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git.ipfire.org Git - thirdparty/asterisk.git/log
Brett Bryant [Tue, 7 Sep 2010 17:49:07 +0000 (17:49 +0000)]
Merged revisions 285194 via svnmerge from
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r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines
Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages.
(closes issue #15726)
Reported by: 298
Patches:
M15726.diff uploaded by junky (license 177)
Tested by: junky
Review: [full review board URL with trailing slash]
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Tilghman Lesher [Mon, 6 Sep 2010 06:55:17 +0000 (06:55 +0000)]
Merged revisions 285088 via svnmerge from
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r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010) | 2 lines
Silly convenience script for BSD platforms.
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Brett Bryant [Fri, 3 Sep 2010 18:15:49 +0000 (18:15 +0000)]
This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response.
(closes issue #17935)
Reported by: alexkuklin
Patches:
iaxshowreg uploaded by alexkuklin (license 1115)
Tested by: alexkuklin
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Terry Wilson [Fri, 3 Sep 2010 16:20:45 +0000 (16:20 +0000)]
Merged revisions 284881 via svnmerge from
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r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines
Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant file. The existing
code treated missing files as though they existed.
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Brett Bryant [Thu, 2 Sep 2010 20:54:33 +0000 (20:54 +0000)]
Merged revisions 284777 via svnmerge from
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r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010) | 7 lines
Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded.
(closes issue #17917)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/883/
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David Vossel [Thu, 2 Sep 2010 16:48:51 +0000 (16:48 +0000)]
Merged revisions 284703 via svnmerge from
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r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines
Removed relatedpeer code from sip_autodestruct
Handling of the relatedpeer structure associated with a
sip_pvt should be done during the final sip_destruction
function, not in sip_autodestruct.
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Tilghman Lesher [Thu, 2 Sep 2010 16:07:19 +0000 (16:07 +0000)]
Fixing build.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284665
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Tilghman Lesher [Thu, 2 Sep 2010 05:30:16 +0000 (05:30 +0000)]
Don't reset queue stats on a module reload.
(closes issue #17535)
Reported by: raarts
Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284631
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Tilghman Lesher [Thu, 2 Sep 2010 03:57:43 +0000 (03:57 +0000)]
Failed to rerun bootstrap.sh after last commit
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Tilghman Lesher [Wed, 1 Sep 2010 22:59:50 +0000 (22:59 +0000)]
Merged revisions 284478 via svnmerge from
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r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
a potential crash bug in all supported releases.
(closes issue #17678)
Reported by: russell
Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/
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Tilghman Lesher [Wed, 1 Sep 2010 18:13:35 +0000 (18:13 +0000)]
Don't warn on floats and timestamps
(closes issue #17082)
Reported by: coolmig
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Tilghman Lesher [Tue, 31 Aug 2010 20:18:32 +0000 (20:18 +0000)]
Merged revisions 284393 via svnmerge from
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r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines
Don't send a devstate change on poke_noanswer if the state did not change.
(closes issue #17741)
Reported by: schmidts
Patches:
chan_sip.c.patch uploaded by schmidts (license 1077)
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Leif Madsen [Tue, 31 Aug 2010 18:59:31 +0000 (18:59 +0000)]
Merged revisions 284316 via svnmerge from
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r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines
Update say.conf.sample to match the rules in say.c
(closes issue #17835)
Reported by: RoadKill
Patches:
say.conf.sample.patch.rules uploaded by RoadKill (license 933)
Tested by: RoadKill
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Tilghman Lesher [Mon, 30 Aug 2010 22:27:06 +0000 (22:27 +0000)]
Fix 3 coding errors:
1) After we close FD, we should not be trying to write to it.
2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child.
3) Use endian, not processor, detection to ensure bytes are written in the correct order.
(closes issue #15706)
Reported by: modelnine
Patches:
asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865)
Tested by: gmartinez
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David Vossel [Fri, 27 Aug 2010 22:27:50 +0000 (22:27 +0000)]
Merged revisions 283960 via svnmerge from
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r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines
Parse all "Accept" headers for SIP SUBSCRIBE requests.
(closes issue #17758)
Reported by: ibc
Patches:
multiple_accept_headers_1.4.diff uploaded by dvossel (license 671)
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Jason Parker [Fri, 27 Aug 2010 20:30:27 +0000 (20:30 +0000)]
Merged revisions 283880 via svnmerge from
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r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | 8 lines
Fix issue with decoding ^-escaped characters in realtime.
(closes issue #17790)
Reported by: denzs
Patches:
17790-chunky.diff uploaded by qwell (license 4)
Tested by: qwell, denzs
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David Vossel [Thu, 26 Aug 2010 15:24:40 +0000 (15:24 +0000)]
Merged revisions 283690 via svnmerge from
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r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines
Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response.
If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response
to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc
compliant and results in confusion at the other endpoint. sip_pretend_ack will ack
and remove all the packets in the retransmit queue. This means that the INVITE will
stop retransmitting, and that any response to that INVITE that comes after the pretend_ack
occurs will be ignored.
Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal
hangup, we should let the protocol stack process the INVITE transaction and terminate
the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag
is used, once the dialog proceeds to an escapable state the transaction will either be
canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do
this any other way is incorrect. If the endpoint is not responding to the INVITE request,
the INVITE must continue to be retransmitted until it times out which will result in the
dialog being destroyed.
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David Vossel [Wed, 25 Aug 2010 22:56:42 +0000 (22:56 +0000)]
Add to and from tags to NOTIFY dialog-info xml body so pickup can occur.
When pedantic mode is used, the dialog-info xml generated during a
ringing event must contain the to and from tag values. Otherwise if
a pickup occurs using INVITE with replaces, Astrisk will not be able
to locate the subscription.
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David Vossel [Wed, 25 Aug 2010 15:52:54 +0000 (15:52 +0000)]
Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used.
Asterisk now dynamically builds the "Supported" header depending
on what is enabled/disabled in sip.conf. Session timers used
to always be advertised as being supported even when they were disabled
in the configuration. This caused problems with some end points.
(issue #17005)
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David Vossel [Tue, 24 Aug 2010 16:07:37 +0000 (16:07 +0000)]
Merged revisions 283380 via svnmerge from
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r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines
This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set.
When the pending bye flag is used, it is possible that the dialog will terminate
and leave the sip_pvt->owner channel up. This is because we never hangup the
ast_channel after sending the SIP_BYE request. When we receive the response for
the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the
next do_monitor loop, but this is not the case. The dialog will only be destroyed
once the owner is hungup even with the need_destroy flag set. This patch sets the
softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the
pending bye flag.
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Tilghman Lesher [Mon, 23 Aug 2010 21:32:14 +0000 (21:32 +0000)]
CDR drivers depend upon res_odbc, not directly on the ODBC libraries
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Richard Mudgett [Fri, 20 Aug 2010 16:48:10 +0000 (16:48 +0000)]
Merged revisions 283123 via svnmerge from
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r283123 | rmudgett | 2010-08-20 11:46:22 -0500 (Fri, 20 Aug 2010) | 9 lines
Merged revision 278274 from
https://origsvn.digium.com/svn/asterisk/trunk
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r278274 | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1 line
Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR.
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Richard Mudgett [Fri, 20 Aug 2010 15:31:03 +0000 (15:31 +0000)]
Merged revisions 283048 via svnmerge from
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r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines
Q931 - Sending PROGRESS after sending ALERTING is a protocol error
The PRI layer in chan_dadhi will check if a PROGRESS message has already
been sent, and not allow sending another (although that is technically
allowed by the Q931 spec), however it does not protect against sending an
ALERTING and then sending a PROGRESS message, which is a violation of the
specification.
Most switches don't seem to care too deeply about this, but some do, and
will disconnect the call when receiving this invalid sequence.
Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure
A.5/Q.931 -- Overview protocol control (network side) point-point
(sheet 3 of 8)"
(closes issue #17874)
Reported by: nic_bellamy
Patches:
asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
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David Vossel [Thu, 19 Aug 2010 21:05:54 +0000 (21:05 +0000)]
Merged revisions 282893 via svnmerge from
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r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines
tos_sip option was not being set correctly
When tos_sip is used, the tos of the sip socket is only set
correctly if the socket binding changes on a reload. If the binding
stays the same but the TOS changes, the new tos value would not take
into effect. This patch fixes that.
(closes issue #17712)
Reported by: nickb
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David Vossel [Thu, 19 Aug 2010 20:31:22 +0000 (20:31 +0000)]
fixes sip peer memory leaks in the peer_by_ip table
(issue #17798)
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Matthew Nicholson [Thu, 19 Aug 2010 19:44:00 +0000 (19:44 +0000)]
Merged revisions 277944 via svnmerge from
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r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines
Regression with T.38 negotiation
Prior to 1.4.26.3 T.38 negotiation worked properly, in the case
of the reporter.
(issue #16852)
Reported by: cfc
(closes issue #16705)
Reported by: mpiazzatnetbug
Patches:
issue16705_2.diff uploaded by ebroad (license 878)
Tested by: vrban, ebroad, c0rnoTa, samdell3
Review: https://reviewboard.asterisk.org/r/754/
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Terry Wilson [Thu, 19 Aug 2010 02:14:28 +0000 (02:14 +0000)]
Merged revisions 282729 via svnmerge from
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r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines
Add some documentation about codec negotiation to sip.conf
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David Vossel [Wed, 18 Aug 2010 14:28:52 +0000 (14:28 +0000)]
fixes crash with notifycid
(closes issue #17868)
Reported by: francesco_r
Patches:
issue_17868.diff uploaded by dvossel (license 671)
Tested by: francesco_r
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Tilghman Lesher [Wed, 18 Aug 2010 07:43:14 +0000 (07:43 +0000)]
Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes.
(closes issue #16770)
Reported by: jamicque
Patches:
20100413__issue16770.diff.txt uploaded by tilghman (license 14)
20100811__issue16770.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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David Vossel [Tue, 17 Aug 2010 21:35:17 +0000 (21:35 +0000)]
fixes no default transport for temp peer creation in chan_sip
(closes issue #17829)
Reported by: falves11
Patches:
issue_17829.rev1.txt uploaded by russell (license 2)
issue_17829.diff uploaded by dvossel (license 671)
Tested by: falves11
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Leif Madsen [Mon, 16 Aug 2010 18:00:09 +0000 (18:00 +0000)]
Add information about creating sounds files using
the sounds tools publically available so that others can create their
own sounds prompts using the same tools we use to generate sounds releases.
This allows people creating their own prompts to sound consistent with
the prompts available from the open source project.
SWP-595
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Terry Wilson [Mon, 16 Aug 2010 17:32:01 +0000 (17:32 +0000)]
Merged revisions 282430 via svnmerge from
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r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines
Send a SRCCHANGE indication when we masquerade
Masquerading a channel means that the src of the audio is potentially
changing, so send a SRCCHANGE so that RTP-based media streams can get
a new SSRC generated to reflect the change. Original patch by addix
(along with lots of testing--thanks!).
(closes issue #17007)
Reported by: addix
Patches:
1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006)
srcchange.diff uploaded by twilson (license 396)
Tested by: addix, twilson
Review: https://reviewboard.asterisk.org/r/862/
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David Vossel [Fri, 13 Aug 2010 18:54:53 +0000 (18:54 +0000)]
only do magic pickup when notifycid is enabled
A new way of doing BLF pickup was introduced into 1.6.2. This feature
adds a call-id value into the XML of a SIP_NOTIFY message sent to alert
a subscriber that a device is ringing. This option should only be enabled
when the new 'notifycid' option is set... but this was not the case. Instead
the call-id value was included for every RINGING Notify message, which
caused a regression for people who used other methods for call pickup.
(closes issue #17633)
Reported by: urosh
Patches:
chan_sip.txt uploaded by urosh (license )
blf_cid_issue.diff uploaded by dvossel (license 671)
Tested by: dvossel, urosh, okrief, alecdavis
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Jason Parker [Thu, 12 Aug 2010 22:50:54 +0000 (22:50 +0000)]
Merged revisions 282129 via svnmerge from
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r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | 1 line
Register CLI commands before parsing config, in case there is a config error.
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Jeff Peeler [Thu, 12 Aug 2010 03:01:38 +0000 (03:01 +0000)]
Merged revisions 281911 via svnmerge from
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r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines
Ensure SSRC is changed when media source is changed to resolve audio delay.
This change causes the SSRC to change right before the channels are bridged,
which is what used to happen. It seems that fixes were made to attempt limiting
SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC
with this change.
There are two other control frames sent in ast_channel_bridge that probably
should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave
this change up to the discretion of resolving issue #17007.
For reference - old review implementing new control frame SRCCHANGE:
https://reviewboard.asterisk.org/r/540
(closes issue #17404)
Reported by: sdolloff
Patches:
bug17404.patch uploaded by jpeeler (license 325)
Tested by: sdolloff
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Leif Madsen [Wed, 11 Aug 2010 21:09:47 +0000 (21:09 +0000)]
Merged revisions 281819 via svnmerge from
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r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) | 6 lines
Add Danish support to say.conf.sample
(closes issue #17836)
Reported by: RoadKill
Patches:
say.conf.sample.patch.dk uploaded by RoadKill (license 933)
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Leif Madsen [Wed, 11 Aug 2010 17:54:09 +0000 (17:54 +0000)]
Merged revisions 281762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) | 6 lines
Allow say.conf to handle large numbers ending with multiple zeros.
(closes issue #17833)
Reported by: RoadKill
Patches:
say.conf.sample.patch.largenumbers uploaded by RoadKill (license 933)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281763
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Tilghman Lesher [Wed, 11 Aug 2010 15:17:20 +0000 (15:17 +0000)]
Only set status TIMEOUT, if we have no digits.
(closes issue #15188)
Reported by: jcovert
Patches:
app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281722
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Russell Bryant [Tue, 10 Aug 2010 18:04:32 +0000 (18:04 +0000)]
Don't move the time threshold for running scheduled events on every iteration.
Instead, only calculate the time threshold each time ast_sched_runq() is called.
(closes issue #17742)
Reported by: schmidts
Patches:
sched.c.patch uploaded by schmidts (license 1077)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281574
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Russell Bryant [Tue, 10 Aug 2010 17:47:13 +0000 (17:47 +0000)]
Merged revisions 281566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines
Reset visible indication after answer.
(closes issue #17641)
Reported by: klaus3000
Patches:
ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65)
Tested by: schmidts
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281567
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David Vossel [Mon, 9 Aug 2010 20:46:50 +0000 (20:46 +0000)]
fixes SIP peers memory leak
We zeroed out the peer's addr before it was removed from the
peers_by_ip container. This made it impossible to be removed
from the container as the addr is the key used by the container
to find the peer.
(closes issue #17774)
Reported by: kkm
Patches:
017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281430
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Jeff Peeler [Mon, 9 Aug 2010 20:07:29 +0000 (20:07 +0000)]
Merged revisions 281390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines
Prevent loss of Caller ID information set on local channel after masquerade.
Caller ID set on the channel before a masquerade occurs when using a local
channel would cause the information to be lost. The problem was that the
information was set on a channel destined to be hung up. The somewhat confusing
fix is to detect if any Caller ID has been set on the channel and if so
preswap the Caller ID data so that basically the masquerade puts the data back.
(closes issue #17138)
Reported by: kobaz
Review: https://reviewboard.asterisk.org/r/847/
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David Vossel [Fri, 6 Aug 2010 21:37:29 +0000 (21:37 +0000)]
Blocked revisions 281185 via svnmerge
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r281185 | dvossel | 2010-08-06 16:34:38 -0500 (Fri, 06 Aug 2010) | 20 lines
chan_sip: fixes provisional keepalive scheduled item crash
There is a scheduler item in chan_sip that keeps sending the
last provisional message in response to an INVITE Request for
a period of time until a final response to that INVITE is
sent. Because of the way this scheduler item works, it requires
a reference to a sip_pvt pointer to work properly. The problem
with this is that it is currently possible (but rare) for the
sip_pvt to get destroyed and that scheduler item to still
exist. When this occurs, the scheduler event fires and attempts
to access a freed sip_pvt which causes a crash.
(closes issue #17497)
Reported by: anonymouz666
Patches:
keepalive_diff_1.4_v2.diff uploaded by dvossel (license 671)
Review: https://reviewboard.asterisk.org/r/849/
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Russell Bryant [Thu, 5 Aug 2010 13:11:32 +0000 (13:11 +0000)]
Cleanup default option value handling for cdr.conf [general].
The default values would differ depending on whether or not cdr.conf exists.
That is no longer the case.
Apply a default value to the unanswered option.
Define all default values as named constants.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281051
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Tilghman Lesher [Thu, 5 Aug 2010 07:40:47 +0000 (07:40 +0000)]
Merged revisions 280982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines
Change context lock back to a mutex, because functionality depends upon the lock being recursive.
(closes issue #17643)
Reported by: zerohalo
Patches:
20100726__issue17643.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo
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Tilghman Lesher [Tue, 3 Aug 2010 20:52:20 +0000 (20:52 +0000)]
Merged revisions 280811 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) | 9 lines
Prevent DAHDI channels from overriding the callerid, once it's been set by the user.
(closes issue #16661)
Reported by: jstapleton
Patches:
20100414__issue16661.diff.txt uploaded by tilghman (license 14)
20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: jstapleton
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Tilghman Lesher [Tue, 3 Aug 2010 18:39:28 +0000 (18:39 +0000)]
Document -B and -W flags and regenerate manpage from sgml
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280739
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Tilghman Lesher [Mon, 2 Aug 2010 21:26:11 +0000 (21:26 +0000)]
Allow the pipe, but also allow the comma
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280671
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Jeff Peeler [Mon, 2 Aug 2010 21:14:20 +0000 (21:14 +0000)]
Change SIP NOTIFY requests to expect a response so authentication will work.
This changes the request to be sent with the transmit type XMIT_RELIABLE so that
sip_ack doesn't return false and cause the 401 to be ignored in cases where
authentication is required.
(closes issue #14255)
Reported by: zktech
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280669
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Tilghman Lesher [Thu, 29 Jul 2010 21:07:03 +0000 (21:07 +0000)]
Off-by-one error
(closes issue #17590)
Reported by: atis
Patches:
20100729__issue17590.diff.txt uploaded by tilghman (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280556
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David Vossel [Thu, 29 Jul 2010 20:42:29 +0000 (20:42 +0000)]
fixes wrong SRV query for TLS connection
(closes issue #17612)
Reported by: marcelloceschia
Patches:
chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079)
chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079)
Tested by: marcelloceschia, st, pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280551
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David Vossel [Thu, 29 Jul 2010 19:05:25 +0000 (19:05 +0000)]
Merged revisions 280448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines
fixes issue with translator frame not getting freed
A translator frame even if it local storage so the translation path
can be freed. This issue prevented g729 licenses from being freed up.
(closes issue #17630)
Reported by: manvirr
Patches:
encoder_fix.diff uploaded by dvossel (license 671)
Tested by: manvirr, dvossel
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Jean Galarneau [Thu, 29 Jul 2010 16:01:35 +0000 (16:01 +0000)]
Merged revisions 280341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines
Fix a dsp structure leak occuring when a local channel is put into a meetme
conference, then masquaraded away.
ABE-2422
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Matthew Nicholson [Thu, 29 Jul 2010 13:45:11 +0000 (13:45 +0000)]
Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported.
ABE-2229
Review: https://reviewboard.asterisk.org/r/813/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280306
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Jason Parker [Wed, 28 Jul 2010 20:02:27 +0000 (20:02 +0000)]
Work around some silly behavior on BSD.
A non-zero exit from a subshell should make the build fail.
(closes issue #17621)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280231
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Richard Mudgett [Wed, 28 Jul 2010 19:57:49 +0000 (19:57 +0000)]
Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280229
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Jason Parker [Wed, 28 Jul 2010 19:54:54 +0000 (19:54 +0000)]
Add sha1sum-sh in case there is no util on the system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280227
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Jason Parker [Wed, 28 Jul 2010 18:05:54 +0000 (18:05 +0000)]
Remove unnecessary subshells. Attempt to make checksumming work.
Also improves readability.
(issue #17621)
Reported by: bjm
Review: https://reviewboard.asterisk.org/r/808/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280193
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Sean Bright [Wed, 28 Jul 2010 16:51:11 +0000 (16:51 +0000)]
Plug a reference leak in app_queue when adding members dynamically.
(closes issue #17738)
Reported by: bobwienholt
Patches:
issue17738.patch uploaded by bobwienholt (license 950)
Tested by: bobwienholt, seanbright
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280160
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Leif Madsen [Wed, 28 Jul 2010 13:51:16 +0000 (13:51 +0000)]
Merged revisions 280088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 Jul 2010) | 1 line
Update help text to be less confusing.
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David Vossel [Tue, 27 Jul 2010 20:54:32 +0000 (20:54 +0000)]
Merged revisions 279945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
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Jason Parker [Tue, 27 Jul 2010 17:54:54 +0000 (17:54 +0000)]
Add SHA1SUM to configure, since we require it for sounds/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279883
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Jason Parker [Tue, 27 Jul 2010 16:48:41 +0000 (16:48 +0000)]
Remove aptly-named EMPTY and BS vars, since they aren't used anymore.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279852
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Jason Parker [Tue, 27 Jul 2010 16:39:16 +0000 (16:39 +0000)]
Simply sounds/Makefile some more.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279849
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Mark Michelson [Tue, 27 Jul 2010 15:13:24 +0000 (15:13 +0000)]
Fix bad behavior of dynamic_exclude_static option in sip.conf.
We were attempting to create a contactdeny rule based on the peer's
IP address before the peer's IP address had been set. By moving the
processing further down in the function, we can ensure stuff works
as we expect for it to.
(closes issue #17717)
Reported by: mmichelson
Patches:
17717.patch uploaded by mmichelson (license 60)
Tested by: DennisD
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279784
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Jason Parker [Mon, 26 Jul 2010 22:59:52 +0000 (22:59 +0000)]
Really fix sounds Makefile (and make it readableish).
There was a rather large syntax error that should have caused ALL versions of GNU make to fail.
I don't know how it worked.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279657
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Tilghman Lesher [Mon, 26 Jul 2010 21:18:17 +0000 (21:18 +0000)]
Dunno why this worked on my machine, but it works better this way.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279609
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Gavin Henry [Mon, 26 Jul 2010 20:25:54 +0000 (20:25 +0000)]
Apply all patches in:
https://issues.asterisk.org/view.php?id=13573
(closes issue #13573)
Reported by: navkumar
Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961)
Tested by: suretec
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279597
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Tilghman Lesher [Mon, 26 Jul 2010 19:15:59 +0000 (19:15 +0000)]
Use a special Makefile for noobs who still have GNU Make 3.80.
(Closes issue #17716)
Reported by: farisraouf
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279561
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Sean Bright [Mon, 26 Jul 2010 15:41:13 +0000 (15:41 +0000)]
Expand the correct value within AST_OPTION_ONLY.
(closes issue #17703)
Reported by: stuarth
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279501
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Bradley Latus [Sat, 24 Jul 2010 23:58:19 +0000 (23:58 +0000)]
Minor update to man page
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279347
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Jeff Peeler [Sat, 24 Jul 2010 23:29:44 +0000 (23:29 +0000)]
Blocked revisions 279344 via svnmerge
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r279344 | jpeeler | 2010-07-24 18:27:22 -0500 (Sat, 24 Jul 2010) | 4 lines
Provide a default value for DAHDI_TRANSCODE so when DAHDI is not installed
menuselect doesn't get confused:
Unknown value '' found in build_tools/menuselect-deps for DAHDI_TRANSCODE
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Richard Mudgett [Fri, 23 Jul 2010 22:11:23 +0000 (22:11 +0000)]
Merged revisions 279206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines
SIP promiscuous redirect could fail to dial the redirect.
The ast_channel was created with one variable to ast_request() but the
call to ast_call() that initiates the outgoing call was using a different
variable. The two variables are not equivalent if the call_forward string
included a channel technology specifier. e.g., SIP/200
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Mark Michelson [Fri, 23 Jul 2010 18:29:15 +0000 (18:29 +0000)]
Backport sip_uri_params_cmp() fix from trunk to 1.6.2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279112
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Russell Bryant [Fri, 23 Jul 2010 18:22:34 +0000 (18:22 +0000)]
remove old properties
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279088
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Russell Bryant [Fri, 23 Jul 2010 18:20:02 +0000 (18:20 +0000)]
Add branch-1.4-merged and branch-1.4-blocked properties to 1.6.2 branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279072
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Tilghman Lesher [Fri, 23 Jul 2010 17:06:17 +0000 (17:06 +0000)]
Merged revisions 278985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010) | 12 lines
Merged revisions 278984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010) | 5 lines
Establish a maximum version for openh323 (i.e. not opal), because chan_h323 will fail to load, even if it links.
(issue #17679)
Reported by: am
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Tilghman Lesher [Fri, 23 Jul 2010 16:44:53 +0000 (16:44 +0000)]
Merged revisions 278982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010) | 15 lines
Merged revisions 278981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) | 8 lines
Avoid race with consolethread on shutdown (on parallel processors).
(closes issue #17080)
Reported by: sybasesql
Patches:
20100721__issue17080.diff.txt uploaded by tilghman (license 14)
Tested by: sybasesql
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Tzafrir Cohen [Fri, 23 Jul 2010 15:23:09 +0000 (15:23 +0000)]
Two more typos to cancell.
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Jeff Peeler [Thu, 22 Jul 2010 19:52:17 +0000 (19:52 +0000)]
Merged revisions 278708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r278708 | jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
Add method for finding XML doc files for systems that don't support GLOB_BRACE.
In particular, Solaris and perhaps others do not support the above mentioned
GNU extension. In this case the paths are simply expanded without the braces
and the calls to glob are made separately.
Note: I could not explain memory allocation failures that were being reported
from within libxml itself when making calls to glob without using GLOB_NOCHECK.
This is the only reason why that flag is being used.
(closes issue #15402)
Reported by: snuffy
Patches:
bug_xmlpatt-v3.diff uploaded by snuffy (license 35),
modified by me
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Richard Mudgett [Thu, 22 Jul 2010 19:32:16 +0000 (19:32 +0000)]
DNID does not get cleard on a new call when using immediate=yes with ISDN signaling.
When you are using chan_dahdi ISDN signaling with immediate=yes and a call
comes in without a DNID then you get the DNID of a previous call.
Chan_dahdi does not touch the DNID field on a new call if it does not have
a DNID.
Made always copy the DNID from the new call.
The patches backport the relevant changes from trunk -r210387.
(closes issue #17568)
Reported by: wuwu
Patches:
issue17568_v1.4.patch uploaded by rmudgett (license 664)
issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
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Mark Michelson [Thu, 22 Jul 2010 15:00:11 +0000 (15:00 +0000)]
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r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul 2010) | 19 lines
Merged revisions 278618 via svnmerge from
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r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines
Allow PLC to function properly when channels use SLIN for audio.
If a channel involved in a bridge was using SLIN audio, then translation
paths were not guaranteed to be set up properly since in all likelihood
the number of translation steps was only 1.
This patch enforces the transcode_via_slin behavior if transcode_via_slin
or generic_plc is enabled and one of the formats to make compatible is
SLIN.
AST-352
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Tzafrir Cohen [Wed, 21 Jul 2010 18:22:25 +0000 (18:22 +0000)]
Fix invalid test for rxisoffhook in FXO channels
This fixes some cases of no outgoing calls on FXO before an incoming call.
Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO
(KS/GS) channels.In some cases the bit would not be initialized properly
before the first inbound call and thus prevent an outgoing call.
If those tests are actually required by anybody, they should define
DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c .
(closes issue #14577)
Reported by: jkroon
Patches:
asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd (license 610)
Tested by: frawd
Review: https://reviewboard.asterisk.org/r/699/
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Russell Bryant [Wed, 21 Jul 2010 16:20:18 +0000 (16:20 +0000)]
Merged revisions 278465 via svnmerge from
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r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines
Use poll() instead of select() in res_timing_pthread to avoid stack corruption.
This code did not properly check FD_SETSIZE to ensure that it did not try to
select() on fds that were too large. Switching to poll() removes the limitation
on the maximum fd value.
(closes issue #15915)
Reported by: keiron
(closes issue #17187)
Reported by: Eddie Edwards
(closes issue #16494)
Reported by: Hubguru
(closes issue #15731)
Reported by: flop
(closes issue #12917)
Reported by: falves11
(closes issue #14920)
Reported by: vrban
(closes issue #17199)
Reported by: aleksey2000
(closes issue #15406)
Reported by: kowalma
(closes issue #17438)
Reported by: dcabot
(closes issue #17325)
Reported by: glwgoes
(closes issue #17118)
Reported by: erikje
possibly other issues, too ...
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Tilghman Lesher [Wed, 21 Jul 2010 15:58:20 +0000 (15:58 +0000)]
Merged revisions 278463 via svnmerge from
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r278463 | tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11 lines
Ensure realtime conferences are treated the same as static conferences when trying to find an empty one.
Also, parse the useropts properly, when retrieving from realtime, and add them
to the existing flags.
(closes issue #17502)
Reported by: kenji
Patches:
20100720__issue17502.diff.txt uploaded by tilghman (license 14)
Tested by: kenji
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Tilghman Lesher [Tue, 20 Jul 2010 22:43:51 +0000 (22:43 +0000)]
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r278275 | tilghman | 2010-07-20 17:40:19 -0500 (Tue, 20 Jul 2010) | 14 lines
Merged revisions 278261 via svnmerge from
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r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines
Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message.
(closes issue #16350)
Reported by: noahisaac
Patches:
20100623__issue16350.diff.txt uploaded by tilghman (license 14)
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Tilghman Lesher [Tue, 20 Jul 2010 22:30:46 +0000 (22:30 +0000)]
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r278272 | tilghman | 2010-07-20 17:26:23 -0500 (Tue, 20 Jul 2010) | 11 lines
Merged revisions 278167 via svnmerge from
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r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines
Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110)
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Tilghman Lesher [Tue, 20 Jul 2010 16:54:20 +0000 (16:54 +0000)]
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r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010) | 14 lines
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r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines
Off-by-one error
(closes issue #16506)
Reported by: nik600
Patches:
20100629__issue16506.diff.txt uploaded by tilghman (license 14)
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Jean Galarneau [Mon, 19 Jul 2010 21:21:20 +0000 (21:21 +0000)]
Merged revisions 277945 via svnmerge from
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r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) | 15 lines
Merged revisions 277906 via svnmerge from
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r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | 7 lines
Avoid trying to pickup a parked extension before the park operation is completed.
A crash could occur if the extension is picked up while the parking extension is
being announced. Testing pu->notquiteyet while searching for a parked extension
resolves this crash.
(ABE-2418)
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Tilghman Lesher [Sat, 17 Jul 2010 17:52:28 +0000 (17:52 +0000)]
Merge issues...
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Tilghman Lesher [Sat, 17 Jul 2010 17:43:50 +0000 (17:43 +0000)]
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r277775 | tilghman | 2010-07-17 12:42:32 -0500 (Sat, 17 Jul 2010) | 12 lines
Merged revisions 277738 via svnmerge from
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r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) | 5 lines
Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not.
(closes issue #17616)
Reported by: pprindeville
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Tilghman Lesher [Sat, 17 Jul 2010 17:41:01 +0000 (17:41 +0000)]
Merged revisions 277773 via svnmerge from
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r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010) | 15 lines
Merged revisions 277568 via svnmerge from
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r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines
Since we split values at the semicolon, we should store values with a semicolon as an encoded value.
(closes issue #17369)
Reported by: gkservice
Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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Tim Ringenbach [Fri, 16 Jul 2010 23:37:28 +0000 (23:37 +0000)]
Merged revisions 277657 via svnmerge from
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r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul 2010) | 16 lines
Merged revisions 277625 via svnmerge from
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r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines
Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer.
ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended
transfer, ast_bridge_call() is called for a second bridge on the same channel,
and it clears that flag, which still needs to get set for when the original
ast_bridge_call() gets control back and checks it.
Review: https://reviewboard.asterisk.org/r/741
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Matthew Nicholson [Fri, 16 Jul 2010 21:31:49 +0000 (21:31 +0000)]
Merged revisions 277530 via svnmerge from
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r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul 2010) | 11 lines
Merged revisions 277497 via svnmerge from
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r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines
Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
FAX-128
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Jeff Peeler [Fri, 16 Jul 2010 21:16:51 +0000 (21:16 +0000)]
Merged revisions 277488 via svnmerge from
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r277488 | jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines
Fix reporting estimated queue hold time.
Just say the number of seconds (after minutes) rather than doing some incorrect
calculation with respect to minutes.
(closes issue #17498)
Reported by: corruptor
Patches:
holdesecs_bug.diff uploaded by corruptor (license 253)
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Richard Mudgett [Fri, 16 Jul 2010 20:35:34 +0000 (20:35 +0000)]
Merged revisions 277467 via svnmerge from
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r277467 | rmudgett | 2010-07-16 15:27:51 -0500 (Fri, 16 Jul 2010) | 22 lines
Merged revisions 277419 via svnmerge from
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r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines
priexclusive in chan_dahdi.conf ignored when reloading dahdi module
During a reload, the priexclusive and outsignalling parameters are not
read in from the config file as intended. Unfortunately, they get set to
defaults as a result. This patch makes sure that they do not get set to
defaults during a reload.
(closes issue #17441)
Reported by: mtryfoss
Patches:
issue17441_v1.4.patch uploaded by rmudgett (license 664)
issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
issue17441_trunk.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
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Tilghman Lesher [Fri, 16 Jul 2010 20:30:40 +0000 (20:30 +0000)]
Merged revisions 277452 via svnmerge from
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r277452 | tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines
Add documentation for MOH realtime fields
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Jeff Peeler [Fri, 16 Jul 2010 19:24:38 +0000 (19:24 +0000)]
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r277366 | jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines
Add missing handling for ringing state for use with queue empty options.
(closes issue #17471)
Reported by: jazzy
Patches:
app_queue.c.diff uploaded by jazzy (license 1056)
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Matthew Nicholson [Fri, 16 Jul 2010 18:33:05 +0000 (18:33 +0000)]
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r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul 2010) | 15 lines
Merged revisions 277327 via svnmerge from
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r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines
Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE.
(closes issue #16035)
Reported by: francesco_r
Patches:
pbx.c.patch uploaded by viniciusfontes (license 978)
Tested by: francesco_r, agx, lawbar
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Tilghman Lesher [Fri, 16 Jul 2010 18:14:56 +0000 (18:14 +0000)]
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r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010) | 12 lines
Merged revisions 277261 via svnmerge from
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r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines
If variable gotten is not set, will segfault on Solaris.
(closes issue #17636)
Reported by: bklang
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