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${ noResults }
21718 Commits (283f63c594d0c4327c51fdae0587a6a9f1cb15d3)
Author | SHA1 | Message | Date |
---|---|---|---|
|
0bd877621e |
Addition of "outofcall_message_context" sip.conf option.
Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
e42402ba2c |
Merged revisions 323154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323154 | lmadsen | 2011-06-13 15:00:41 -0400 (Mon, 13 Jun 2011) | 6 lines Tweak documentation for AGI Hangup command. (closes issue ASTERISK-17999) Reported by: Ben Klang Patches: hangup-doc.diff - uploaded by Ben Klang (License #5876) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323155 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
42cb4cf514 |
MOH for only user not working with ConfBridge
This adds the playing_moh flag to the conference_bridge_user struct that signifies when MOH should be playing so code doesn't have to guess whether MOH is playing. This change also adds the necessary checking to ensure that MOH continues playing for a single user in a conference after the join sound is played when configured to do so. (closes ASTERISK-17988) Review: https://reviewboard.asterisk.org/r/1263/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323107 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
cd15477923 |
ConfBridge: Use of bridge or user profiles that don't exist
Bridge and user profiles are not checked for existence before use. The lack of a fully formed bridge profile can cause a segfault when sounds are accessed. This change ensures that bridge and user profiles exist prior to usage attempts. Review: https://reviewboard.asterisk.org/r/1264/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323106 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
4c459c2c85 |
Merged revisions 323040 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop. (closes issue ASTERISK-17798) tested by mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
58ca560291 |
Merged revisions 322981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) | 11 lines Avoid a DB1 infinite loop bug Explicity check the last entry in the DB and make sure that we don't iterate past it. Since there can be no duplicates, this just makes sure that we stop after matching the last key. This patch also refactors the code to get away from some code duplication. A previous patch added many astdb tests and this patch passed them. Review: https://reviewboard.asterisk.org/r/1259/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6017de6292 |
Merged revisions 322923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322923 | twilson | 2011-06-09 19:33:23 -0700 (Thu, 09 Jun 2011) | 2 lines Add some astdb unit tests ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5eb1d79d40 |
Merged revisions 322865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322865 | twilson | 2011-06-09 15:29:20 -0700 (Thu, 09 Jun 2011) | 4 lines Correct ast_db_deltree documentation ast_db_deltree returns -1 on error, otherwise the number of deletions ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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53ef4bfc16 |
Merged revisions 322807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun 2011) | 5 lines don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0a8f9d2cf0 |
Merged revisions 322749 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f0deadaf60 |
Blocked revisions 322585 via svnmerge
........ r322585 | jrose | 2011-06-09 09:06:42 -0500 (Thu, 09 Jun 2011) | 11 lines Adds ast_escape_encoded utility to properly handle escaping of quoted field before uri. This commit backports a feature in trunk affecting initreqprep so that display name won't be encoded improperly. Also includes unit tests for the ast_escape_quoted function. This patch gives 1.8 a much improved outlook in countries which don't use standard ASCII characters. (closes issue ASTERISK-16949) Reported by: Örn Arnarson Review: https://reviewboard.asterisk.org/r/1235/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7df5d0d416 |
Add autoanswer to skinny.
Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER. Initial value must be the time to autoanswer in ms, then optionally :BEEP to play a tone when answered and :MUTE to mute the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and play a beep. just 3000 would answer afer 3 secs of ringing with no beep and full two way audio. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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67dc7a4c93 |
Merged revisions 322484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines Ring all queue with more than 255 agents will cause crash. 1. Create a ring-all queue with 500 permanent agents. 2. Call it. 3. Asterisk will crash. The watchers array in app_queue.c has a hard limit of 255. Bounds checking is not done on this array. No sane person should put 255 people in a ring-all queue, but we should not crash anyway. * Added bounds checking to the watchers array. JIRA AST-464 JIRA SWP-2903 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322485 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4b773e2ed9 |
Merged revisions 322425 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) | 16 lines SRV lookup attempted for SIP peers listed as an IP address. Asterisk attempts to SRV lookup a host name even if the host name is an IP address. Regression introduced when IPv6 support was added. * Restored the check in ast_dnsmgr_lookup() to see if the given host name is an IP address. The IP address could be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815) Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621) Review: https://reviewboard.asterisk.org/r/1240/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322426 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9598e5bc2f |
Remove skinny do_monitor and use ast_sched_start instead
The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything. Review: https://reviewboard.asterisk.org/r/1256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322381 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4cd9bc43c2 |
Merged revisions 322322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines Make handle_request_publish do dialog expiration and destruction. This patch fixes handle_request_publish so that it does dialog expiration and destruction. Without this patch the incoming PUBLISH requests will get stuck in the dialog list. Restarting asterisk is the only way to remove them. Personal observation on one system the server hung up while looping through the channels rendering asterisk unusable and all sip phones unregisterd when they try reregister more requests are added. (closes issue #18898) Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: https://reviewboard.asterisk.org/r/1253 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322323 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ba625fa7d5 |
Correct some whitespace and a reference debug message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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8755236193 |
Actually check the "sendtodialplan" option setting for xmpp.
(closes issue ASTERISK-17978) Reported by: elguero Patches: stop_messages_going_to_dialplan.patch (license #5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322244 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5cb2775480 |
Merged revisions 322189 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot' (closes issue ASTERISK-17915) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322190 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2cfe89a7fd |
Remove Unused Var Warning rt_handle_member_record
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322128 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cfb10e99b5 |
Refactor rt_handle_member_record
Review: https://reviewboard.asterisk.org/r/1172 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4ab3825fe4 |
Merged revisions 322069 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | 8 lines Fixes level toggling for logger set levels since it was reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H Review: https://reviewboard.asterisk.org/r/1244/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322070 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c8548bad22 |
Merged revisions 321926 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) | 18 lines Asterisk crash when unloading cdr_radius/cel_radius. The rc_openlog() API call is passed a string that is used by openlog() to format log messages. The openlog() does not copy the string it just keeps a pointer to it. When the module is unloaded, the string is gone from memory. Depending upon module load order and if the other module then has an error, a crash happens. * Pass rc_openlog() a strdup'd string with the understanding that there will be a small memory leak if the cdr_radius/cel_radius modules are unloaded. * Call rc_destroy() to free the rc handle memory when the module is unloaded. JIRA AST-483 JIRA SWP-3062 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321927 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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31bcafab5b |
Merged revisions 321924 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) | 5 lines Be more explicit for CCSS generic device state event subscription. Make CCSS generic device state event subscription specify the AST_EVENT_IE_STATE ie exists to be safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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85aa126b34 |
Merged revisions 321871 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) | 27 lines Event subscription fixes. Must commit the subscription fixes together with the integration subscription tests. The subscription fixes cause an erroneously passing test to fail. The new subscription tests detect errors without the subscription fixes. * Added missing event_names[] table entry. * Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to correctly detect if a subscriber exists for the proposed event. * Made match_ie_val() and match_sub_ie_val_to_event() check the buffer length for RAW payload types. * Fixed error handling memory leak in ast_event_sub_activate(), ast_event_unsubscribe(), and ast_event_queue(). * Made ast_event_new() and ast_event_check_subscriber() better protect themselves from an invalid payload type. * Added container lock protection between removing old cache events and adding the new cached event in ast_event_queue_and_cache()/event_update_cache(). * Added new event subscription tests. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321872 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 321812-321813 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription description parameter string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321814 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c55fb048b2 |
Blocked revisions 321753 via svnmerge
........ r321753 | russell | 2011-06-03 13:32:45 -0500 (Fri, 03 Jun 2011) | 2 lines Backport an astobj2 unit test so that it runs on 1.8 as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321754 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6357719a82 |
Fix some astobj2 iterator breakage, add another unit test.
Review: https://reviewboard.asterisk.org/r/1254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321752 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a0468ca7fa |
Merged revisions 321685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) | 5 lines Also document the 'queue-minute' option. (closes issue #19386) Reported by: juanmol ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9cd3cf2e71 |
Fix message destination extension.
Don't send all messages to 's'. Get the destination from the request URI. (Found using automated test cases). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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49927bcbb8 |
Merged revisions 321547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321547 | rmudgett | 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011) | 1 line CDR comment tweaks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3f4d0e8743 |
Support routing text messages outside of a call.
Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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eca8a0a625 |
Merged revisions 321537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 Jun 2011) | 8 lines This patch fixes an issue with using the wrong voicemail folders with greetings. (closes issue #17871) Reported by: edhorton Patches: digium_bug_17871_2 uploaded by fhackenberger (license 592) Tested by: edhorton, fhackenberger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ea01c3b4fa |
Merged revisions 321528 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines Fix double alerting, add forced alerting before answer Fix double alerting (it wasn't fixed here by issue #18542) Add forced alerting before connect (if it wasn't before) Try to send all packets from outgoing queue rather than one only Call goes into clearing state when disconnect command is received (closes issue #19361) Reported by: vmikhelson Patches: issue19361-3.patch uploaded by may213 (license 454) Tested by: vmikhelson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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17b8521836 |
Merged revisions 321517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011) | 1 line Update some comments. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321518 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3588746c75 |
Merged revisions 321515 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) | 12 lines Chan_local locking cleanup. This patch removes all of the unnecessary deadlock avoidance loops that occur in chan_local. It also resolves an issue with a deadlock triggered by local channel optimizations. (issue #18028) Review: https://reviewboard.asterisk.org/r/1231/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321516 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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42907d40cd |
Merged revisions 321511 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) | 8 lines Enhance NOTICE message to know who couldn't access the dialplan. (closes issue #19390) Reported by: lmadsen Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10) Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5da4161283 |
Merged revisions 321436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321436 | rmudgett | 2011-05-27 19:27:52 -0500 (Fri, 27 May 2011) | 4 lines Some hagi launch cleanup. Inspired by issue 19256. This patch would also fix the crash. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321445 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9d8943868c |
Merged revisions 321392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) | 12 lines Crash when using hagi and no servers are available. When none of the servers returned by the SRV querey respond, asterisk crashes. The problem is that if the loop over all the SRV entries finishes then the srv_context has already been cleaned up. * Make ast_srv_cleanup() check to see if the context is already cleaned up. (closes issue #19256) Reported by: byronclark ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cdee44e992 |
Merged revisions 321337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Blocked revisions 321335 via svnmerge
........ r321335 | lmadsen | 2011-05-27 17:54:54 -0400 (Fri, 27 May 2011) | 7 lines Fix issue with playback of H.261 video. (closes issue #19379) Reported by: neutrino88 Patches: videoprompt.patch uploaded by neutrino88 (license 297) (changes by russell) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321336 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a2ca0997a6 |
Merged revisions 321333 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) | 7 lines Allow parking lot hints and musicclass to be set. (closes issue #19378) Reported by: sboily_proformatique Patches: pf_parkinghint_music_fix uploaded by sboily proformatique (license 206) Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321334 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d211be98ed |
Add note about PrivacyManager to UPGRADE.txt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321332 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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83439d0581 |
Merged revisions 321330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. The trunk(v1.10) version will remove the unused options position. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0caae96609 |
Merged revisions 321273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | 3 lines markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 321211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May 2011) | 16 lines Fix *8 directed pickup locks system during pickupsound play out move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2 threads trying to write audio to same channel. In addition fixes choppy audio beep in issue 19177. (issue #18654) (issue #19177) Reported by: Docent Patches: review1232-1.8.diff.txt alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1232/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 321155 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic. Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails. (closes issue #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321156 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 321100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null pointer Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables (closes issue #19346) Reported by: kobaz Patches: netsock2.patch uploaded by kobaz (license 834) Tested by: kobaz, Marquis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321101 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 321044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011) | 1 line Update ast_sockaddr comment with an important note. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321045 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 321042 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321042 | twilson | 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011) | 6 lines Initialize stack-allocated ast_sockaddrs before use It is important to always initialize ast_sockaddrs before use--even if they are passed to ast_sockaddr_copy as the underlying storage could be bigger than what ends up being copied--leaving part of the data unitialized. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321043 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |