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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.17-rc1@301945 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.17-rc1@301944 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.17-rc1@301943 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.17-rc1@301942 f38db490-d61c-443f-a65b-d21fe96a405b
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r301869 | lmadsen | 2011-01-14 14:21:00 -0600 (Fri, 14 Jan 2011) | 7 lines
Fix issue with cross-compiling failing
(closes issue #18301)
Reported by: abelbeck
Patches:
asterisk-1.4-bugid18301.patch.txt uploaded by abelbeck (license 946)
Tested by: abelbeck, russellb
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Fix amatuer type mistake
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301848 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301842 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18607)
Reported by: cjacobsen
Patches:
polycom_331.diff uploaded by cjacobsen (license 1029)
Tested by: lathama
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301730 f38db490-d61c-443f-a65b-d21fe96a405b
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When merging another SUBSCRIBE fix from 1.4, some braces were put in
the wrong place. This patch fixes that.
(closes issue #18597)
Reported by: thsgmbh
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301682 f38db490-d61c-443f-a65b-d21fe96a405b
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ms_t member from the mansession_session structure.
Merged revisions 301591 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines
Don't store the thread id for the manager session in the structure we pass to
the thread for the manager session.
ABE-2543
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines
Fix CPU spike when pressing DTMF after agent login.
The problem here is that DTMF was being continuously deferred and requeued
since ast_safe_sleep is called in a loop. There are serveral other places in the
code that sleeps and then loops in a similar fashion. Because of this fact I
opted to not defer DTMF any more, which will not affect the original fix:
https://reviewboard.asterisk.org/r/674
(closes issue #18130)
Reported by: rgj
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301310 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines
Prevent buffer overflows in ast_uri_encode()
ABE-2705
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Sounds files included in the Asterisk tarball were being ignored and
re-downloaded. Users wanting to cache the files can still override the setting
using the --with-sounds-cache option.
(closes issue #18589)
Reported by: pabelanger
Patches:
issue18589.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
Review: https://reviewboard.asterisk.org/r/1074/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301220 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18586)
Reported by: kshumard
Patches:
app_verbose.c.patch uploaded by kshumard (license 92)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301176 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18182)
Reported by: dimas
Patches:
v1-18182.patch uploaded by dimas (license 88)
Tested by: dimas
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301089 f38db490-d61c-443f-a65b-d21fe96a405b
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I had actually already fixed this in 295200 in 1.4 and thought it wasn't
missing in the other branches for some reason.
(closes issue #18358)
Reported by: cabal95
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301046 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines
Ensure good bye prompt in voicemail is played at the correct time.
Specifically in the case of timing out but not leaving voicemail nothing
should be heard. And when leaving voicemail it should be heard.
ABE-2647
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) | 10 lines
Use the sanity check in place of the disconnect/connect cycle.
The disconnect/connect cycle has the potential to cause random crashes.
(closes issue #18243)
Reported by: ks3
Patches:
res_odbc.patch uploaded by ks3 (license 1147)
Tested by: ks3
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Also removed latter part of message
Discussed on #asterisk-dev
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300574 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18547)
Reported by: jcovert
Patches:
xmldoc.c.patch uploaded by jcovert (license 551)
chan_iax2.c.doc.patch uploaded by jcovert (license 551)
chan_sip.c.patch uploaded by jcovert (license 551)
chan_agent.c.patch uploaded by jcovert (license 551)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300520 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18531)
Reported by: lathama
Patches:
users.conf.sample2.diff uploaded by lathama (license 1028)
Tested by: lathama
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300431 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011) | 4 lines
Update the autosupport script from Digium support.
(closes AST-395)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines
Don't authenticate SUBSCRIBE re-transmissions
This only skips authentication on retransmissions that are already
authenticated. A similar method is already used for INVITES. This
is the kind of thing we end up having to do when we don't have a
transaction layer...
(closes issue #18075)
Reported by: mdu113
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: twilson, mdu113
Review: https://reviewboard.asterisk.org/r/1005/
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* Misc formatting changes.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300165 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299864 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) | 5 lines
Move check for extension existence below variable inheritance, due to the possible use of an eswitch.
(closes issue #16228)
Reported by: jlaguilar
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299533 f38db490-d61c-443f-a65b-d21fe96a405b
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accepted
(closes issue #18438)
Reported by: mariner7
Tested by: moy
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299530 f38db490-d61c-443f-a65b-d21fe96a405b
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from the "s" extension as used by the AEL macros.
(closes issue #18480)
Reported by: nivek
Patches:
20101215__issue18480__2.diff.txt uploaded by tilghman (license 14)
Tested by: nivek
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299448 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines
Respond as soon as possible with a 202 Accepted to refer requests.
This change also plugs a few memory leaks that can occur when parking sip calls.
ABE-2656
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r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines
Remove changes to via processing that were not supposed to go into the last commit.
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r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines
Use ast_free() instead of free()
ABE-2656
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299136 f38db490-d61c-443f-a65b-d21fe96a405b
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Also, due to a copy and paste error, a request for the answer field would have
given the start value, instead.
(closes issue #18460)
Reported by: joscas
Patches:
20101215__issue18460.diff.txt uploaded by tilghman (license 14)
Tested by: joscas
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299130 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15758)
Reported by: mmurdock
Tested by: mmurdock, seanbright
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299087 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299003 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298962 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines
Let Asterisk find better backtrace information with libbfd.
The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search
for better symbol information within both the Asterisk binary, as well as
loaded modules, to assist when using inline backtraces to track down problems.
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some platforms.
(closes issue #18493)
Reported by: pprindeville
Patches:
asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347)
Tested by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298817 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) | 2 lines
After recording only silence for a voicemail prepending, restore backup files.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines
Fix improper hangup when doing an attended transfer to queue.
Had to indicate ringing in wait_for_answer so the attended transfer code would
not try and hang up the local channel it created, which would kill the call.
ABE-2624
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010) | 14 lines
Only increment the pointer once per loop, otherwise we corrupt the value.
(closes issue #18251)
Reported by: bcnit
Patches:
20101110__issue18251.diff.txt uploaded by tilghman (license 14)
Tested by: trev, jthurman, elguero
(closes issue #18279)
Reported by: zerohalo
Patches:
20101109__issue18279.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298481 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18091)
Reported by: bunny
Patches:
20101006__issue18091.diff.txt uploaded by tilghman (license 14)
Tested by: bunny
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298477 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010) | 8 lines
Unregister before shutting down the connection, to avoid a race.
(closes issue #18481)
Reported by: pabelanger
Patches:
20101215__issue18481.diff.txt uploaded by tilghman (license 14)
Tested by: pabelanger
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r298345 | seanbright | 2010-12-15 16:28:29 -0500 (Wed, 15 Dec 2010) | 6 lines
Fix reference and container leaks when running 'astobj2 test.'
We need to make sure that ao2_iterator_destroy is called once for each time that
ao2_iterator_init is called. Also make sure to unref a newly allocated object
that we've linked into a container.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) | 19 lines
Outgoing PRI/BRI calls cannot do DTMF triggered transfers.
Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING
message is not received. The debug output shows that the DTMF begin event
is seen, but the DTMF end event is missing. When the DTMF begin happens,
the call is muted so we now have one way audio (until a DTMF end event is
somehow seen).
* Made set the proceeding flag when the PRI_EVENT_ANSWER event is
received.
* Made absorb the DTMF begin and DTMF end events if we are overlap dialing
and have not seen a PROCEEDING message.
* Added a debug message when absorbing a DTMF event.
JIRA SWP-2690
JIRA ABE-2697
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Also detect the required structure element, because OpenSolaris defines
SIOCGIFHWADDR, but without support for IP sockets.
(closes issue #18442)
Reported by: ranjtech
Patches:
20101209__issue18442.diff.txt uploaded by tilghman (license 14)
Tested by: ranjtech
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298050 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines
Ignore spurious REGISTER requests
If a REGISTER request with a Call-ID matching an existing transaction is received
it was possible that the REGISTER request would overwrite the initreq of the
private structure. This info is used to generate messages for other responses in
the transaction. This patch ignores REGISTER requests that match non-REGISTER
transactions.
(closes issue #18051)
Reported by: eeman
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/1050/
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(from an internal Digium discussion)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines
Revert code that changed SSRC for DTMF.
Some previous behavior was attempted to be restored, but mistakingly I did
not realize that the previous behavior was incorrect. This fixes DTMF not
being detected since DTMF shouldn't cause the SSRC to change.
(related to issue #17404)
(closes issue #18189)
(closes issue #18352)
Reported by: marcbou
Tested by: cmbaker82
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010) | 4 lines
Use non-deprecated APIs for CoreAudio
Review: https://reviewboard.asterisk.org/r/1040/
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