[asterisk-bugs] [Asterisk 0018140]: SRTP enable disable from dialplan
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Oct 19 13:24:02 CDT 2010
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=18140
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Reported By: chodorenko
Assigned To:
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Project: Asterisk
Issue ID: 18140
Category: Resources/res_srtp
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.8.0-rc3
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2010-10-14 07:17 CDT
Last Modified: 2010-10-19 13:24 CDT
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Summary: SRTP enable disable from dialplan
Description:
variable
secure_bridge_signaling
secure_bridge_media
secure_signaling
secure_media
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(0128202) chodorenko (reporter) - 2010-10-19 13:24
https://issues.asterisk.org/view.php?id=18140#c128202
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Yes , misspelling is my Bug , also i not retrieve answer on all bugs
1.1 -- Executing [8888 at default:7] Set("SIP/test2-0000005f",
"CHANNEL(secure_bridge_media)=0") in new stack
[Oct 14 17:04:58] NOTICE[19465]: chan_sip.c:4052 sip_setoption: Unknown
option: 9
Why ? its example of secure-call.tex
2.2 Variable secure_media is 1 and before and after change
secure_bridge_media
Please note that the call after "CHANNEL(secure_bridge_media)=0" go
unencrypted Although the appointment of a variable passed with error
new
5. quote=>twilson > _SIPSRTP_CRYPTO is no longer used.
in res_srtp.c header present indication of the need to use, please correct
this and give link to manual in code
6. "sip show peer NAME" do not report capability peer of encryption and
setting encryption
Please Sorry me by Spell error in configs in previous report
Issue History
Date Modified Username Field Change
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2010-10-19 13:24 chodorenko Note Added: 0128202
2010-10-19 13:24 chodorenko Status closed => new
2010-10-19 13:24 chodorenko Resolution no change required =>
reopened
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