mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
18436 Commits (104974cdb4c57f697e78e4073e68ecb88ecea12c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
bd2e4dc229 |
Merged revisions 203375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines Fix a case where CDR answer time could be before the start time involving parking. (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203376 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
3814caadad |
Merged revisions 203311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009) | 2 lines Don't try to free NULL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
5c7da226e4 |
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/trunk@203304 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
afa8db54a0 |
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/trunk@203258 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
694c1dcd76 |
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/trunk@203231 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ae87ba45b5 |
Add support for multicast RTP paging.
(closes issue #11797) Reported by: macbrody Review: https://reviewboard.asterisk.org/r/270/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ca3a181c33 |
Pass a logmsg to ast_log_vsyslog instead of separate arguments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ce70b28f38 |
Insure ring cadence is set for fxs ports
Moved SETCADENCE ioctl call to before call into new analog signal module to insure that it gets set. (closes issue #15381) Reported by: alecdavis Patches: fix15381.diff uploaded by dbailey (license 819) Tested by: dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
c6a986222e |
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/trunk@203116 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
80822297d4 |
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/trunk@203037 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
0a915a84e6 |
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/trunk@202967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
4c07c7a6b2 |
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/trunk@202925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
0ed7219ffe |
Ignore some files generated when asterisk.pdf is created.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
1fa4796b19 |
Update sample cdr_tds configuration to try and eliminate some confusion.
Also change the preferred configuration option from 'hostname' (which was misleading because it didn't actually treat the value as a hostname) to 'connection' and added some verbage explaining that the user would need to refer to their freetds.conf file for those settings. 'hostname' was kept as a backwards compatible configuration parameter. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202887 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
5059530d62 |
Change some section names in the CDR tex documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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233f1bc8da |
Remove some trailing whitespace before making content changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1fe9c73c19 |
Clean up section hierarchy for the CDR chapter.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2a68d05b96 |
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/trunk@202761 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2ce62b35cf |
Make outgoing_colp=2 misdn.conf port parameter not send redirecting or transfer messages.
If the outgoing_colp parameter is set to not send COLP information, then it does not make sense to send redirecting or transfer messages announcing new COLP information that is blocked. The service provider may supply the listed number for that line when it passes the messages to the next hop. Why tell the switch that these events happened when the information is otherwise suppressed? Also blocked the number of previous redirects that may have occurred to calls going out the port when outgoing_colp is 2. Follow on to JIRA ABE-1853. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202755 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ef667f2f82 |
If we delete the info, lets also delete the lines
(closes issue #14509) Reported by: timeshell Patches: 20090504__bug14509.diff.txt uploaded by tilghman (license 14) Tested by: awk, timeshell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202753 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5f73ab9f4e |
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/trunk@202672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0c64f73f0a |
Blocked revisions 202601 via svnmerge
........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202603 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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008621ab84 |
Blocked revisions 202572 via svnmerge
........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0bbd5c9424 |
Ignore voicemail messages that are just silence.
(closes issue #2264) Reported by: pfn Patches: silent-vm-1.6.2.txt uploaded by pfn (license 810) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2affa3e999 |
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/trunk@202497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8d3fb80928 |
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/trunk@202417 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e2bfdbac0a |
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/trunk@202415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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87c8658912 |
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/trunk@202410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f142cbe10c |
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/trunk@202343 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e68e6f9d75 |
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/trunk@202337 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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356c0e2f8c |
Note a bug in cdr_sqlite3_custom so I don't forget about it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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eccfeaf9b4 |
Fix possibility of crashiness during reload in custom fields handling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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32fdcc00b2 |
Standardize return values of load_config() so reload() doesn't report an error on success.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202258 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f57752d380 |
Leave a note about some unsafe code in cdr_manager
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2fc4832e85 |
Fix version detection for API changes in spandsp.
(closes issue #15355) Reported by: deuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b390327dd7 |
Remove unnecessary usleep() from a couple of module unload callbacks.
In passing, also tweak cdr_unregister() to hold the list lock a bit less time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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55c6789f74 |
Use sched_yield() instead of usleep(1)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7339d85ee3 |
Blocked revisions 202022 via svnmerge
........ r202022 | mnicholson | 2009-06-19 16:21:15 -0500 (Fri, 19 Jun 2009) | 4 lines 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/trunk@202026 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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05da5f14d9 |
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/trunk@201994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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25ccb5e850 |
Add note about the addition of calendar support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6b53ec413d |
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/trunk@201904 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e85296e244 |
Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201902 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ad0d1bfd9e |
Merged revisions 201828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines 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/trunk@201829 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f341418988 |
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/trunk@201783 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e1fa477ba7 |
Clarify CUT code, and in the process, fix a bug in trunk only
(closes issue #15320) Reported by: chappell Patches: cut_fix.patch uploaded by chappell (license 8) cut_clarify.patch uploaded by chappell (license 8) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201745 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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21ad428d0d |
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/trunk@201717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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dcfe69ec64 |
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/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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730e60e583 |
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/trunk@201610 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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dce6a54a4a |
Trunk implementation of setting an alternate RTP source.
This contains the interface by which we can let an rtp instance know that it might start receiving audio from a new source. This is similar in nature to revision 197588 of Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a11ac5ae2f |
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/trunk@201570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |