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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318106 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317996 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317970 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317868 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317866 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317862 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317860 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317843 f38db490-d61c-443f-a65b-d21fe96a405b
automatically stopped when applications are executed, so this shouldn't cause
any problems.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317806 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317803 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317721 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317671 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317596 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317531 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@317479 f38db490-d61c-443f-a65b-d21fe96a405b
I excluded the part of this patch that used the HOME environment variable since
the built-in editline library goes to great lengths to disallow that. Instead
only settings the EDITRC environment variable will use a user specified file.
Also, the default environment variable use to determine the edit more is
AST_EDITMODE instead of AST_EDITOR (although the latter is still supported).
(closes issue #15929)
Reported by: kkm
Patches:
astcli-editrc-v2.diff uploaded by kkm (license 888)
015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888)
Tested by: seanbright
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317395 f38db490-d61c-443f-a65b-d21fe96a405b
skinny_hold moved to setsubstate_hold and skinny_unhold integrated into
setsubstate_connected. Removed sub->onhold and replaced with
SUBSTATE_HOLD.
Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on
answering a SUBSTATE_RINGIN sub (was a typo).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317382 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May 2011) | 10 lines
Don't duplicate our data on the stack and just use the MYSQL_ROW directly.
With large result sets we were blowing out the stack.
(closes issue #19090)
Reported by: mickecarlsson
Patches:
issue19090_trunk_svn.patch uploaded by seanbright (license 71)
Tested by: mickecarlsson
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317377 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | 10 lines
Resolves a deadlock that occurs during sip_new
This is based on an uncommitted patch by jpeeler for the issue. Instead of
relocking and then unlocking the channel though, we keep the lock on the channel
until we are finished doing what we need to the channel.
(closes issue #18441)
Reported by: Alric
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317334 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317281 | russell | 2011-05-05 13:39:44 -0500 (Thu, 05 May 2011) | 29 lines
Merged revisions 317255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r317255 | russell | 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) | 22 lines
Merged revisions 317211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines
chan_sip: fix broken realtime peer count, fix memory leak
This patch addresses two bugs in chan_sip:
1) The count of realtime peers and users was off. The increment checked the
value of the caching option, while the decrement did not.
2) Add a missing regfree() for a regex.
(closes issue #19108)
Reported by: vrban
Patches:
missing_regfree.patch uploaded by vrban (license 756)
sip_object_counter.patch uploaded by vrban (license 756)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317282 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May 2011) | 8 lines
Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer
abruptly disappears. This mostly occurs after a successful registration.
(closes issue #17544)
Reported by: marcelloceschia
Patches:
(modified) tcptls.patch uploaded by st (license 907)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317198 f38db490-d61c-443f-a65b-d21fe96a405b
mnicholson