asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	e30d2c1d45	Locking issue in action_bridge and bridge_exec action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it. issue# 14296 Review: http://reviewboard.digium.com/r/167/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	7949345352	Fixed error where a check for an zero length, terminated string was needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	fa3dec169b	Need to take into account the \0 terminator of the old string to determine the amount available. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	0fe1df19df	This patch fixes merge_contexts_and_delete so it does not deadlock when hints are present.	17 years ago
Russell Bryant	a844cfa904	Fix a number of incorrect uses of strncpy(). The big problem here is that the 3rd argument provided in these uses of strncpy() did not reserve a byte for the null terminator, leaving the potential for writing one byte past the end of the buffer. Aside from this, there were coding guidelines violations with regards to spacing, as well as hard coded lengths being used instead of sizeof(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	3dcdaa5d05	suppress smoothers for Siren codecs as well as Speex and G.723.1 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	f40edf2793	Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 \| jpeeler \| 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) \| 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c461d29b0b	Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	044cf691fe	Significantly improve scheduler performance under high load. This patch changes the scheduler to use a max-heap to store pending scheduler entries instead of a fully sorted doubly linked list. When the number of entries in the scheduler gets large, this will perform much better. For much more detailed information on this change, see the review request. Review: http://reviewboard.digium.com/r/160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	bb03ef8d47	Add an implementation of the heap data structure. A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	184872fdfd	Fix a race condition that caused device states to become incorrect for hints. The problem here is that the hint processing code was subscribed to the wrong event type. So, it started processing state for a hint too soon, before the device state cache had been updated. Also, fix a similar bug in app_queue, as it was also subscribed to the wrong event type. (closes issue #14461) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	fa26904074	If there are no realtime engines, there's no reason to check for realtime families git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	76b100354d	Might want to update the buffer pointer after a realloc (or we crash) (closes issue #14485) Reported by: davevg git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	0381d94d14	Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 \| kpfleming \| 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) \| 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 \| kpfleming \| 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) \| 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e32921415f	Assist proper thread synchronization when stopping the logger thread. I was finding that on my dev box, occasionally attempting to "stop now" in trunk would cause Asterisk to hang. I traced this to the fact that the logger thread was waiting on a condition which had already been signalled. The logger thread also need to be sure to check the value of the close_logger_thread variable. The close_logger_thread variable is only checked when the list of logmessages is empty. This allows for the logger thread to print and free any pending messages before exiting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	96326f5aa1	Make the causes array static, and remove the type name as it is not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8c75380f52	Make ast_sched_report() and ast_sched_dump() thread safe. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ca9d3b8ac9	Fix a number of problems with ast_sched_report(). 1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	6a0773602a	add missing </para> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2a53f2ec98	Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip must decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9e0f5d2afe	Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 \| tilghman \| 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) \| 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	a46d290802	Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 \| jpeeler \| 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) \| 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	66e88633a5	Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 \| jpeeler \| 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) \| 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	12f02a8c11	Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 \| russell \| 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) \| 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3a9d79f056	Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	47ebea6a8d	Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	8435535300	Tell the device state core a change happened when a channel is freed but not a specific state. We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e909bfcea9	Fix an fd leak that would occur in HTTP AMI sessions The explanation behind this fix is a bit complicated, and I've already typed it up in the code as a huge comment inside of manager.c, so I'll give the abridged version here. We needed a way to separate action-specific data from session-specific data. Unfortunately, the only way to maintain API compatibility and to not have to change every single manager action was to rename the current mansession structure and wrap it inside a new mansession structure which actually contains action- specific data. (closes issue #14364) Reported by: awk Patches: 14364_better.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/148/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	23939e54f3	improve slinfactory API to remove implicit sample rate and require explicit sample rate selection by creator of the slinfactory git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	b7dc074376	Merged revisions 174583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174583 \| mnicholson \| 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) \| 18 lines Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b2484d7db9	Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached. (closes issue #14414) Reported by: bluecrow76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0cbada4f96	Add a common implementation of a scheduler context with a dedicated thread. This commit expands the Asterisk scheduler API to include a common implementation of a scheduler context being processed by a dedicated thread. chan_iax2 has been updated to use this new code. Also, as a result, this resolves some race conditions related to the previous chan_iax2 scheduler handling. Related to rev 171452 which resolved the same issues in 1.4. Code from team/russell/sched_thread2 Review: http://reviewboard.digium.com/r/129/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1d46bb0d0f	Resolve a memory leak that would occur on an invalid channel given to Action: Status git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	39ec5d1576	Merged revisions 173211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 \| jpeeler \| 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) \| 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f2fd6528a2	When using a socket as a FILE *, the stdio functions will sometimes try to do an fseek() on the stream, which is an invalid operation for a socket. Turning off buffering explicitly lets the stdio functions know they cannot do this, thus avoiding a potential error. (closes issue #14400) Reported by: fnordian Patches: tcptls.patch uploaded by fnordian (license 110) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	bd244342e2	Fix a problem where file playback would cause fds to remain open forever The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f90021fdd0	Ensure that commas placed in the middle of extension character classes do not interfere with correct parsing of the extension. Also, if an unterminated character class DOES make its way into the pbx core (through some other method), ensure that it does not crash Asterisk. (closes issue #14362) Reported by: Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e179e613f7	1. Make OS X compile cleanly with app_stack. 2. Use curl to download sound files, as curl is installed natively on OS X, whereas wget and fetch are not. (closes issue #14332) Reported by: oej Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	34be09bf5c	Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 \| twilson \| 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) \| 2 lines Fix a feature inheritance bug I added after code review ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c668cbfbfc	Reverting commit number 173028 as there are some potential issues git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7db67f9ca7	Add a CLI command to log out a manager user (closes issue #13877) Reported by: eliel Patches: cli_manager_logout.patch.txt uploaded by eliel (license 64) Tested by: eliel, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	53d9b77898	This reverts the changes I made for 11583; will	17 years ago
Steve Murphy	c61e8a7865	This change allows the disconnect feature (as in "one-touch" in features.c) to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	8d782f96b8	Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 \| twilson \| 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) \| 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	80f91f6ab7	Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 \| tilghman \| 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) \| 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	3b08de23df	Remove tabs from comment git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	b79a12e929	- Make sure we set setvar= variables on outbound calls too, not only inbound calls. - Also, change a function in app.c to return a userful value instead of always returning 0. Patch by fnordian, changed by Corydon76 and myself. This does not close the bug report, as fnordian had an additional change we're still discussing. (related to issue #14059) Reported by: fnordian Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) Tested by: fnordian, Corydon76, oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	268ac221a2	Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 \| murf \| 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) \| 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	fc7455fa44	Merged revisions 171621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171621 \| mmichelson \| 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) \| 18 lines Prevent a crash from occurring when a jitter buffer interpolated frame is removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

1 2 3 4 5 ...

2146 Commits (e30d2c1d450a6d09ec5cc0f2759e8b9f256eb950)