61 Commits (3fb6ac7fb7925197a085cc573ac0af2ec9102c68)

Author	SHA1	Message	Date
Russell Bryant	3fb6ac7fb7	In ast_channel_make_compatible(), just return if the channels' read and write ... formats already match up. There are code paths that call this function on a pair of channels multiple times. This made calls fail that were using g729 in some cases. The reason is that codec_g729a will unregister itself from the list of available translators will all licenses are in use. So, the first time the function got called, the right translation path was allocated. However, the second time it got called, the code would not find a translation path to/from g729 and make the call fail, even if the channel actually already had a g729 translation path allocated. (SPD-32) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	080acbfd16	Merged revisions 68682 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 lines Improve deadlock handling of the channel list. (issue #8376 reported by one47) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	ac65d7f143	Fix logic when doing a name based channel search for a structure when you want to start from a specific point in the channel list. (issue #9324 reported by slavon) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	e7dd69eea8	Merged revisions 67715 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines We have some bug reports showing crashes due to a double free of a channel. Add a sanity check to ast_channel_free() to make sure we don't go on trying to free a channel that wasn't found in the channel list. (issue #8850, and others...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	9dd7be2f58	Handle a failure in malloc() in ast_safe_string_alloc() ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	6d6cd29a87	Fix a problem that showed itself by causing Zap channel names to be completely ... bogus on my machine. ast_safe_string_alloc() was broken. It called vsnprintf() on a va_args list twice without re-initializing it. After the first usage, va_end() and va_start() must be called again. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	a4c142d73d	if the string field init fails, clean up the stuff that was allocated already ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	11d6381ec0	Check the result of ast_string_field_init() in ast_channel_alloc() ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	aff63fba76	Fix scenario where if a phone that simply called Echo() put itself on hold it could never get off hold. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	7583e21359	Add hangupcause when we lack codecs for transcoding ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	7deba0a857	Use the DTMF frame on the channel when returning a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	da55121999	Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the ... code in that if a channel does not have a send_digit_begin() callback, it only cares about DTMF END events. (pointed out by Michael Neuhauser on the asterisk-dev list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	e9fab07122	Only call ast_senddigit_begin() in ast_senddigit() if the channel has a ... send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the wrong thing to do, because that flag indicates that a *bridged* channel only wants DTMF END events coming from this channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	99cdfb2542	Merged revisions 63285 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 lines Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	796acdbbdb	Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending). ... This set of changes came from a debugging session I had with Dwayne Hubbard. When he called into his home FXO, ran the Echo application, and pressed a digit, the digit would be echoed back and would never end. This is fixed, along with a couple other little improvements. * When chan_zap is in the middle of playing a digit to a channel, it feeds back null frames, not voice frames. So, I have modified ast_read to check the timing on emulated DTMF when it receives null frames, in addition to where it was doing this on voice frames. * Make a tweak to setting the duration on emulated DTMF digits. If there was no duration specified, it set it to be the minimum, instead of the default. * Instead of timing the emulated digits off of the number of samples in audio frames that pass through, just use time values. Now there is no code in this section that assumes 8kHz audio. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	ff3e3ba129	Merge changes from team/russell/inband_dtmf ... ... Fix some issues related to generating inband DTMF. There are two changes here: 1) The list of DTMF tones in the senddigit_begin() function explicitly specified 100ms of the tone followed by 100ms of silence. This really broke things with the way that Asterisk now wants complete control over when the digit begins and ends. So, regardless of what Asterisk really wanted to do, this was going to play out the tone at the length it wanted to. This caused various problems like DTMF translation to inband to be extremely unreliable. The list of tones has been changed so that the correct DTMF tone is played indefinitely until Asterisk tells it to stop. 2) ast_write() had to be modified to let a DTMF_END frame get processed even when a generator is present. This is how the tone will finally get stopped. (issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for the testing and feedback!) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	55f4eb3e3d	a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	230dc4ad88	Missed an ast_app_group_discard during merge. Thanks blitzrage! ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	1d4adc0174	Merged revisions 61804 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	456cad8a47	Improve DTMF handling in ast_read() even more in response to a discussion on ... the asterisk-dev mailing list. I changed the enforced minimum length of a digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in between digits. These values are not configurable in a configuration file right now, but they can be easily changed near the top of main/channel.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	baab073219	Ensure that digits passing through Asterisk have a reasonable minimum length. ... It is currently 100 ms. If someone thinks this should be different, feel free to speak up. (related to issues #8944, #9250, and #9348) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	7d5a79a0b9	This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	798039b4d8	several changes via kpflemings review ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	9c69e34f62	These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated from transfer situations. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	185d7f5a3a	Merged revisions 56684 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines Issue 9130 - If prev is the last item on the channel list, then evaluating additional conditions (e.g. name prefix) will cause a NULL dereference. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	4565c1483c	Merged revisions 56230 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	7d029cabf6	Add G722 to ast_best_codec. If anyone disagrees with it's placement, feel free to change it. (issue #9045 reported by gork) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Paul Cadach	3a68e31940	Provide correct DTMF duration ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	9440d428a0	Merged revisions 51843 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | 6 lines Fix an issue related to synchronization of recordings when using Monitor(). The bug is a miscalculation of the amount to seek the stream for writing to disk when the number of samples coming in and out of a channel do not match up. (issue #8298, #8887, report and patch by guillecabeza, patch files created and testing done by whoiswes) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	33235b40d6	Merge the changes from the /team/group/vldtmf_fixup branch. ... The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	76eb4e76dc	Add some more checks for option_debug before ast_log(LOG_DEBUG, ...) calls. ... Issue 8832, patch(es) by tgrman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	eab59cabdb	Fix an issue with deprecated commands ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	e4de85c4f5	Only write a frame out to the channel if one exists. There are cases where one may not and would therefore cause the channel driver to segfault. (issue #8434 reported by slimey) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	ac75028a50	Don't keep repeating the warning over and over when the end of the call is reached. (issue #8724 reported by xrg) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	a299485952	since these variables all have static duration, none of them need initializers (they default to zero anyway) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	cb11263607	We need to ensure timelimit stuff is included as well so warnings get played. (issue #8050 reported by KNK) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	729ce4d791	This update fixes 7531 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	517978fd5f	These mods are to solve the problem in bug 7506. It's a lot of rework to solve a fairly small problem... such is life. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	e05a2752e8	Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments" ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	de9c3aaa8d	Pass through a frame if we don't know what it is, rather than trying to pass a ... NULL, which will segfault a channel driver (Bug 8149) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	0a78fea2ac	Don't attempt to access private data members of the pthread_mutex_t object, ... because this does not work on all linux systems. Instead, just access the reentrancy field in the ast_mutex_info struct when DEBUG_THREADS is enabled. If DEBUG_CHANNEL_LOCKS is enabled, the developer probably has DEBUG_THREADS on as well. (issue #8139, me) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	227d415709	optimize the 'quick response' code a bit more... no more malloc() or memset() for each response ... expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Paul Cadach	53024e3508	CHANNEL() function sometime mix parameter and value ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	439f15bdb9	Merged revisions 43778 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines Fix a problem that occurred if a user entered a digit that matched a bridge feature that was configured using multiple digits, and the digit that was pressed timed out in the feature digit timeout period. For example, if blind transfer is configured as '##', and a user presses just '#'. In this situation, the call would lock up and no longer pass any frames. (issue #7977 reported by festr, and issue #7982 reported by michaels and valuable input provided by mneuhauser and kuj. Fixed by me, with testing help and peer review from Joshua Colp). There are a couple of issues involved in this fix: 1) When ast_generic_bridge determines that there has been a timeout, it returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls ast_generic_bridge over again with the same timestamp for the next event. This results in an endless loop of nothing until the call is terminated. This is resolved by simply changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it sees a timeout. 2) I also changed ast_channel_bridge such that if in the process of calculating the time until the next event, it knows a timeout has already occured, to immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the channels anyway. 3) In the process of testing the previous two changes, I ran into a problem in res_features where ast_channel_bridge would return because it determined that there was a timeout. However, ast_bridge_call in res_features would then determine by its own calculation that there was still 1 ms before the timeout really occurs. It would then proceed, and since the bridge broke out and did *not* return a frame, it interpreted this as the call was over and hung up the channels. The reason for this was because ast_bridge_call in res_features and ast_channel_bridge in channel.c were using different times for their calculations. channel.c uses the start_time on the bridge config, which is the time that the feature digit was recieved. However, res_features had another time, 'start', which was set right before calling ast_channel_bridge. 'start' will always be slightly after start_time in the bridge config, and sometimes enough to round up to one ms. This is fixed by making ast_bridge_call use the same time as ast_channel_bridge for the timeout calculation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	d95288fdf8	Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	53ff7ae7cd	all the Linux systems I have don't use '__m_count' for this field, so I don't know where this came from... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	fcb999c01c	merge qwell's CLI verbification work ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	0f3eaaf703	Merged revisions 42600 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r42600 | file | 2006-09-09 16:24:19 -0400 (Sat, 09 Sep 2006) | 2 lines Only truly consider the channel in the same format if the format matches the raw format OR if a translation path already exists to translate between them. (issue #7887 reported by softins & issue #7803 reported by alvaro_palma_aste). Thanks goes to stubert for giving me access to a box and showing me a scenario where this occured. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	160cea6139	Merged revisions 42452 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r42452 | file | 2006-09-08 14:50:43 -0400 (Fri, 08 Sep 2006) | 2 lines Swap spies during masquerading ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	d597b76c36	whentohangup is already in seconds, just need to convert to milliseconds ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago