[asterisk-bugs] [Asterisk 0012737]: Monitor fails to record call after second Dial(...)
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed May 28 08:00:53 CDT 2008
The following issue has been SUBMITTED.
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http://bugs.digium.com/view.php?id=12737
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Reported By: viraptor
Assigned To:
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Project: Asterisk
Issue ID: 12737
Category: Applications/app_mixmonitor
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.4.20
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 05-28-2008 08:00 CDT
Last Modified: 05-28-2008 08:00 CDT
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Summary: Monitor fails to record call after second Dial(...)
Description:
Using dialplan like this:
...
exten => _501,1,MixMonitor(recording.wav)
exten => _501,n,Dial(SIP/501)
...
exten => _555,1,Answer()
exten => _555,n,Dial(Local/500,10,tr)
exten => _555,n,Dial(Local/501,10,tr)
exten => _555,n,Hangup()
(happens also with normal Monitor)
if 501 answers I get only http://bugs.digium.com/view.php?id=5#1KB file instead
of whole call recorded. If
Monitor is placed after Answer, everything works as expected, but in this
dialplan monitor fails every time.
Console output when using MixMonitor:
--->8---
-- Called 501
== Begin MixMonitor Recording Local/501 at default-84a1,2
-- SIP/501-081b6f40 is ringing
-- Local/501 at default-84a1,1 is ringing
-- SIP/501-081b6f40 is ringing
-- SIP/501-081b6f40 is ringing
-- SIP/501-081b6f40 answered Local/501 at default-84a1,2
-- Local/501 at default-84a1,1 answered SIP/502-081a0860
== Spawn extension (default, 501, 2) exited non-zero on
'Local/501 at default-84a1,2'
-- Executing [h at default:1] Set("Local/501 at default-84a1,2",
"CDR(userfield)=") in new stack
== End MixMonitor Recording Local/501 at default-84a1,2
-- fixed jitterbuffer created on channel SIP/502-081a0860
== Spawn extension (default, 555, 3) exited non-zero on
'SIP/502-081a0860'
-- Executing [h at default:1] Set("SIP/502-081a0860",
"CDR(userfield)=ssrc=2014719930;themssrc=3346387491;lp=0;rxjitter=0.001814;rxcount=1406;txjitter=0.000000;txcount=1394;rlp=0;rtt=0.000000")
in new stack
-- fixed jitterbuffer destroyed on channel SIP/502-081a0860
--->8---
So it looks like monitor is closed down just before actual channel is
created.
Tested on versions between 1.4.11 and 1.4.20.1
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Issue History
Date Modified Username Field Change
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05-28-08 08:00 viraptor Asterisk Version => 1.4.20
05-28-08 08:00 viraptor SVN Branch (only for SVN checkou => N/A
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