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Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723
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This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub
have loaded properly before attempting to load any modules that depend
on them since the module loader system is not currently capable of
resolving module dependencies on its own.
ASTERISK-24312 #close
Reported by: Dafi Ni
Review: https://reviewboard.asterisk.org/r/4062/
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Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 425691 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Replace code using ast_uuid_generate() with simpler and faster code using
ast_uuid_generate_str(). The new code avoids a malloc(), free(), and
copy.
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Merged revisions 424103 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 424105 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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"module show" .
ASTERISK-23919 #close
Reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/3802
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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An md5 hash is 32 bytes long. The char buffer must be at least 33 bytes to
avoid clobbering of the stack. This patch also fixes a potential clobbering
in test_utils.c.
Thanks to Andrew Nagy for reporting and testing this out in #asterisk-dev
Reported by: Andrew Nagy
Tested by: Andrew Nagy
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Merged revisions 404843 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When switching to using a vector for authentication, I initialized
the vector for the artificial endpoint to be of size 1. However, this
does not result in AST_VECTOR_SIZE() returning 1 since there isn't
actually anything in the vector.
Rather than trifle with the vector by putting unnecessary elements in,
I simply changed the callback in res_pjsip_authenticator_digest.c to
explicitly report that the artificial endpoint requires authentication.
Thanks to Joshua Colp for pointing this out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Since Asterisk has a vector API now, places where arrays are manually
resized don't really make sense any more. Since the auth work in PJSIP
was freshly-written, it was easy to reform it to use a vector.
Review: https://reviewboard.asterisk.org/r/3044
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Previously, regardless of whether failure to authenticate was due to
lacking any authentication or actually failing authentication, the
Digest Authenticator would simply return that a challenge was still
needed. It will continue to do that when no authentication information
is in the received SIP digest, but when authentication information
is present and does not pass authentication, that will be treated as
an authentication error. This is to ensure that PJSIP will issue
security events indicated failed auths.
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Merged revisions 402537 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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With this change, if no realm is specified in an outbound auth
section, then we will simply match the realm that was present
in the 401/407 challenge.
(closes issue ASTERISK-22471)
Reported by George Joseph
(closes issue ASTERISK-22386)
Reported by Rusty Newton
Patches:
outbound_auth_realm_v4.patch uploaded by George Joseph (License #6322)
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Merged revisions 399059 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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