asterisk

Commit Graph

Author	SHA1	Message	Date
Jeff Peeler	21ca773c28	Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. Because using the ast prefix calls are a better choice, ast_free_ptr is the new wrapper for free to pass to functions. Also, a little bit of clean up was done to avoid the debug macros intentionally being redefined. (closes issue #13593) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	5436d8709f	Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	426cee0362	This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a5198f55e0	Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-15 are not defined, these bits are never turned off. In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue #14283) Reported by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c66921ec43	Document the return value of the update method (as requested on -dev list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	de8f6bab86	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/branches/1.4@176701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	db4dc67740	fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	3cc3863d28	Zaptel is not DAHDI. Rather, Zaptel is actually Zaptel. (in case you're confused, DAHDI is still DAHDI) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	4e069885ce	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/branches/1.4@172517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	13a60eba0c	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/branches/1.4@172030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e2d1540779	AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is whether autoconf detected a static initializer value. This fixes rwlocks on all such platforms (mainly, Mac OS X). (closes issue #13767) Reported by: jcovert Patches: 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) Tested by: jcovert, Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d671bb1404	Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9161b7fc87	Revert unnecessary indications API change from rev 122314 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	acd0af78ea	Re-work ref count handling of MoH classes using astobj2 to resolve crashes. (closes issue #13566) Reported by: igorcarneiro Tested by: russell Review: http://reviewboard.digium.com/r/106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@166262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7fdf99803e	Backport of AUDIOHOOK_INHERIT for Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@166157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	e3700a13a4	This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@166093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e0bec5d67d	(closes issue #13480 ) Reported by: tzafrir Replace a bunch of if defined checks for Zaptel/DAHDI through several new defines in dahdi_compat.h. This removes a lot of code duplication. Example from bug: #ifdef HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd = open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd = open(DAHDI_FILE_PSEUDO, O_RDRW); git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	083047a794	Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was being used within the context of the thread local data destructors. We would go off and allocate more thread local data while the pthread lib was in the middle of destroying it all. This led to a memory leak. Another issue was an invalid argument being provided to the the object_add API call. (closes issue #13678) Reported by: ys Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ccf5565ccd	Add the deadlock note to ast_spawn_extension as well git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164422 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a43cf62956	Add notes to autoservice and pbx doxygen regarding a potential deadlock scenario so that it is avoided in the future git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b80ffd6d26	Use autoconf logic to determine whether the system has timersub or not. Do not blindly assume Solaris does not. (closes issue #13838) Reported by: ano git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c518ed3be1	Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@163448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4cad4bff2c	Update to stringfield handling so that side-effects on parameters are not evaluated multiple times. An example where this caused a problem was in chan_sip.c, with the line ast_string_field_set(p, fromdomain, ++fromdomain); This patch was originally uploaded to issue #13783 by jamessan. While the issue was closed for other reasons, this patch is valid and fixes a separate problem, and is thus being committed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c2446000d9	Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	a36a0d7dbd	(closes issue #14019 ) Reported by: ckjohnsonme Patches: 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, murf This crash was the result of a few small errors that would combine in 64-bit land to result in a crash. 32-bit land might have seen these combine to mysteriously drop the args to an application call, in certain circumstances. Also, in trying to find this bug, I spotted a situation in the flex input, where, in passing back a 'word' to the parser, it would allocate a buffer larger than necessary. I changed the usage in such situations, so that strdup was not used, but rather, an ast_malloc, followed by ast_copy_string. I removed a field from the pval struct, in u2, that was never getting used, and set in one spot in the code. I believe it was an artifact of a previous fix to make switch cases work invisibly with extens. And, for goto's I removed a '!' from before a strcmp, that has been there since the initial merging of AEL2, that might prevent the proper target of a goto from being found. This was pretty harmless on its own, as it would just louse up a consistency check for users. Many thanks to ckjohnsonme for providing a simplified and complete set of information about the bug, that helped considerably in finding and fixing the problem. Now, to get aelparse up and running again in trunk, and out of its "horribly broken" state, so I can run the regression suite! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	ffc0c7e4ae	Merged revisions 161421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r161421 \| seanbright \| 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) \| 8 lines Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@161426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	a3724485b4	make compile with dev mode and malloc debug git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	1653a9ef65	Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	50515ed372	update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	07741a3261	Add ao2_trylock() to go along with ao2_lock() and ao2_unlock() git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	929c14e432	System call ioperm is non-portable, so check for its existence in autoconf. (Closes issue #13863) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159025 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	c7dcb086dd	(closes issue #13786 ) Reported by: tzafrir When compiling against Zaptel dahdi_compat will now only define all the DAHDI defines if the Zaptel define is present. Also, there is no such thing as DAHDI_PRI. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	a0bf88537a	When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	6a632a5a2b	There was an issue when attempting to reference an embedded frame in a freed ast_filestream. This patch makes use of the ao2 functions to make sure that we do not free an ast_filestream structure until the embedded ast_frame has been "freed" as well. (closes issue #13496) Reported by: fst-onge Patches: filestream_frame_1_4.diff uploaded by putnopvut (license 60) Tested by: putnopvut Closes AST-89 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	3429c9de5e	Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@157305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Sean Bright	f2ecc4c80e	Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@155553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	0d1441526e	features.h depends on linkedlists.h, so include it git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@153651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	6280e04736	Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@153095 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	961bc7e758	The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152535 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	c19ca186c1	Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	fc2e08f823	(closes issue #13425 ) Reported by: mdu113 Tested by: mdu113 Similar to r143204, masquerade the channel in the case of Park being called from AGI. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@146129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	d10c4137fb	fix some minor issues with rev 144924 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	50a28e9853	improve header inclusion process in a few small ways: - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled - simplify the usage of some of these headers in the AEL-related stuff in the utils directory git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	eccd14d7f0	Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@142675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	28b9f5b384	Fix compilation when DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@139553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	e2fc623d88	Backport support for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not doing so just makes it difficult for people with channels that are in alarm when Asterisk starts up to get them going once the alarm is cleared (closes issue #12160) Reported by: tzafrir Patches: asterisk-chanalarms_14.patch uploaded by tzafrir (license 46) Tested by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@139145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	a763147ee6	add DAHDI_POLICY_WHEN_FULL compatability define for Zaptel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@138949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	e797e4ed72	add support for Zaptel versions that contain the new transcoder interface git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@137677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	6d7a2a5fa4	Merged revisions 136945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r136945 \| tilghman \| 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) \| 2 lines Regression fixes for Solaris ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@136946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	195fe34304	(closes issue #13236 )	18 years ago

1 2 3 4 5 ...

1298 Commits (3cbc42e2e458dac6ccc68e68ada7dfcb416fd111)