You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.

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
..
Makefile	…
func_aes.c	…
func_audiohookinherit.c	Merged revisions 373466,373468 via svnmerge from	14 years ago
func_base64.c	…
func_blacklist.c	…
func_callcompletion.c	…
func_callerid.c	…
func_cdr.c	…
func_channel.c	Merged revisions 370643 via svnmerge from	14 years ago
func_config.c	…
func_curl.c	…
func_cut.c	Merged revisions 370643 via svnmerge from	14 years ago
func_db.c	…
func_devstate.c	Multiple revisions 377838,377842,377848,377882,377923,377947,377992,378037,378089,378093,378120,378218,378286,378320	13 years ago
func_dialgroup.c	…
func_dialplan.c	…
func_enum.c	…
func_env.c	…
func_extstate.c	…
func_frame_trace.c	…
func_global.c	Merged revisions 370643 via svnmerge from	14 years ago
func_groupcount.c	…
func_iconv.c	…
func_jitterbuffer.c	Merged revisions 375894 via svnmerge from	13 years ago
func_lock.c	…
func_logic.c	Merged revisions 370643 via svnmerge from	14 years ago
func_math.c	Merged revisions 372621,372625,372629 via svnmerge from	14 years ago
func_md5.c	…
func_module.c	…
func_odbc.c	Merged revisions 371591 via svnmerge from	14 years ago
func_pitchshift.c	…
func_presencestate.c	…
func_rand.c	…
func_realtime.c	…
func_sha1.c	…
func_shell.c	Merged revisions 370384 via svnmerge from	14 years ago
func_speex.c	…
func_sprintf.c	…
func_srv.c	…
func_strings.c	Merged revisions 370643 via svnmerge from	14 years ago
func_sysinfo.c	…
func_timeout.c	…
func_uri.c	…
func_version.c	…
func_vmcount.c	…
func_volume.c	Merged revisions 376916,376920 via svnmerge from	13 years ago