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 :: [email protected]
--
_____________________________________________________________________
-- 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
