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r158959 | seanbright | 2008-11-24 20:01:49 -0500 (Mon, 24 Nov 2008) | 8 lines
This is basically a complete rollback of r155401, as it was determined that
it would be best to maintain API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.
Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
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r158808 | twilson | 2008-11-24 12:11:08 -0600 (Mon, 24 Nov 2008) | 8 lines
This patch adds a new application for sending MWI to phones via Asterisk's event subsystem. Also, the minivm documentation is all converted to use xmldocs.
(closes issue #13946)
Reported by: Marquis
Patches:
minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32)
Tested by: otherwiseguy, Marquis
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r158756 | seanbright | 2008-11-22 22:36:52 -0500 (Sat, 22 Nov 2008) | 6 lines
If you enabled 'notifycid' one of the limitations is that the calling channel
is only found if it dialed the extension that was subscribed to. You can now
specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as
it's value implies, ignore the current context of the caller when doing the
lookup.
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r158754 | seanbright | 2008-11-22 22:30:46 -0500 (Sat, 22 Nov 2008) | 3 lines
No need to use a separate structure for this since we can just pass
our sip_pvt pointer in directly.
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r158723 | mvanbaak | 2008-11-22 18:17:33 +0100 (Sat, 22 Nov 2008) | 4 lines
last commit worked on OpenBSD but still generated warning on Ubuntu.
Initialise a variable so --enable-dev-mode does not complain
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r158694 | mvanbaak | 2008-11-22 17:57:11 +0100 (Sat, 22 Nov 2008) | 8 lines
dont send reorder tone after a device is hungup if a dialout is abandoned or failed.
Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box
(closes issue #13948)
Reported by: wedhorn
Patches:
switch.diff uploaded by wedhorn (license 30)
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r158606 | murf | 2008-11-21 16:40:46 -0700 (Fri, 21 Nov 2008) | 19 lines
Merged revisions 158603 via svnmerge from
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r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) | 11 lines
In reference to the fix made for 13871, I was
merging the fix into 1.6.0 and realized I missed
the code in the h-exten block, and didn't catch it
because my test case had the h-exten commented out.
So, this corrects the code I missed, as a
preventative against another crash report.
Tested with the h-exten defined, all is well.
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r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines
Merged revisions 158600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines
The passed extension may not be the same in the list as the current entry,
because we strip spaces when copying the extension into the structure.
Therefore, use the copied item to place the item into the list.
(found by lmadsen on -dev, fixed by me)
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r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | 19 lines
Merged revisions 158483 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | 11 lines
(closes issue #13871)
Reported by: mdu113
This one is totally my fault. The code doesn't even
create a bridge CDR if the channel CDR has POST_DISABLED.
I didn't check for that at the end of the bridge.
Fixed with a few small insertions. Tested. Looks
good. No cdr generated, no crash, no unnecc. data
objects created either.
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r158374 | twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines
Reloading the config and having no changes still initialized some settings to 0. Initialize settings after doing all of the cfg checks.
(closes issue #13942)
Reported by: davidw
Patches:
cdr_diff.txt uploaded by otherwiseguy (license 396)
Tested by: davidw
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r158307 | mmichelson | 2008-11-21 09:25:58 -0600 (Fri, 21 Nov 2008) | 12 lines
Blocked revisions 158306 via svnmerge
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r158306 | mmichelson | 2008-11-21 09:24:19 -0600 (Fri, 21 Nov 2008) | 5 lines
This change had somehow gotten reverted due to a
completely unrelated commit. Thanks to Theo Belder
on the Asterisk-dev list for pointing this out.
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r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines
Use some magic constants to get the right size
for this sscanf statement. Thanks Richard!
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r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines
Use a more expressive constant for a 64-bit scanned int
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r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines
Fix the build for 32-bit systems. %lu is only 32-bits
on 32-bit systems, so we need to use %llu instead. Of course
%llu is 128-bits on 64-bit systems, so we have to cast to
unsigned long long. No harm, but it's sure annoying.
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r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines
Change the remote user agent session version variable
from an int to a uint64_t. This prevents potential comparison
problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could
not properly place a call on hold since the version in the
SDP of the re-INVITE to place the call on hold greatly
exceeded INT_MAX.
This also aligns with RFC 2327 better since it recommends
using an NTP timestamp for the version (which is a
64-bit number).
(closes issue #13531)
Reported by: sgofferj
Patches:
13531.patch uploaded by putnopvut (license 60)
Tested by: sgofferj
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r158188 | seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 lines
Fix one case where the application argument was not converted from a pipe to
a comma. This was causing problems with switch statements with empty expressions.
(closes issue #13901)
Reported by: smurfix
Patches:
20081118_bug13901.diff uploaded by seanbright (license 71)
Tested by: seanbright
Reviewed by: murf
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r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines
Merged revisions 158071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
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r158062 | jpeeler | 2008-11-20 11:37:31 -0600 (Thu, 20 Nov 2008) | 6 lines
(closes issue #12929)
Reported by: snyfer
This handles the case for a zero length file to attempt to be streamed. Instead of failing from not playing any data, go ahead and return success as ast_streamfile should consider playing nothing a success when there is nothing to play.
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r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines
Merged revisions 158053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
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r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov 2008) | 13 lines
Merged revisions 157859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
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r157870 | tilghman | 2008-11-19 15:54:39 -0600 (Wed, 19 Nov 2008) | 10 lines
Two new functions, REALTIME_FIELD, and REALTIME_HASH, which should make
querying realtime from the dialplan a little more consistent and easy to use.
The original REALTIME function is preserved, for those who are already
accustomed to that interface.
(closes issue #13651)
Reported by: Corydon76
Patches:
20081119__bug13651__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage, Corydon76
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r157818 | twilson | 2008-11-19 13:25:14 -0600 (Wed, 19 Nov 2008) | 2 lines
Fix checking for CONFIG_STATUS_FILEINVALID so that modules don't crash upon trying to parse an invalid config
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r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines
make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
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r157639 | tilghman | 2008-11-18 19:02:45 -0600 (Tue, 18 Nov 2008) | 7 lines
Starting with a change to ensure that ast_verbose() preserves ABI compatibility
in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also
deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy
in core functions. va_copy() is C99, anyway, and we already require C99 for
other purposes, so this isn't really a big change anyway. This change also
simplifies some of the core ast_str_* functions.
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r157632 | mmichelson | 2008-11-18 18:59:48 -0600 (Tue, 18 Nov 2008) | 10 lines
If malloc returns NULL, we need to return NULL immediately or
else Asterisk will crash when attempting to dereference the NULL
pointer
(closes issue #13858)
Reported by: eliel
Patches:
astmm.c.patch uploaded by eliel (license 64)
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r157600 | seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 lines
Fix a few build problems on Solaris (and check for an md5 utility in
configure instead of the icky loop I was doing before).
(closes issue #13842)
Reported by: snuffy
Patches:
bug13842_20081106.diff uploaded by snuffy (license 35)
13842.diff uploaded by seanbright (license 71)
Tested by: snuffy
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r157592 | mmichelson | 2008-11-18 17:59:02 -0600 (Tue, 18 Nov 2008) | 10 lines
This change prevents a crash from occurring if res_musiconhold.so
is unloaded and then Asterisk is stopped. The problem was that
we are not unregistering the ast_moh_destroy function at exit.
(closes issue #13761)
Reported by: eliel
Patches:
res_musiconhold.c.patch uploaded by eliel (license 64)
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