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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010) | 6 lines
Don't treat warnings as errors for muted.
muted supports OS X, but uses functions marked as deprecated in 10.6. However,
the functions are still supported, so just ignore the warnings for now and
allow the build to proceed.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines
Additional extensions.ael global variable fixes.
Fixing up a couple more overlapping global variable namespaces shared with
extensions.conf.sample. Also noticed a few of the lines that were commented
out didn't have the closing semi-colon so I added that as well.
(issue #17035)
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Previously, values that began with whitespace were silently treated as 'no',
and all non-'yes' values were also treated as 'no'. Now the supplied value
is specifically checked for a 'yes' or 'no' (or equivalent) value, after skipping
leading whitespace. If the value is not valid, then a warning message is generated.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Peer reviewed out-of-band by file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by philipp64 in #asterisk-dev.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010) | 2 lines
Uh, yeah. Umask. I'm stupid.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines
Update extensions.ael file to not overlap extensions.conf.
Updated the extensions.ael file so the global variables don't overlap
those that we have in extensions.conf (sample files). This way unexpected
things won't happed hopefully if both pbx_ael and res_config are loaded.
(closes issue #17035)
Reported by: pprindeville
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to where it's not executed on non-Darwin systems.
(closes issue #17028)
Reported by: pabelanger
Patches:
issue17028_20100315.patch uploaded by seanbright (license 71)
20100315__issue17028.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #17025)
Reported by: smurfix
Patches:
sip.patch uploaded by smurfix (license 547)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252366 | tilghman | 2010-03-14 20:39:00 -0500 (Sun, 14 Mar 2010) | 2 lines
Typo
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010) | 4 lines
Launch Asterisk on Mac OS X with launchd.
Reviewboard: https://reviewboard.asterisk.org/r/551/
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They added a sqlite3_log() function which was conflicting with our function
names.
(closes issue #17017)
Reported by: alephlg
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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added response to roundtrip delay requests from opposite side
added roundtrip delay request sending to opposite side after answer,
added options for sending request (interval between request and
count of unreplied requests before forced call hangup)
(closes issue #16976)
Reported by: vmikhelson
Patches:
rtdr-1.6.0-2.patch uploaded by may213 (license 454)
Tested by: vmikhelson, may213
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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On Linux (glibc), regcomp() does not return an error for an empty string.
However, the version on OSX will return an error. The test for channel group
matching by regex now passes on the mac, as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This resolves the last compile failure on bamboo.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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alteration.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #16849)
Reported by: ip-rob
Patches:
20100311__issue16849.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010) | 1 line
Make chan_dahdi wakeup_sub() prototype not conditional.
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quotes during parsing.
(closes issue #16905)
Reported by: ip-rob
Patches:
20100303__issue16905.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
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quotes before trying to execute.
(closes issue #16842)
Reported by: ip-rob
Patches:
20100310__issue16842.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Since 1.6.1 CLI help reports that option p(n) 'initial pause' is available.
Supporting code was never implemented.
(closes issue #16751)
Reported by: alecdavis
Patches:
directory_pause.trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/481/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Copied from reviewboard)
Tests the following:
1. Basic allocation and setting of string fields.
2. Shrinking a string field and re-expanding it.
3. Growing the last allocation in a string field pool.
4. Setting a string to a large value such that a new string field pool must be
allocated.
In each part, we make sure that the string field is accurate (has the correct
value in it), make sure that the 2 bytes before the string field has the correct
capacity for the field, and for tests 2-4, we make sure that the string field is
where we expect it to be in memory.
Also tested:
5. Shrinking a string field and partially re-expanding it.
6. Setting strings in such a way as to create three separate string field pools
and then removing the middle pool.
There is a bug fix in the init function, which ensures the embedded_pool is set
to NULL which is important for stack allocated structures.
Review: https://reviewboard.asterisk.org/r/185/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #16900)
Reported by: bluecrow76
Patches:
asterisk-1.6.2.4-func_strings.diff uploaded by bluecrow76 (license 270)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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For some reason the documentation for the 'k' application in trunk
and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them all
to match. The wording in 1.6.2 and trunk was ambiguous, so you could
interpret the wording the mean that recording would continue upon hangup
indefinitely, or you could interpret it to mean that the recorded
data would not be discarded upon hangup. This change makes it clear
we mean the latter, and not the former.
Came from a discussion in #asterisk on IRC.
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The patch ensures that if a peer does not exist, parking settings are read from
the channel. A unit test has been written to ensure proper operation for both
standard parking and parking using masquerades.
(closes issue #16592)
Reported by: mwyres
Patches:
bug_16592.diff uploaded by snuffy (license 35)
Review: https://reviewboard.asterisk.org/r/539/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue #16900)
Reported by: bluecrow76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Three changes made here:
1) Do not fail if a previous log does not exist (in fact, this is probably
expected).
2) Ensure that the file descriptor to write to gets assigned properly. I am at
a loss as to why assigning safe_fd outside the if fixes this, but it makes
the if statement slightly less complicated anyway.
3) Move up the failure message so that the errno of the failure is not
overwritten by fclose.
(closes issue #16917)
Reported by: Artem
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Also avoid potential crash because cause could be NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Rework HAVE_PRI_SERVICE_MESSAGES to not use the active values directly
from the database. Database access is likely expensive. Database access
now only happens on initialization, destruction, and when the B channel is
taken in or out of service.
This change is not related to call waiting but it would cause the search
for a call waiting interface to be very expensive and slow down D channel
message servicing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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X 10.6.
(closes issue #16997)
Reported by: jquinn
Patches:
20100309__issue16997__2.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, russell
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r251410 | seanbright | 2010-03-09 14:29:39 -0500 (Tue, 09 Mar 2010) | 8 lines
Use ast_strlen_zero to avoid a crash when a Dial() string isn't passed to ParkAndAnnounce
(closes issue #16731)
Reported by: sebele67
Patches:
issue16731_20100129.diff uploaded by seanbright (license 71)
Tested by: sebele67
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r251309 | lmadsen | 2010-03-08 12:07:44 -0600 (Mon, 08 Mar 2010) | 13 lines
Fix Debian init script to not use -c.
When using the init script as-is currently, it could cause issues on Debian
such as high CPU usage. This fix has worked for several people so I'm
implementing the change.
(closes issue #16784)
Reported by: pabelanger
Tested by: pabelanger, mnick, davidw, mutineer612
(closes issue #16887)
Reported by: jlpedrosa
Tested by: jlpedrosa, mutineer612
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Small patch changing skinny_set_rtp_peer to use transmit_stopmediatransmission and to use new transmit_startmediatransmission.
Basic testing on 30VIP's by wedhorn
Basic testing on 7960 by me
(closes issue #16956)
Reported by: wedhorn
Patches:
skinny-clean05b.diff uploaded by wedhorn (license 30)
Tested by: wedhorn,mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Broke the various functions included in transmit_callstate to their own functions. Transmit_callstate now just transmits callstate.
Generally left the functionality as it was, which highlight some minor code issues (eg multiple transmit_callstate's). I did however revise the hint code usage of the old transmit_callstate as it it not appropriate to put a device on hook based on the change of a hinted device.
(closes issue #16939)
Reported by: wedhorn
Patches:
skinny-clean04.diff uploaded by wedhorn (license 30)
Tested by: mvanbaak,wedhorn
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The first real test added to the external test suite found a pretty nasty crash
that occurred in Asterisk trunk. The crash was due to a race condition between
the REFER handling and channel destruction in the channel thread. After the
transfer has been completed, we go back to the transferrer channel and try to
lock it so we can fire off a CEL event. However, there was no guarantee that
the channel was still around at that point since it's racing against the channel
thread.
Since ast_channel is a reference counted object, the fix is simple. The code
unlocks the transferrer channel before finally completing the transfer with
an async goto. At this point the channel thread is going to start call tear
down and the channel will eventually be destroyed. To ensure that the channel
is valid when we want to fire off the CEL event, increase the channel's
reference count.
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The PITCH_SHIFT function can be used on a channel to independently
modify the pitch of both rx and tx audio streams. Now you can
improve your conference calls by assigning a random pitch effect
to everyone entering a meetme room, or just make your day more
interesting by making your co-workers sound funny. These are just
some of the numerious practical uses for this function. Enjoy!
https://reviewboard.asterisk.org/r/526/
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Review: https://reviewboard.asterisk.org/r/541/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Fixes regression introduced in 140167 that uses the wrong variable names.
(closes issue #16930)
Reported by: ianc
Patches:
fix_reload_followme.diff uploaded by ianc (license 998)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The get_local_address() function for an RTP instance was used when building an
SDP, but the results were not honored. The RTP engine activate() function was
not being used once we have determined that media will now flow.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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