[asterisk-bugs] [Asterisk 0013898]: [patch] Limit connect file and others will not play warnings
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Nov 13 21:31:50 CST 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=13898
======================================================================
Reported By: alecdavis
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 13898
Category: Applications/app_dial
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 155671
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 2008-11-13 19:11 CST
Last Modified: 2008-11-13 21:31 CST
======================================================================
Summary: [patch] Limit connect file and others will not play
warnings
Description:
LIMIT_CONNECT_FILE
LIMIT_WARNING_FILE
LIMIT_TIMEOUT_FILE
All appear to have corrupt pointers when the time comes to play the file.
They show correctly for the call.
The test macro limits the call to 20 seconds, warning at 15, frequency 10
seconds. (Note to myself, frequency too long!)
======================================================================
----------------------------------------------------------------------
(0094875) alecdavis (reporter) - 2008-11-13 21:31
http://bugs.digium.com/view.php?id=13898#c94875
----------------------------------------------------------------------
app_dial.bug13898.diff.txt uploaded
Fix doesn't feel right, as using ast_strdupa(var) causes the problem, but
it's ussed every where.
dialplan code:
exten => s,1,ChanIsAvail(${ARG1})
exten => s,2,Goto(5) ; give some room for error handling after 102
; cut away session id: "-<ID>". separator defaults to '-':
exten => s,5,Set(AVAILCHAN_NOSESSION=${CUT(AVAILCHAN,-,1)})
exten => s,n,Set(LIMIT_CONNECT_FILE=beep)
exten => s,n,Set(LIMIT_WARNING_FILE=beep)
exten => s,n,Set(LIMIT_TIMEOUT_FILE=terminated)
exten => s,n,Dial(${AVAILCHAN_NOSESSION}/${ARG2}#,,rL(20000:12000:4000))
exten => s,n,Congestion
exten => s,n,Hangup()
console output with patch applied:
-- Executing [4888 at trusted:1] Macro("SIP/8622-08446a40",
"dialout,DAHDI/g0,4888") in new stack
-- Executing [s at macro-dialout:1] ChanIsAvail("SIP/8622-08446a40",
"DAHDI/g0") in new stack
-- Hungup 'DAHDI/1-1'
-- Executing [s at macro-dialout:2] Goto("SIP/8622-08446a40", "5") in new
stack
-- Goto (macro-dialout,s,5)
-- Executing [s at macro-dialout:5] Set("SIP/8622-08446a40",
"AVAILCHAN_NOSESSION=DAHDI/1") in new stack
-- Executing [s at macro-dialout:6] Set("SIP/8622-08446a40",
"LIMIT_CONNECT_FILE=beep") in new stack
-- Executing [s at macro-dialout:7] Set("SIP/8622-08446a40",
"LIMIT_WARNING_FILE=beep") in new stack
-- Executing [s at macro-dialout:8] Set("SIP/8622-08446a40",
"LIMIT_TIMEOUT_FILE=terminated") in new stack
-- Executing [s at macro-dialout:9] Dial("SIP/8622-08446a40",
"DAHDI/1/4888#,,rL(20000:12000:4000)") in new stack
-- Limit Data for this call:
> timelimit = 20000
> play_warning = 12000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 4000
> start_sound = beep
> warning_sound = beep
> end_sound = terminated
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1/4888#
> Saved useragent "Grandstream GXP2000 1.1.6.37" for peer 8620
-- DAHDI/1-1 is proceeding passing it to SIP/8622-08446a40
!! Unknown IE 50 (cs5, Unknown Information Element)
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 answered SIP/8622-08446a40
-- <SIP/8622-08446a40> Playing 'beep.gsm' (language 'en')
-- <SIP/8622-08446a40> Playing 'beep.gsm' (language 'en')
-- <SIP/8622-08446a40> Playing 'beep.gsm' (language 'en')
-- <SIP/8622-08446a40> Playing 'terminated.slin' (language 'en')
-- Hungup 'DAHDI/1-1'
== Spawn extension (macro-dialout, s, 9) exited non-zero on
'SIP/8622-08446a40' in macro 'dialout'
== Spawn extension (macro-dialout, s, 9) exited non-zero on
'SIP/8622-08446a40'
Issue History
Date Modified Username Field Change
======================================================================
2008-11-13 21:31 alecdavis Note Added: 0094875
======================================================================
More information about the asterisk-bugs
mailing list