4117 Commits (33ce9ae08e85ceabd179c8bdef108d1ff5778ad2)

Author	SHA1	Message	Date
Matthew Jordan	c2d84a5515	Fix compilation error introduced in r388815 ... r388815 didn't not cleanly apply the intended patch. As a result, app_voicemail had a small issue in that it didn't compile. Whoops. This patch pulls in the correct variable declarations such that gcc can figure out what 'urgent_index' is all about. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@397704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	5ecc000465	Fix VM snapshot handling for combined INBOX. ... The snapshot API contains an option that allow for combining of new and old messages within a single snapshot. New messages, however, include options beyond just 'INBOX' - it also includes the Urgent folder. A previous patch that combined INBOX and Urgent accidentally impacted snapshots that attempted to gain messages from just the Old folder. This patch fixes the snapshot gathering such that the API returns the appropriate messages for the folder selected, with and without the combine option. This should make it more clear about what's happening. Review: https://reviewboard.asterisk.org/r/2539/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@388815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	f6df3bddea	Merged revisions 378689 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r378689 | rmudgett | 2013-01-08 17:55:57 -0600 (Tue, 08 Jan 2013) | 4 lines app_queue: Fix incorrect assertion. (issue ASTERISK-16115) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@378705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	859edde66d	Merged revisions 378683 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r378683 | rmudgett | 2013-01-08 16:55:43 -0600 (Tue, 08 Jan 2013) | 29 lines app_queue: Fix multiple calls to a queue member that is in only one queue. When ringinuse=no queue members can receive more than one call if these calls happen at nearly the same time. * Fix so a queue member does not receive more than one call from a queue. NOTE: This fix does not prevent multiple calls to a member if the member is in more than one queue. * Did some refactoring to eliminate some code redundancy. (issue ASTERISK-16115) Reported by: nik600 Patches: jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett Modified * Revert the -r341580 and -r341599 changes adding the queues.conf check_state_unknown option as it was added in an attempt to fix this problem. The fix did not need to be optional. The fix should not have tried to explicitly set the device state. Setting the device state by something other than the device introduces a race condition. I also could not see how the change would be effective other than delaying the app_queue code long enough for the device state to propagate to app_queue. ........ Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@378686 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	0c0daa1ce0	Multiple revisions 377838,377842,377848,377882,377923,377947,377992,378037,378089,378093,378120,378218,378286,378320 ... ........ r377838 | rmudgett | 2012-12-11 14:42:59 -0600 (Tue, 11 Dec 2012) | 16 lines Cleanup taskprocessor on exit. * Cleanup CLI commands on exit. * v10 only: Merged v1.8 -r374177 change to taskprocessor.c missed in v10 -r374178. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: taskprocessor-cleanup-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell taskprocessor-cleanup-10-only.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377842 | mmichelson | 2012-12-11 14:48:16 -0600 (Tue, 11 Dec 2012) | 13 lines Fix crash that can occur if CLI registration fails for an aliased command. A recent memory leak fix in main/cli.c causes an ast_cli_entry's command field to be freed and NULLed if ast_cli_register() fails. res_clialiases was ignoring the return value of ast_cli_register() and was then passing the NULL command off to a a hash function. This resulted in a crash. The fix is not to ignore the erroneous return value. If ast_cli_register() fails, then we do not continue trying to process the current alias. ........ Merged revisions 377840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377848 | rmudgett | 2012-12-11 15:07:47 -0600 (Tue, 11 Dec 2012) | 14 lines Cleanup udptl on exit. * Cleanup CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377882 | rmudgett | 2012-12-11 15:57:44 -0600 (Tue, 11 Dec 2012) | 10 lines Cleanup CLI commands on exit for several files. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: unregister-cli-multiple-all.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 377881 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377923 | newtonr | 2012-12-12 16:41:24 -0600 (Wed, 12 Dec 2012) | 12 lines Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to 1.4.12 for new Extra Sounds releases See CHANGES-* files in English extra 1.4.12 tarballs for new sound prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan (closes AST-755) Reported by: John Bigelow ........ Merged revisions 377922 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377947 | kmoore | 2012-12-13 07:48:32 -0600 (Thu, 13 Dec 2012) | 17 lines Ensure Min-SE is included in outbound INVITEs Asterisk now includes Min-SE in outbound INVITEs when the value is not 90 (the default) and session timers are not disabled. This has the effect of Asterisk following RFC4028 more closely with regard to 422 responses and preventing situations in which Asterisk would be forced to temporarily accept a call to tear it down based on a Session-Expires below the locally configured Min-SE. (issue SWP-5051) Review: https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey Moore Patch-by: Kinsey Moore ........ Merged revisions 377946 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377992 | rmudgett | 2012-12-13 14:52:26 -0600 (Thu, 13 Dec 2012) | 29 lines confbridge: Fix MOH on simultaneous user entry to a new conference. When two users entered a new conference simultaneously, one of the callers hears MOH. This happened if two unmarked users entered simultaneously and also if a waitmarked and a marked user entered simultaneously. * Created a confbridge internal MOH API to eliminate the inlined MOH handling code. Note that the conference mixing bridge needs to be locked when actually starting/stopping MOH because there is a small window between the conference join unsuspend MOH and actually joining the mixing bridge. * Created the concept of suspended MOH so it can be interrupted while conference join announcements to the user and DTMF features can operate. * Suspend any MOH until the user is about to actually join the mixing bridge of the conference. This way any pre-join file playback does not need to worry about MOH. * Made post-join actions only play deferred entry announcement files. Changing the user/conference state during that time is not protected or controlled by the state machine. (closes issue ASTERISK-20606) Reported by: Eugenia Belova Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2232/ ........ r378037 | rmudgett | 2012-12-14 15:30:46 -0600 (Fri, 14 Dec 2012) | 20 lines app_queue: Revert bad ringinuse=no patch. With the option ringinuse=no set, the patch committed for ASTERISK-16115 causes non-SIP queue members to never be called because the device state is checked after a channel is created to determine if the member is busy. These queue members always get the "Member %s is busy, cannot dial" message. Most channel drivers other than chan_sip use the default device state handling. The default device-state state is considered in use or unknown if the channel exists or not respectively. (closes issue ASTERISK-20801) Reported by: rmudgett Patches: jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r378089 | rmudgett | 2012-12-17 16:57:10 -0600 (Mon, 17 Dec 2012) | 16 lines Make chan_local module references tied to local_pvt lifetime. The chan_local module references were manually tied to the existence of the ;1 and ;2 channel links. * Made chan_local module references tied to the existence of the local_pvt structure as well as automatically take care of the module references. * Tweaked the wording of the local_fixup() failure warning message to make sense. Review: https://reviewboard.asterisk.org/r/2181/ ........ Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r378093 | rmudgett | 2012-12-17 17:08:40 -0600 (Mon, 17 Dec 2012) | 5 lines Fix potential double free when unloading a module. ........ Merged revisions 378092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r378120 | kmoore | 2012-12-18 11:38:22 -0600 (Tue, 18 Dec 2012) | 11 lines Add test events for time limit-related hangups This patch adds hangup-related test events in order to support testing of time-limited bridges. This aids in testing the S() and L() bridge options. (issue SWP-4713) ........ Merged revisions 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r378218 | kmoore | 2012-12-31 08:43:26 -0600 (Mon, 31 Dec 2012) | 12 lines Ensure chan_sip rejects encrypted streams without crypto info This ensures that Asterisk rejects encrypted media streams (RTP/SAVP audio and video) that are missing cryptographic keys and ensures that the incoming SDP is consistent with RFC4568 as far as having a crypto attribute present for any SAVP streams. Review: https://reviewboard.asterisk.org/r/2204/ ........ Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r378286 | mjordan | 2013-01-02 09:23:57 -0600 (Wed, 02 Jan 2013) | 30 lines Resolve crashes due to large stack allocations when using TCP Asterisk had several places where messages received over various network transports may be copied in a single stack allocation. In the case of TCP, since multiple packets in a stream may be concatenated together, this can lead to large allocations that overflow the stack. This patch modifies those portions of Asterisk using TCP to either favor heap allocations or use an upper bound to ensure that the stack will not overflow: * For SIP, the allocation now has an upper limit * For HTTP, the allocation is now a heap allocation instead of a stack allocation * For XMPP (in res_jabber), the allocation has been eliminated since it was unnecesary. Note that the HTTP portion of this issue was independently found by Brandon Edwards of Exodus Intelligence. (issue ASTERISK-20658) Reported by: wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches: ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r378320 | mjordan | 2013-01-02 11:40:28 -0600 (Wed, 02 Jan 2013) | 27 lines Prevent exhaustion of system resources through exploitation of event cache Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) ........ Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377838,377842,377848,377882,377923,377947,377992,378037,378089,378093,378120,378218,378286,378320 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@378661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	5e65377992	Multiple revisions 377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772 ... ........ r377136 | rmudgett | 2012-12-03 14:33:08 -0600 (Mon, 03 Dec 2012) | 17 lines Cleanup core main on exit. * Cleanup time zones on exit. * Make exit clean/unclean report consistent for AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch uploaded by Corey Farrell core-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377135 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377166 | rmudgett | 2012-12-03 16:53:58 -0600 (Mon, 03 Dec 2012) | 15 lines Cleanup ast_run_atexits() atexits list. * Convert atexits list to a mutex instead of a rd/wr lock. The lock is only write locked. * Move CLI verbose Asterisk ending message to where AMI message is output in really_quit() to avoid further surprises about using stuff already shutdown. (issue ASTERISK-20649) Reported by: Corey Farrell ........ Merged revisions 377165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377212 | rmudgett | 2012-12-04 16:31:02 -0600 (Tue, 04 Dec 2012) | 1 line confbridge: Update online XML documentation. ........ r377227 | rmudgett | 2012-12-04 18:49:53 -0600 (Tue, 04 Dec 2012) | 29 lines confbridge: Fix several small issues. * Made func_confbridge_helper() allow an empty value when setting options. You previously could not Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the dialplan. * Made func_confbridge_helper() handle its datastore better if multiple threads attempt to set the first CONFBRIDGE option value on the channel. * Made the func_confbridge_helper() only output one diagnostic message concerning the option. * Made the bridge video_mode able to repeatedly change in the config file and CONFBRIDGE dialplan function. The video_mode option values are an enum and not independent of each other. * Made handle_cli_confbridge_show_bridge_profile() better handle the video_mode option. * Simplified datastore handling code in conf_find_user_profile() and conf_find_bridge_profile(). * Made parse_bridge(), parse_user(), and parse_menu() use var->file instead of CONFBRIDGE_CONFIG because the var could have been from an include file. (closes issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter ........ r377241 | rmudgett | 2012-12-04 20:09:13 -0600 (Tue, 04 Dec 2012) | 4 lines * Fix registering core show codecs/codec CLI commands twice. * Fix registering atexit format_attr_shutdown() more than once. ........ r377258 | file | 2012-12-05 10:49:33 -0600 (Wed, 05 Dec 2012) | 19 lines Fix a SIP request memory leak with TLS connections. During the TLS re-work in chan_sip some TLS specific code was moved into a separate function. This function operates on a copy of the incoming SIP request. This copy was never deinitialized causing a memory leak for each request processed. This function is now given a SIP request structure which it can use to copy the incoming request into. This reduces the amount of memory allocations done since the internal allocated components are reused between packets and also ensures the SIP request structure is deinitialized when the TLS connection is torn down. (closes issue ASTERISK-20763) Reported by: deti ........ Merged revisions 377257 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377261 | jrose | 2012-12-05 10:57:26 -0600 (Wed, 05 Dec 2012) | 15 lines res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session When srtp_create fails, the session may be dealloced or just not alloced. At the same time though, the session pointer might not be set to NULL in this process and attempting to srtp_dealloc it again will cause a segfault. This patch checks for failure of srtp_create and sets the session pointer to NULL if it fails. (closes issue ASTERISK-20499) Reported by: tootai Review: https://reviewboard.asterisk.org/r/2228/ ........ Merged revisions 377256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377354 | rmudgett | 2012-12-06 17:56:45 -0600 (Thu, 06 Dec 2012) | 24 lines confbridge: Fix some resource leaks on conference teardown. * Made destroy_conference_bridge() destroy a missed ast_mutex_t and ast_cond_t. * Made join_conference_bridge() init the ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can destroy them unconditionally. * Made join_conference_bridge() abort if the new conference could not be added to the conferences container. * Made leave_conference() discard any post-join actions if join_conference_bridge() had to abort early. * Made the join_conference_bridge() diagnostic messages better describe what happened. * Renamed leave_conference_bridge() to leave_conference() and made it only take a conference user pointer. The conference pointer was redundant. * Made conf_bridge_profile_copy() use struct copy instead of memcpy(). * No need to lock the conference in start_conf_record_thread() since all of the callers already have it locked. ........ r377382 | kmoore | 2012-12-07 15:58:21 -0600 (Fri, 07 Dec 2012) | 17 lines codec_dahdi: Fix output of "transcoder show" CLI command. In r306010 "Asterisk media architecture conversion - no more format bitfields", the logic for incrementing encoders and decoders when opening transcoder channels was changed without making the corresponding change when decrementing encoder / decoder channels. The result being that when a channel was destroyed, codec_dahdi couldn't properly tell if it was an encoder or decoder, and the default case is to assume it was a decoder. This could result in negative numbers for decoders in use like in: VOIP6*CLI> transcoder show 2/-2 encoders/decoders of 92 channels are in use. (closes issue ASTERISK-19921) Patch-by: Shaun Ruffell ........ r377399 | rmudgett | 2012-12-07 17:42:03 -0600 (Fri, 07 Dec 2012) | 5 lines MALLOC_DEBUG: Only wait if we want atexit allocation dumps. ........ Merged revisions 377398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377432 | rmudgett | 2012-12-07 18:29:23 -0600 (Fri, 07 Dec 2012) | 14 lines Fix order of SIP allow/disallow in MySQL contrib script. Using the contrib sippeers.sql script to create the sippeers MySQL table would result in being unable to place calls if you set the disallow value to all. (closes issue ASTERISK-20756) Reported by: Andre Luis Patches: sippeers.patch patch uploaded by Andre Luis ........ Merged revisions 377431 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377504 | tilghman | 2012-12-09 19:24:41 -0600 (Sun, 09 Dec 2012) | 5 lines Remove some dead code and additionally handle a case that wasn't handled. ........ Merged revisions 377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377510 | tilghman | 2012-12-09 19:39:58 -0600 (Sun, 09 Dec 2012) | 16 lines Improve documentation by making all of the colors used readable, no matter what the background color is. Dark blue on a black background is unreadable, as is yellow on a light background. This patch turns on the bright attribute for colors when on a dark background and turns *off* the bright attribute when the -W command line option is used (indicating a _light_ background). This ensures that text is readable in both cases. Patch by: tilghman Review: https://reviewboard.asterisk.org/r/2224 ........ Merged revisions 377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377558 | igorg | 2012-12-09 23:04:36 -0600 (Sun, 09 Dec 2012) | 8 lines Fix crash on transfer initiated from insreeen menu on Unistim phones. Removed CDR-related code that moved to do_masquarade before. (closes issue ASTERISK-20417) Reported by: Rudolf Migalin ........ Merged revisions 377557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377592 | igorg | 2012-12-10 00:41:47 -0600 (Mon, 10 Dec 2012) | 9 lines Fix codec mismatch Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations. (issue ASTERISK-20183) ........ Merged revisions 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377624 | kmoore | 2012-12-10 08:40:26 -0600 (Mon, 10 Dec 2012) | 14 lines Handle Session-Expires less than local Min-SE in 200 OK Ensure that a call is immediately torn down if a Session-Expires value received in a 200 OK is less than the local Min-SE. This also prevents Asterisk from allowing calls with Session-Expires below the RFC4028-mandated minimum (90s). (closes issue ASTERISK-20653) Review: https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore ........ Merged revisions 377623 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377656 | kmoore | 2012-12-10 10:53:16 -0600 (Mon, 10 Dec 2012) | 14 lines Ensure ReceiveFax provides a CED tone via T.38 When using res_fax_digium, the T.38 CED tone was not being provided properly which would cause some incoming faxes to fail. This was not an issue with res_fax_spandsp since it does not strictly honor the send_ced flag and sends the CED tone whenever receiving a T.38 fax. (closes issue FAX-343) Reported-by: Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377705 | rmudgett | 2012-12-10 18:32:40 -0600 (Mon, 10 Dec 2012) | 14 lines Cleanup dnsmgr on exit. * Cleanup dnsmgr thread and CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377709 | rmudgett | 2012-12-10 19:00:05 -0600 (Mon, 10 Dec 2012) | 15 lines Cleanup event on exit. * Cleanup CLI commands on exit. * v10 only: Merged v1.8 -r374177 change to event.c missed in v10 -r374178. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: event_shutdown-10-only.patch (license #5909) patch uploaded by Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 377708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377741 | rmudgett | 2012-12-10 20:11:29 -0600 (Mon, 10 Dec 2012) | 19 lines Cleanup indications on exit. * Made ast_unregister_indication_country() unlink the found tone zone before selecting a new default_tone_zone to make it impossible to select the tone zone being unregistered again. * Ringcadence is no longer parsed twice in store_config_tone_zone(). * Cleanup CLI commands and destroy default_tone_zone on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377740 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377772 | rmudgett | 2012-12-10 20:42:34 -0600 (Mon, 10 Dec 2012) | 13 lines Cleanup logger on exit. * Cleanup CLI commands, destroy verbosers and logchannels lists on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: logger-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@378659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	45b5a9b572	Fix incorrect pointer manipulation that caused mixmonitor recording to fail. ... (closes ASTERISK-20834) reported by Philippe Lindheimer Patches: ASTERISK-20834.patch uploaded by Mark Michelson (License #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@378655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	a2ae62c0ff	Merged revisions 376658 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r376658 | rmudgett | 2012-11-27 14:36:45 -0600 (Tue, 27 Nov 2012) | 21 lines Remove unnecessary channel module references. * Removed call to ast_module_user_hangup_all() in res_config_mysql.c since it is effectively a noop. No channels can attach a reference to that module. * Removed call to ast_module_user_hangup_all() in app_celgenuserevent.c. The caller of unload_module() has already called it. * Removed redundant channel module references in pbx_dundi.c. The registered dialplan function callback dispatchers for the read/read2/write callbacks already reference the module before calling. * pbx_dundi: Moved unregistering CLI commands, DUNDi switch, and dialplan functions to the first thing the unload_module() does. This will reduce the chance of new channels using DUNDi services while the module is being torn down. ........ Merged revisions 376657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@376686 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	fa81008f3a	Merged revisions 376414 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r376414 | mjordan | 2012-11-18 08:22:39 -0600 (Sun, 18 Nov 2012) | 8 lines Add a test event that reports changes in ConfBridge state This patch adds a test event to ConfBridge that reports transitions between states in ConfBridge. This is used by tests in the Asterisk Test Suite that verify state changes based on the entering/leaving of conference participants. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@376427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	da618f42ea	Merged revisions 376308,376315 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r376308 | jrose | 2012-11-15 16:55:04 -0600 (Thu, 15 Nov 2012) | 17 lines app_meetme: Fix channels lingering when hung up under certain conditions Channels would get stuck and MeetMe would repeatedly display an Unable to write frame to channel error in the conf_run function if hung up during certain sound prompts such as during user count announcements. This patch fixes that by reintroducing a hangup check in the meetme's main loop (also in conf_run). (closes issue ASTERISK-20486) Reported by: Michael Cargile Review: https://reviewboard.asterisk.org/r/2187/ Patches: meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 376307 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r376315 | dlee | 2012-11-15 17:17:54 -0600 (Thu, 15 Nov 2012) | 28 lines Migrate hashtest/hashtest2 to be unit tests. Both hashtest and hashtest2 are manual testing apps that thrash hash tables (hashtab and ao2 containers, respectively), by spinning up several threads that randomly insert, delete, lookup and iterate over the hash table. If the app doesn't crash, the hash table probably passes the test. Those utils are not a part of the typical Asterisk build, so they do not usually get compiled. This all makes them less that useful. This patch removes those manual test programs and replaces them with Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also attempts to make the tests more deterministic. * Rather than spinning up some number of threads that operate on the hash table randomly, spin up four threads that concurrenly add, remove, lookup and iterate over the hash table. * Each thread checks the state of the hash table both during and after execution, and indicates a test failure if things are not as expected. * Each thread times out after 60 seconds to prevent deadlocking the unit test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged revisions 376306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@376338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	b5239761f9	Merged revisions 376263 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r376263 | newtonr | 2012-11-14 19:50:54 -0600 (Wed, 14 Nov 2012) | 10 lines (issue ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo Takebe Tested by: Rusty Newton Patches: asterisk20280.patch uploaded by Rusty Newton (license 5829) ........ Merged revisions 376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@376281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	a54fdc3e43	Merged revisions 375995 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r375995 | mmichelson | 2012-11-07 11:16:24 -0600 (Wed, 07 Nov 2012) | 41 lines Multiple revisions 375993-375994 ........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@376013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	6181402839	Merged revisions 375485,375496 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r375485 | jrose | 2012-10-30 13:55:58 -0500 (Tue, 30 Oct 2012) | 8 lines mixmonitor: Add a test event This test event is being used to fix the mixmonitor_audiohook_inherit test. ........ Merged revisions 375484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r375496 | rmudgett | 2012-10-30 14:20:28 -0500 (Tue, 30 Oct 2012) | 8 lines Fix ConfBridge crash if no timing module loaded. (closes issue ASTERISK-19448) Reported by: feyfre Patches: smfix.patch (license #6099) patch uploaded by feyfre Modified for coding guidelines. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@375510 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	8919c7068e	Merged revisions 375470 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r375470 | jrose | 2012-10-30 09:42:29 -0500 (Tue, 30 Oct 2012) | 7 lines confbridge: Fix a bug which made conferences not record with AMI/CLI commands (closes issue ASTERISK-20601) Reported by: Vilius Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose (license 6182) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@375481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	b7ea4e0c78	Merged revisions 375451 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r375451 | mjordan | 2012-10-29 21:14:01 -0500 (Mon, 29 Oct 2012) | 14 lines Ensure that the Queue application tracks busy members in off nominal situations There are a few code paths where the Queue application fails to count a paused or in use queue member as being 'busy'. This can cause callers to get stuck in the Queue until a paused agent unpauses themselves. (closes issue ASTERISK-20623) Reported by: Bryan Walters patches: app_queue.patch uploaded by Bryan Walters (license 5851) ........ Merged revisions 375450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@375468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	239161dbb5	Merged revisions 375217 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r375217 | jrose | 2012-10-18 16:09:29 -0500 (Thu, 18 Oct 2012) | 15 lines app_queue: Make ordering of rrmemory/rrordered persist over add/remove members Prior to this patch, adding, removing or reloading members to rrmemory would cause the order to become completely jumbled. Now it behaves more or less like rrordered other than the fact that it stores the members on a hash table rather than a linked list. This patch also prevents removal of members and member reloads from jumbling rrordered queues. (issue AST-989) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged revisions 375216 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@375241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	f31dd4fb9b	Merged revisions 375026 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r375026 | mmichelson | 2012-10-15 16:06:42 -0500 (Mon, 15 Oct 2012) | 22 lines Fix some potential misuses of ast_str in the code. Passing an ast_str pointer by value that then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or ast_str_append_va() can result in the pointer originally passed by value being invalidated if the ast_str had to be reallocated. This fixes places in the code that do this. Only the example in ccss.c could result in pointer invalidation though since the other cases use a stack-allocated ast_str and cannot be reallocated. I've also updated the doxygen in strings.h to include notes about potential misuse of the functions mentioned previously. Review: https://reviewboard.asterisk.org/r/2161 ........ Merged revisions 375025 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@375043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	65d82246a9	Avoid a segfault on invalid format names ... If a format name was not found by ast_getformatbyname, a NULL pointer would be passed into ast_format_rate and immediately dereferenced. This ensures that a valid pointer is used since the structure is already allocated on the stack. (closes issue DPH-523) Reported-by: Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	3b57018627	Merged revisions 374803 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374803 | rmudgett | 2012-10-10 15:55:44 -0500 (Wed, 10 Oct 2012) | 30 lines app_queue: Made pass connected line updates from the caller to ringing queue members. Party A calls Party B Party B puts Party A on hold. Party B calls a queue. Ringing queue member D sees Party B identification. Party B transfers Party A to the queue. Queue member D does not get a connected line update for Party A. Queue member D answers the call and still sees Party B information. However, if Party A later transfers the call to Party C then queue member D gets a connected line update for Party C. * Made pass connected line updates from the caller to queue members while the queue members are ringing. (closes issue AST-1017) Reported by: Thomas Arimont (closes issue ABE-2886) Reported by: Thomas Arimont Tested by: rmudgett ........ Merged revisions 374801 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 374802 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1f39d3738f	Merged revisions 374652 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r374652 | mjordan | 2012-10-08 08:46:27 -0500 (Mon, 08 Oct 2012) | 46 lines Resolve issues in ConfBridge regarding marked, waitmarked, and unmarked users Thank's to Neil Tallim (flan)'s tireless testing, issue reporting, and patches it became clear that app_confbridge had some complex logic in how it handled interactions between marked, waitmarked, and unmarked users. In particular, there were some areas in which the interactions between the users resulted in inconsistent behavior, and app_confbridge was missing logic in how to handle some corner cases. Some areas included: * Poor handling of mixing unmarked and waitmarked users * Inconsistencies in how MOH and muting was applied to various users * Handling of various announcements for different user profile options flan's patches seem to fix the various issues, but highlighted how hard the code could be to maintain. In an attempt to make things easier to maintain and to more fully enumerate the various cases that exist, this patch breaks up the logic into a state machine-like setup. Please note that the various state transitioned are documented on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes Review: //https://reviewboard.asterisk.org/r/2072/ Note that for the following issues, mjordan uploaded the patch, although it was written by twilson. Any contributor license discrepency is due to that. (closes issue ASTERISK-19562) Reported by: flan Tested by: flan, mjordan, jrose patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) (closes issue ASTERISK-19726) Reported by: flan Tested by: flan patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) (closes issue ASTERISK-20181) Reported by: Jonathan White Tested by: Jonathan White patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	322e224ec4	Merged revisions 374132,374135 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374132 | seanbright | 2012-10-01 12:27:22 -0500 (Mon, 01 Oct 2012) | 2 lines Use ast_copy_string instead of strncpy to guarantee a NUL terminated string. ................ r374135 | seanbright | 2012-10-01 12:52:38 -0500 (Mon, 01 Oct 2012) | 23 lines app_queue: Support persisting and loading of long member lists. Greenlight in #asterisk brought up that he was receiving an error message "Could not create persistent member string, out of space" when running app_queue in Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to store the generated string, but with queues that have large member lists this is not always the case. This patch removes the limitation and uses ast_str instead of a fixed sized buffer. The complicating factor comes from the fact that ast_db_get requires a buffer and buffer size argument, which doesn't let us pull back more than what we pass in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d copy of the value from astdb. As an aside, I did some testing on the maximum size of data that we can store in the BDB library we distribute and was able to store a 10MB string and retrieve it with no problems, so I feel this is a safe patch. Review: https://reviewboard.asterisk.org/r/2136/ ........ Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	0915adaf03	Merged revisions 373946 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373946 | rmudgett | 2012-09-27 17:12:47 -0500 (Thu, 27 Sep 2012) | 14 lines Fix SendDTMF crash and channel reference leak using channel name parameter. The SendDTMF channel name parameter has two issues. 1) Crashes if the channel name does not exist. 2) Leaks a channel reference if the channel is the current channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF documentation. * Renamed app to senddtmf_name and tweaked the type. ........ Merged revisions 373945 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f6ae5b7142	Merged revisions 373816 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373816 | rmudgett | 2012-09-26 13:15:50 -0500 (Wed, 26 Sep 2012) | 18 lines Fixed meetme tab completion and command documentation. * Removed unnecessary case sensitivity in meetme list, lock, unlock, mute, unmute, and kick commands. * Separated meetme lock/unlock, mute/unmute, and kick commands into their own registered commands to simplify tab completion and parameter checking. meetme_lock_cmd(), meetme_mute_cmd(), and meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue AST-1006) Reported by: John Bigelow Tested by: rmudgett ........ Merged revisions 373815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	6128a98e50	Merged revisions 373737 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373737 | mmichelson | 2012-09-25 16:12:40 -0500 (Tue, 25 Sep 2012) | 11 lines Fix error where improper IMAP greetings would be deleted. (closes issue ASTERISK-20435) Reported by: fhackenberger Patches: asterisk-20435-imap-del-greeting.diff uploaded by Michael L. Young (License #5026) (with suggested modification made by me) ........ Merged revisions 373735 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	5dcc32ba72	Merged revisions 373665,373675 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373665 | twilson | 2012-09-25 12:35:30 -0500 (Tue, 25 Sep 2012) | 21 lines Properly handle UAC/UAS roles for SIP session timers The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder. This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties. (closes issue AST-922) Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged revisions 373652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373675 | kmoore | 2012-09-25 13:20:04 -0500 (Tue, 25 Sep 2012) | 13 lines "show" completion option for "queue" shouldn't appear twice When tab-completing CLI commands starting with "queue", "show" appeared twice in the list due to the way that Asterisk's tab completion functions and the order in which the commands were registered. The registration order has been altered to resolve this issue. (closes issue AST-940) Reported-by: Steve Pitts ........ Merged revisions 373666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	3123eb71c7	Merged revisions 373466,373468 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373466 | rmudgett | 2012-09-24 15:44:27 -0500 (Mon, 24 Sep 2012) | 33 lines Fix potential reentrancy problems in chan_sip. Asterisk v1.8 and later was not as vulnerable to this issue. * Made find_call() lock each private as it processes the found dialogs. (Primary cause of ABE-2876) * Made the other functions that traverse the dialogs container lock each private as it examines them. * Fix race condition in sip_call() if the thread that sent the INVITE is held up long enough for a response to be processed. The p->initid for the INVITE retransmission could be added after it was canceled by the response processing. * Made __sip_destroy() clean up resource pointers after freeing. This is primarily defensive in case someone has a stale private pointer. * Removed redundant memset() in reqprep(). The call to init_req() already does the memset() and is the first reference to req in reqprep(). * Removed useless set of req.method in transmit_invite(). The calls to initreqprep() and reqprep() have to do this because they memset() the req. JIRA ABE-2876 .......... Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 373424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373468 | jrose | 2012-09-24 16:05:44 -0500 (Mon, 24 Sep 2012) | 13 lines func_audiohookinherit: Document some missed sources. This patch also mentions that AUDIOHOOK_INHERIT can be used to transfer MixMonitor audiohooks. There is also wiki that addresses audiohooks and the use of AUDIOHOOK_INHERIT at the following link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........ Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	c11c20e464	Merged revisions 373300 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373300 | jrose | 2012-09-21 10:07:38 -0500 (Fri, 21 Sep 2012) | 12 lines app_queue: Make queue reload members and variants of that work Prior to this patch, 'queue reload members' cli command did not work at all. This also affects the manager function 'QueueReload' when supplied with the 'members: yes' field. (closes issue AST-956) Reported by: John Bigelow ........ Merged revisions 373298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	22c0ba7287	Merged revisions 373237,373245 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373237 | mjordan | 2012-09-20 13:42:51 -0500 (Thu, 20 Sep 2012) | 18 lines When processing RFC 2833 DTMF, accomodate increasing timestamps in End events While endpoints should not be changing the source timestamp between DTMF event packets, the fact is there exists those endpoints that do exactly that. To work around this, we absorb timestamps within the expected re-transmit period. Note that this period only affects End of Event packets, so it should not prevent the detection of new DTMF digits that happen to arrive right on top of each other. (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson Tested by: mjordan, Vladimir Mikhelson Review: https://reviewboard.asterisk.org/r/2124 ........ Merged revisions 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373245 | file | 2012-09-20 14:14:31 -0500 (Thu, 20 Sep 2012) | 15 lines Fix incorrect MeetME conference bridge reference count decrementing and sometimes premature destruction. When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one. This operation left around a pointer to the last created conference bridge still containing participants. When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of and the reference count of the conference bridge decremented. If there was only a single participant in the conference bridge it was ultimately destroyed prematurely. (closes issue AST-994) Reported by: John Bigelow ........ Merged revisions 373242 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	d26918bd63	Merged revisions 373196 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r373196 | mjordan | 2012-09-19 21:35:13 -0500 (Wed, 19 Sep 2012) | 12 lines Ensure that all ConfBridge sounds can be set using CONFBRIDGE function The CONFBRIDGE function can be used to set the sounds in a ConfBridge bridge profile. Unfortunately, three sounds were missed in the portion of the code that applies the settings passed in from the function: sound_only_one, join, and leave. This patch makes sure that the sounds passed from the function are applied to the bridge profile. (closes issue ASTERISK-20404) Reported by: univ Tested by: mjordan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	7c19870a4a	Merged revisions 372764,372767 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372764 | kmoore | 2012-09-10 13:32:51 -0500 (Mon, 10 Sep 2012) | 12 lines Warn on CLI when UDPTL init fails This adds a CLI warning when a SDP offer is rejected due to UDPTL initialization failure. Previously, there was no indication of the reason for offer rejection in this case. (closes issue ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi ........ Merged revisions 372763 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r372767 | jrose | 2012-09-10 13:41:45 -0500 (Mon, 10 Sep 2012) | 8 lines app_meetme: Document that 'p' option will continue in dialplan. (closes issue AST-991) Reported by John Bigelow ........ Merged revisions 372765 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	fe4486d684	Merged revisions 372621,372625,372629 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372621 | rmudgett | 2012-09-07 16:24:39 -0500 (Fri, 07 Sep 2012) | 18 lines Fix VoicemailUserEntry event headers ServerEmail and MailCommand reported values. The AMI action VoicemailUsersList VoicemailUserEntry event headers ServerEmail and MailCommand did not report the global values if they were not overridden. The VoicemailUserEntry event header ServerEmail was not populated with the global value if the voicemail user did not override it. The VoicemailUserEntry event header MailCommand was never populated with a value. * Removed unused struct ast_vm_user member mailcmd[]. (closes issue AST-973) Reported by: John Bigelow Tested by: rmudgett ........ Merged revisions 372620 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r372625 | rmudgett | 2012-09-07 16:49:16 -0500 (Fri, 07 Sep 2012) | 10 lines Fix exception path typo in app_queue.c try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy Pepper Patches: fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper ........ Merged revisions 372624 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r372629 | rmudgett | 2012-09-07 17:07:59 -0500 (Fri, 07 Sep 2012) | 8 lines Remove annoying unconditional debug message from INC/DEC functions. (closes issue AST-1001) Reported by: Guenther Kelleter ........ Merged revisions 372628 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	60a5b2ef2a	Merged revisions 372582 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372582 | mjordan | 2012-09-06 21:25:36 -0500 (Thu, 06 Sep 2012) | 13 lines Free ast_str objects when temp file fails to be created in MiniVM The previous commit (r372554) was from a patch that was written before r366880, which ensured that ast_str objects allocated in the sendmail routine were free'd in off nominal paths. This commit frees the string objects in the off nominal path introduced in r372554. (issue ASTERISK-17133) Reported by: Tzafrir Cohen ........ Merged revisions 372581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	72742944b7	Merged revisions 372555 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372555 | mjordan | 2012-09-06 21:11:46 -0500 (Thu, 06 Sep 2012) | 22 lines Fix file descriptor leak and pointer scope issue in MiniVM when sending mail When MiniVM sends an e-mail and it has the volgain option set, it will spawn sox in a separate process to handle the manipulation of the sound file. In doing so, it creates a temporary file. There are two problems here: 1) The file descriptor returned from mkstemp is leaked 2) The finalfilename character pointer points to a buffer that loses scope once volgain processing is finished. Note that in r316265, Russell fixed some gcc warnings by using the return value of the mkstemp call. A warning was placed in minivm that the file descriptor was going to be leaked. This patch reverts that change, as it handles the leak and 'uses' the file descriptor returned from mkstemp. (closes issue ASTERISK-17133) Reported by: Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir Cohen (license #5035) ........ Merged revisions 372554 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	6690d0f289	Merged revisions 372518,372522 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372518 | kmoore | 2012-09-06 16:40:50 -0500 (Thu, 06 Sep 2012) | 14 lines Ensure listed queues are not offered for completion When using tab-completion for the list of queues on "queue reset stats" or "queue reload {all|members|parameters|rules}", the tab-completion listing for further queues erroneously listed queues that had already been added to the list. The tab-completion listing now only displays queues that are not already in the list. (closes issue AST-963) Reported-by: John Bigelow ........ Merged revisions 372517 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r372522 | rmudgett | 2012-09-06 17:10:04 -0500 (Thu, 06 Sep 2012) | 22 lines Fix loss of MOH on an ISDN channel when parking a call for the second time. Using the AMI redirect action to take an ISDN call out of a parking lot causes the MOH state to get confused. The redirect action does not take the call off of hold. When the call is subsequently parked again, the call no longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on repeated AST_CONTROL_HOLD frames if it is already in a state where it is supposed to be sending MOH. The MOH may have been stopped by other means. (Such as killing the generator.) This simple fix is done rather than making the AMI redirect action post an AST_CONTROL_UNHOLD unconditionally when it redirects a channel and thus potentially breaking something with an unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches: jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 372521 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	d58e162ac9	Merged revisions 372445 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372445 | kmoore | 2012-09-06 09:29:35 -0500 (Thu, 06 Sep 2012) | 14 lines Ensure "rules" is tab-completable for "queue show" Previously, tabbing at the end of "queue show" produced a list of available queues about which information could be shown, but did not include an alternative command, "rules", to access information about queue rules. The "rules" item should now be shown in the list of tab-completable items. (closes issue AST-958) Reported-by: John Bigelow ........ Merged revisions 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	cf2a598cab	Merged revisions 372391 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372391 | mjordan | 2012-09-05 19:56:47 -0500 (Wed, 05 Sep 2012) | 24 lines Allow configured numbers for FollowMe to be greater than 90 characters When parsing a 'number' defined in followme.conf, FollowMe previously parsed the number in the configuration file into a buffer with a length of 90 characters. This can artificially limit some parallel dial scenarios. This patch allows for numbers of any length to be defined in the configuration file. Note that Clod Patry originally wrote a patch to fix this problem and received a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256 characters. Instead, the patch being committed duplicates the string in the config file on the stack before parsing it for consumption by the application. (closes issue ASTERISK-16879) Reported by: Clod Patry Tested by: mjordan patches: followme_no_limit.diff uploaded by Clod Patry (license #5138) Slightly modified for this commit. ........ Merged revisions 372390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	6ee7c728ee	Merged revisions 372288 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372288 | mjordan | 2012-09-05 08:42:54 -0500 (Wed, 05 Sep 2012) | 27 lines Fix memory leaks in app_voicemail when using IMAP storage or realtime config This patch fixes two memory leaks: 1. When find_user is called with NULL as its first parameter, the voicemail user returned is allocated on the heap. The inboxcount2 function uses find_user in such a fashion when counting new messages, and fails to free the resulting voicemail user object. 2. When populate_defaults is called on a voicemail user, it wipes whatever flags have been set on the object by copying over the global flags object. If the VM_ALLOCED flag was ste on the voicemail user prior to doing so, that flag is removed. This leaks the voicemail user when free_user is later called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277) Patch slightly modified for this commit. Review: https://reviewboard.asterisk.org/r/2096 ........ Merged revisions 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1db640cdbc	Merged revisions 372090 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372090 | mmichelson | 2012-08-30 15:53:09 -0500 (Thu, 30 Aug 2012) | 17 lines Prevent crash on shutdown due to refcount error on queues container. When app_queue is unloaded, the queues container has its refcount decremented, potentially to 0. Then the taskprocessor responsible for handling device state changes is unreferenced. If the taskprocessor happens to be just about to run its task, then it will create and destroy an iterator on the queues container. This can cause the refcount on the queues container to increase to 1 and then back to 0. Going back to 0 a second time results in double frees. This failure was seen periodically in the testsuite when Asterisk would shut down. ........ Merged revisions 372089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	415082bc75	Merged revisions 372049 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372049 | mmichelson | 2012-08-30 13:33:37 -0500 (Thu, 30 Aug 2012) | 16 lines Help prevent ringing queue members from being rung when ringinuse set to no. Queue member status would not always get updated properly when the member was called, thus resulting in the member getting multiple calls. With this change, we update the member's status at the time of calling, and we also check to make sure the member is still available to take the call before placing an outbound call. (closes issue ASTERISK-16115) reported by nik600 Patches: app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409) ........ Merged revisions 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	33c5db3259	Merged revisions 371920 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371920 | jrose | 2012-08-29 15:58:21 -0500 (Wed, 29 Aug 2012) | 5 lines app_meetme: Adding test events for following activity in MeetMe. ........ Merged revisions 371919 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	57a546d181	Merged revisions 371861 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371861 | rmudgett | 2012-08-29 13:24:54 -0500 (Wed, 29 Aug 2012) | 15 lines Fix hangup cause passthrough regression. The v1.8 -r369258 change to fix the F and F(x) action logic introduced a regression in passing the hangup cause from the called channel to the caller channel. (closes issue ASTERISK-20287) Reported by: Konstantin Suvorov Patches: app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged revisions 371860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	30708a6a2c	Merged revisions 371591 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371591 | mmichelson | 2012-08-21 15:40:18 -0500 (Tue, 21 Aug 2012) | 22 lines Fix misuses of asprintf throughout the code. This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 ........ Merged revisions 371590 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	0083779480	Merged revisions 371313 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371313 | mmichelson | 2012-08-15 18:19:09 -0500 (Wed, 15 Aug 2012) | 20 lines Fix bug where final queue member would not be removed from memory. If a static queue had realtime members, then there could be a potential for those realtime members not to be properly deleted from memory. If the queue's members were loaded from realtime and then all the members were deleted from the backend, then the queue would still think these members existed. The reason was that there was a short- circuit in code such that if there were no members found in the backend, then the queue would not be updated to reflect this. Note that this only affected static queues with realtime members. Realtime queues with realtime members were unaffected by this issue. (closes issue ASTERISK-19793) reported by Marcus Haas ........ Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f7c4ca17f4	Merged revisions 371199,371203 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371199 | mmichelson | 2012-08-13 14:51:19 -0500 (Mon, 13 Aug 2012) | 5 lines Fix problem where incorrect pointer was checked for nullity. ........ Merged revisions 371198 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r371203 | kmoore | 2012-08-13 15:04:15 -0500 (Mon, 13 Aug 2012) | 13 lines Add test instrumentation This adds test instrumentation for loading and unloading of modules and for certain actions in MeetMe to be used in the testsuite or any other consumer of AMI events. These will only be generated when Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	36e8019901	Merged revisions 371142 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371142 | mmichelson | 2012-08-10 16:23:52 -0500 (Fri, 10 Aug 2012) | 18 lines Fix a couple of documentation problems in app_queue.c * The RemoveQueueMember app made mention of options that could be passed in, but no options are supported. I have removed the listing of options from the documentation. * The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value that could be set. (closes issue AST-949) reported by Steve Pitts (closes issue AST-954) reported by Steve Pitts ........ Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	60a5d5f510	Merged revisions 370986,370989 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370986 | kmoore | 2012-08-09 12:39:52 -0500 (Thu, 09 Aug 2012) | 11 lines Correct documentation for the MeetMe x flag The documentation for the x flag for MeetMe incorrectly described its function as closing down the conference when the last marked user left. It actually causes the users with that flag to leave the conference when the last marked user exits. The functionality of this flag is not changing. ........ Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370989 | may | 2012-08-09 13:05:34 -0500 (Thu, 09 Aug 2012) | 5 lines change opening h323 logfile with append mode instead of overwrite ........ Merged revisions 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	0e1ac09a2b	Merged revisions 370954 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370954 | elguero | 2012-08-08 17:42:05 -0500 (Wed, 08 Aug 2012) | 26 lines Fix Not Unreferencing A Spied Channel When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	86776cb4d5	Merged revisions 370643 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370643 | kmoore | 2012-07-31 14:57:09 -0500 (Tue, 31 Jul 2012) | 12 lines Clean up and ensure proper usage of alloca() This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ec96eda533	Merged revisions 370547 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r370547 | jrose | 2012-07-30 09:50:34 -0500 (Mon, 30 Jul 2012) | 5 lines app_meetme: Change app_meetme support level to extended from deprecated (closes issue ASTERISK-20134) Reported by: Leif Madsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	8d9bfd3342	Merged revisions 370184 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370184 | kpfleming | 2012-07-18 12:13:07 -0500 (Wed, 18 Jul 2012) | 10 lines Ensure that all ast_datastore_info structures are 'const'. While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago