asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	fb74294b92	Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default. With ICE support enabled in chan_sip and a large number of interfaces on the system it was possible for the produced SDP to be truncated due to some fixed size buffers. These buffers have now been changed so they will dynamically grow as needed. ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is no longer enabled by default there. (closes issue ASTERISK-20643) Reported by: coopvr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	0e7a64e55e	Fix a "set but not used" warning on newer gccs. Turns out the "helpful" setting of ms and res in this macro is completely useless after the timeout antipattern fix. If you're a new guy looking to write code, don't write a macro like this one. ........ Merged revisions 376087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376088 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c7e2bf3187	chan_dahdi/SS7: Made reject incoming call for an in-alarm or blocked channel. If a SS7 call comes in requesting a CIC that is in-alarm, the call is accepted and connects if the extension exists in the dialplan. The call does not have any audio. * Made release the call immediately with circuit congestion cause. (closes issue ASTERISK-20204) Reported by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376059 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1bba4b00b2	Add MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc(). It will return a memory block filled with 0x55. A nonzero value. * Makes free() fill the released memory block and boundary fence's with 0xdeaddead. Any pointer use after free is going to have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is usually an invalid memory address so a crash is expected. * Puts the freed memory block into a circular array so it is not reused immediately. * When the circular array rotates out a memory block to the heap it checks that the memory has not been altered from 0xdeaddead. * Made the astmm_log message wording better. * Made crash if the DO_CRASH menuselect option is enabled and something is found. * Fixed a potential alignment issue on 64 bit systems. struct ast_region.data[] should now be aligned correctly for all platforms. * Extracted region_check_fences() from __ast_free_region() and handle_memory_show(). * Updated handle_memory_show() CLI usage help. Review: https://reviewboard.asterisk.org/r/2182/ ........ Merged revisions 376029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376030 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a65fbf8012	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 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	647810f1d0	Fix stuck DTMF when bridge is broken. When a bridge is broken by an AMI Redirect action or the ChannelRedirect application, an in progress DTMF digit could be stuck sending forever. * Made simulate a DTMF end event when a bridge is broken and a DTMF digit was in progress. (closes issue ASTERISK-20492) Reported by: Jeremiah Gowdy Patches: bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy Modified to jira_asterisk_20492_v1.8.patch jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2169/ ........ Merged revisions 375964 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375965 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	5f28931a1f	Fix a bug where our Motif ICE candidates were not quite proper, and make us more forgiving. An issue was reported on the mailing list where calling would result in an "Incomplete ICE-UDP candidate received on session" error message. This is the result of the ICE-UDP candidate code not placing a "network" attribute within the candidates. This is now done. To increase compatibility though I have removed the requirement for the "network" attribute to exist within ICE-UDP candidates that are received since we don't actually require the value. Reported on the mailing list by Jean-Denis Girard. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f0cd27e027	Refactor ast_timer_ack to return an error and handle the error in timer users Currently, if an acknowledgement of a timer fails Asterisk will not realize that a serious error occurred and will continue attempting to use the timer's file descriptor. This can lead to situations where errors stream to the CLI/log file. This consumes significant resources, masks the actual problem that occurred (whatever caused the timer to fail in the first place), and can leave channels in odd states. This patch propagates the errors in the timing resource modules up through the timer core, and makes users of these timers handle acknowledgement failures. It also adds some defensive coding around the use of timers to prevent using bad file descriptors in off nominal code paths. Note that the patch created by the issue reporter was modified slightly for this commit and backported to 1.8, as it was originally written for Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches: jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 375893 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375894 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	463cbc55ac	Add safety NULL pointer check in module user references. Made __ast_module_user_remove() check for NULL pointers. ........ Merged revision 375860 from C.3 ........ Merged revisions 375862 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375863 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	1d054dbbba	chan_sip: Document a change to user-field encoding introduced with r303509 The change in question was added to improve compliance with RFC3261, but at the time of commit, it wasn't adequately documented in the UPGRADE notes. (closes issue ASTERISK-20561) Reported by: Deniz Review: https://reviewboard.asterisk.org/r/2177/ ........ Merged revisions 375846 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	769d9aa259	Don't attempt to purge sessions when no sessions exist Manager's tcp/tls objects have a periodic function that purge old manager sessions periodically. During shutdown, the underlying container holding those sessions can be disposed of and set to NULL before the tcp/tls periodic function is stopped. If the periodic function fires, it will attempt to iterate over a NULL container. This patch checks for whether or not the sessions container exists before attempting to purge sessions out of it. If the sessions container is NULL, we simply return. Note that this error was also caught by the Asterisk Test Suite. ........ Merged revisions 375800 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375801 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e4afa2c0e2	Only deref a reserved gateway session if we actually reserved one Its perfectly acceptable to have a gateway session unreserved when we go to first allocate one. Unreffing the reserved gateway session - when its NULL - will result in an assertion error. This problem was caught by the Asterisk Test Suite (once we had enough of the debugging flags enabled) ........ Merged revisions 375797 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d38c42aab8	Properly clean up manager resources on exit This patch does two things: 1) It properly unregisters the manager CLI commands 2) It cleans up AMI users on exit. Prior to this patch, the AMI users were not being disposed of properly, resulting in a memory leak. (closes issue ASTERISK-20646) Reported by: Corey Farrell patches: manager_shutdown.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 375793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375794 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	260da253be	Properly finalize prepared SQLite3 statements to prevent memory leak The AstDB uses prepared SQLite3 statements to retrieve data from the SQLite3 database. These statements should be finalized during Asterisk shutdown so that the SQLite3 database can be properly closed. Failure to finalize the statements results in a memory leak and a failure when closing the database. This patch fixes those issues by ensuring that all prepared statements are properly finalized at shutdown. (closes issue ASTERISK-20647) Reported by: Corey Farrell patches: astdb-sqlite3_close.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 375761 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	4483067638	Blocked revisions 375759 ........ Fix memory leak when unloading XML documentation This patch is a modified version of a patch originally committed for the Asterisk 11 branch in r375756. A portion of that patch, that fixed the memory leak during unloading XML documentation, applies to branches 1.8 and 10 as well. The patch for this issue was modified for these two branches. (issue ASTERISK-20648) Reported by: Corey Farrell Tested by: mjordan patches: xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 375758 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	717068e362	Fix memory leaks in XML documentation This patch fixes two memory leaks: 1) When building XML documentation items, the 'name' attribute was extracted from XML elements but not properly freed after being copied into the item being built. 2) When unloading XML documentation, the doctree container objects were not properly freed. This patch corrects these memory leaks. Note that this patch was modified slightly for this commmit, as the case where the 'name' attribute doesn't exist also wasn't handled in the item construction. This patch also checks for that attribute not existing. (closes issue ASTERISK-20648) Reported by: Corey Farrell Tested by: mjordan patches: xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	69777a9376	Prevent multiple CDR batches from conflicting when scheduling the CDR write The Asterisk Test Suite caught an error condition where a scheduled CDR batch write can be deleted twice if two channels attempt to post their CDRs at the same time. The batch CDR mutex is locked while the CDRs are appended to the current batch list; however, it is unlocked prior to actually scheduling the CDR write. As such, two threads can attempt to remove the currently scheduled batch write at the same time, resulting in an assertion error. This patch extends the time that the mutex is locked to encompass actually scheduling the write. This prevents two threads from unscheduling the currently scheduled write at the same time. ........ Merged revisions 375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375728 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	794eb78090	Doxygen Updates Replace links to missing text files removed in the 1.6.x series with links to the wiki. Doxygen can handle URLs fine, don't atempt to quote them. Also update the wiki link in the Readme to get everyone on the same page. (issue ASTERISK-20259) ........ Merged revisions 375698 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375699 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	93d85a0087	Things don't need to be that const. ........ Merged revisions 375658 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375659 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	f4fb271601	Fix for chan_skinny leaving RTP ports open Skinny wasn't closing RTP sockets. This patch includes ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes the problem. Also add destroy for VRTP (which I believe is unused, but exists). Review: https://reviewboard.asterisk.org/r/2176/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	35e96f995e	Multiple revisions 375519-375524 ........ r375519 \| rmudgett \| 2012-10-30 16:06:15 -0500 (Tue, 30 Oct 2012) \| 11 lines chan_misdn: Timer primitives must be handled first. The frm->addr is a different "address space" than the stack/instance address of other Lx primitives. The test for B channel instance address could fail. Patches: patch01_timers.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375520 \| rmudgett \| 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) \| 10 lines chan_misdn: Free memory in error paths and other memory leaks. The one line commented with BUG is not easily fixable because there is no de-init function one can call. Patches: patch02_memory.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375521 \| rmudgett \| 2012-10-30 16:38:41 -0500 (Tue, 30 Oct 2012) \| 14 lines chan_misdn: ISDN NT L2 de-establish/establish * An NT-PTMP cannot de/establish L2 since it doesn't know the TEIs. * On NT-PTP L2 is started when L1 is finally active in handle_l1. * L2 deactivation logging cleanup. * L2 aggregate link status is unknown for NT-PTMP, show as "UNKN". * Removed unused functions and code for L2 handling. Patches: patch03_L2estab.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 \| rmudgett \| 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) \| 22 lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH prim via lower_id layer (3 or 1) simply does not work. For TE (3) it returns an error (len=-6) which is not evaluated by handle_l1(), so the L1 layer status ends up wrong. Instead PH must be sent via L4, only then does it reach L1 without an error message. And NT PH prims only reach L1 when they are sent to layer 2 id. --> use upper_id to send PH primitives. * Check for errors in PH_(DE)ACTIVATE \| CONFIRM. * Debug messages are improved. * The lower_id is now not used for anything, except: Why is lower_id layer deleted when it wasn't created? I removed this code since it looks very wrong. Patches: patch04_l1activation.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375523 \| rmudgett \| 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) \| 31 lines chan_misdn: Fix loss of B channels if L1 is down. If you make 2 calls out an NT PTMP port which is not connected to any phone, the B channel associated with that call becomes unusable until Asterisk is restarted. The problem is the EVENT_SETUP is queued when L1 is not up in misdn_lib_send_event(). If L1 cannot be activated the event won't be dequeued. It gets even worse when the call is hung up. The queued EVENT_SETUP will be overwritten by an EVENT_DISCONNECT. The reserved B channel then will never be freed. If later someone connects a phone to the port, L1 will eventually activate and the queued EVENT_DISCONNECT is sent down the stack. However, it is ignored because it is the wrong call state. The real fix would be that activation and queueing for a new SETUP is done by the NT stack. But since it doesn't, the workaround must be removed because it doesn't always work. Fix: The event is no longer queued but immediately sent to the stack. If L1 cannot be activated, the L3 state machine that was started by the EVENT_SETUP will do its work, i.e. a timeout will release the B channel properly. The SETUP possibly cannot be sent the first time but is resent by T303 in case L1 could be activated. Patches: patch05_bchan-loss.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 \| rmudgett \| 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) \| 13 lines chan_misdn: Remove some calls to exit(). Try proper cleanup when something goes wrong in misdn_lib_init(). Especially do not call exit()! * Fix memory leak because stack_destroy() does not free the stack struct. Patches: patch06_cleanup-init.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ Merged revisions 375519-375524 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 375625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375626 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	2fce31c09a	Fix Wrong Result In Debug Message For SDP Origin Processing While looking at some debug logs, I noticed that it was being reported that the SDP origin line was unsupported or failed. Upon looking into this on my local machine, I found that I too was getting this debug message yet everything seemed to be getting processed properly. What was discovered is, that, the variable to determine what is displayed in the debug message for the SDP line that was processed, was not being set for the origin line when the result was successful. This patch fixes this and was tested on local machine. ........ Merged revisions 375594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375601 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	509f348639	chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry A regression was introduced in chan_sip by changes to sip reload introduced by r349097. That patch moved peer purging from the beginning of the reload to after the general configuration was finished. This patch fixes that by undoing the repositioning of the original peer purging code and using a similar function after performing general configuration that purges only autocreated peers that were created when persist mode isn't enabled. (closes issue ASTERISK-20611) Reported by: Alisher Review: https://reviewboard.asterisk.org/r/2171/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	05be2e8bee	Fix an issue with res_http_websocket where the chan_sip WebSocket handler could not be registered. On some systems the optional API support uses the GCC compiler attribute "weakref" to provide its functionality. This code changes the function names and prefixes "__" to the front. The res_http_websocket exports file did not take this into account, thereby not allowing those functions to be global and ultimately found. (closes issue ASTERISK-20631) Reported by: danjenkins git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7cedfef4fd	Properly extract the Body information of an EWS calendar item Unlike all other calendar modules, res_calendar_ews fails to extract the Body information for a calendar item. This is due, in part, to a quirk in the schema in the XML - not only does a CalendarItem contain a Body element, but the CalendarItem exists as a descendant of a different Body element. The neon parser was erroneously skipping all Body elements. This patch fixes that by bypassing Body elements that are not a child of CalendarItem, and parsing the Body element out if it is a child. Note that the original patch by Terry Wilson only needed slight modifications to make it properly pull the Body information out; as such, while I've linked to the patch that I uploaded for Dmitry, I've attributed the patch to Terry. (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff uploaded by Terry Wilson (license 6283) ........ Merged revisions 375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375531 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1fc6b54a5d	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. ........ Merged revisions 375496 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375506 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	d7d9a99bd6	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 ........ Merged revisions 375485 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	e1da5dbd57	confbridge: Fix a bug which made conferences not record with AMI/CLI commands When confbridge was changed to handle conference status with a state machine in r374658. The function responsible for starting recording for a conference was refactored with the function actually responsible for launching the recording thread being split into a function with another name. The old function name was still used for manually started recordings through AMI or CLI. This patch fixes that by switching which function is used to start recording the conference. (closes issue ASTERISK-20601) Reported by: Vilius Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 375470 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3ac0ced049	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 ........ Merged revisions 375451 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	d51cc27812	Prevent resetting of NATted realtime peer address on reload. If a "sip reload" is issued for a SIP peer, then his IP address will be cleared, thus resulting in forgetting the public IP address. Asterisk will then attempt to route SIP traffic to the private IP address. The fix here is to make "sip reload" ignore realtime peers when "host = dynamic" is spotted. Realtime peers can now only have their IP address reset if they have gone from being not dynamic to being dynamic. (closes issue ASTERISK-18203) reported by daren ferreira (closes issue ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff uploaded by JoshE (license #6075) ........ Merged revisions 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375417 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	48f0d85bd1	Fix the Park 'r' option when a channel parks itself. When a channel uses the Park appliation to park itself with the 'r' option, the channel hears music-on-hold instead of the requested ringing. * Added a missing check for the 'r' option when a channel parks itself. (closes issue ASTERISK-19382) Reported by: James Stocks Patches by: dsessions Review: https://reviewboard.asterisk.org/r/2148/ ........ Merged revisions 375388 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375389 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	421fbee8d8	chan_dahdi: Fix segfault dereferencing a NULL tech_pvt. The tech support customer was using the AMI Redirect action shortly after a call was placed. While the channel tried to do an ast_read(), the masquerade resulting from the channel redirect took place. The masquerade in the middle of the ast_read() resulted in the segfault. (closes issue AST-1025) Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 375361 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375362 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	8deafe9dd0	ast_tls_cert script: Better response for various exit conditions to openssl (closes issue ASTERISK-20260) Reported by: Daniel O'Connor Patches: ast_tls_cert-update.diff uploaded by Daniel O'Connor (license 6419) ........ Merged revisions 375325 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375326 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	7ad1f16efd	core: Fix a memory leak in app.c from an early return ast_app_group_match_get_count allocates memory with the regcomp function and we previously forgot to free it when bailing out due to a regex compilation failure against category. (closes issue AST-1018) Reported by: Guenther Kelleter Patches: regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372) ........ Merged revisions 375299 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375300 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	bcd4e3e8ac	GSM: Fix encoding problems with GSM (closes issue ASTERISK-20457) Reported by: Richard Miller Patches: code.patch uploaded by Richard Miller (license 5685) ........ Merged revisions 375272 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375273 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c5485ddb4f	app_queue: add upgrade notes for 375216 Adds UPGRADE notes describing behavioral changes to rrmemory strategy caused by 375216 (issue AST-989) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	9aab271f60	Blocked revisions 375245 ........ app_queue: add upgrade notes for 375216 Adds notes describing behavioral changes to rrmemory strategy caused by 375216 (issue AST-989) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	8d0143f4a6	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 ........ Merged revisions 375217 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7c69310497	build_tools: Allow Asterisk to report git SHAs in version string. Make git more attractive for managing work-in-progress. Especially convenient when a potential patch set needs to be tested on multiple platforms since one can use git to keep all the test environments in sync independent of a subversion server. Now the Asterisk version will show the exact git SHA5 that was used when building (still appended by "M" if there are local modifications) from a git clone of the Asterisk repository so the developer can more easily know what is actually under test. You will now get this: $ asterisk -V Asterisk GIT-1698298 Instead of this: $ asterisk -V Asterisk UNKNOWN__and_probably_unsupported This has zero impact for those not using git with the exception of an extra test in the configure script to gather git's path. This is necessary to prevent "sudo make install" from failing since git may not be in the path in make's shell environment. (closes issue ASTERISK-20483) Reported by: Shaun Ruffell Patches: 0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch (license #5417) patch uploaded by Shaun Ruffell Modified ........ Merged revisions 375189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375190 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	09d951514d	Ensure Asterisk fails TCP/TLS SIP calls when certificate checking fails When placing a call to a TCP/TLS SIP endpoint whose certificate is not signed by a configured CA certificate, Asterisk would issue a warning and continue to process the call as if there was not an issue with the certificate. Asterisk now properly fails the call if the certificate fails verification or if the certificate does not exist when certificate checking is enabled (the default behavior). (closes issue ASTERISK-20559) Reported by: kmoore Review: https://reviewboard.asterisk.org/r/2163/ ........ Merged revisions 375146 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375147 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	0ee22cfd14	Fixes to the fd-oriented SIP TCP reads. Don't crash on large user input. Allow SIP headers without space. Optimize code a bit. Review: https://reviewboard.asterisk.org/r/2162 ........ Merged revisions 375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375112 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	5fc8671fb7	Update sip_request_call SIP dial string documentation. This was missed when merging review r1859. ........ Merged revisions 375074 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375078 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	a318db28e3	Remove a log message that was left in accidentally from call-id logging development. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	94c0fa9098	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 ........ Merged revisions 375026 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	5b1a89e1b1	Fix underscreen buttons warnings apeared while transfer process	13 years ago
Tzafrir Cohen	cabdb471fb	Update config.guess and config.sub: 2012-10-10 Update config.guess and config.sub to revision fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM 64bit). config.guess:timestamp='2012-09-25' config.sub:timestamp='2012-10-10' ........ Merged revisions 374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374991 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	09dac916ba	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/11@374932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	ccf01fbfdc	Do not use a FILE handle when doing SIP TCP reads. This is used to solve an issue where a poll on a file descriptor does not necessarily correspond to the readiness of a FILE handle to be read. This change makes it so that for TCP connections, we do a recv() on the file descriptor instead. Because TCP does not guarantee that an entire message or even just one single message will arrive during a read, a loop has been introduced to ensure that we only attempt to handle a single message at a time. The tcptls_session_instance structure has also had an overflow buffer added to it so that if more than one TCP message arrives in one go, there is a place to throw the excess. Huge thanks goes out to Walter Doekes for doing extensive review on this change and finding edge cases where code could fail. (closes issue ASTERISK-20212) reported by Phil Ciccone Review: https://reviewboard.asterisk.org/r/2123 ........ Merged revisions 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374906 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	963f94e99f	Fix a bug where audio on Google Voice would not work due to ignoring candidates. Instead of ignoring parts of the message that are not known just ignore the ones we know may be present and that would cause a problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	59d02d37de	Remove code that should not have gotten in. (issue ASTERISK-20554) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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