[asterisk-bugs] [Asterisk 0018761]: SoftHangup not effective on 1.8.2.3
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Feb 10 13:15:39 CST 2011
The following issue has been CLOSED
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https://issues.asterisk.org/view.php?id=18761
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Reported By: jkister
Assigned To:
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Project: Asterisk
Issue ID: 18761
Category: Applications/app_softhangup
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: 1.8.2.3
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: open
Fixed in Version:
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Date Submitted: 2011-02-07 09:45 CST
Last Modified: 2011-02-10 13:15 CST
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Summary: SoftHangup not effective on 1.8.2.3
Description:
I am trying to use SoftHangup in my dialplan, but it's either not
working or I'm not using it correctly.
when i'm on the console, i see:
pbx1*CLI> core show channels
Channel Location State Application(Data)
SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line))
SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,,
2 active channels
1 active call
194 calls processed
pbx1*CLI>
in my dialplan, i have:
exten => s,1,Set(CHAN=${SHELL(asterisk -rx "core show channels" | awk
'/^SIP\/vgw1-/ { print $1 }' | head -1)})
exten => s,n,SoftHangup(${CHAN})
exten => s,n,Wait(2)
When I dial the extension to invoke the above dialplan code, the console
shows:
-- Executing [s at nineoneone:10] SoftHangup("SIP/111-000000a3",
"SIP/vgw1-000000a2") in new stack
but the SIP/vgw1-000000a2 is still active. If I use 'channel request
hangup SIP/vgw1-000000a2', the call is dropped instantly.
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(0131820) lmadsen (administrator) - 2011-02-10 13:15
https://issues.asterisk.org/view.php?id=18761#c131820
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Closed per the reporter. Thanks for the follow up!
Issue History
Date Modified Username Field Change
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2011-02-10 13:15 lmadsen Note Added: 0131820
2011-02-10 13:15 lmadsen Status feedback => closed
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