revert Mark's change that caused a memory leak (cap_set_proc() does not free the capability structure so we always need to call cap_free())
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44152 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44125 | russell | 2006-10-01 14:30:06 -0400 (Sun, 01 Oct 2006) | 6 lines
Fix a problem that cuased AST_DATA_DIR in defaults.h to be empty. The cause is
that since ASTDATADIR is explicitly exported using "export ASTDATADIR" at the
top of the Makefile, make no longer considers the variable "undefined", so the
Makefile can't use ?= to set ASTDATADIR if not yet set.
(issue #8063, reported by akohlsmith, fixed by me)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44126 f38db490-d61c-443f-a65b-d21fe96a405b
- with AST_DEVMODE, building codecs/lpc10 fails because of lots
of warnings, and the configure step in editline fails as well.
Fix this by removing the -Werror in these steps.
- on FreeBSD (but probably on other platforms as well), the final
link of asterisk fails because AST_LIBS was not exported to the
subdirs Makefiles. Add a proper fix in the top-level Makefile
(a possible alternative way is to add "export AST_LIBS" near
the beginning of the file).
With this fix, i believe that some of the platform-specific
conditionals in main/Makefile are redundant (because they should
be already dealt with in the top level Makefile) but i don't
have a platform to check.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44081 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44078 | pcadach | 2006-09-30 22:12:23 +0600 (Сбт, 30 Сен 2006) | 6 lines
Fix issue #7928 correctly. Next is a comment of previous fix:
Issue #7928 - Don't send both 404 and 503. Fix by phsultan with
a small fix by me, myself or I. Thanks, Philippe!
(This was caused by my changes to the transaction handling)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44079 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) | 14 lines
Found some buggy SIP clients (phones Planet VIP-153T firmware
1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK
message only (when remote party answers) but on RINGING message
too, so when we send 200 OK message, we get unidentified ACK
message (because INVITE acknowledged on RINGING message already),
so 200 OK retransmits within its retransmission interval then
call gets dropped.
If someone else knows how to provide workaround for such cases,
please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44069 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44034 | pcadach | 2006-09-30 02:43:13 +0600 (Сбт, 30 Сен 2006) | 1 line
Fake display name by called number on incoming calls (until passing connected number/connected name is not implemented)
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r44042 | pcadach | 2006-09-30 03:05:43 +0600 (Сбт, 30 Сен 2006) | 1 line
Set TON/PRESENTATION information more carefully when no CallingNumber IE available
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r44043 | pcadach | 2006-09-30 03:09:10 +0600 (Сбт, 30 Сен 2006) | 1 line
Compile first, please
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44044 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43899 | bweschke | 2006-09-28 12:41:05 -0400 (Thu, 28 Sep 2006) | 11 lines
Merged revisions 43897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43897 | bweschke | 2006-09-28 12:37:15 -0400 (Thu, 28 Sep 2006) | 3 lines
app_queue is comparing the device names incorrectly while checking their statuses. It's internal list of interfaces includes the dial string, while the argument passed to this function does not have the dial string (/n for a local channel). This causes it to ignore the device state changes because it thinks it belongs to none of its members. (#8040 reported and patch by tim_ringenbach)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43902 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | 1 line
Put attribute tag at correct place
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r43862 | pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line
Force remote side to start media on outgoing PROGRESS message
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43863 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43852 | crichter | 2006-09-28 13:03:05 +0200 (Do, 28 Sep 2006) | 9 lines
Merged revisions 43764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43764 | crichter | 2006-09-27 14:51:03 +0200 (Mi, 27 Sep 2006) | 1 line
fixed a bug which led to chan_list zombies, when the call could not be properly established in misdn_call. also removed the ACK_HDLC stuff which is not really needed.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43855 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) | 1 line
Fix ASN1 description of non-standard Cisco extensions
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r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't treat unknown control frames as voice
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r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't warn on HOLD/UNHOLD control frames
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r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28 Сен 2006) | 1 line
Do not open transmit channel until TCS is received
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43853 f38db490-d61c-443f-a65b-d21fe96a405b
