Age | Commit message (Collapse) | Author |
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bitfield.
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dialplan function. Also make fax_rate_str_to_int() return an unsigned int and return 0 instead of -1 in the event of an error.
FAX-202
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines
Fix problem with RFC 2833 DTMF not being accepted.
A recent check was added to ensure that we did not erroneously
detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that
the seqno of an RTP stream will roll over back to 0 after hitting
65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(closes issue #17571)
Reported by: mdeneen
Patches:
rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
Tested by: richardf, maxochoa, JJCinAZ
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r274093 | mnicholson | 2010-07-06 08:52:28 -0500 (Tue, 06 Jul 2010) | 2 lines
Make get_member_status return QUEUE_NO_MEMBERS instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work again. This regression was introduced in 273639. Also fixed whitespace.
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r273981 | tilghman | 2010-07-05 14:48:42 -0500 (Mon, 05 Jul 2010) | 2 lines
Command 'stop gracefully' doesn't.
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r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines
Remove extra line breaks from 'core show config mappings'
(closes issue #17583)
Reported by: pabelanger
Patches:
issue17583.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
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r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
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r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines
Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup.
(closes issue #17564)
Reported by: ramonpeek
Patches:
20100630__issue17564.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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r273639 | tilghman | 2010-07-02 10:46:27 -0500 (Fri, 02 Jul 2010) | 8 lines
If all members are paused, the wrong status is indicated.
(closes issue #17576)
Reported by: ramonpeek
Patches:
diff.txt uploaded by ramonpeek (license 266)
Tested by: ramonpeek
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force directly to the default case.
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(Also fix the typos in the comments)
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r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines
Don't return a partially initialized datastore.
If memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen.
(related to ABE-2415)
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r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines
Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following:
conf => 2345,,6666
did not prompt for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew the correct
pin. The original bug report was submitted claiming that the blank user pin
should deny entry into the conference. I think a better way to handle this
would be with a feature enhancement that used the following syntax:
conf => 2345,X,6666 - where X denotes no acceptable pin allowed
(closes issue #15704)
Reported by: modelnine
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FAX-177
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A failure when calling the get_destination can mean multiple things. If
the extension is not found, a 404 error is appropriate, but if the URI
scheme is incorrect, a 404 is not approperiate. This patch adds the
get_destination_result enum to differentiate between these and other failure
types. The only logical difference in this patch is that we now send a "416
Unsupported URI scheme" response instead of a "404" when the scheme is not
recognized. This indicates to the initiator of the INVITE to retry the request
with a correct URI.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines
Ensure channel placed in meetme in ringing state is properly hung up.
An outgoing channel placed in meetme while still ringing which was then hung up
would not exit meetme and the channel was not properly destroyed. Specifically
checking for this scenario by looking at the appropriate control frames resolves
the issue.
(closes issue #15871)
Reported by: Ivan
Patches:
meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229)
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(closes issue #16430)
Reported by: azbest
Tested by: azbest
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packets. This regression was introduced in r48338.
AST-359
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Also clean up some coding errors.
(closes issue #17469)
Reported by: wdoekes
Patches:
astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, pabelanger
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(closes issue #17550)
Reported by: kenner
Patches:
manager.c.diff uploaded by kenner (license 1040)
Tested by: kenner
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(closes issue #17560)
Reported by: Nick_Lewis
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r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines
Allow the "useragent" value to be restored into memory from the realtime backend.
This value is purely informational. It does not alter configuration at all.
(closes issue #16029)
Reported by: Guggemand
Patches:
realtime-useragent.patch uploaded by Guggemand (license 897)
Tested by: Guggemand
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r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) | 4 lines
_Really_ skip the channel... don't just retry for another 200 cycles.
(Closes issue SWP-1652, ABE-2240)
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Otherwise, it goes to all manager sessions and may exclude the current session,
if the Events mask excludes it.
(closes issue #17504)
Reported by: rrb3942
Patches:
showdialplan_patch.diff uploaded by rrb3942 (license 1003)
Tested by: rrb3942
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RFC 2361 section 24.4.1 send a 400 Bad Request if the request
can not be understood due to malformed syntax. Currently we
simply ignore a packet with a missing callid, to, from, or
via header. Instead of ignoring we now send the 400 Bad request.
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r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) | 8 lines
Don't change ownership/group/permissions on run directory, if it already exists.
(closes issue #17076)
Reported by: stuarth
Patches:
20100324__issue17076.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
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r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) | 8 lines
Change the way that we read include files, to accommodate for changes in GCC 4.4.
(closes issue #17472)
Reported by: seandarcy
Patches:
config2.patch uploaded by nivan (license 1066)
Tested by: nivan
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r272922 | tilghman | 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines
Also trim trailing blanks on #includes
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RFC 3261 section 8.2.2.3 states that if any unsupported options
are found in the Require header field, a "420 (Bad Extension)"
response should be sent with an Unsupported header field containing
only the unsupported options.
This is not currently being done correctly. Right now, if Asterisk
detects any unsupported sip options in a Require header the entire
list of options are returned in the Unsupported header even if some
of those options are in fact supported. This patch fixes that by
building an unsupported options character buffer when parsing the
options that can be sent with the 420 response. A unit test verifying
this functionality has been created. Some code refactoring was required.
Review: https://reviewboard.asterisk.org/r/680/
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r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines
Decode URI in contact header of 302 response.
ABE-2352
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I am doing work in this function. I noticed a large number of
coding guidline fixes that needed to be made. Rather than have
those changes distract from my functional changes I decided
to separate these into a separate patch.
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r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines
Make the structure of the table specified before match the queries and results.
(closes issue #17557)
Reported by: cmaj
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RFC3261 states that Timer A should start at 500ms (T1) by default.
In chan_sip this value initially started at 1000ms and I changed
it to 500ms recently. After doing that I noticed in my packet
captures that it still occasionally retransmitted starting at
1000ms instead of 500ms like I told it to. This occurs because
the scheduler runs in the do_monitor thread. If a new retransmission
is added while the do_monitor thread is sleeping then it may not
detect that retransmission for nearly 1000ms. To fix this I just
poke the do_monitor thread to wake up when a new packet is sent
reliably requiring retransmits. The thread then detects the new
scheduler entry and adjusts its sleep time to account for it.
Review: https://reviewboard.asterisk.org/r/747
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are on the same system.
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r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines
ss_thread calls pri_grab without lock during overlap dial
Recent changes to chan_dahdi with relation to overlap dialing call
pri_grab without first obtaining a lock.
(closes issue #17414)
Reported by: pdf
Patches:
bug17414.patch uploaded by jpeeler (license 325)
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The external test suite stops Asterisk using the "core stop gracefully" command.
The logs from the tests show that there are a number of problems with Asterisk
trying to cleanly shut down. This patch addresses the following type of error
that comes from chan_iax2:
[Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy:
chan_iax2.c line 11371 (iax2_process_thread_cleanup):
Error destroying mutex &thread->lock: Device or resource busy
For an example in the context of a build, see:
http://bamboo.asterisk.org/browse/AST-TRUNK-739/log
The primary purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits from a point
where it is not holding a lock. While testing that, I encountered various
crashes due to the order of operations in unload_module() being problematic.
I reordered some things there, as well.
Review: https://reviewboard.asterisk.org/r/736/
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This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers.
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r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines
Send AgentComplete manager events in the event of blind and attended transfers.
(closes issue #16819)
Reported by: elbriga
Patches:
app_queue.diff uploaded by elbriga (license 482)
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the config file also changes.
(closes issue #16982)
Reported by: dmitri
Patches:
res_musiconhold.patch uploaded by dmitri (license 1001)
Tested by: atis
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(closes issue #17215)
Reported by: vazir
Patches:
20100518__issue17215.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines
First caller into a dynamic conference now enter pin once.
If MeetMe is configured to use dynamic conference
numbers, then the first caller (which creates the
conference) had to enter the PIN number twice.
(closes issue #15878)
Reported by: shawkris
Patches:
issue15878.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
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(closes issue #16991)
Reported by: pprindeville
Patches:
with_netsnmp.patch.txt uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/739/
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