Jeff Peeler [Tue, 6 Oct 2009 19:24:59 +0000 (19:24 +0000)]
Fix crash during destruction of second channel when variable set with setvar.
The setvar line in chan_dahdi.conf is shared among all the channels, so make
sure to only free the resources only when the last channel is destroyed.
(closes issue #15899)
Reported by: tzafrir
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222298
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Tilghman Lesher [Tue, 6 Oct 2009 19:17:11 +0000 (19:17 +0000)]
When we call a gosub routine, the variables should be scoped to avoid contaminating the caller.
This affected the ~~EXTEN~~ hack, where a subroutine might have changed the
value before it was used in the caller.
Patch by myself, tested by ebroad on #asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222273
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Tzafrir Cohen [Tue, 6 Oct 2009 16:17:30 +0000 (16:17 +0000)]
Make sure digit events are not reported as "ERROR"
dahdievent_to_analogevent used a simple switch statement to convert DAHDI
event numbers to "ANALOG_*" event numbers. However "digit" events
(DAHDI_EVENT_PULSEDIGIT, DAHDI_EVENT_DTMFDOWN, DAHDI_EVENT_DTMFUP)
are accompannied by the digit in the low word of the event number.
This fix makes dahdievent_to_analogevent() return the event number as-is
for such an event.
This is also required to fix #15924 (in addition to r222108).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222237
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Kevin P. Fleming [Tue, 6 Oct 2009 01:24:24 +0000 (01:24 +0000)]
Recorded merge of revisions 222152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176
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Kevin P. Fleming [Mon, 5 Oct 2009 19:45:00 +0000 (19:45 +0000)]
Allow non-compliant T.38 endpoints to be supportable via configuration option.
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.
In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).
(issue #15586)
Reported by: globalnetinc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110
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Jeff Peeler [Mon, 5 Oct 2009 19:20:36 +0000 (19:20 +0000)]
Add a few missing events to analog_handle_event.
The reported bug was actually only for pulsedigit, dtmfup, and dtmfdown
handling. Also added recognition for fax events (just some verbose output) and
fixed handling for the ec_disabled_event. In order to make comparing the analog
version of events to the DAHDI events easier, the ordering has been changed to
follow that of the DAHDI events.
(closes issue #15924)
Reported by: tzafrir
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222108
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David Vossel [Fri, 2 Oct 2009 17:34:07 +0000 (17:34 +0000)]
Merged revisions 222026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines
Removes unnecessary unlock, clarifies a memcpy.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222030
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Tilghman Lesher [Fri, 2 Oct 2009 16:59:57 +0000 (16:59 +0000)]
Merged revisions 221970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines
Ensure the result of the hash function is positive. Negative array offsets suck.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221971
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Tilghman Lesher [Fri, 2 Oct 2009 03:04:34 +0000 (03:04 +0000)]
Initialize a variable that we check immediately upon startup.
(closes issue #15973)
Reported by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221920
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Richard Mudgett [Fri, 2 Oct 2009 01:49:25 +0000 (01:49 +0000)]
Whitespace change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221881
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Richard Mudgett [Fri, 2 Oct 2009 01:46:51 +0000 (01:46 +0000)]
Whitespace change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221880
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Richard Mudgett [Fri, 2 Oct 2009 01:09:31 +0000 (01:09 +0000)]
Merged revisions 221769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines
Occasionally losing use of B channels in chan_misdn.
I have not been able to reproduce the problem of losing channels.
However, I have seen in the code a reentrancy problem that might give
these symptoms.
The reentrancy patch does several things:
1) Guards B channel and B channel structure allocation.
2) Makes the B channel structure find routines more precise in locating records.
3) Never leave a B channel allocated if we received cause 44.
The last item may cause temporary outgoing call problems, but they should
clear when the line becomes idle.
(closes issue #15490)
Reported by: slutec18
Patches:
issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: rmudgett, slutec18
(closes issue #15458)
Reported by: FabienToune
Patches:
issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: FabienToune, rmudgett, slutec18
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221844
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Tilghman Lesher [Fri, 2 Oct 2009 00:08:21 +0000 (00:08 +0000)]
One more off-by-one in trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221781
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Tilghman Lesher [Thu, 1 Oct 2009 23:59:15 +0000 (23:59 +0000)]
Merged revisions 221776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines
Fix a bunch of off-by-one errors
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221777
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Richard Mudgett [Thu, 1 Oct 2009 20:18:29 +0000 (20:18 +0000)]
Move DAHDI/ISDN channel naming note from CHANGES to UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221709
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Tilghman Lesher [Thu, 1 Oct 2009 20:09:46 +0000 (20:09 +0000)]
Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221705
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Richard Mudgett [Thu, 1 Oct 2009 19:48:58 +0000 (19:48 +0000)]
Prevent deadlock if chan_dahdi attempts to change PRI channel names.
The PRI channels can no longer change the channel name if a different B
channel is selected during call negotiation. To prevent using the channel
name to infer what B channel a call is using and to avoid name collisions,
the channel name format is changed.
The new channel naming for PRI channels is:
DAHDI/ISDN-<span>-<sequence-number>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221701
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David Vossel [Thu, 1 Oct 2009 19:33:33 +0000 (19:33 +0000)]
outbound tls connections were not defaulting to port 5061
(closes issue #15854)
Reported by: dvossel
Patches:
sip_port_config_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/357/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221697
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Kevin P. Fleming [Thu, 1 Oct 2009 16:27:05 +0000 (16:27 +0000)]
Sync up UPGRADE.txt with the 1.6.2 version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221627
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Kevin P. Fleming [Thu, 1 Oct 2009 16:16:09 +0000 (16:16 +0000)]
Remove ability to control T.38 FAX error correction from udptl.conf.
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221592
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Matthew Nicholson [Thu, 1 Oct 2009 15:26:20 +0000 (15:26 +0000)]
Merged revisions 221588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines
Use unsigned ints for portinuri flags.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221589
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Olle Johansson [Thu, 1 Oct 2009 07:00:04 +0000 (07:00 +0000)]
Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221554
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Matthew Nicholson [Wed, 30 Sep 2009 23:04:03 +0000 (23:04 +0000)]
Cleaned up merge from r221432
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221484
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Matthias Nick [Wed, 30 Sep 2009 21:15:01 +0000 (21:15 +0000)]
Prevents from division by zero
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221436
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Matthew Nicholson [Wed, 30 Sep 2009 20:40:20 +0000 (20:40 +0000)]
Merged revisions 221360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines
Fix SRV lookup and Request-URI generation in chan_sip.
This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur.
(closes issue #14418)
Reported by: klaus3000
Tested by: klaus3000, mnicholson
Review: https://reviewboard.asterisk.org/r/369/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221432
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Matthias Nick [Wed, 30 Sep 2009 19:42:36 +0000 (19:42 +0000)]
Merged revisions 221153,221157,221303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines
check bounds - prevents for buffer overflow
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r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines
added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf
(closes issue #15471)
Reported by: dkerr
Patches:
csv_quote_14.txt uploaded by mnick (license )
Tested by: mnick
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r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines
changed the prototype definition of csv_quote
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221368
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Terry Wilson [Wed, 30 Sep 2009 18:47:53 +0000 (18:47 +0000)]
Remove spurious debug
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221300
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Terry Wilson [Wed, 30 Sep 2009 18:21:03 +0000 (18:21 +0000)]
Use rtp properties instead of adding a callback
Thanks, Josh.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278
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Terry Wilson [Wed, 30 Sep 2009 17:52:30 +0000 (17:52 +0000)]
Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266
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Tilghman Lesher [Wed, 30 Sep 2009 16:56:42 +0000 (16:56 +0000)]
Merged revisions 221200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines
Avoid a potential NULL dereference.
(closes issue #15865)
Reported by: kobaz
Patches:
20090915__issue15865.diff.txt uploaded by tilghman (license 14)
Tested by: kobaz
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221201
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Sean Bright [Wed, 30 Sep 2009 15:11:21 +0000 (15:11 +0000)]
Modify VoiceMailMain()'s a() argument to allow mailboxes to be specified by name.
(closes issue #14740)
Reported by: pj
Patches:
issue14740_09022009.diff uploaded by seanbright (license 71)
Tested by: seanbright, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221090
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Sean Bright [Wed, 30 Sep 2009 14:47:58 +0000 (14:47 +0000)]
Clarify documentation for VoiceMailMain()'s a() option.
We require box numbers, not names as the documentation implies.
(issue #14740)
Reported by: pj
Patches:
__20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10)
Tested by: seanbright, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221085
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Tilghman Lesher [Wed, 30 Sep 2009 04:32:36 +0000 (04:32 +0000)]
Allow locks to be inherited through a masquerade without causing starvation.
(closes issue #14859)
Reported by: atis
Patches:
20090821__issue14859.diff.txt uploaded by tilghman (license 14)
20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: atis, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221044
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Mark Michelson [Tue, 29 Sep 2009 21:28:04 +0000 (21:28 +0000)]
Fix channel reference leak.
ast_cel_report_event would geet a reference to the
bridged channel. However, certain return paths, such
as if CEL was not enabled, would result in a reference
leak. All return paths now properly unref the channel.
(closes issue #15991)
Reported by: mmichelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220995
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Matthew Nicholson [Tue, 29 Sep 2009 20:22:43 +0000 (20:22 +0000)]
Blocked revisions 220907 via svnmerge
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r220907 | mnicholson | 2009-09-29 15:14:29 -0500 (Tue, 29 Sep 2009) | 10 lines
Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked.
(closes issue #15965)
Reported by: atis
Patches:
chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96)
Tested by: atis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220928
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Mark Michelson [Tue, 29 Sep 2009 20:20:48 +0000 (20:20 +0000)]
Get rid of annoying and cryptic debug messages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220920
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Tilghman Lesher [Tue, 29 Sep 2009 19:57:37 +0000 (19:57 +0000)]
Merged revisions 220873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines
Reduce CPU usage related to building a peer merely for devicestates.
This fixes a 100% CPU problem in the SIP driver, found by profiling
the driver while the problem was occurring.
(closes issue #14309)
Reported by: pkempgen
Patches:
20090924__issue14309.diff.txt uploaded by tilghman (license 14)
Tested by: pkempgen, vrban
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220906
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Matthew Nicholson [Tue, 29 Sep 2009 19:49:02 +0000 (19:49 +0000)]
Fix options 'm' and 's'. They were swapped in the code. Also document the fact that app_confbridge does not automatically answer the channel.
(closes issue #15964)
Reported by: shrift
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220904
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Jeff Peeler [Tue, 29 Sep 2009 16:58:29 +0000 (16:58 +0000)]
Make deletion of temporary greetings work properly with IMAP_STORAGE
When imapgreetings was set to yes, the message was being deleted but wasn't
actually being expunged. When imapgreetings was set to no, the file based
message was not being deleted at all. All good now!
(closes issue #14949)
Reported by: noahisaac
Patches:
vm_tempgreeting_removal.patch uploaded by noahisaac (license 748),
modified by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220833
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Richard Mudgett [Mon, 28 Sep 2009 21:02:20 +0000 (21:02 +0000)]
Miscellaneous minor changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220792
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Sean Bright [Mon, 28 Sep 2009 19:11:20 +0000 (19:11 +0000)]
Merged revisions 220717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines
When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler
so we override any default optimization levels for a particular install.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220721
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Jeff Peeler [Mon, 28 Sep 2009 19:10:10 +0000 (19:10 +0000)]
Fix building of registration entry in build_peer when using callbackextension
Check for remotesecret option was unintentionally always true, which therefore
caused the secret option to never be used. Thanks to dvossel for pointing out
the exact fix.
(closes issue #15943)
Reported by: tpsast
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220718
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Richard Mudgett [Mon, 28 Sep 2009 15:27:46 +0000 (15:27 +0000)]
Locking issues dealing with service_lock.
* Removed unneeded and uninitialized service_lock.
* Fixed potential locking imbalance in pri_dchannel():PRI_EVENT_RESTART.
* Fixed verbose message typo in pri_dchannel():PRI_EVENT_RESTART.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220672
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Michiel van Baak [Sun, 27 Sep 2009 20:40:16 +0000 (20:40 +0000)]
add name argument for the CALLERID dialplan function to the xml documentation.
Pointed out to me on IRC by snuff-home. Thanks
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220629
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Tilghman Lesher [Sat, 26 Sep 2009 15:10:28 +0000 (15:10 +0000)]
Allow AES to compile, when OpenSSL is not present.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220586
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Richard Mudgett [Fri, 25 Sep 2009 19:56:18 +0000 (19:56 +0000)]
Reduce indentation in sig_pri_available().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220543
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Kevin P. Fleming [Fri, 25 Sep 2009 14:50:29 +0000 (14:50 +0000)]
Eliminate unnecessary include of version.h in manager.c.
Including version.h here causes this file to get recompiled after
every commit or update, which is not needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220496
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Kevin P. Fleming [Fri, 25 Sep 2009 14:44:40 +0000 (14:44 +0000)]
Correct sense of logic test committed in revision 220494.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220495
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Kevin P. Fleming [Fri, 25 Sep 2009 14:38:41 +0000 (14:38 +0000)]
Don't use hash-based lookups for ast_channel_get_by_name_prefix().
ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based
channel lookups, but this will not work properly when the channel's full
name was not supplied; for name-prefix searches, there is no value in
doing a hash-based lookup, and in fact doing so could result in many
channels being skipped.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220494
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Philippe Sultan [Fri, 25 Sep 2009 10:54:42 +0000 (10:54 +0000)]
Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channels
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457
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Tilghman Lesher [Thu, 24 Sep 2009 22:53:23 +0000 (22:53 +0000)]
Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417
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David Vossel [Thu, 24 Sep 2009 20:37:20 +0000 (20:37 +0000)]
fixes tcptls_session memory leak caused by ref count error
(closes issue #15939)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/375/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220365
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Jeff Peeler [Thu, 24 Sep 2009 20:29:51 +0000 (20:29 +0000)]
Add bridge related dial flags to the bridge app
Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.
(closes issue #13165)
Reported by: tim_ringenbach
Patches:
app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
modified by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344
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Olle Johansson [Thu, 24 Sep 2009 19:57:23 +0000 (19:57 +0000)]
Documentation in the commit messages is soon forgotten, please add it to the docs in the product.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220295
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Tilghman Lesher [Thu, 24 Sep 2009 19:41:02 +0000 (19:41 +0000)]
Merged revisions 220288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines
Implicitly sending a progress signal breaks some applications.
Call Progress() in your dialplan if you explicitly want progress to be sent.
(Reverts change 216430, closes issue #15957)
Reported by: Pavel Troller on the Asterisk-Dev mailing list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220289
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Sean Bright [Thu, 24 Sep 2009 18:19:41 +0000 (18:19 +0000)]
Merged revisions 220213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep 2009) | 1 line
Resolve parallel build warnings.
Reported by Klaus Darilion on the asterisk-dev mailing list.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220217
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Matthew Nicholson [Thu, 24 Sep 2009 16:33:20 +0000 (16:33 +0000)]
Ensure the numeric portion of the P-Asserted-Identity header is properly escaped.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220174
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Sean Bright [Thu, 24 Sep 2009 14:44:08 +0000 (14:44 +0000)]
Merged revisions 220099 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep 2009) | 2 lines
Remove the remaining bashisms in the Makefile/mkpkgconfig
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220100
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Michiel van Baak [Thu, 24 Sep 2009 08:36:18 +0000 (08:36 +0000)]
Merged revisions 220027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) | 7 lines
mkpkgconfig does not need bash so make it use /bin/sh
This fixes building on all systems that don't have bash
at /bin/bash
Reported by _ys on #asterisk-dev
Tested by _ys on #asterisk-dev
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220028
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Tilghman Lesher [Thu, 24 Sep 2009 07:39:44 +0000 (07:39 +0000)]
Fix two possible crashes, one only in 1.6.1 and one in 1.6.1 forward.
(closes issue #15739)
Reported by: DLNoah, jeffg
Patches:
20090914__issue15739.diff.txt uploaded by tilghman (license 14)
20090922__issue15739.diff.txt uploaded by tilghman (license 14)
Tested by: DLNoah, jeffg
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219987
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Tilghman Lesher [Wed, 23 Sep 2009 23:38:19 +0000 (23:38 +0000)]
Add support for 'setvar=' for MGCP device lines, like other channel drivers provide.
(closes issue #14818)
Reported by: alea-soluciones
Patches:
chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219952
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Tilghman Lesher [Wed, 23 Sep 2009 23:19:14 +0000 (23:19 +0000)]
Update fax number to the legal fax, not the generic fax.
(closes issue #15946)
Reported by: jtodd
Patches:
leif-is-a-wuss.txt uploaded by jtodd (license 870)
Tested by: jparker, tilghman, jtodd, russellb, mmichelson, seanbright, kpfleming, and the rest of the usual suspects
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219951
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Leif Madsen [Wed, 23 Sep 2009 17:46:46 +0000 (17:46 +0000)]
Add Mantis work flow documention.
This commit adds the doxygen changes that I've made to describe the Mantis
work flow documentation for the open source issue tracker. This should make
it easier to determine the flow of issues through the issue tracker, and what
those statuses mean.
(closes issue #15902)
Reported by: lmadsen
Patches:
mantisworkflow.h uploaded by lmadsen (license 10)
Review: https://reviewboard.asterisk.org/r/367/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219895
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Tilghman Lesher [Tue, 22 Sep 2009 21:43:22 +0000 (21:43 +0000)]
Merged revisions 219816 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines
When IMAP variables were changed during a reload, Voicemail did not use the new values.
This change introduces a configuration version variable, which ensures that
connections with the old values are not reused but are allowed to expire
normally.
(closes issue #15934)
Reported by: viniciusfontes
Patches:
20090922__issue15934.diff.txt uploaded by tilghman (license 14)
Tested by: viniciusfontes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219818
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David Vossel [Mon, 21 Sep 2009 16:59:05 +0000 (16:59 +0000)]
Merged revisions 219720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines
Reverting merge 219520. This change was not necessary.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219721
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Tilghman Lesher [Sun, 20 Sep 2009 17:55:49 +0000 (17:55 +0000)]
Merged revisions 219653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines
Really stop the stream, when ast_closestream() is called.
(closes issue #15129)
Reported by: bmh
Patches:
20090918__issue15129.diff.txt uploaded by tilghman (license 14)
Review:
https://reviewboard.asterisk.org/r/372/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219654
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Russell Bryant [Sat, 19 Sep 2009 02:59:52 +0000 (02:59 +0000)]
Merged revisions 219586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) | 6 lines
Make sure the iax_pvt exists before dereferencing it.
This fixes the latest crash posted on issue 15609.
(issue #15609)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219587
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David Vossel [Fri, 18 Sep 2009 23:20:58 +0000 (23:20 +0000)]
Merged revisions 219519 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines
iax2 frame double free
The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219520
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David Vossel [Fri, 18 Sep 2009 16:20:41 +0000 (16:20 +0000)]
Merged revisions 219450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines
via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's invite_branch
variable must be updated so CANCEL requests use the correct branch id.
(closes issue #15262)
Reported by: maniax
Patches:
asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608)
invite_new_branch_trunk.diff uploaded by dvossel (license 671)
Tested by: maniax, dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219451
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Tilghman Lesher [Fri, 18 Sep 2009 13:54:51 +0000 (13:54 +0000)]
Missing value setting line for maxsecs/maxmessage
(closes issue #15696)
Reported by: fhackenberger
Patches:
maxsecs.patch uploaded by fhackenberger (license 592)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219412
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David Vossel [Thu, 17 Sep 2009 22:37:28 +0000 (22:37 +0000)]
fixes deadlock when performing directed pickup w Invite/replaces
(closes issue #15340)
Reported by: lmsteffan
Patches:
deadlock.patch uploaded by lmsteffan (license 779)
Tested by: lmsteffan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219371
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Mark Michelson [Thu, 17 Sep 2009 22:22:01 +0000 (22:22 +0000)]
Merged revisions 219320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines
Send a 100 Trying response when we detect a spiral.
This was problematic during spiral tests at SIPit...
along with some other things as well.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219324
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David Vossel [Thu, 17 Sep 2009 21:59:21 +0000 (21:59 +0000)]
Merged revisions 219303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines
INVITE w/Replaces deadlock fix
This patch cleans up the locking logic in chan_sip.c's
handle_invite_replaces() function as well as making use
of ast_do_masquerade() rather than forcing the masquerade
on an ast_read(). The code had several redundant unlocks
that would result in 'freed more times than we've locked!'
errors. I cleaned these up as well as moving all the unlock
logic to the end of the function. This patch should also
resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls.
(closes issue #15151)
Reported by: irroot
Patches:
invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
Tested by: irroot, dvossel
Review: https://reviewboard.asterisk.org/r/371/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219304
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Joshua Colp [Thu, 17 Sep 2009 19:57:39 +0000 (19:57 +0000)]
Ensure no spaces exist before "refresher=" when doing the comparison.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219264
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Sean Bright [Thu, 17 Sep 2009 16:25:38 +0000 (16:25 +0000)]
Get this compiling under dev-mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219230
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Matthew Nicholson [Thu, 17 Sep 2009 15:18:01 +0000 (15:18 +0000)]
Merged revisions 219136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316)
Reported by: vmarrone
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219139
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Tilghman Lesher [Thu, 17 Sep 2009 00:58:10 +0000 (00:58 +0000)]
Add the 'E' option to exit ChanSpy, once the single channel it spied upon hangs up.
In addition, there's a bit of cleanup to the arguments and documentation, in which
I discovered that the last feature added to this application duplicated an option
(oops!) and changed that option so that it now works.
(closes issue #14909)
Reported by: junky
Patches:
__20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10)
Tested by: amilcar, junky, flujan, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219105
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Tilghman Lesher [Wed, 16 Sep 2009 23:42:12 +0000 (23:42 +0000)]
Merged revisions 219023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines
Properly deal with quotes in the arguments of '#exec' includes.
(closes issue #15583)
Reported by: pkempgen
Patches:
20090726__issue15583.diff.txt uploaded by tilghman (license 14)
20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169)
Tested by: pkempgen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219061
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Tilghman Lesher [Wed, 16 Sep 2009 23:15:43 +0000 (23:15 +0000)]
Detect whether we actually have the long double type, before looking for those functions.
(closes issue #15017)
Reported by: tzafrir
Patches:
20090916__issue15017.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219007
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Sean Bright [Wed, 16 Sep 2009 20:32:50 +0000 (20:32 +0000)]
Remove some unused defines from res_jabber.
(closes issue #15359)
Reported by: snuffy
Patches:
bug_res_jabber_unused_defines.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218973
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Mark Michelson [Wed, 16 Sep 2009 19:25:36 +0000 (19:25 +0000)]
Reverse order of args to fread.
This way, we don't always write a null byte into
byte 1 of the buffer
(closes issue #15905)
Reported by: ebroad
Patches:
freadfix.patch uploaded by ebroad (license 878)
Tested by: ebroad
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218933
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Joshua Colp [Wed, 16 Sep 2009 18:31:47 +0000 (18:31 +0000)]
On TCP and TLS connections do not attempt to stop retransmission of the packet internally.
This was preventing responses from being properly processed because the packet was not being found
causing handle_response to return prematurely.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218918
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David Brooks [Wed, 16 Sep 2009 18:06:42 +0000 (18:06 +0000)]
Merged revisions 218867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines
Fixes CID pattern matching behavior to mirror that of extension pattern matching.
Pattern matching for extensions uses a type of scoring system, giving values for
specificity to each character in the pattern. Unfortunately, this is done character
by character, in order. This does lead to some less specific patterns being first
in line for matching, but it will usually get the job done.
This patch merely brings CID matching to the same level as extension matching.
This patch does not attempt to tackle the problem shared by extension matching.
(closes issue #14708)
Reported by: klaus3000
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218868
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Russell Bryant [Wed, 16 Sep 2009 13:34:41 +0000 (13:34 +0000)]
Merged revisions 218798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines
Remove the IAXy firmware from Asterisk.
The firmware can now be found on downloads.digium.com, where the rest of our
binary downloads live. This was the last part of our Asterisk tarballs that
was considered non-free by Debian. :-)
(closes issue #15838)
Reported by: paravoid
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218799
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Tilghman Lesher [Tue, 15 Sep 2009 22:33:10 +0000 (22:33 +0000)]
Merged revisions 218730 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines
If the user enters the same password as before, don't signal an error when the change does nothing.
(closes issue #15492)
Reported by: cbbs70a
Patches:
20090713__issue15492.diff.txt uploaded by tilghman (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218731
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David Vossel [Tue, 15 Sep 2009 19:22:37 +0000 (19:22 +0000)]
upward bound checking for port string to int conversion
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218687
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Jeff Peeler [Tue, 15 Sep 2009 16:30:43 +0000 (16:30 +0000)]
Blocked revisions 218623 via svnmerge
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r218623 | jpeeler | 2009-09-15 11:29:27 -0500 (Tue, 15 Sep 2009) | 3 lines
Fix small memory leak in handle_init_event by always destroying the pthread
attr before returning.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218629
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Matthew Nicholson [Tue, 15 Sep 2009 16:15:02 +0000 (16:15 +0000)]
Merged revisions 218578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines
Send request contact header field with response to registrer queries instead of the address of record.
(closes issue #14438)
Reported by: ravindrad
Patches:
regquerypatch uploaded by ravindrad (license 684)
Tested by: ravindrad
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218586
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Jeff Peeler [Tue, 15 Sep 2009 16:12:49 +0000 (16:12 +0000)]
Add some changes related to 218430.
* Remove thread_spawned in handle_init_event since it was never used
* Always check handle_init_event in case a channel is destroyed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218583
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Tilghman Lesher [Tue, 15 Sep 2009 16:04:41 +0000 (16:04 +0000)]
Merged revisions 218577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines
Ensure FollowMe sets language in channels it creates.
Also, not in the original bug report, but related fields are accountcode and
musicclass, and the inheritance of datastores.
(closes issue #15372)
Reported by: Romik
Patches:
20090828__issue15372.diff.txt uploaded by tilghman (license 14)
Tested by: cervajs
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218579
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Mark Michelson [Tue, 15 Sep 2009 15:40:14 +0000 (15:40 +0000)]
Use a better method of ensuring null-termination of the buffer
while reading the SDP when using TCP.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218566
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Mark Michelson [Tue, 15 Sep 2009 15:05:53 +0000 (15:05 +0000)]
Ensure that SDP read from TCP socket is null-terminated.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218504
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Kevin P. Fleming [Tue, 15 Sep 2009 15:02:21 +0000 (15:02 +0000)]
Merged revisions 218497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep 2009) | 1 line
Use proper hostname for downloading sound files.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218500
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Mark Michelson [Tue, 15 Sep 2009 14:59:50 +0000 (14:59 +0000)]
Fix off-by-one error when reading SDP sent over TCP.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218499
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Tzafrir Cohen [Tue, 15 Sep 2009 10:24:55 +0000 (10:24 +0000)]
Fix false error message on DAHDI_EVENT_REMOVED (RESULT_SUCCESS == 0)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218465
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Jeff Peeler [Mon, 14 Sep 2009 22:38:25 +0000 (22:38 +0000)]
Merged revisions 218401 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines
Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor.
After talking to rmudgett about some of his recent iflist locking changes, it
was determined that the only place that would destroy a channel without being
explicitly to do so was in handle_init_event. The loop to walk the interface
list has been modified to wait to destroy the channel until the dahdi_pvt of
the channel to be destroyed is no longer needed.
(closes issue #15378)
Reported by: samy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218430
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Richard Mudgett [Mon, 14 Sep 2009 20:08:11 +0000 (20:08 +0000)]
Add support for multiple interface lists.
Also unlink the sig_pri_pri.pvts[] pointer in destroy_dahdi_pvt().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218365
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Tilghman Lesher [Mon, 14 Sep 2009 19:29:48 +0000 (19:29 +0000)]
Recorded merge of revisions 218331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines
Don't say "Please try again" if we don't give the user another chance to try again.
(issue #15055, SWP-129)
Reported by: jthurman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218361
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Joshua Colp [Mon, 14 Sep 2009 18:16:39 +0000 (18:16 +0000)]
Do not attempt to add a parking extension if an error occurred while reading the configuration.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218295
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Matthew Nicholson [Mon, 14 Sep 2009 14:57:23 +0000 (14:57 +0000)]
Merged revisions 218223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines
Ensure we don't pickup ourselves when doing pickup by exten.
(closes issue #15100)
Reported by: lmsteffan
Patches:
(modified) pickup.patch uploaded by lmsteffan (license 779)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218224
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