342 Commits (091f850a5831399bc2c427cb2b2b3d4399e5ab17)

Author	SHA1	Message	Date
David Vossel	091f850a58	fixes sample rate conversion issue with Monitor application ... When using ast_seekstream with the read/write streams of a monitor, the number of samples we are seeking must be of the same rate as the stream or the jump calculation will be incorrect. This patch adds logic to correctly convert the number of samples to jump to the sample rate the read/write stream is using. For example, if the call is G722 (16khz) and the read/write stream is recording a 8khz wav, seeking 320 samples of 16khz audio is not the same as seeking 320 samples of 8khz audio when performing the ast_seekstream on the stream. ABE-2044 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	7d5d0311c1	Merged revisions 246545 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) | 16 lines lock channel during datastore removal On channel destruction the channel's datastores are removed and destroyed. Since there are public API calls to find and remove datastores on a channel, a lock should be held whenever datastores are removed and destroyed. This resolves a crash caused by a race condition in app_chanspy.c. (closes issue #16678) Reported by: tim_ringenbach Patches: datastore_destroy_race.diff uploaded by tim ringenbach (license 540) Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	72c1b76038	Merged revisions 244070 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010) | 16 lines Revert previous chan_local fix (r236981) and fix instead by destroying expired frames in the queue. (closes issue #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt uploaded by tilghman (license 14) 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: kobaz, atis (closes issue #16581) Reported by: ZX81 (closes issue #16681) Reported by: alexr1 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	c277952cea	Merged revisions 243258 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243258 | jpeeler | 2010-01-26 12:19:10 -0600 (Tue, 26 Jan 2010) | 2 lines Remove unnecessary code in ast_read as issue 16058 has been fully solved now. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	568c057c4c	Extend max call limit duration from 24.8 days to 292+ million years. ... If the limit was set past MAX_INT upon answering, the call was immediately hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup). The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been changed to return an int64_t to prevent overflow. Also the reporter suggested adding a message indicating the reason for the call hanging up. Given that the new limit is so much higher, the message (which would only really be useful in the overflow scenario) has been made a debug message only. (closes issue #16006) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	9228fba7d7	Fix broken call pickup ... The problem was the OUTGOING flag was not getting set properly on the channel, resulting in pickup failing as ast_read thought the call was inbound. Refer to 170393 for a more verbose description as this is the same exact change. (closes issue #16539) Reported by: syspert Patches: bug16539.patch uploaded by jpeeler (license 325) Tested by: syspert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	03529837cc	add silence gen to wait apps ... asterisk.conf's 'transmit_silence' option existed before this patch, but was limited to only generating silence while recording and sending DTMF. Now enabling the transmit_silence option generates silence during wait times as well. To achieve this, ast_safe_sleep has been modified to generate silence anytime no other generators are present and transmit_silence is enabled. Wait apps not using ast_safe_sleep now generate silence when transmit_silence is enabled as well. (closes issue #16524) Reported by: kobaz (closes issue #16523) Reported by: kobaz Tested by: dvossel Review: https://reviewboard.asterisk.org/r/456/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a575a50cd5	fixes ast_transfer stall until hangup if called with a channel that doesn't support transfers ...	16 years ago
Jeff Peeler	cf7b67d9d3	Merged revisions 235635 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	473837a4ab	Change match criteria existence in ast_channel_cmp_cb to use ast_strlen_zero. ... (closes issue #16161) Reported by: may213 Patches: core-show-channel.patch uploaded by may213 (license 454) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	97bb26bf75	Merged revisions 233092 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) | 7 lines Only do frame payload check for HOLD frames. This code was added for helping to debug the source of invalid HOLD frames. However, a side effect of this is that it will incorrectly report errors for frames that have an integer payload. Make the check for this block specific to the HOLD frame case. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	d9f37a67e1	Merged revisions 231911 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) | 12 lines Fix crash with invalid frame data The crash was happening as a result of a frame containing an invalid data pointer, but was set with data length of zero. The few times the issue was reproduced it _seemed_ that the frame was queued properly, that is the data pointer was set to NULL. I never could reproduce the crash so as a last resort the crash has been fixed, but a check in __ast_read has been added to give as much information about the source of problematic frames in the future. (closes issue #16058) Reported by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f98c12a57d	Merged revisions 231298 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) | 2 lines After a frame duplication failure, unlock the channel before returning. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5e2aa190fe	Display a list of channel variables in each channel-oriented event. ... (Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	fef304e641	Merged revisions 228896 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) | 6 lines Update WARNING message. Update a WARNING message to give a suggested fix when encountered. (closes issue #16198) Reported by: atis Tested by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	10fa8fe8d0	Merged revisions 228692 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) | 9 lines fixes audiohook write crash occuring in chan_spy whisper mode. After writing to the audiohook list in ast_write(), frames were being freed incorrectly. Under certain conditions this resulted in a double free crash. (closes issue #16133) Reported by: wetwired (closes issue #16045) Reported by: bluecrow76 Patches: issue16045.diff uploaded by dvossel (license 671) Tested by: bluecrow76, dvossel, habile ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. ... Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1174a61612	Add support for calling and called subaddress. Partial support for COLP subaddress. ... The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	cdd1f9e296	Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it. ... This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the case where multiple results need to be returned; OBJ_NODATA mode already was supported). In addition, it converts ast_channel_iterators (only the targeted versions, not the ones that iterate over all channels) to use this method. During this work, I removed the 'ao2_flags' arguments to the ast_channel_iterator constructor functions; there were no uses of that argument yet, there is only one possible flag to pass, and it made the iterators less 'opaque'. If at some point in the future someone really needs an ast_channel_iterator that does not lock the container, we can provide constructor(s) for that purpose. Review: https://reviewboard.asterisk.org/r/379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	0d4726b0a2	Merged revisions 223225 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9456ab2724	Deadlock in channel masquerade handling ... Channels are stored in an ao2_container. When accessing an item within an ao2_container the proper locking order is to first lock the container, and then the items within it. In ast_do_masquerade both the clone and original channel must be locked for the entire duration of the function. The problem with this is that it attemptes to unlink and link these channels back into the ao2_container when one of the channel's name changes. This is invalid locking order as the process of unlinking and linking will lock the ao2_container while the channels are locked!!! Now, both the channels in do_masquerade are unlinked from the ao2_container and then locked for the entire function. At the end of the function both channels are unlocked and linked back into the container with their new names as hash values. This new method of requiring all channels and tech pvts to be unlocked before ast_do_masquerade() or ast_change_name() required several changes throughout the code base. (closes issue #15911) Reported by: russell Patches: masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, atis (closes issue #15618) Reported by: lmsteffan Patches: deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671) Tested by: lmsteffan, dvossel Review: https://reviewboard.asterisk.org/r/387/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a4ece92018	Merged revisions 221200 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	8c30540269	Correct sense of logic test committed in revision 220494. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	aabdc575a5	Don't use hash-based lookups for ast_channel_get_by_name_prefix(). ... ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based channel lookups, but this will not work properly when the channel's full name was not supplied; for name-prefix searches, there is no value in doing a hash-based lookup, and in fact doing so could result in many channels being skipped. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	b27a54b8de	Merged revisions 219136 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	40c13bd1b0	Merged revisions 214701 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines Modify comment to be a bit more accurate. We have kept this comment around long enough, that it's pretty clear that we're keeping the code, because changing the code would require a pretty fundamental architectural shift. We've also taken criticism in some quarters, because it was believed that it was referring to the code being nasty. No, the code isn't nasty, just the operation itself is rather odd. Fixed for eternity (probably not). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2794b198ce	Merged revisions 214194 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) | 19 lines ast_write() ignores ast_audiohook_write() results In ast_write(), if a channel has a list of audiohooks, those lists are written to and the resulting frame is what ast_write() should continue with. The problem was the returned audiohook frame was not being handled at all, and the original frame passed into it did not contain the mixed audio, so essentially audio was being lost. One result of this was chan_spy's whisper mode no longer worked. To complicate the issue, frames passed into ast_write may either be a single frame, or a list of frames. So, as the list of frames is processed in the audiohook_write, the returned frames had to be added to a new list. (closes issue #15660) Reported by: corruptor Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	642bec4d6f	AST-2009-005 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	724c1239fc	Fix up some issues with getting a channel by "name". ... Even though the get_channel_by_name() API advertised that you could search by name or uniqueid (just as the old API did), searching by uniqueid was not actually implemented. This patch fixes that problem. The ast_channel_get_full() function now makes a second search attempt by uniqueid if the parameter was a name. The channel comparison function also now knows how to compare by unqieueid. Finally, a bug was fixed in passing where OBJ_POINTER was being passed in some scenarios where it should not have been. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	feced6672c	Merged revisions 210913 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009) | 7 lines Because channel information can be accessed outside of the channel thread, we must lock the channel prior to modifying it. (closes issue #15397) Reported by: caspy Patches: 20090714__issue15397.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	28ad5ced1a	Initial minimum ast_party_caller support. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	214453904e	Fix order and redundancy of channel rename manager events in ast_do_masquerade. ... Patch contributed by Mark Spencer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	0a6e06c7ff	Rework of T.38 negotiation and UDPTL API to address interoperability problems ... Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	44301c95d2	Merged revisions 207360 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	8b878c8303	Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. ... This change allows applications that request T.38 negotiation on a channel that does not support it to get the proper indication that it is not supported, rather than thinking that negotiation was started when it was not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	fd6a49beac	Moved trigger for BRIDGE_END CEL event so that it is more accurate. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b5f6eac49e	Allow trunk to once again compile under MALLOC_DEBUG ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	59c1998d67	Improve T.38 negotiation by exchanging session parameters between application and channel. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. ... CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2affa3e999	Merged revisions 202496 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d8cc968adc	Merged revisions 201450 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4c0265664e	Merged revisions 200991 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	afcbf2e14f	Merged revisions 200360 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	dabfa94fdc	Release the allocated channel decreasing the reference counter. ... When allocating the channel use ao2_ref(-1) to release it, instead of calling ast_free(). Also avoid freeing structures inside that channel (on error) if they will be released by the channel destructor being called if the reference counter reachs 0. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	554456f0fc	Use ast_channel_unref to instead of ast_free on a newly created channel. ... Also I removed an unnecessary free of a cid_name. This will be freed properly in the channel destructor. Reported by mnicholson in #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	c42344b319	ast_call_forward() todo notes and originate flag copy. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	3830c415c7	Generic call forward api, ast_call_forward() ... The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	298d745fb4	Add the ability to execute connected line interception macros. ... When connected line updates are received or generated in the middle of an application call, it is now possible to execute a macro to manipulate the connected line data. This way, phone numbers may be manipulated to be more presentable to users, names may be changed for...whatever reason, or whatever else needs to be done may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8580871fd4	Constify the ast_frame arg to ast_queue_frame(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	c8b0c41ed8	Merged revisions 198068 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago