Age | Commit message (Collapse) | Author |
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(closes issue #12891)
Reported by: chris-mac
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(closes issue #12888)
Reported by: slavon
Patches:
app_queue_transfer_patch_trunk.diff uploaded by slavon (license 288)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) | 8 lines
Add support for saying numbers in Hebrew.
(closes issue #11662)
Reported by: greenfieldtech
Patches:
say.c.patch-12042008 uploaded by greenfieldtech (license 369)
Hebrew-Sounds.ods uploaded by greenfieldtech
(with signficant changes to the spreadsheet by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008) | 7 lines
Set the variables top-down, so that if a script sets a variable more than once,
the last one will take precedence.
(closes issue #12673)
Reported by: phber
Patches:
20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)
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removed. No telling when it happened. Anyway, it's back in now
and works properly.
(Based on issue reported on mailing list)
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with the attended transfer logging.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123650 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #12840)
Reported by: pputman
Patches:
app_dial_threadsafe3.patch uploaded by pputman (license 81)
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counts so that they only get destroyed when all threads are done using
them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #12149)
Reported by: davevg
Patches:
app_talkdetect.c.diff uploaded by davevg (license 209)
(Plus a few additional cleanups by moi)
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a memory leak when running the astobj2 test CLI command. After
searching, it appears the leak was in the command handler itself.
Each object was allocated (recount = 1) and then linked into
a container (refounct = 2). Then at the end of the function,
the container was unreffed, causing all the objects to have
their refcount decremented by one, leaving the refcount for
all objects allocated in that function at 1. I've now added
an extra unref to the mix so that the refcount equals zero
when the container is unreffed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun 2008) | 4 lines
Make chan_sip build under dev mode with compilers >= GCC 4.2
Thanks to jpeeler for alerting me of this
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in 12494. Due to changes in trunk to use the astobj2 i/f in the sip channel driver, the order of the entries in the config file was lost, thus the output was in a random order, but no longer.
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situations where 'settings' is not set to a pointer
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008) | 3 lines
Fix 3 more places where failure to lock the structure could cause the wrong lock to be
unlocked. (Closes issue #12795)
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(closes issue #12835)
Reported by: ys
Patches:
pbx.c.diff uploaded by ys (license 281)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines
Cisco BTS sends SIP responses with a tab between the Cseq number and
SIP request method in the Cseq: header. Asterisk did not handle this
properly, but with this patch, all is well.
(closes issue #12834)
Reported by: tobias_e
Patches:
12834.patch uploaded by putnopvut (license 60)
Tested by: tobias_e
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun 2008) | 12 lines
davidw pointed out that the holdtime calculation used by
app_queue does not use "boxcar" filtering as the comments
say. The term "boxcar" means that the number of samples used
to calculate stays constant, with new samples replacing the
oldest ones. The queue holdtime calculation uses all holdtime
samples collected since the queue was loaded, so the comment
has been changed to be accurate.
(closes issue #12781)
Reported by: davidw
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines
Fix a memory leak in astobj2 that was pointed out by seanbright. When a container
got destroyed, the underlying bucket list entry for each object that was in the
container at that time did not get free'd.
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do_monitor and creating its own thread.
Added RP-AS mwi message generation using patches from meneault as a basis.
(closes issue #8587)
Reported by: meneault
Tested by: meneault
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Reported by: ys
Many thanks to ys for doing the research on this problem.
I didn't think it would be best to unlock the contexts
and then relock them after the remove_extension2() call,
so I added an extra arg to remove_extension2() and set it
appropriately in each call. There were not that many.
I considered forcing the code to lock the contexts before
the call to remove_extension2(), but that would require
a slightly greater degree of changes, especially since
the find_context_locked is local to pbx.c
I did a simple sanity test to make sure the code doesn't
mess things up in general.
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delimited. closes issue #12804
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines
Port "hasvoicemail" change from SIP to other channel drivers
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines
People expect that if "hasvoicemail" is set in users.conf, even if "mailbox"
isn't set, that SIP will detect a mailbox.
(closes issue #12855)
Reported by: PLL
Patches:
20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: PLL
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does not require DAHDI. It's called "pthread" because it uses a pthread
API call in the timing thread for sleeping and ensuring we wake up at
an appropriate time. I wasn't sure what else to call it. :)
The timing API requires a file descriptor that can be polled on. So,
when you open a timer, this module creates a pipe and returns the read
end of the pipe. There is a background thread that wakes up every 10ms
and checks to see if any of the currently open timers need a 'tick' and
writes to the appropriate pipe.
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50 ticks per second, and then counts to see how many ticks it actually
gets in a second.
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- Convert the last part of channel.c over to use the timing API. This would
not have made a difference when using the dahdi timing module. I noticed
it when trying to use another timing source. Oops. :)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines
Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP.
(closes issue #12803)
Reported by: lanzaandrea
Patches:
chan_sip.c.diff uploaded by lanzaandrea (license 496)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines
Don't send a BYE on a dialog that is already gone during a REFER.
(closes issue #12865)
Reported by: flefoll
Patches:
chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll (license 244)
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function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
Reported by: mostyn
Patches:
iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
(with some additional cleanup by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun 2008) | 9 lines
Short circuit the loop in autoservice_run if there are no channels to poll.
If we continued, then the result would be calling poll() with a NULL
pollfd array. While this is fine with POSIX's poll(2) system call, those
who use Asterisk's internal poll mechanism (Darwin systems) would have
a failed assertion occur when poll is called.
(related to issue #10342)
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r122663 | jpeeler | 2008-06-13 13:57:24 -0500 (Fri, 13 Jun 2008) | 1 line
fixed dahdi compatability header from assuming either dahdi or zaptel is installed (may not have either)
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r122613 | jpeeler | 2008-06-13 12:36:56 -0500 (Fri, 13 Jun 2008) | 6 lines
(closes issue #12846)
Reported by: Netview
Tested by: jpeeler
Use correct location to search for tonezone.
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