[asterisk-commits] phsultan: branch phsultan/gmail-voice-video r241459 - in /team/phsultan/gmail...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 20 04:02:53 CST 2010
Author: phsultan
Date: Wed Jan 20 04:02:24 2010
New Revision: 241459
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=241459
Log:
Merged revisions 220457,220494-220496,220543,220586,220629,220672,220718,220721,220792,220833,220904,220906,220920,220928,220995,221044,221085,221090,221201,221266,221278,221300,221368,221432,221436,221484,221554,221589,221592,221627,221697,221701,221705,221709,221777,221781,221844,221880-221881,221920,221971,222030,222108,222110,222176,222237,222273,222298,222309,222351,222394,222398-222399,222463,222543,222548,222614-222615,222652,222692,222761,222799,222873,222879-222880,222947,222981,223015-223016,223053,223088,223132,223136,223144,223206,223211,223215,223273,223330,223370,223413-223415,223449,223487,223553,223617,223652,223693,223756,223832,223874-223875,223911-223912,223992,224035,224074,224109,224144,224178,224225,224261,224331,224335,224403,224446,224448,224491,224527,224562,224567,224637,224671,224738,224774,224856,224930,224932,225003,225033-225034,225048,225089,225102,225104,225170,225172,225244-225245,225307,225357,225360,225405-225406,225440,225445-225446,225483,225485,225515,225582,225650,225690,225692,225727,225767,225803,225836,225872,225912,225955-225956,226018,226060,226099,226159,226184,226227,226270,226305,226378,226384,226453,226490,226532,226606,226648,226687,226689,226748,226812,226882,226890,226970,226973-226974,227049,227091,227162,227167,227237-227238,227276-227277,227298,227361,227368,227372,227424,227435,227448,227462-227464,227509,227545,227579-227580,227614-227615,227643,227645-227646,227712,227739,227759,227824,227829,227897-227898,227914,227945,228015,228049,228080,228145,228189,228191,228196,228233,228268,228273,228339,228410,228420,228441,228499,228548,228616,228620-228621,228658-228659,228661,228691,228693,228766,228798,228858,228897,228947,228979,229015,229050,229093,229102,229168,229228,229282,229351,229356,229361,229431,229460,229499,229568,229606-229607,229639,229670,229750,229753-229754,229788,229819,229840,229871,229912,229966,229970,230039,230111,230145,230217,230247,230314,230343,230381,230438,230509,230583-230584,230628,230697,230726,230773,230876-230877,230881,230964,230994,231025,231058,231095,231134,231189,231234,231236,231299,231369,231401,231436,231438-231439,231491,231556,231602,231616,231637,231688,231692,231741,231814,231850,231867,231927,232008,232012,232017,232091,232164,232166,232230,232269,232345,232351,232356,232365,232442,232445,232510,232544,232576,232580,232582,232587,232657,232660-232661,232700,232738,232771,232853-232854,232916,232950,232982,233046,233050,233059,233089,233093,233100,233121,233196-233198,233234-233235,233239,233280,233358,233393-233394,233468,233472,233545,233577,233610-233611,233619,233692,233718,233732,233783,233880,233967,234008,234028,234051,234053,234055,234129,234173,234210,234256,234380,234458,234526,234572,234631,234700,234776,234820,234855,234893,234897,234940,234976,235010,235053,235132,235226,235229,235265,235298,235342,235382,235422,235521,235573,235656,235660,235740,235774,235822,235904,235941,236027-236028,236063,236144,236183,236185-236186,236300,236304,236306,236308,236312,236358,236434,236510,236613,236667,236713,236756,236802,236847,236891,236893,236902,236982,237050,237098,237136,237213,237250,237284,237319,237323,237327,237406,237410,237414,237494,237574,237656,237699,237749,237802-237804,237839,237882,237920,237968,238010,238014,238091,238134,238181,238231,238313,238361,238405,238410,238412,238492,238527,238583,238630,238635,238716,238754,238795,238835,238916,239000,239037,239074,239111,239113-239114,239152,239191,239231,239245,239270,239308,239389,239427,239473,239520,239525,239571,239624-239625,239663,239665,239712,239719,239797,239834,239839,239920,239957,239996-239997,240039,240078,240129,240175,240179,240226,240271,240328-240329,240368,240411,240415,240419-240421,240499-240500,240505,240548,240552,240629,240667,240717,240769,240842,240887,240892,240969,240971,240973-240974,241012,241016,241097-241098,241143,241187-241188,241229-241230,241314-241315,241364,241366,241416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r220457 | phsultan | 2009-09-25 12:54:42 +0200 (Fri, 25 Sep 2009) | 13 lines
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/
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r220494 | kpfleming | 2009-09-25 16:38:41 +0200 (Fri, 25 Sep 2009) | 9 lines
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.
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r220495 | kpfleming | 2009-09-25 16:44:40 +0200 (Fri, 25 Sep 2009) | 3 lines
Correct sense of logic test committed in revision 220494.
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r220496 | kpfleming | 2009-09-25 16:50:29 +0200 (Fri, 25 Sep 2009) | 6 lines
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.
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r220543 | rmudgett | 2009-09-25 21:56:18 +0200 (Fri, 25 Sep 2009) | 1 line
Reduce indentation in sig_pri_available().
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r220586 | tilghman | 2009-09-26 17:10:28 +0200 (Sat, 26 Sep 2009) | 2 lines
Allow AES to compile, when OpenSSL is not present.
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r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009) | 3 lines
add name argument for the CALLERID dialplan function to the xml documentation.
Pointed out to me on IRC by snuff-home. Thanks
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r220672 | rmudgett | 2009-09-28 17:27:46 +0200 (Mon, 28 Sep 2009) | 6 lines
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.
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r220718 | jpeeler | 2009-09-28 21:10:10 +0200 (Mon, 28 Sep 2009) | 10 lines
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
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r220721 | seanbright | 2009-09-28 21:11:20 +0200 (Mon, 28 Sep 2009) | 10 lines
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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r220792 | rmudgett | 2009-09-28 23:02:20 +0200 (Mon, 28 Sep 2009) | 1 line
Miscellaneous minor changes.
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r220833 | jpeeler | 2009-09-29 18:58:29 +0200 (Tue, 29 Sep 2009) | 12 lines
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
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r220904 | mnicholson | 2009-09-29 21:49:02 +0200 (Tue, 29 Sep 2009) | 5 lines
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
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r220906 | tilghman | 2009-09-29 21:57:37 +0200 (Tue, 29 Sep 2009) | 16 lines
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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r220920 | mmichelson | 2009-09-29 22:20:48 +0200 (Tue, 29 Sep 2009) | 3 lines
Get rid of annoying and cryptic debug messages.
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r220928 | mnicholson | 2009-09-29 22:22:43 +0200 (Tue, 29 Sep 2009) | 15 lines
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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r220995 | mmichelson | 2009-09-29 23:28:04 +0200 (Tue, 29 Sep 2009) | 11 lines
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
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r221044 | tilghman | 2009-09-30 06:32:36 +0200 (Wed, 30 Sep 2009) | 8 lines
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
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r221085 | seanbright | 2009-09-30 16:47:58 +0200 (Wed, 30 Sep 2009) | 9 lines
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
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r221090 | seanbright | 2009-09-30 17:11:21 +0200 (Wed, 30 Sep 2009) | 8 lines
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
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r221201 | tilghman | 2009-09-30 18:56:42 +0200 (Wed, 30 Sep 2009) | 14 lines
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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r221266 | twilson | 2009-09-30 19:52:30 +0200 (Wed, 30 Sep 2009) | 32 lines
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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r221278 | twilson | 2009-09-30 20:21:03 +0200 (Wed, 30 Sep 2009) | 4 lines
Use rtp properties instead of adding a callback
Thanks, Josh.
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r221300 | twilson | 2009-09-30 20:47:53 +0200 (Wed, 30 Sep 2009) | 2 lines
Remove spurious debug
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r221368 | mnick | 2009-09-30 21:42:36 +0200 (Wed, 30 Sep 2009) | 23 lines
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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r221432 | mnicholson | 2009-09-30 22:40:20 +0200 (Wed, 30 Sep 2009) | 17 lines
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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r221436 | mnick | 2009-09-30 23:15:01 +0200 (Wed, 30 Sep 2009) | 2 lines
Prevents from division by zero
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r221484 | mnicholson | 2009-10-01 01:04:03 +0200 (Thu, 01 Oct 2009) | 2 lines
Cleaned up merge from r221432
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r221554 | oej | 2009-10-01 09:00:04 +0200 (Thu, 01 Oct 2009) | 3 lines
Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE.
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r221589 | mnicholson | 2009-10-01 17:26:20 +0200 (Thu, 01 Oct 2009) | 9 lines
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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r221592 | kpfleming | 2009-10-01 18:16:09 +0200 (Thu, 01 Oct 2009) | 12 lines
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.
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r221627 | kpfleming | 2009-10-01 18:27:05 +0200 (Thu, 01 Oct 2009) | 1 line
Sync up UPGRADE.txt with the 1.6.2 version.
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r221697 | dvossel | 2009-10-01 21:33:33 +0200 (Thu, 01 Oct 2009) | 10 lines
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/
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r221701 | rmudgett | 2009-10-01 21:48:58 +0200 (Thu, 01 Oct 2009) | 10 lines
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>
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r221705 | tilghman | 2009-10-01 22:09:46 +0200 (Thu, 01 Oct 2009) | 2 lines
Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers.
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r221709 | rmudgett | 2009-10-01 22:18:29 +0200 (Thu, 01 Oct 2009) | 1 line
Move DAHDI/ISDN channel naming note from CHANGES to UPGRADE.txt.
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r221777 | tilghman | 2009-10-02 01:59:15 +0200 (Fri, 02 Oct 2009) | 9 lines
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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r221781 | tilghman | 2009-10-02 02:08:21 +0200 (Fri, 02 Oct 2009) | 2 lines
One more off-by-one in trunk
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r221844 | rmudgett | 2009-10-02 03:09:31 +0200 (Fri, 02 Oct 2009) | 33 lines
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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r221880 | rmudgett | 2009-10-02 03:46:51 +0200 (Fri, 02 Oct 2009) | 1 line
Whitespace change.
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r221881 | rmudgett | 2009-10-02 03:49:25 +0200 (Fri, 02 Oct 2009) | 1 line
Whitespace change.
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r221920 | tilghman | 2009-10-02 05:04:34 +0200 (Fri, 02 Oct 2009) | 4 lines
Initialize a variable that we check immediately upon startup.
(closes issue #15973)
Reported by: atis
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r221971 | tilghman | 2009-10-02 18:59:57 +0200 (Fri, 02 Oct 2009) | 9 lines
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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r222030 | dvossel | 2009-10-02 19:34:07 +0200 (Fri, 02 Oct 2009) | 9 lines
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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r222108 | jpeeler | 2009-10-05 21:20:36 +0200 (Mon, 05 Oct 2009) | 12 lines
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
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r222110 | kpfleming | 2009-10-05 21:45:00 +0200 (Mon, 05 Oct 2009) | 25 lines
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
................
r222176 | kpfleming | 2009-10-06 03:24:24 +0200 (Tue, 06 Oct 2009) | 27 lines
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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r222237 | tzafrir | 2009-10-06 18:17:30 +0200 (Tue, 06 Oct 2009) | 12 lines
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).
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r222273 | tilghman | 2009-10-06 21:17:11 +0200 (Tue, 06 Oct 2009) | 5 lines
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
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r222298 | jpeeler | 2009-10-06 21:24:59 +0200 (Tue, 06 Oct 2009) | 9 lines
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
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r222309 | tilghman | 2009-10-06 21:31:39 +0200 (Tue, 06 Oct 2009) | 10 lines
Change schema query to involve the use of an optional schema parameter.
This change is done in such a way as to allow the driver to continue to
function with older databases which don't have these features.
(closes issue #16000)
Reported by: jamicque
Patches:
20091002__issue16000.diff.txt uploaded by tilghman (license 14)
20091002__issue16000__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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r222351 | jpeeler | 2009-10-06 22:35:19 +0200 (Tue, 06 Oct 2009) | 9 lines
Fix 222298 (crash during destruction of second channel when variable set with
setvar).
I mistakenly reasoned that setvar would be used on all channels. Since it can
be set per channel, give each dahdi channel a copy of the variable.
(related to #15899)
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r222394 | jpeeler | 2009-10-07 00:27:44 +0200 (Wed, 07 Oct 2009) | 17 lines
Blocked revisions 222393 via svnmerge
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r222393 | jpeeler | 2009-10-06 17:27:13 -0500 (Tue, 06 Oct 2009) | 11 lines
Fix potential crash when entire span request is received.
The variable index used in this scenario for accessing the dahdi_pvts was
wrong and was most likely copied from the several other places it is used
correctly.
(closes issue #15998)
Reported by: tsearle
Patches:
dahdi_reset_crash.patch uploaded by tsearle (license 373)
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r222398 | dvossel | 2009-10-07 00:39:56 +0200 (Wed, 07 Oct 2009) | 21 lines
contact header port ignored transport when using externip
This patch adds support for TCP/TLS in the Contact header when using
NAT, specifically externip or externhost. The original issue was that
Asterisk sent 5060 as the port in the contact header whether TLS was
used or not. Additionally, this patch adds 2 config options to sip.conf,
specifically externtcpport and externtlsport. This allows a user to
specify different external ports for TCP and TLS other than those used
internally, this is especially useful in in a PAT/port redirection setup.
Thanks to ebroad for reporting the issue and providing the patch!
(closes issue #15880)
Reported by: ebroad
Patches:
portmap.patch uploaded by ebroad (license 878)
externtXXport_v2.patch uploaded by ebroad (license 878)
Tested by: ebroad
Review: https://reviewboard.asterisk.org/r/392/
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r222399 | dvossel | 2009-10-07 00:49:30 +0200 (Wed, 07 Oct 2009) | 3 lines
Updates CHANGES to reflect the new externtcpport and externtlsport sip options
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r222463 | jpeeler | 2009-10-07 01:56:01 +0200 (Wed, 07 Oct 2009) | 14 lines
Merged revisions 222462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) | 8 lines
Add missing unlock(s) in dahdi_read
(two cases in trunk)
(closes issue #15683)
Reported by: alecdavis
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r222543 | dvossel | 2009-10-07 19:44:52 +0200 (Wed, 07 Oct 2009) | 14 lines
Merged revisions 222542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) | 8 lines
crash on transfer
handle_invite_replaces() attempts to uplock a pvt's
owner channel without first verifing that it exists.
(issue #16027)
........
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r222548 | qwell | 2009-10-07 20:04:56 +0200 (Wed, 07 Oct 2009) | 7 lines
Remove 'keepstats' queue option from sample config, as it's no longer used.
https://reviewboard.asterisk.org/r/115/
(closes issue #15820)
Reported by: kshumard
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r222614 | oej | 2009-10-07 20:55:25 +0200 (Wed, 07 Oct 2009) | 2 lines
Use extref for doxygen references to external libraries (in this case PostgreSQL)
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r222615 | oej | 2009-10-07 20:57:29 +0200 (Wed, 07 Oct 2009) | 2 lines
Formatting, moving error messages to ERROR, removing references to unexisting debug output. No functionality changes.
................
r222652 | jpeeler | 2009-10-07 22:08:14 +0200 (Wed, 07 Oct 2009) | 8 lines
Change ringt (ring timeout) styles to be consistent across chan_dahdi.
(closes issue #15684)
Reported by: alecdavis
Patches:
chan_dahdi.bug15684.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
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r222692 | rmudgett | 2009-10-07 23:56:36 +0200 (Wed, 07 Oct 2009) | 21 lines
Merged revisions 222691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines
chan_misdn.c:process_ast_dsp() memory leak
misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not
occur.
The translated frame "f2" when passing through ast_dsp_process() is not
freed whenever it is not used further in process_ast_dsp(). Then in the
end it is never ever freed.
Patches:
translate.patch
JIRA ABE-1993
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r222761 | dvossel | 2009-10-08 00:58:38 +0200 (Thu, 08 Oct 2009) | 35 lines
Deadlock in channel masquerade handling
Channels are stored in an ao2_container. When accessing an item within
an ao2_container the proper locking order is to first lock the container,
and then the items within it.
In ast_do_masquerade both the clone and original channel must be locked
for the entire duration of the function. The problem with this is that
it attemptes to unlink and link these channels back into the ao2_container
when one of the channel's name changes. This is invalid locking order as
the process of unlinking and linking will lock the ao2_container while
the channels are locked!!! Now, both the channels in do_masquerade are
unlinked from the ao2_container and then locked for the entire function.
At the end of the function both channels are unlocked and linked back
into the container with their new names as hash values.
This new method of requiring all channels and tech pvts to be unlocked
before ast_do_masquerade() or ast_change_name() required several
changes throughout the code base.
(closes issue #15911)
Reported by: russell
Patches:
masq_deadlock_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel, atis
(closes issue #15618)
Reported by: lmsteffan
Patches:
deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671)
Tested by: lmsteffan, dvossel
Review: https://reviewboard.asterisk.org/r/387/
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r222799 | rmudgett | 2009-10-08 18:44:33 +0200 (Thu, 08 Oct 2009) | 19 lines
Merged revisions 222797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) | 12 lines
Fix memory leak if chan_misdn config parameter is repeated.
Memory leak when the same config option is set more than once in an
misdn.conf section. Why must this be considered? Templates! Defining a
template with default port options and later adding to or overriding some
of them.
Patches:
memleak-misdn.patch
JIRA ABE-1998
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r222873 | dvossel | 2009-10-08 21:35:30 +0200 (Thu, 08 Oct 2009) | 6 lines
fixes an ast_netsock_list memory leak.
ABE-1998
Review: https://reviewboard.asterisk.org/r/395/
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r222879 | dvossel | 2009-10-08 21:46:15 +0200 (Thu, 08 Oct 2009) | 11 lines
Blocked revisions 222877 via svnmerge
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r222877 | dvossel | 2009-10-08 14:45:15 -0500 (Thu, 08 Oct 2009) | 6 lines
fixes an ast_netsock_list memory leak.
ABE-1998
Review: https://reviewboard.asterisk.org/r/395/
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r222880 | russell | 2009-10-08 21:52:03 +0200 (Thu, 08 Oct 2009) | 51 lines
Merged revisions 222878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines
Make filestream frame handling safer by isolating frames before returning them.
This patch is related to a number of issues on the bug tracker that show
crashes related to freeing frames that came from a filestream. A number of
fixes have been made over time while trying to figure out these problems, but
there re still people seeing the crash. (Note that some of these bug reports
include information about other problems. I am specifically addressing
the filestream frame crash here.)
I'm still not clear on what the exact problem is. However, what is _very_
clear is that we have seen quite a few problems over time related to unexpected
behavior when we try to use embedded frames as an optimization. In some cases,
this optimization doesn't really provide much due to improvements made in other
areas.
In this case, the patch modifies filestream handling such that the embedded frame
will not be returned. ast_frisolate() is used to ensure that we end up with a
completely mallocd frame. In reality, though, we will not actually have to malloc
every time. For filestreams, the frame will almost always be allocated and freed
in the same thread. That means that the thread local frame cache will be used.
So, going this route doesn't hurt.
With this patch in place, some people have reported success in not seeing the
crash anymore.
(SWP-150)
(AST-208)
(ABE-1834)
(issue #15609)
Reported by: aragon
Patches:
filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2)
Tested by: aragon, russell
(closes issue #15817)
Reported by: zerohalo
Tested by: zerohalo
(closes issue #15845)
Reported by: marhbere
Review: https://reviewboard.asterisk.org/r/386/
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r222947 | dvossel | 2009-10-08 22:53:14 +0200 (Thu, 08 Oct 2009) | 6 lines
makes externtcpport and externtlsport static variables
externtcpport and externtlsport need to be declared as static
variables. Thanks to russell for finding and pointing this out.
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r222981 | dvossel | 2009-10-09 00:04:41 +0200 (Fri, 09 Oct 2009) | 13 lines
Deadlock between ast_cel_report_event and ast_do_masquerade
chan_sip calls pbx_exec on a pvt's owner channel while only the
pvt lock is held. Since pbx_exec calls ast_cel_report_event which
attempts to lock the channel, invalid locking order occurs. Channels
should be locked before pvt's.
(closes issue #15512)
Reported by: lmsteffan
Patches:
ast_cel_deadlock_15512.diff uploaded by dvossel (license 671)
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r223015 | dvossel | 2009-10-09 00:57:53 +0200 (Fri, 09 Oct 2009) | 2 lines
fixed comment line for do_magic_pickup
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r223016 | twilson | 2009-10-09 01:11:23 +0200 (Fri, 09 Oct 2009) | 8 lines
Remove global variable that makes dlopen unhappy
This isn't the best way to do this, but it is the easiest. There are some
limitations that are going to need to be addressed at some point with reloads
and when I (or someone else) work on that, then the API can be updated to
handle passing the private config data that the calendar tech modules need in
a better way as well.
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r223053 | twilson | 2009-10-09 17:00:49 +0200 (Fri, 09 Oct 2009) | 2 lines
Don't add Attendees during copy, replace them
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r223088 | dvossel | 2009-10-09 17:49:30 +0200 (Fri, 09 Oct 2009) | 14 lines
p->peerauth is always empty in transmit_register()
When using callbackextension or specifing the peer name
in a registration string, the peer's specific auth settings
set by the "auth=" strings within the peer definition are not
used by the registration. Thanks to ebroad for reporting the
issue and providing the patch.
(closes issue #15955)
Reported by: ebroad
Patches:
regauthfix.patch uploaded by ebroad (license 878)
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r223132 | dvossel | 2009-10-09 18:54:02 +0200 (Fri, 09 Oct 2009) | 9 lines
'auth=' did not parse md5 secret correctly
(closes issue #15949)
Reported by: ebroad
Patches:
authparsefix.patch uploaded by ebroad (license 878)
15949_trunk.diff uploaded by dvossel (license 671)
Tested by: ebroad
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r223136 | mnicholson | 2009-10-09 19:14:38 +0200 (Fri, 09 Oct 2009) | 8 lines
Don't close the sqlite database when reloading. Only close the database when unloading.
(closes issue #15953)
Reported by: frawd
Patches:
sqlite3_rev220097.diff uploaded by frawd (license 610)
Tested by: frawd
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r223144 | dvossel | 2009-10-09 19:19:18 +0200 (Fri, 09 Oct 2009) | 15 lines
Blocked revisions 223142 via svnmerge
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r223142 | dvossel | 2009-10-09 12:18:54 -0500 (Fri, 09 Oct 2009) | 9 lines
'auth=' did not parse md5 secret correctly
(closes issue https://issues.asterisk.org/view.php?id=15949)
Reported by: ebroad
Patches:
authparsefix.patch uploaded by ebroad (license 878)
15949_trunk.diff uploaded by dvossel (license 671)
Tested by: ebroad
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r223206 | dvossel | 2009-10-09 19:53:37 +0200 (Fri, 09 Oct 2009) | 13 lines
Merged revisions 223205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
fixes sip registration using authuser in user.conf
(closes issue #14954)
Reported by: tornblad
Tested by: mmichelson, tornblad, dvossel
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r223211 | mmichelson | 2009-10-09 20:13:57 +0200 (Fri, 09 Oct 2009) | 5 lines
Fix potential memory leaks.
ABE-1998
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r223215 | mmichelson | 2009-10-09 20:17:34 +0200 (Fri, 09 Oct 2009) | 9 lines
Recorded merge of revisions 223213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines
Fix potential memory leak in app_dial.c
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r223273 | mnicholson | 2009-10-09 20:34:08 +0200 (Fri, 09 Oct 2009) | 14 lines
Merged revisions 223225 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines
Signal timeouts by returning AST_CONTROL_RINGING when originating calls.
(closes issue #15104)
Reported by: nblasgen
Patches:
manager-timeout1.diff uploaded by mnicholson (license 96)
Tested by: nblasgen, mnicholson
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r223330 | kpfleming | 2009-10-09 22:58:44 +0200 (Fri, 09 Oct 2009) | 10 lines
Initiate T.38 switchover when acting as called party, regardless of FAX direction.
SendFAX() and ReceiveFAX() can be given options to indicate whether they should
act as the calling or called party; this mode should be used to decide whether
[... 87737 lines stripped ...]
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