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r318549 | twilson | 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) | 27 lines
Merged revisions 318548 via svnmerge from
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r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) | 19 lines
Clean up several chan_sip reference leaks
Several situations in the code could lead to peers or sip_pvt references
being leaked. This would cause RTP ports to never be destroyed (leading
to exhaustion of all available RTP ports) and memory leaks.
The original patch for this issue from rgagnon was the result of an
obscene amount of testing and hard work, for which I am very grateful. I
did some cleanup and added a few additional refcount fixes that I found.
(closes issue #17255)
Reported by: kvveltho
Patches:
tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202)
Tested by: rgagnon, twilson, wdoekes, loloski
Review: https://reviewboard.asterisk.org/r/1101/
Review: https://reviewboard.asterisk.org/r/1207/
Review: https://reviewboard.asterisk.org/r/1210/
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r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines
Unable to pickup DAHDI/PRI call because call state is reported as DIALING.
The channel state is not updated to RINGING when an ALERTING message is
received. Regression caused when sig_pri.c (also sig_ss7.c) extracted
from chan_dahdi.c.
* Added missing channel state update to RINGING when the
AST_CONTROL_RINGING frame is queued for ISDN and SS7.
(closes issue #19257)
Reported by: alecdavis
Patches:
issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
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r318436 | russell | 2011-05-10 10:13:16 -0500 (Tue, 10 May 2011) | 2 lines
chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read().
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r318337 | twilson | 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) | 18 lines
Merged revisions 318331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines
Don't offer video to directmedia callee unless caller offered it as well
Make sure that when directmedia is enabled, that video is not offered to the
callee even if it supports it. p->vrtp will not exist since the caller didn't
offer video.
(closes issue #19195)
Reported by: one47
Patches:
sip_cant_add_video_rtp uploaded by one47 (license 23)
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r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) | 6 lines
Remove references to res_features and its export file.
The contents of res/res_features.c was moved to into main/features.c
awhile ago. There is no longer any need for the res/Makefile to reference
res_features or the res_features linker exports file to exist.
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r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) | 24 lines
Hangup extension executed twice.
When a user hangs up a call, in certain circumstances, the hangup
extension can end up being executed twice:
1) If a call is bridged and the 'h' extension executes the Hangup
application, then the 'h' extension will be executed twice.
2) If a call is bridged within a macro (Dial or Queue), it has its own 'h'
extension, the main context also has an 'h' extension, and the macro 'h'
extension executes the Hangup application, then both 'h' extensions will
be executed.
* Revert originally commited fix for #16106 and just set
AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call(). The
bridge code just executed an 'h' extension so the main PBX loop does not
need to execute one as well.
(issue #16106)
Reported by: ajohnson
(issue #16548)
Reported by: hajekd
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r318233 | dvossel | 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) | 14 lines
Merged revisions 318230 via svnmerge from
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r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) | 7 lines
Fixes cases where sip_set_rtp_peer can return too early during media path reset.
(closes issue #19225)
Reported by: one47
Patches:
sip_set_rtp_peer.patch uploaded by one47 (license 23)
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r318231 | rmudgett | 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) | 41 lines
Don't get early media for ISDN on outgoing calls.
It looks to be a long-standing misinterpretation of the progress indicator
ie values:
1 - Call is not end-to-end ISDN; further call progress information may be
available in-band.
8 - In-band information or an appropriate pattern is now available.
Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled
as early media probably because the meaning of the second half of it's
description was overlooked.
* Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or
PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path.
(closes issue #18868)
Reported by: isrl
Patches:
issue18868_19246_v1.8.patch uploaded by rmudgett (license 664)
Tested by: satish_lx
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No inband progress on PRI_EVENT_RINGING even if inband flag set.
My ISDN-PRI provider sends an ALERTING with "Inband information or
appropriate pattern now available", but Asterisk only generates and passes
the RING to the SIP extension, not the inband message. Unfortunately, the
inband message is not a ringback tone but a prompt that says the number is
not in service. The SIP extension then hears two rings and the call is
hungup which confuses the caller.
* Post an AST_CONTROL_PROGRESS as well as opening the media path if inband
audio is indicated with an ALERTING message.
(closes issue #19246)
Reported by: cristiandimache
Patches:
issue19246_v1.8.patch uploaded by rmudgett (license 664)
Tested by: cristiandimache
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(closes issue #19238)
Reported by: byronclark
Patches:
increase_prepend_filename_length.patch uploaded by byronclark (license 1200)
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r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines
Documenting an observed behavior of features in features.conf. Since parkinglots use an
integer for the parkinglot extensions, leading zeros specified in the configuration file
are ignored.
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r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May 2011) | 9 lines
Make indicate/control frames WRITE events on framehooks. Also, if a framehook
returns a non-control frame, don't forward it to the channel.
(closes issue #19251)
Reported by: irroot
Patches:
(modified) framehook_indicate.patch2 uploaded by irroot (license 52)
Tested by: irroot
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When invoking the app parkedcall, the argument can now include '@parkinglot' after the
extension.
(closes issue #18777)
Reported by: cartama
Patches:
0018777.diff uploaded by cartama (license 1157)
Review: https://reviewboard.asterisk.org/r/1209/
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If a call is made to a line that already has a call and the device is
offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN.
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r318057 | russell | 2011-05-07 18:35:37 -0500 (Sat, 07 May 2011) | 8 lines
res_config_curl: fix a crash with static realtime.
(closes issue #18413)
Reported by: jmls
Patches:
20101202__issue18413.diff.txt uploaded by tilghman (license 14)
Tested by: jmls
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r318055 | russell | 2011-05-07 18:24:18 -0500 (Sat, 07 May 2011) | 7 lines
chan_iax2: Don't overwrite port found with an SRV lookup.
(closes issue #17291)
Reported by: jcovert
Patches:
chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert (license 551)
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(closes issue #17901)
Reported by: salecha
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Improve readability of code, eg, (sub->parent == d->activeline) becomes
(sub->line == d->activeline).
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Long time coming, finally moving the hookstate from line to device.
This may fix some issues where a device has multiple lines. Previously
we had to run through all lines on a device to see if it was actually
onhook or not.
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r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines
Use the right variable to print the time in a debug message.
The original patch also increased some buffer sizes, but that was already
done in this version.
(closes issue #17034)
Reported by: sysreq
Patches:
asterisk-issue-17034.patch uploaded by sysreq (license 1009)
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r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) | 2 lines
Fix some more "set but unused" compiler warnings.
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r317918 | dvossel | 2011-05-06 16:06:55 -0500 (Fri, 06 May 2011) | 7 lines
Fixes missing colon from To/From headers in RTCP manager events.
(closes issue #18221)
Reported by: clegall_proformatique
Patches:
18221_1.patch uploaded by ebroad (license 878)
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r317917 | russell | 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) | 7 lines
Fix calculation of free RAM to make minmemfree option work.
(closes issue #17124)
Reported by: loic
Patches:
pbx_c.diff uploaded by loic (license 1020)
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(closes issue #17036)
Reported by: precisenetworks
Patches:
import-cdr-csv-mysql.pl uploaded by precisenetworks (license 1010)
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(closes issue #16962)
Reported by: jlpedrosa
Patches:
patch.diff uploaded by jlpedrosa (license 1002)
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r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) | 10 lines
chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer.
Don't duplicate variables on the sip_pvt. Just reset the variable list each
time.
(closes issue #19202)
Reported by: wdoekes
Patches:
issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717)
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r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) | 11 lines
chan_sip: fix a deadlock in check_rtp_timeout.
Don't block doing silly deadlock avoidance. Just return and try again later.
The funciton gets called often enough that it's fine. Also, this change was
already made in trunk.
(closes issue #18791)
Reported by: irroot
Patches:
chan_sip.rtptimeout.patch uploaded by irroot (license 52)
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r317861 | russell | 2011-05-06 14:35:00 -0500 (Fri, 06 May 2011) | 11 lines
URI encode less characters in the RPID and Contact headers.
If this change causes any problems, we will need to backport the more extensive
uri encoding and decoding handling changes that are in trunk/1.10.
(closes issue #18686)
Reported by: wolfgang
Patches:
quick-and-dirty.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, devellow, wolfgang, mav3rick
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r317858 | mnicholson | 2011-05-06 14:31:50 -0500 (Fri, 06 May 2011) | 6 lines
pbx_lua autoservice fixes
Don't start an autoservice in pbx_lua if pbx_lua already started one and don't
stop one if we didn't start one. Also start and stop the autoservice when
transferring control from and to the pbx.
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This change is already implemented in trunk.
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r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) | 11 lines
Fix a crash in the MySQL() application.
This code was not handling channel datastores safely. The channel
must be locked.
(closes issue #17964)
Reported by: wuwu
Patches:
issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71)
Tested by: wuwu
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r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) | 7 lines
Add a new sipfriends.sql for MySQL that has more fields in it.
(closes issue #16399)
Reported by: pabelanger
Patches:
sipfriends.sql.v3 uploaded by pabelanger (license 224)
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automatically stopped when applications are executed, so this shouldn't cause
any problems.
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Make autoservice_start() and autoservice_stop() return nothing. Also check if
the autoservice flag is set before starting or stopping the autoservice and
stop and start the autoservice when returning control to and getting control
from the pbx engine.
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jumps
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This code will actually detect any dialplan jump from any application that
calls ast_explicit_goto(). This change is only being done in trunk as it may
change the way some dialplans execute.
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r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines
Fix SIP connected line updates.
This patch fixes a couple SIP connected line update problems:
1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured. Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.
2) The connected line should not be updated on initial connect if there is
no connected line information. Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.
(closes issue #18367)
Reported by: GeorgeKonopacki
Patches:
issue18367_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1199/
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r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines
Merged revisions 317575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines
Merged revisions 317574 via svnmerge from
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r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines
Re-fix queue round-robin
This part of the change for r315596 was incorrect. No bridge occurs
when doing a roundrobin dial and no one answers, so this code shouldn't
have been removed.
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r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011) | 10 lines
If the configure script runs, force a rebuild of menuselect-tree.
Some contents in the menuselect tree are dependent on configure script
parameters, namely --enable-dev-mode.
(closes issue #17219)
Reported by: Nick_Lewis
Patches:
issue_17219.rev1.txt uploaded by russell (license 2)
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r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011) | 9 lines
Fix some more realtime MySQL schema issues.
(closes issue #18537)
Reported by: denzs
Patches:
sipfriends.sql.svndiff uploaded by denzs (license 1182)
queue_log.sql.svndiff uploaded by denzs (license 1182)
meetme.sql.svndiff uploaded by denzs (license 1182)
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r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011) | 8 lines
Fix some errors in sample MySQL realtime schema files.
(closes issue #18915)
Reported by: Dovid
Patches:
sipfriends.patch uploaded by Dovid (license 652)
meetme.patch uploaded by Dovid (license 652)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #18246)
Reported by: junky
Patches:
calendar_types.diff uploaded by junky (license 177)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #18462)
Reported by: joscas
Patches:
bug_18462.diff uploaded by snuffy (license 35)
cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317480 | russell | 2011-05-05 18:00:55 -0500 (Thu, 05 May 2011) | 8 lines
Don't lose cdr_syslog config on a reload.
(closes issue #18679)
Reported by: enegaard
Patches:
issue18679_seanbright.patch uploaded by seanbright (license 71)
Tested by: enegaard
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines
Fix some consistency issues with jitterbuffer config.
Store the defaults noted in the sample config files in the jitterbuffer config
data structure. This makes the CLI commands that output these settings show
the right thing. Also only show the settings that are relevant in the settings
CLI commands, based on which jitterbuffer is selected and whether it's enabled.
(closes issue #19083)
Reported by: rgagnon
Patches:
issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317476 | russell | 2011-05-05 17:47:57 -0500 (Thu, 05 May 2011) | 8 lines
Add a datastore fixup to fix a pbx_lua crash.
(closes issue #19055)
Reported by: jamhed
Patches:
lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, jamhed
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317474 | russell | 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) | 2 lines
Fix more "set but unused" warnings.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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