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2010-03-27corrections in gk interface, small fixes in call clearing.may6-67/+83
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255199 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-27We need to inclde sys/wait.h on OpenBSD to get WEXITSTATUS.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255158 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-27inotify support for pbx_spooltilghman1-4/+143
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: http://svn.digium.com/svn/asterisk/trunk@255117 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-26Replace some documentation from 1.6.x back into trunk.lmadsen1-0/+2
This documentation associated wth tlsbindaddr is still useful so lets synchronize it between trunk and 1.6.x branches. (issue #17054) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255066 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-26Update confusing documentation for tlsbindaddr.lmadsen1-4/+2
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: http://svn.digium.com/svn/asterisk/trunk@255021 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-26Work around a bug in dash on Ubuntu by checking the number of arguments ↵seanbright1-1/+3
before shift'ing. Reported and tested by pabelanger. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254976 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Use "local" instead of "system" header file inclusion.kpfleming33-89/+89
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: http://svn.digium.com/svn/asterisk/trunk@254931 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Fix a number of other build problems on Mac OS X.russell2-17/+21
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254884 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Merged revisions 254800 via svnmerge from qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@254802 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Resolve compiler warning on FreeBSD.russell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254801 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Fix chan_ooh323 so it works on Mac OS X, as well.russell4-8/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254799 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25chan_usbradio depends on alsa.russell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254718 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Blocked revisions 254714 via svnmergeqwell0-0/+0
........ 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: http://svn.digium.com/svn/asterisk/trunk@254715 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Bump cleancount due to ast_channel change.kpfleming1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254638 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Remove no-longer-used (and unsafe) field in ast_channel for linked lists.kpfleming1-1/+0
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: http://svn.digium.com/svn/asterisk/trunk@254637 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Get chan_ooh323 building again after recent build system changes.kpfleming1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254636 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Add unit test for testing ACL functionality.mmichelson1-0/+244
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: http://svn.digium.com/svn/asterisk/trunk@254557 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Merged revisions 254552 via svnmerge from mmichelson1-7/+176
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: http://svn.digium.com/svn/asterisk/trunk@254553 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Add new rtpsource options to the CHANNEL function.mmichelson1-3/+39
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: http://svn.digium.com/svn/asterisk/trunk@254551 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Recorded merge of revisions 254452 via svnmerge from mmichelson1-13/+37
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: http://svn.digium.com/svn/asterisk/trunk@254454 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Merged revisions 254451 via svnmerge from twilson1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@254453 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Improve handling of T.38 re-INVITEs that arrive before a T.38-capablekpfleming6-100/+207
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: http://svn.digium.com/svn/asterisk/trunk@254450 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25handle_speechset has 4 arguments.lmadsen1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@254446 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25remove unneeded explicit channel in dahdi ioctlstzafrir1-17/+13
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: http://svn.digium.com/svn/asterisk/trunk@254406 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-24Fix potential invalid reads that could occur in pbx.cmmichelson1-2/+21
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: http://svn.digium.com/svn/asterisk/trunk@254362 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-24Allow configuration of minsecs and nextaftercmd per mailbox.jpeeler3-10/+193
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: http://svn.digium.com/svn/asterisk/trunk@254321 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-24Merged revisions 254235 via svnmerge from jpeeler1-19/+21
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: http://svn.digium.com/svn/asterisk/trunk@254277 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23make 'core show settings' should show all settable directoriestzafrir1-0/+7
(closes issue #17086) Reported by: tzafrir Patches: asterisk_extra_settings_dirs.diff uploaded by tzafrir (license 46) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254162 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Put test output for a failure in a CDATA section in the XML results.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254159 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Exit native bridging early for greater timing accuracy with warningsjpeeler1-2/+5
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: http://svn.digium.com/svn/asterisk/trunk@254050 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Blocked revisions 254046 via svnmergeqwell0-0/+0
........ 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: http://svn.digium.com/svn/asterisk/trunk@254047 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Remove unused structure member in app_queue.seanbright1-2/+0
(closes issue #15494) Reported by: makoto git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254045 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Change the name of the category 'TEST' to match the name of the subdirtzafrir1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254001 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Don't act like an http write failed when it didn'ttwilson1-1/+1
fwrite returns the number of items written, not the number of bytes git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253958 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Change per-file debug and verbose levels to be per-module, the waykpfleming10-84/+115
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: http://svn.digium.com/svn/asterisk/trunk@253917 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-22Initialize channels prior to loading "preload" modules.mmichelson1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@253872 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-22Merged revisions 253799 via svnmerge from mnicholson1-3/+1
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: http://svn.digium.com/svn/asterisk/trunk@253800 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-22Update query should be an UPDATE, not a SELECT.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253758 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-22Return the list for later manipulation. This fixes an issue with the update ↵tilghman1-1/+1
procedure. Debugging with mmichelson. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253755 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-22Accomodate equal signs in DSNs and add documentation, based upon ↵tilghman2-2/+3
mmichelson's feedback. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253712 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Fix memory corruption found by unit tests.russell1-1/+1
ast_str_reset() was being called on a potentially uninitialized pointer. Valgrind is my hero, once again. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253579 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Resolve more compiler warnings on FreeBSD.russell8-7/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253540 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Include sys/wait.h on FreeBSD to get the WEXITSTATUS() macro.russell1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253539 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Resolve compiler warnings on FreeBSD.russell2-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Resolve a compiler warning on FreeBSD.russell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253537 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Use SHRT_MAX instead of MAXSHORT.russell1-3/+2
These changes fix build issues I had with this module on FreeBSD. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253536 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-19prevent segfault if bad magic number is encountered.alecdavis1-1/+4
internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic number', but internal_ao2_ref continues on, causing segfault. Although AO2_MAGIC number is checked by INTERNAL_OBJ before internal_ao2_ref is called, A02_MAGIC is being destroyed (or a wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ. internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad magic number. (issue #17037) Reported by: alecdavis Patches: bug17037.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253490 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-18Update comment to reflect new timeout value.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253378 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-18Increase CLI command output timeout for asterisk -rx to 60 seconds.russell1-1/+1
(closes issue #17049) Reported by: russell Tested by: russell Review: https://reviewboard.asterisk.org/r/573/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253357 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-18Change usage of pipe to comma in UserEvent docs.lmadsen1-1/+1
Change the example usage of pipe as a separator to comma in the UserEvent documentation. (closes issue #16961) Reported by: jlpedrosa git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253345 f38db490-d61c-443f-a65b-d21fe96a405b
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