[asterisk-bugs] [Asterisk 0018674]: [patch] Unable to choose which SRTP suite to offer
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Feb 2 16:45:33 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18674
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Reported By: bbeers
Assigned To:
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Project: Asterisk
Issue ID: 18674
Category: Channels/chan_sip/SRTP
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 303637
Request Review:
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Date Submitted: 2011-01-25 09:56 CST
Last Modified: 2011-02-02 16:45 CST
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Summary: [patch] Unable to choose which SRTP suite to offer
Description:
Setting encryption=yes in sip.conf will cause asterisk to
generate a line in SIP INVITE SDP:
a=crypto: AES_CM_128_HMAC_SHA1_80 ...
There is no way to specify that asterisk should offer
AES_CM_128_HMAC_SHA1_32 instead of
AES_CM_128_HMAC_SHA1_80.
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(0131422) c1nco (reporter) - 2011-02-02 16:45
https://issues.asterisk.org/view.php?id=18674#c131422
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Applied new patch, however, I am still having issues please see below
links--
external caller to phonerlite client: http://pastebin.com/xtEkJe2n
phonerlite trying to dial Voice Mail: http://pastebin.com/2Y3wjZ90
Both internal and external calls are still being dropped.
Thanks for your assistance!
Issue History
Date Modified Username Field Change
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2011-02-02 16:45 c1nco Note Added: 0131422
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