goto(play,s,1) [play] exten => s,1,Noop(play) exten => s,2,Saydigits(123579) Anyone can help me? Thanks Enrico. -- -- Pasqualotto Enrico cell. +39 3473292620 skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto http://www.netspin.it :: e.pasqualo...@netspin.it -- _____________________________________________________________________ -- 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