asterisk

Commit Graph

Author	SHA1	Message	Date
David M. Lee	c5aee60db5	Fix XML encoding of 'identity display' in NOTIFY messages. XML encoding in chan_sip is accomplished by naively building the XML directly from strings. While this usually works, it fails to take into account escaping the reserved characters in XML. This patch adds an 'ast_xml_escape' function, which works similarly to 'ast_uri_encode'. This is used to properly escape the local_display attribute in XML formatted NOTIFY messages. Several things to note: * The Right Thing(TM) to do would probably be to replace the ast_build_string stuff with building an ast_xml_doc. That's a much bigger change, and out of scope for the original ticket, so I refrained myself. * It is with great sadness that I wrote my own ast_xml_escape function. There's one in libxml2, but it's knee-deep in libxml2-ness, and not easily used to one-off escape a string. * I only escaped the string we know is causing problems (local_display). At least some of the other strings are URI-encoded, which should be XML safe. Rather than figuring out what's safe and escaping what's not, it would be much cleaner to simply build an ast_xml_doc for the messages and let the XML library do the XML escaping. Like I said, that's out of scope. (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by: Guenther Kelleter Review: http://reviewboard.digium.internal/r/365/ ........ Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	f6404e6e15	Fix end condition in ast_rtp_lookup_mime_multiple2. The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag in the debug output. (closes issue ASTERISK-20772) Reported by: Xavier Hienne git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	d1321a2a77	Replace errant tabs with spaces in causes.h. (closes issue ASTERISK-20826) Reported by: snuffy Patches: notabs.dif uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7ba4fe48ca	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	6a94f73341	res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create Under some circumstances, libsrtp's srtp_create function deallocates memory that it wasn't initially responsible for allocating. Because we weren't initially aware of this behavior, this memory was still used in spite of being unallocated during the course of the srtp_unprotect function. A while back I made a patch which would set this value to NULL, but that exposed a possible condition where we would then try to check a member of the struct which would cause a segfault. In order to address these problems, ast_srtp_unprotect will now set an error value when it ends without a valid SRTP session which will result in the caller of srtp_unprotect observing this error and hanging up the relevant channel instead of trying to keep using the invalid session address. (closes issue ASTERISK-20499) Reported by: Tootai Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	d06773e931	Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field On a multihomed server when sending a NOTIFY message, we were not figuring out which network should be used to contact the peer. This patch fixes the problem by calling ast_sip_ouraddrfor() and then build_via() so that our NOTIFY message contains the correct IP address. Also, a debug message is being added to help follow the call-id changes that occur. This was helpful for confirming that the IP address was set properly since the call-id contains the IP address. It also will be helpful for troubleshooting purposes when following a call in the debug logs. (closes issue ASTERISK-20805) Reported by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches: asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2255/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f5012d8d0d	Don't pass STUN packets through the SRTP unprotect function. (closes issue AST-1036) Reported by: jbigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	afd0961a3c	Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present When the "h" extension is present within the context of the queue, all calls are being reported COMPLETECALLER even when the agent is hanging up the call. This patch checks to see if the agent hung-up or not instead of only relying on checking if the queue (caller) channel hung-up or not. It would appear that having the h extension in the mix, the pbx goes to the h extension, "hanging-up" the queue channel and triggering the reporting of COMPLETECALLER. (closes issue ASTERISK-20743) Reported by: call Tested by: call, Michael L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	4477902eb1	chan_agent: Fix wrapup time wait response. * Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup time expires. agent_cont_sleep() had tried but returned the wrong value to stop waiting. * Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void pointer for better type safety. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	540cb4eedb	chan_agent: Misc code cleanup. * Fix off-nominal path resource cleanup in agent_request(). * Create agent_pvt_destroy() to eliminate inlined versions in many places. * Pull invariant code out of loop in add_agent(). * Remove redundant module user references in login_exec(). * Remove unused struct agent_pvt logincallerid[] member. * Remove some redundant code in agent_request(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	22c5eea34b	Add missing test event This test event was missing from channel.c causing the dial_LS_options test to fail intermittently because of a race condition where most code paths emitted the test event but this one did not. The dial_LS_options test should stop bouncing now. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ba65f21e73	chan_agent: Fix agent_indicate() locking. Avoid deadlock potential with local channels and simplify the locking. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	eb0d0eb4d1	Prevent crashes from occurring when reading from data sources with large values When reading configuration data from an Asterisk .conf file or when pulling data from an Asterisk RealTime backend, Asterisk was copying the data on the stack for manipulation. Unfortunately, it is possible to read configuration data or realtime data from some data source that provides a large blob of characters. This could potentially cause a crash via a stack overflow. This patch prevents large sets of data from being read from an ARA backend or from an Asterisk conf file. (issue ASTERISK-20658) Reported by: wdoekes Tested by: wdoekes, mmichelson patches: * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674) * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1457bf7bf0	Fix AMI redirect action with two channels failing to redirect both channels. The AMI redirect action can fail to redirect two channels that are bridged together. There is a race between the AMI thread redirecting the two channels and the bridge thread noticing that a channel is hungup from the redirects. * Made the bridge wait for both channels to be redirected before exiting. * Made the AMI redirect check that all required headers are present before proceeding with the redirection. * Made the AMI redirect require that any supplied ExtraChannel exist before proceeding. Previously the code fell back to a single channel redirect operation. (closes issue ASTERISK-18975) Reported by: Ben Klang (closes issue ASTERISK-19948) Reported by: Brent Dalgleish Patches: jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5996dd6a23	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a73c5b80ff	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	367bed142a	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fe02f96c77	Give the causes[] a struct name. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	c9e7b549b1	Adjust RTP instance's available_formats callback to return the correct type. The RTP engine public function that gets the available formats expects a format_t to be returned; however when calling into an RTP instance's callback to get the available formats, the callback returned an int. This never was noticed in Asterisk because the two RTP engines included do not provide an available_formats callback. This introduces an API change, and the proposal for this change was brought up on the Asterisk developers mailing list [1]. There was no public objection to this change, so it is now being put in. (closes AST-1054) reported by Doug Bailey [1] http://lists.digium.com/pipermail/asterisk-dev/2012-December/058058.html git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	16983cf3b5	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	091b93b98e	Fix potential double free when unloading a module. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7700095727	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c75a2efd1b	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	663f68273f	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	724c3c620e	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c7ad36cf26	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e5c5d7abe8	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	20aa0aa476	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	440f4e1dcc	Cleanup taskprocessor on exit. * Cleanup CLI commands on exit. (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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	014ba80cdc	Cleanup pbx on exit. * Cleanup CLI commands on exit. * Unreference hints and statecbs containers on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	11a0764011	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	2f14338cfb	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f1821b41fd	Cleanup frame on exit. * Cleanup CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: frame-cleanup-1_8-only.patch (license #5909) patch uploaded by Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	65c767d630	Cleanup event on exit. * Cleanup CLI commands on exit. (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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	5dc35f3a29	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	681a4d4269	Cleanup astdb on exit. * Cleanup astdb thread and CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: db-cleanup-1_8-only.patch (license #5909) patch uploaded by Corey Farrell Modified git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	7412e73962	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	bc11470697	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	32d4c02de2	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	9fc52cdaa3	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tilghman Lesher	dc63fd67d3	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tilghman Lesher	6769e186b9	Remove some dead code and additionally handle a case that wasn't handled. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ad7af7db6b	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	46581779aa	MALLOC_DEBUG: Only wait if we want atexit allocation dumps. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	40dfa4b748	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	711f4c99f4	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9cd2528b7e	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	83c6a754e6	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	570e415591	Cleanup config cache on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: config-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	2e036745a6	Cleanup CLI resources on exit and CLI command registration errors. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch uploaded by Corey Farrell cli-leaks-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

1 2 3 4 5 ...

22334 Commits (c5aee60db58c688c80c91313fa69ba26880c8c4c) All Branches Search

22334 Commits (c5aee60db58c688c80c91313fa69ba26880c8c4c)

All Branches