asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	8842f76a7f	Fix local channel chains optimizing themselves out of a call. * Made chan_local.c:check_bridge() check the return value of ast_channel_masquerade(). In long chains of local channels, the masquerade occasionally fails to get setup because there is another masquerade already setup on an adjacent local channel in the chain. * Made the outgoing local channel (the ;2 channel) flush one voice or video frame per optimization attempt. * Made sure that the outgoing local channel also does not have any frames in its queue before the masquerade. * Made do the masquerade immediately to minimize the chance that the outgoing channel queue does not get any new frames added and thus unconditionally flushed. * Made block indication -1 (Stop tones) event when the local channel is going to optimize itself out. When the call is answered, a chain of local channels pass down a -1 indication for each bridge. This blizzard of -1 events really slows down the optimization process. (closes issue ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec Davis Review: https://reviewboard.asterisk.org/r/1894/ ........ Merged revisions 365313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365320 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	404b890f49	Fix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report. These three all are in RTP code that attempts to print the number of sequence number cycles in an RTCP RR report. The code was masking out the upper 16 bits and then shifting the number right by 16 bits. This led to an all zero result in all cases. The fix is to do the shift without the bit masking. (issue ASTERISK-19649) ........ Merged revisions 365298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365299 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	fdf3cd0e2e	Update security events unit tests The security events framework API was changed in Asterisk 10 but the unit tests were not updated at the same time. This patch does the following: * Adds two more security events that were added to the API * Add challenge, received_challenge and received_hash in the inval_password security event unit test (Closes issue ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael L. Young Patches: issue-asterisk-19760-trunk.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1897/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	69e2a2b265	Update documentation references in CHANGES to reflect the correct pages on the wiki. The current CHANGES file refers to doc/ in many places and those files no longer exist. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	0fe5050553	Fix warning of Coverity Static analysis, change H225ProtocolIdentifier from value to pointer per functions that use this. (close issue ASTERISK-19670) Reported by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415) ........ Merged revisions 365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365160 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	474612d7f7	Add IPv6 support to ExternalIVR. Review: https://reviewboard.asterisk.org/r/1896/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	244922cf2e	Fix coverity static analysis warning, allocate full ie structure instead of without data buffer (close issue ASTERISK-19674) Reported by: Matt Jordan Patches: ASTERISK-19674.patch (License #5415) ........ Merged revisions 365143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365155 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	07309e586c	Multiple revisions 365006,365068 ........ r365006 \| twilson \| 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) \| 12 lines Fix a CEL LINKEDID_END race and local channel linkedids This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes the race condition by no longer scanning the channel list for "other" channels with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings and uses the refcount of the string as a counter of how many channels with the linkedid exist. Not only does this eliminate the race condition, but it also allows us to look up the linkedid by the hashed key instead of traversing the entire channel list. Review: https://reviewboard.asterisk.org/r/1895/ ........ r365068 \| twilson \| 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) \| 11 lines Don't leak a ref if out of memory and can't link the linkedid If the ao2_link fails, we are most likely out of memory and bad things are going to happen. Before those bad things happen, make sure to clean up the linkedid references. This patch also adds a comment explaining why linkedid can't be passed to both local channel allocations and combines two ao2_ref calls into 1. Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365083 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	ff06b448b8	Blocked revisions 365014 ........ Update security events unit tests The security events framework API was changed in Asterisk 10 but the unit tests were not updated at the same time. This patch does the following: * Adds two more security events that were added to the API * Add challenge, received_challenge and received_hash in the inval_password security event unit test (issue ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael L. Young Patches: issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1877/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365016 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	067064bd65	Save the address on which a MESSAGE was received, so it can be used in MESSAGE() This is useful in cases where chan_sip may be listening on multiple addresses. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	54143892af	Only log a failure to get read/write samples from factories if it didn't happen In audiohook_read_frame_both, anytime samples are obtained from the read/write factories a debug statement is logged stating that samples were not obtained from the factories. This statement used to only occur if option_debug was turned on and no samples were obtained; in some refactoring when the option_debug statement was removed, the "else" clause was removed as well. This patch makes it so that those debug log statements only occur if the condition leading up to them actually happened. ........ Merged revisions 364965 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	355a6d6f37	Remove a function that has been marked unused since Asterisk 1.6.0. The reason I'm removing this is that Coverity reported a STRAY_SEMICOLON issue here. Since the function has been unused for so long, I just elected to remove it altogether. (closes issue ASTERISK-19660) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1420522c6e	Fixed __ao2_ref() validating user_data twice. (closes issue ASTERISK-19755) Reported by: Gunther Kelleter Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter ........ Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364903 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	b5f0647fc8	Fix Coverity-reported ARRAY_VS_SINGLETON error. As it turned out, this wasn't a huge deal. We were calling ast_app_parse_options() for a set of options of which none took arguments. The proper thing to do for this case is to pass NULL for the "args" parameter here. We were instead passing a seemingly-randomly chosen char * from the function. While this would never get written to, you can rest assured things would have gotten bad had new options (which took arguments) been added to func_volume. (closes issue ASTERISK-19656) ........ Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364900 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2b7e1affac	* Fix error path resouce leak in local_request(). * Restructure local_request() to reduce indentation. ........ Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364845 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	885fbf6b04	Prevent a potential crash when using manager hooks. Found by me while poking at DPMA-127. ........ Merged revisions 364841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364842 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	a965f18695	Play conf-placeintoconf message to the correct channel Correct the code in app_confbridge to play the conf-placeintoconf message to the marked user entering the bridge instead of to the conference while the marked user hears silence. (closes issue ASTERISK-19641) Reported-by: Mark A Walters ........ Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364787 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	cefff2e52c	Fix bad check in voicemail functions for ast_inboxcount2_func Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes issue ASTERISK-19718) Reported by: Corey Farrell Patches: ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364777 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	6eb1ea3b79	Revert revision 360862. Revision 360862 was intended to improve identities sent in dialog-info NOTIFY requests. Some users reported that hint became broken once this was done. It's not clear exactly what part of the patch has caused this regression, but broken hints are bad. For now, this revision is being reverted so that the next releases of Asterisk do not have bad behavior in them. The original reported issue will have to be fixed differently in the next version of Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364707 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	f3cde589cd	Merged revisions 364635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r364635 \| markm \| 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) \| 10 lines Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs (closes issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark Murawski ........ ........ Merged revisions 364650 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	6d87267542	Fix use freed pointer in return value from call thread (issue ASTERISK-19663) Reported by: Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415) ........ Merged revisions 364649 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364651 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	60d9ca687c	Fix error that caused truncate operations to fail Another very inappropriate placement of a ')' (again introduced in r362151) caused the various truncate operations to attempt to truncate the sound file at a position of '0'. (issue ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec ........ Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364579 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	2cbcbc7f6b	Fix configuring custom sound_leader_has_left in confbridge.conf The configuration option to specify a custom sound_leader_has_left file for a conference bridge was not being parsed. This patch fixes it so that a custom sound file will now be used. (closes issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380) Review: https://reviewboard.asterisk.org/r/1884/ ........ Merged revisions 364536 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	ae1502be33	Add support for lightweight NAT keepalive. If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	19097a4b62	md5: supress some compiler warnings. md5.c: In function ‘MD5Final’: md5.c:154:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] md5.c:155:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] There is an md5 unit test and it still passes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	eebdf35159	res_corosync: Fix build against corosync 2.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	a498ef2aa0	app_minivm: Fix a couple compiler warnings. The warnings were about argv[0] being used uninitialized, which is correct. Just remove setting username to this value, since username is set again before it actually gets used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	386c2c6edf	features: Add FEATURE() and FEATUREMAP() functions. Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the [general] section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon. See the built-in documentation for details. Review: https://reviewboard.asterisk.org/r/1871/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e8a6e0ef0e	PreDial - Ability to run dialplan on callee and caller channels before Dial. Thanks to Mark Murawski for the initial patch and feature definition. (closes issue ASTERISK-19548) Reported by: Mark Murawski Review: https://reviewboard.asterisk.org/r/1878/ Review: https://reviewboard.asterisk.org/r/1229/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f7c174ff84	Multiple revisions 364365,364369 ........ r364365 \| twilson \| 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) \| 11 lines Fix ast_parse_arg numeric type range checking and add tests ast_parse_arg wasn't checking for strto* parse errors or limiting the results by the actual range of the numeric types. This patch fixes that and adds unit tests as well. Review: https://reviewboard.asterisk.org/r/1879/ ........ Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364369 \| twilson \| 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) \| 2 lines Add missing test_config.c ........ Merged revisions 364365,364369 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	1a58b3b775	Don't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails. (closes issue ASTERISK-18321) Reported by Dan Lukes Patches: ASTERISK-18321.patch by Mark Michelson (license #5049) ........ Merged revisions 364341 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364342 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	50c959580c	Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds, between two timeval structs, and return the difference in a 64-bit integer. Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval struct are large enough to hold the calculated values before it returns. On 64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit machines, however, a long may be less (32-bits), in which case, the calculation can overflow. This overflow caused significant problems in MixMonitor, which uses the method to determine if an audio factory, which has not presented audio to an audiohook, is merely late in providing said audio or will never provide audio. In an overflow situation, the audiohook would incorrectly determine that an audio factory that will never provide audio is merely late instead. This led to situations where a MixMonitor never recorded any audio. Note that this happened most frequently when that MixMonitor was started by the ConfBridge application itself, or when the MixMonitor was attached to a Local channel. (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski Tested by: Michael L. Young Patches: 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) (closes issue ASTERISK-19471) Reported by: feyfre Tested by: feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1889/ ........ Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364285 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	83cf78deda	Allow SIP pvts involved in Replaces transfers to fall out of reference sooner Unref the SIP pvt stored in the refer structure as soon as it is no longer needed so that the pvt and associated file descriptors can be freed sooner. This change makes a reference decrement unnecessary in code that handles SIP BYE/Also transfers which should not touch the reference anyway. (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski Tested by: Maciej Krajewski ........ Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364259 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	103031330a	Allow for reloading SRTP crypto keys within the same SIP dialog As a continuation of the patch in r356604, which allowed for the reloading of SRTP keys in re-INVITE transfer scenarios, this patch addresses the more common case where a new key is requested within the context of a current SIP dialog. This can occur, for example, when certain phones request a SIP hold. Previously, once a dialog was associated with an SRTP object, any subsequent attempt to process crypto keys in any SDP offer - either the current one or a new offer in a new SIP request - were ignored. This patch changes this behavior to only ignore subsequent crypto keys within the current SDP offer, but allows future SDP offers to change the keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Review: https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364204 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	14b52ff9da	fix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt use utc. This change uses the same timezone from the start time. ........ Merged revisions 364163 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	238640dc1b	Update Pickup application documentation. (With feeling this time.) ........ Merged revisions 364108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364109 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b22874415e	Fix DTMF atxfer running h exten after the wrong bridge ends. When party B does an attended transfer of party A to party C, the attending bridge between party B and C should not be running an h exten when the bridge ends. Running an h exten now sets a softhangup flag to ensure that an AGI will run in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the attending bridge between party B and C. (closes issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario ........ Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364065 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	49a49a51ef	Add more constness to the end_buf pointer in the netconsole issue ASTERISK-18308 Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 364046 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364047 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	e5c20ccb76	Code formatting fixes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	7bf6a01cfa	Fix reference leaks involving SIP Replaces transfers The reference held for SIP blind transfers using the Replaces header in an INVITE was never freed on success and also failed to be freed in some error conditions. This caused a file descriptor leak since the RTP structures in use at the time of the transfer were never freed. This reference leak and another relating to subscriptions in the same code path have now been corrected. (closes issue ASTERISK-19579) ........ Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363987 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	5746e0d2ac	chan_sip: [general] maxforwards, not checked for a value greater than 255 The peer maxforwards is checked for both '< 1' and '> 255', but the default 'maxforwards' in the [general] section is only checked for '< 1' alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1888/ ........ Merged revisions 363934 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363935 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9d655bd0e8	Update Pickup application documentation. (Even better) ........ Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363876 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e736a4fed3	* Put more information in pickup_exec() LOG_NOTICE. * Delay duplicating a string on the stack in pickup_exec(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0986873128	Update Pickup application documentation. ........ Merged revisions 363788 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363789 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	af39a4374e	Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call. Some switches may not handle the call-deflection/call-rerouting message if the call is disconnected too soon after being sent. Asteisk was not waiting for any reply before disconnecting the call. * Added a 5 second delay before disconnecting the call to wait for a potential response if the peer does not disconnect first. (closes issue ASTERISK-19708) Reported by: mehdi Shirazi Patches: jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363734 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d2ac624b87	Clear ISDN channel resetting state if the peer continues to use it. Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in response to a RESTART request. * Made the second SETUP received after sending a RESTART request clear the channel resetting state as if the peer had sent the expected RESTART ACKNOWLEDGE before continuing to process the SETUP. The peer may not be sending the expected RESTART ACKNOWLEDGE. (issue ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified) ........ Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363688 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	04ddb5714f	Add documentation Thanks Tilghman! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	f102aecf12	Formatting changes only git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	a8e755700e	Use the DEFINED value for musicclass length. For some reason, features.c has it's own definition. Should propably be fixed too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	7aa0c3c64b	Make it possible to change the minimum DTMF duration in asterisk.conf Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

1 2 3 4 5 ...

22493 Commits (8842f76a7f486cd3d223a2c28ace91fcab5e8a12) All Branches Search

22493 Commits (8842f76a7f486cd3d223a2c28ace91fcab5e8a12)

All Branches