mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
17675 Commits (05030c4a00a2379bdbaeba5fba4f4ae0c9a9b2f5)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Jeff Peeler
|
05030c4a00 |
Blocked revisions 203304 via svnmerge
........ r203304 | jpeeler | 2009-06-25 14:54:12 -0500 (Thu, 25 Jun 2009) | 6 lines New signaling module to handle PRI/BRI operations in chan_dahdi This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into sig_pri.c. Functionality in theory should not change (mostly). A few trivial changes were made in sig_analog with verbose messages and commenting. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jason Parker
|
027b94dce0 |
Merged revisions 203258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) | 10 lines Unmute when we get a dtmfup (we muted on dtmfdown) event. This would occasionally cause one-way audio when using hardware DTMF detection. (closes issue #14761) Reported by: tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) Tested by: tzafrir, dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203274 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
f710b0b128 |
Merged revisions 203231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203231 | mmichelson | 2009-06-25 13:53:12 -0500 (Thu, 25 Jun 2009) | 8 lines Blocked revisions 203230 via svnmerge ........ r203230 | mmichelson | 2009-06-25 13:52:22 -0500 (Thu, 25 Jun 2009) | 3 lines Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203233 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
8ac0deae26 |
Merged revisions 203116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines Merged revisions 203115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Richard Mudgett
|
482ffa8830 |
Merged revisions 203037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203057 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
9d35f9503b |
Merged revisions 202967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun 2009) | 9 lines Merged revisions 202966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding the same thing in-line. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202969 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
10d49a7cc8 |
Merged revisions 202925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 | file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines Ensure the default settings are applied for T.38 when we set it up for a peer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202927 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Fredrickson
|
a6208dc59d |
Merged revisions 202761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) | 1 line I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
|
f2441e1d3d |
Merged revisions 202672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) | 18 lines Merged revisions 202671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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26c525065d |
Blocked revisions 202603 via svnmerge
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r202603 | mmichelson | 2009-06-23 10:23:00 -0500 (Tue, 23 Jun 2009) | 8 lines
Blocked revisions 202601 via svnmerge
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r202601 | mmichelson | 2009-06-23 10:22:35 -0500 (Tue, 23 Jun 2009) | 3 lines
Fix more memory leaks that may result if rtp is not successfully allocated.
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17 years ago |
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Mark Michelson
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8814cab7da |
Blocked revisions 202574 via svnmerge
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r202574 | mmichelson | 2009-06-23 10:11:47 -0500 (Tue, 23 Jun 2009) | 8 lines
Blocked revisions 202572 via svnmerge
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r202572 | mmichelson | 2009-06-23 10:08:27 -0500 (Tue, 23 Jun 2009) | 3 lines
Fix potential memory leak in chan_sip when video rtp is not allocated properly.
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17 years ago |
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Russell Bryant
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f46fa522ad |
Merged revisions 202497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) | 11 lines Merged revisions 202496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Sean Bright
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3511936a18 |
Merged revisions 202417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid potential crashes during reload. Pointed out by Russell while working on the CEL branch. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202472 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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9bce657f84 |
Merged revisions 202415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) | 9 lines Merged revisions 202414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines Make Polycom subscription type override check more explicit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202418 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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4abee612d0 |
Merged revisions 202410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 | dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines attempting to load running modules Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202412 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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1606795a78 |
Merged revisions 202343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines Fix a situation in which Asterisk would not stop retransmitting 487s. If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line left from previous commit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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ee91773ea8 |
Merged revisions 202337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines Merged revisions 202336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines Fix a possible infinite loop in SDP parsing during glare situation. There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202339 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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2df709b3c9 |
Merged revisions 202262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 | russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines Fix possibility of crashiness during reload in custom fields handling. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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6a6dad000d |
Merged revisions 202258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 | russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines Standardize return values of load_config() so reload() doesn't report an error on success. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Sean Bright
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a0e8f439bf |
Merged revisions 202183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 | seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5 lines Fix version detection for API changes in spandsp. (closes issue #15355) Reported by: deuffy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202185 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Nicholson
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e8a03ddfdd |
Added deadlock protection to try_suggested_sip_codec in chan_sip.c.
Review: https://reviewboard.asterisk.org/r/287/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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2e9d5788e8 |
Merged revisions 201994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500 (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines timestamp was being converted to host order as a short rather than a long (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Matthew Nicholson
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19d82d90ae |
Blocked revisions 201717 via svnmerge
........ r201717 | mnicholson | 2009-06-18 12:41:09 -0500 (Thu, 18 Jun 2009) | 4 lines Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/285/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
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0d6dbd3774 |
Merged revisions 201904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009) | 4 lines Fix 2 typos and add support for wide character types. Reported by Benny Amorsen via the asterisk-users mailing list. http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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03b87a486e |
If the "h" extension fails, give it another chance in main/pbx.c.
If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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0801832977 |
Merged revisions 201783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 | tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines One of the changes in 1.6.1 was to allow app_directory to use functionality within app_voicemail for directory functions. It is therefore no longer necessary for app_directory to be linked against the ODBC libraries (and it never was necessary for app_directory to be linked against IMAP, though it was). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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09b65fe917 |
Merged revisions 201678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201680 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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c6c5ca6454 |
Merged revisions 201610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201610 | russell | 2009-06-18 10:27:10 -0500 (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines Fix memory corruption and leakage related reloads of non files mode MoH classes. For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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c2e5311110 |
Merged revisions 201570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 | dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines parsing extension correctly from sip register lines If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'. (closes issue #15111) Reported by: ffs Patches: chan_sip.c_register-parser.patch uploaded by ffs (license 730) Tested by: ffs, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201601 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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ded37caf1a |
Merged revisions 201531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009) | 7 lines Initialize additional variables, to prevent a possible crash. (closes issue #15186) Reported by: ajohnson Patches: 20090528__issue15186.diff.txt uploaded by tilghman (license 14) Tested by: ajohnson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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0a92ebc9bd |
Merged revisions 201462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines Fix problem with no audio due to ignoring the SDP. A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201464 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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311e463f48 |
Merged revisions 201458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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31ddd9da2b |
Merged revisions 201453 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 | dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201456 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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1644daa9aa |
Merged revisions 201445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines StopMixMonitor race condition (not giving up file immediately) StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201448 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Brooks
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c33eb64920 |
Merged revisions 201381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) | 16 lines Merged revisions 201380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201444 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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a3d2d156ee |
Merged revisions 201344 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 | dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines SIP registry ref count error During a sip reload, the list of sip_registry objects are supposed to be traversed, unlinked, and destroyed, but destruction never takes place due to a ref counting error. This causes a memory leak when registry items are removed from sip.conf and reloaded. While the registries are removed from the global list, they are not removed from the scheduler. Because of this, SIP register attempts continue to be sent out for the item even though it may no longer be in the .conf. (closes issue #15295) Reported by: amorsen Review: https://reviewboard.asterisk.org/r/282/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
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1502bfdb70 |
Merged revisions 201262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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f5fca5c8e1 |
Merged revisions 201223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201225 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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bbfd73e692 |
Merged revisions 201056 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201096 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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21da1c5fde |
Merged revisions 201090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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6b120bb149 |
Merged revisions 200985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 | kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7 lines Fix problems with new compiler attribute checking in configure script. The last changes to ast_gcc_attribute.m4 caused some problems checking for various attributes, because the scope of the symbol the attribute is applied to can be important; this patch allows the scope to be specified for the check. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200989 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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8e5e00bd07 |
Merged revisions 200946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Michiel van Baak
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12e5744655 |
Merged revisions 200943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail can only use one storage module at the moment. Because it's unclear that selecting one of the storage modules in menuselect will disable filesystem storage we now have a FILE_STORAGE option that conflicts with the other modules. (closes issue #15333) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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c4acb78d1e |
Merged revisions 200764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200766 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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a498e2f0fe |
Merged revisions 200726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines Document the new automatic 'ignoresdpversion' behavior. Asterisk will now automatically ignore incorrect incoming SDP version numbers when necessary to complete a T.38 re-INVITE operation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200728 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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7375533824 |
Merged revisions 165180,200689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200707 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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369810c36c |
Merged revisions 200514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200516 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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e0aa37c39a |
Merged revisions 200361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines Merged revisions 200360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Sean Bright
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852a3835ee |
Merged revisions 199781 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 | seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 lines Fix all of the parallel build warnings issued when running make -j#. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Terry Wilson
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713d775a87 |
Don't access rtp->rtcp->* if rtp->rtcp is null
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |