asterisk

Commit Graph

Author	SHA1	Message	Date
Leif Madsen	2de9cd0d38	Add documentation clarifying when 't' and 'T' can be used. (closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c6a759fdfa	Merged revisions 255409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255409 \| russell \| 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) \| 2 lines Don't kill Asterisk if the H323 listener does not start. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	be672528e6	Merged revisions 255322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255322 \| russell \| 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) \| 2 lines Don't make Asterisk not start if pbx_dundi fails to initialize. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jared Smith	c34ec47577	This patch adds custom device state handling for ConfBridge conferences, matching the devstate handling of the MeetMe conferences. Review: https://reviewboard.asterisk.org/r/572/ Closes issue #16972 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	899f995703	Remove a debugging log entry. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	89e4c15783	corrections in gk interface, small fixes in call clearing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	b8aeb50b7b	We need to inclde sys/wait.h on OpenBSD to get WEXITSTATUS. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	11f57c6806	inotify support for pbx_spool This should give a good speed boost, in that one particular thread isn't waking up once a second to read directory contents. Reviewboard: https://reviewboard.asterisk.org/r/137/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	8a3576d16c	Replace some documentation from 1.6.x back into trunk. This documentation associated wth tlsbindaddr is still useful so lets synchronize it between trunk and 1.6.x branches. (issue #17054) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	aae9d51510	Update confusing documentation for tlsbindaddr. Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	a165ccc10f	Work around a bug in dash on Ubuntu by checking the number of arguments before shift'ing. Reported and tested by pabelanger. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	fff17a25dd	Use "local" instead of "system" header file inclusion. Now that these files are in the tree, they should prefer the tree's local copy of all Asterisk headers over any that may be installed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0ec49aa381	Fix a number of other build problems on Mac OS X. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	6c693bc2b1	Merged revisions 254800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254800 \| qwell \| 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) \| 1 line Don't remove local copies of utils in uninstall. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c23ff728ea	Resolve compiler warning on FreeBSD. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	b69245f8e3	Fix chan_ooh323 so it works on Mac OS X, as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0703c27015	chan_usbradio depends on alsa. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	3cfab3bb6c	Blocked revisions 254714 via svnmerge ........ r254714 \| qwell \| 2010-03-25 14:39:23 -0500 (Thu, 25 Mar 2010) \| 4 lines Fix DEBUG_THREADS issue with out-of-tree modules. Take 2, without ABI breakage this time. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	5cb45e4970	Bump cleancount due to ast_channel change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	32938f653f	Remove no-longer-used (and unsafe) field in ast_channel for linked lists. The ast_channel structure had a field used for linking a channel into a linked list, but now that ast_channel structures are ao2 objects, this is no longer needed, and could be harmful as ao2 objects really shouldn't ever be placed into linked lists (since those lists don't assist with reference count management on the objects). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	66dd3e295d	Get chan_ooh323 building again after recent build system changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	49e2cd4291	Add unit test for testing ACL functionality. There are two unit tests contained here. 1. "Invalid ACL" This attempts to read a bunch of badly formatted ACL entries and add them to a host access rule. The goal of this test is to be sure that all invalid entries are rejected as they should be. 2. "ACL" This sets up four ACLs. One is a permit all, one is a deny all, and the other two have specific rules about which subnets are allowed and which are not. Then a set of test addresses is used to determine whether we would allow those addresses to access us when each ACL is applied. This test, by the way, was what resulted in AST-2010-003's creation. Review: https://reviewboard.asterisk.org/r/532 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a1e426cd2a	Merged revisions 254552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254552 \| mmichelson \| 2010-03-25 12:33:35 -0500 (Thu, 25 Mar 2010) \| 5 lines Add doxygen for acl.h Review: https://reviewboard.asterisk.org/r/528 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	54f5e1f840	Add new rtpsource options to the CHANNEL function. This adds rtpsource options analogous to the rtpdest functions that already exist. In addition, this fixes potential crashes which could result due to trying to read values from nonexistent RTP streams. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bd716c50fd	Recorded merge of revisions 254452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254452 \| mmichelson \| 2010-03-25 10:59:56 -0500 (Thu, 25 Mar 2010) \| 44 lines Several fixes regarding RFC2833 DTMF detection. Here is a copy and paste of the details from my request on reviewboard that dealt with these changes: Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too: seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF. Fix 2. The second change in place is to fix an issue like the following: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) packet lost seqno 4: DTMF 1 (end) packet lost seqno 5: DTMF 1 (end) packet lost seqno 6: DTMF 2 When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list. Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	408ba24fad	Merged revisions 254451 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254451 \| twilson \| 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) \| 2 lines Handle new SRCCHANGE control message here too ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	42577406fd	Improve handling of T.38 re-INVITEs that arrive before a T.38-capable application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame can fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	0eb71bccf1	handle_speechset has 4 arguments. Update code to reflect that handle_speechset has 4 arguments. (closes issue #17093) Reported by: gpatri Patches: res_agi.patch uploaded by gpatri (license 1014) Tested by: pabelanger, mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	8acb32460a	remove unneeded explicit channel in dahdi ioctls This patch removes some cases where the channel number for an ioctl was passed as a member in a struct rather then through the file descriptor. The gain setting functions passed around a channel which is always 0, and thus this parameter is simply dropped. Review: https://reviewboard.asterisk.org/r/584/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	10d65ad6b1	Fix potential invalid reads that could occur in pbx.c Here is a cut and paste of my review request for this change: This past weekend, Russell ran our current suite of unit tests for Asterisk under valgrind. The PBX pattern match test caused valgrind to spew forth two invalid read errors. This patch contains two changes that shut valgrind up and do not cause any new memory leaks. Change 1: In ast_context_remove_extension_callerid2, valgrind reported an invalid read in the for loop close to the function's end. Specifically, one of the the strcmp calls in the loop control was reading invalid memory. This was because the caller of ast_context_remove_extension_callerid2 (__ast_context destroy in this case) passed as a parameter a shallow copy of an ast_exten's exten field. This same ast_exten was what was destroyed inside the for loop, thus any iterations of the for loop beyond the destruction of the ast_exten would result in invalid reads. My fix for this is to make a copy of the ast_exten's exten field and pass the copy to ast_context_remove_extension_callerid2. In addition, I have also acted similarly with the ast_exten's matchcid field. Since in this case a NULL is handled quite differently than an empty string, I needed to be a bit more careful with its handling. Change 2: In __ast_context_destroy, we iterated over a hashtab and called ast_context_remove_extension_callerid2 on each item. Specifically, the hashtab over which we were iterating was an ast_exten's peer_table. Inside of ast_context_remove_extension_callerid2, we could possibly destroy this ast_exten, which also caused the hashtab to be freed. Attempting to call ast_hashtab_end_traversal on the hashtab iterator caused an invalid read to occur when trying to read the iterator->tab->do_locking field since iterator->tab had already been freed. My handling of this problem is a bit less straightforward. With each iteration over the hashtab's contents, we set a variable called "end_traversal" based on the return of ast_context_remove_extension_callerid2. If 0 is ever returned, then we know that the extension was found and destroyed. Because of this, we cannot call ast_hashtab_end_traversal because we will be guaranteeing a read of invalid memory. In such a case, we forego calling ast_hashtab_end_traversal and instead call ast_free on the hashtab iterator. Review: https://reviewboard.asterisk.org/r/585 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	560d5c6099	Allow configuration of minsecs and nextaftercmd per mailbox. Previously only configurable globally. A unit test has also been written to provide protection against parse failures for supported mailbox options. (closes issue #16864) Reported by: kobaz Patches: voicemail2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/555/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5990fe07b8	Merged revisions 254235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254235 \| jpeeler \| 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) \| 72 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248860. As such the dialplan test has been extended: ; non absolute path, not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) exten => 5040, n, dial(sip/5001) ; absolute path, not combined exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, dial(sip/5001) ; combined: changemonitor from no path to non absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before exten => 5044, n, dial(sip/5001) ; non absolute path, combined exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, dial(sip/5001) ; absolute path, combined exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, dial(sip/5001) ; no path, combined exten => 5047, 1, monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, dial(sip/5001) ; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, dial(sip/5001) ; combined: changemonitor from no path to absolute exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, dial(sip/5001) ; combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; not combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, dial(sip/5001) ; not combined: changemonitor from no path to non absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, dial(sip/5001) ; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, dial(sip/5001) ; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, dial(sip/5001) ; not combined: changemonitor from no path to absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, n, dial(sip/5001) ; not combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	5a5599a764	make 'core show settings' should show all settable directories (closes issue #17086) Reported by: tzafrir Patches: asterisk_extra_settings_dirs.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	d9b1ff23ba	Put test output for a failure in a CDATA section in the XML results. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	48edf2c78a	Exit native bridging early for greater timing accuracy with warnings This changes native bridging to break one millisecond early so that the more accurate timeval calculations done in the generic bridge can be performed using the bridge config. Currently the time between exiting native bridging slightly late can sometimes cause a large enough discrepancy for warnings to be missed. For the record, 1.4 does not attempt to native bridge at all when warnings are enabled. (closes issue #15815) Reported by: adomjan Review: https://reviewboard.asterisk.org/r/577/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	61e7dbdddd	Blocked revisions 254046 via svnmerge ........ r254046 \| qwell \| 2010-03-23 16:07:54 -0500 (Tue, 23 Mar 2010) \| 9 lines Allow out-of-tree modules to load, regardless of DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences. This can be guaranteed by forcing the ABI to no longer change when these compiler flags are set. An unfortunate side-effect to this is that there is an ABI change here. However, there is some mitigation. Existing modules will fail to load since they would require functions that no longer exist. Review: https://reviewboard.asterisk.org/r/508/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	9461bac812	Remove unused structure member in app_queue. (closes issue #15494) Reported by: makoto git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	69f7901fc0	Change the name of the category 'TEST' to match the name of the subdir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	66053b8a58	Don't act like an http write failed when it didn't fwrite returns the number of items written, not the number of bytes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ae6008ef3a	Change per-file debug and verbose levels to be per-module, the way users expect them to work. 'core set debug' and 'core set verbose' can optionally change the level for a specific filename; however, this is actually for a specific source file name, not the module that source file is included in. With examples like chan_sip, chan_iax2, chan_misdn and others consisting of multiple source files, this will not lead to the behavior that users expect. If they want to set the debug level for chan_sip, they want it set for all of chan_sip, and not to have to also set it for reqresp_parser and other files that comprise the chan_sip module. This patch changes this functionality to be module-name based instead of file-name based. To make this work, some Makefile modifications were required to ensure that the AST_MODULE definition is present in each object file produced for each module as well. Review: https://reviewboard.asterisk.org/r/574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f9e4d024c9	Initialize channels prior to loading "preload" modules. We can have bad results when a module, upon being loaded, attempts to reference the channels container if the container hasn't yet been initialized. I saw this happen by trying to preload pbx_config.so and having a hint defined which referenced a non-existent SIP peer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	8acef966ef	Merged revisions 253799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253799 \| mnicholson \| 2010-03-22 14:50:00 -0500 (Mon, 22 Mar 2010) \| 4 lines Unconditionally copy the caller's account code to the called party. (related to issue #16331) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	fdef7e5deb	Update query should be an UPDATE, not a SELECT. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	957dfc8c6d	Return the list for later manipulation. This fixes an issue with the update procedure. Debugging with mmichelson. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	589c397065	Accomodate equal signs in DSNs and add documentation, based upon mmichelson's feedback. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	008930a3f2	Fix memory corruption found by unit tests. ast_str_reset() was being called on a potentially uninitialized pointer. Valgrind is my hero, once again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	3da9f8ed19	Resolve more compiler warnings on FreeBSD. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	a5b4b429f1	Include sys/wait.h on FreeBSD to get the WEXITSTATUS() macro. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	33aa72d592	Resolve compiler warnings on FreeBSD. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	b6e5e180b9	Resolve a compiler warning on FreeBSD. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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19634 Commits (2de9cd0d38b2da1876c09fd1a7ea302bd3012bb0) All Branches Search

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