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r109447 | twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines
Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not
actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
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r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar 2008) | 41 lines
Merged revisions 108737 via svnmerge from
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r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines
Fix a race condition in the SIP packet scheduler which could cause a crash.
chan_sip uses the scheduler API in order to schedule retransmission of reliable
packets (such as INVITES). If a retransmission of a packet is occurring, then the
packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if
a response is received from the packet as previously transmitted, then when we
ACK the response, we will remove the packet from the scheduler and free the packet.
The problem is that both the ACK function and retrans_pkt attempt to acquire the
same lock at the beginning of the function call. This means that if the ACK function
acquires the lock first, then it will free the packet which retrans_pkt is about to
read from and write to. The result is a crash.
The solution:
1. If the ACK function fails to remove the packet from the scheduler and the retransmit
id of the packet is not -1 (meaning that we have not reached the maximum number of
retransmissions) then release the lock and yield so that retrans_pkt may acquire the
lock and operate.
2. Make absolutely certain that the ACK function does not recursively lock the lock in
question. If it does, then releasing the lock will do no good, since retrans_pkt will
still be unable to acquire the lock.
(closes issue #12098)
Reported by: wegbert
(closes issue #12089)
Reported by: PTorres
Patches:
12098-putnopvutv3.patch uploaded by putnopvut (license 60)
Tested by: jvandal
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r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008) | 18 lines
Merged revisions 108530 via svnmerge from
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r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines
Make a tweak that gets the LEDs on polycom phones to blink when an extension that
has been subscribed to goes on hold. Otherwise, they just stay on like it does
when an extension is in use.
(closes issue #11263)
Reported by: russell
Patches:
notify_hold.rev1.txt uploaded by russell (license 2)
Tested by: russell
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r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar 2008) | 22 lines
Merged revisions 108288 via svnmerge from
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r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines
Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip.
The scheduler callback will always return 0. This means that this id
is never rescheduled, so it makes no sense to loop trying to delete
the id from the scheduler queue. If we fail to remove the item from the
queue once, it will fail every single time.
(Yes I realize that in this case, the macro would exit early because the
id is set to -1 in the callback, but it still makes no sense to use
that macro in favor of calling ast_sched_del once and being done with it)
This is the first of potentially several such fixes.
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r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar 2008) | 14 lines
Merged revisions 108086 via svnmerge from
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r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines
if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP
closes issue #11475
Reported by: andrebarbosa
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r105734 | russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines
Fix some bugs in the SIP tcp helper thread.
- fix a spot where a lock wouldn't get unlocked in an error condition
- call ast_mutex_destroy() on the lock before freeing its memory
(related to issue #11972)
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r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | 16 lines
Merged revisions 105674 via svnmerge from
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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson
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automatically generated file like it used to be. This still needs to be there
for modules that have to check it to compile against multiple asterisk versions.
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r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6 lines
Make it so a users.conf user creates both a SIP peer and a SIP user. The user will be used for inbound authentication for the device, and peer will be used for placing calls to the device.
(closes issue #9044)
Reported by: queuetue
Patches:
sip-gui-friend.diff uploaded by qwell (license 4)
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r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6 lines
If a resubscription comes in for a dialog we no longer know about tell the remote side that the dialog does not exist so they subscribe again using a new dialog.
(closes issue #10727)
Reported by: s0l4rb03
Patches:
10727-2.diff uploaded by file (license 11)
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r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6 lines
Due to recent changes tag will no longer be NULL if not present so we have to use ast_strlen_zero to see if it's actually blank.
(closes issue #12061)
Reported by: flefoll
Patches:
chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll (license 244)
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r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) | 9 lines
When a SIP channel is being auto-destroyed, it's possible for it to still be
in bridge code. When that happens, we crash. Delay the RTP destruction until
the bridge is ended.
(closes issue #11960)
Reported by: norman
Patches:
20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: norman
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(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
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r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8 lines
Handle ACK and CANCEL in an invite transaction - even if we get INFO transactions during the actual call setup.
(closes issue #10567)
Reported by: jacksch
Tested by: oej
Patch by: oej inspired by suggestions from neutrino88 in the bug tracker
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r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008) | 5 lines
Change the SDP_SAMPLE_RATE macro. It turns out that even though G.722 is 16 kHz,
it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed
to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but
people follow it anyway, because it's the spec ...)
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