34 Commits (71c19043bd88e3196318d657124cd8fa54138041)

Author	SHA1	Message	Date
Jason Parker	5e65377992	Multiple revisions 377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772 ... ........ r377136 | rmudgett | 2012-12-03 14:33:08 -0600 (Mon, 03 Dec 2012) | 17 lines Cleanup core main on exit. * Cleanup time zones on exit. * Make exit clean/unclean report consistent for AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch uploaded by Corey Farrell core-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377135 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377166 | rmudgett | 2012-12-03 16:53:58 -0600 (Mon, 03 Dec 2012) | 15 lines Cleanup ast_run_atexits() atexits list. * Convert atexits list to a mutex instead of a rd/wr lock. The lock is only write locked. * Move CLI verbose Asterisk ending message to where AMI message is output in really_quit() to avoid further surprises about using stuff already shutdown. (issue ASTERISK-20649) Reported by: Corey Farrell ........ Merged revisions 377165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377212 | rmudgett | 2012-12-04 16:31:02 -0600 (Tue, 04 Dec 2012) | 1 line confbridge: Update online XML documentation. ........ r377227 | rmudgett | 2012-12-04 18:49:53 -0600 (Tue, 04 Dec 2012) | 29 lines confbridge: Fix several small issues. * Made func_confbridge_helper() allow an empty value when setting options. You previously could not Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the dialplan. * Made func_confbridge_helper() handle its datastore better if multiple threads attempt to set the first CONFBRIDGE option value on the channel. * Made the func_confbridge_helper() only output one diagnostic message concerning the option. * Made the bridge video_mode able to repeatedly change in the config file and CONFBRIDGE dialplan function. The video_mode option values are an enum and not independent of each other. * Made handle_cli_confbridge_show_bridge_profile() better handle the video_mode option. * Simplified datastore handling code in conf_find_user_profile() and conf_find_bridge_profile(). * Made parse_bridge(), parse_user(), and parse_menu() use var->file instead of CONFBRIDGE_CONFIG because the var could have been from an include file. (closes issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter ........ r377241 | rmudgett | 2012-12-04 20:09:13 -0600 (Tue, 04 Dec 2012) | 4 lines * Fix registering core show codecs/codec CLI commands twice. * Fix registering atexit format_attr_shutdown() more than once. ........ r377258 | file | 2012-12-05 10:49:33 -0600 (Wed, 05 Dec 2012) | 19 lines Fix a SIP request memory leak with TLS connections. During the TLS re-work in chan_sip some TLS specific code was moved into a separate function. This function operates on a copy of the incoming SIP request. This copy was never deinitialized causing a memory leak for each request processed. This function is now given a SIP request structure which it can use to copy the incoming request into. This reduces the amount of memory allocations done since the internal allocated components are reused between packets and also ensures the SIP request structure is deinitialized when the TLS connection is torn down. (closes issue ASTERISK-20763) Reported by: deti ........ Merged revisions 377257 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377261 | jrose | 2012-12-05 10:57:26 -0600 (Wed, 05 Dec 2012) | 15 lines res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session When srtp_create fails, the session may be dealloced or just not alloced. At the same time though, the session pointer might not be set to NULL in this process and attempting to srtp_dealloc it again will cause a segfault. This patch checks for failure of srtp_create and sets the session pointer to NULL if it fails. (closes issue ASTERISK-20499) Reported by: tootai Review: https://reviewboard.asterisk.org/r/2228/ ........ Merged revisions 377256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377354 | rmudgett | 2012-12-06 17:56:45 -0600 (Thu, 06 Dec 2012) | 24 lines confbridge: Fix some resource leaks on conference teardown. * Made destroy_conference_bridge() destroy a missed ast_mutex_t and ast_cond_t. * Made join_conference_bridge() init the ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can destroy them unconditionally. * Made join_conference_bridge() abort if the new conference could not be added to the conferences container. * Made leave_conference() discard any post-join actions if join_conference_bridge() had to abort early. * Made the join_conference_bridge() diagnostic messages better describe what happened. * Renamed leave_conference_bridge() to leave_conference() and made it only take a conference user pointer. The conference pointer was redundant. * Made conf_bridge_profile_copy() use struct copy instead of memcpy(). * No need to lock the conference in start_conf_record_thread() since all of the callers already have it locked. ........ r377382 | kmoore | 2012-12-07 15:58:21 -0600 (Fri, 07 Dec 2012) | 17 lines codec_dahdi: Fix output of "transcoder show" CLI command. In r306010 "Asterisk media architecture conversion - no more format bitfields", the logic for incrementing encoders and decoders when opening transcoder channels was changed without making the corresponding change when decrementing encoder / decoder channels. The result being that when a channel was destroyed, codec_dahdi couldn't properly tell if it was an encoder or decoder, and the default case is to assume it was a decoder. This could result in negative numbers for decoders in use like in: VOIP6*CLI> transcoder show 2/-2 encoders/decoders of 92 channels are in use. (closes issue ASTERISK-19921) Patch-by: Shaun Ruffell ........ r377399 | rmudgett | 2012-12-07 17:42:03 -0600 (Fri, 07 Dec 2012) | 5 lines MALLOC_DEBUG: Only wait if we want atexit allocation dumps. ........ Merged revisions 377398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377432 | rmudgett | 2012-12-07 18:29:23 -0600 (Fri, 07 Dec 2012) | 14 lines Fix order of SIP allow/disallow in MySQL contrib script. Using the contrib sippeers.sql script to create the sippeers MySQL table would result in being unable to place calls if you set the disallow value to all. (closes issue ASTERISK-20756) Reported by: Andre Luis Patches: sippeers.patch patch uploaded by Andre Luis ........ Merged revisions 377431 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377504 | tilghman | 2012-12-09 19:24:41 -0600 (Sun, 09 Dec 2012) | 5 lines Remove some dead code and additionally handle a case that wasn't handled. ........ Merged revisions 377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377510 | tilghman | 2012-12-09 19:39:58 -0600 (Sun, 09 Dec 2012) | 16 lines Improve documentation by making all of the colors used readable, no matter what the background color is. Dark blue on a black background is unreadable, as is yellow on a light background. This patch turns on the bright attribute for colors when on a dark background and turns *off* the bright attribute when the -W command line option is used (indicating a _light_ background). This ensures that text is readable in both cases. Patch by: tilghman Review: https://reviewboard.asterisk.org/r/2224 ........ Merged revisions 377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377558 | igorg | 2012-12-09 23:04:36 -0600 (Sun, 09 Dec 2012) | 8 lines Fix crash on transfer initiated from insreeen menu on Unistim phones. Removed CDR-related code that moved to do_masquarade before. (closes issue ASTERISK-20417) Reported by: Rudolf Migalin ........ Merged revisions 377557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377592 | igorg | 2012-12-10 00:41:47 -0600 (Mon, 10 Dec 2012) | 9 lines Fix codec mismatch Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations. (issue ASTERISK-20183) ........ Merged revisions 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377624 | kmoore | 2012-12-10 08:40:26 -0600 (Mon, 10 Dec 2012) | 14 lines Handle Session-Expires less than local Min-SE in 200 OK Ensure that a call is immediately torn down if a Session-Expires value received in a 200 OK is less than the local Min-SE. This also prevents Asterisk from allowing calls with Session-Expires below the RFC4028-mandated minimum (90s). (closes issue ASTERISK-20653) Review: https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore ........ Merged revisions 377623 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377656 | kmoore | 2012-12-10 10:53:16 -0600 (Mon, 10 Dec 2012) | 14 lines Ensure ReceiveFax provides a CED tone via T.38 When using res_fax_digium, the T.38 CED tone was not being provided properly which would cause some incoming faxes to fail. This was not an issue with res_fax_spandsp since it does not strictly honor the send_ced flag and sends the CED tone whenever receiving a T.38 fax. (closes issue FAX-343) Reported-by: Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377705 | rmudgett | 2012-12-10 18:32:40 -0600 (Mon, 10 Dec 2012) | 14 lines Cleanup dnsmgr on exit. * Cleanup dnsmgr thread and CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377709 | rmudgett | 2012-12-10 19:00:05 -0600 (Mon, 10 Dec 2012) | 15 lines Cleanup event on exit. * Cleanup CLI commands on exit. * v10 only: Merged v1.8 -r374177 change to event.c missed in v10 -r374178. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: event_shutdown-10-only.patch (license #5909) patch uploaded by Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 377708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377741 | rmudgett | 2012-12-10 20:11:29 -0600 (Mon, 10 Dec 2012) | 19 lines Cleanup indications on exit. * Made ast_unregister_indication_country() unlink the found tone zone before selecting a new default_tone_zone to make it impossible to select the tone zone being unregistered again. * Ringcadence is no longer parsed twice in store_config_tone_zone(). * Cleanup CLI commands and destroy default_tone_zone on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377740 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377772 | rmudgett | 2012-12-10 20:42:34 -0600 (Mon, 10 Dec 2012) | 13 lines Cleanup logger on exit. * Cleanup CLI commands, destroy verbosers and logchannels lists on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: logger-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@378659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	13b65b5dce	Merged revisions 376431 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r376431 | mjordan | 2012-11-18 14:18:24 -0600 (Sun, 18 Nov 2012) | 49 lines Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@376446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	8aef9fdedd	Merged revisions 369005 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369005 | kpfleming | 2012-06-15 11:07:08 -0500 (Fri, 15 Jun 2012) | 22 lines Multiple revisions 369001-369002 ........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	b0e27a76f0	Multiple revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 ... ........ r361208 | jrose | 2012-04-04 14:30:09 -0500 (Wed, 04 Apr 2012) | 10 lines Make 'help devstate change' display properly (get rid of excess comma) (closes issue ASTERISK-19444) Reported by: Makoto Dei Patches: devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027) ........ Merged revisions 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361211 | jrose | 2012-04-04 15:00:23 -0500 (Wed, 04 Apr 2012) | 12 lines Fix some stuff involving calls to memcpy and memset The important parts of the patch were already applied through other updates. (closes issue ASTERISK-19445) Reported by: Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto Dei (license 5027) ........ Merged revisions 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361270 | jrose | 2012-04-05 11:53:35 -0500 (Thu, 05 Apr 2012) | 10 lines Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined There were a few instances of restarting music on hold in meetme that would cause Asterisk to revert to the default class of music on hold for no adequate reason. Review: https://reviewboard.asterisk.org/r/1844/ ........ Merged revisions 361269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361330 | kmoore | 2012-04-06 08:31:51 -0500 (Fri, 06 Apr 2012) | 11 lines Remove unnecessary error message in app_dial.c The error message for failure to stop autoservice after a gosub or macro call during a dial was removed for macro while Asterisk 1.4 was still being actively developed. The corresponding gosub error message was never removed. (closes issue ASTERISK-19551) ........ Merged revisions 361329 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361333 | mjordan | 2012-04-06 09:01:33 -0500 (Fri, 06 Apr 2012) | 11 lines Fix a typo in the warning messages for an ignored media stream Added a '\n' to the warning messages when we ignore a media stream due to the port number being '0'. (closes issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........ Merged revisions 361332 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361381 | russell | 2012-04-06 10:49:19 -0500 (Fri, 06 Apr 2012) | 5 lines Remove a few more files related to chan_usbradio and app_rpt. ........ Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361422 | pabelanger | 2012-04-06 11:31:18 -0500 (Fri, 06 Apr 2012) | 14 lines Multiple revisions 361403,361412 ........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412 | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ Merged revisions 361403,361412 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361472 | kmoore | 2012-04-06 13:13:04 -0500 (Fri, 06 Apr 2012) | 5 lines Add missing newlines to CLI logging ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361522 | rmudgett | 2012-04-06 14:47:29 -0500 (Fri, 06 Apr 2012) | 8 lines Don't add an empty MESSAGE_DATA(key) header if it doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key header did not already exist. If it already existed it would delete it. * Made msg_set_var_full() exit early if the named variable did not already exist and the value to set is empty. ........ r361560 | mjordan | 2012-04-06 15:32:13 -0500 (Fri, 06 Apr 2012) | 13 lines Fix memory leak when using MeetMeAdmin 'e' option with user specified A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command (eject last user that joined) is used in conjunction with a specified user. Regardless of the command being executed, if a user is specified for the command, MeetMeAdmin will look up that user. Because the 'e' option kicks the last user that joined, as opposed to the one specified, the reference to the user specified by the command would be leaked when the user variable was assigned to the last user that joined. ........ Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361607 | mjordan | 2012-04-06 17:00:11 -0500 (Fri, 06 Apr 2012) | 12 lines Fix memory leak in res_calendar_ews when event email address node is empty If the XML calendar data returned by a Microsoft Exchange Web Service specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address is provided, a condition existed where an ast_calendar_attendee struct would be allocated but not appended to the list of attendees. Because of that, the memory associated with the attendee would never be freed. This patch frees the memory if no e-mail address is provided. ........ Merged revisions 361606 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361658 | mjordan | 2012-04-09 14:42:53 -0500 (Mon, 09 Apr 2012) | 15 lines Change SHARED function to use a safe traversal when modifying a variable When the SHARED function modifies a variable, it removes it from its list of variables and reinserts the new value at the head of the list of variables. Doing this inside a standard list traversal can be dangerous, as the standard list traversal does not account for the list being changed. While the code in question should not cause a use after free violation due to its breaking out of the loop after freeing the variable, it could lead to a maintenance issue if the loop was modified. This also fixes a violation reported by a static analysis tool, which also makes this code easier to maintain in the future. ........ Merged revisions 361657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361706 | mjordan | 2012-04-09 15:54:55 -0500 (Mon, 09 Apr 2012) | 17 lines Prevent invalid access of free'd memory if DAHDI channel during an MWI event In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. This patch makes it so that we only free the caller ID structure if a DAHDI channel is successfully created, and we bump the gains back up if we fail to make a DAHDI channel. ........ Merged revisions 361705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361754 | mjordan | 2012-04-09 16:44:30 -0500 (Mon, 09 Apr 2012) | 12 lines Allow func_curl to exit gracefully if list allocation fails during write If the global_curl_info data structure could not be allocated, the datastore associated with the operation would be free'd, but the function would not return. This would later dereference the datastore, almost certainly causing Asterisk to crash. With this patch, if the data structure is not allocated the method will return an error code, and not attempt any further operation. ........ Merged revisions 361753 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361804 | mjordan | 2012-04-10 14:57:30 -0500 (Tue, 10 Apr 2012) | 10 lines Fix crash caused by unloading or reloading of res_http_post When unlinking itself from the registered HTTP URIs, res_http_post could inadvertently free all URIs registered with the HTTP server. This patch modifies the unregister method to only free the URI that is actually being unregistered, as opposed to all of them. ........ Merged revisions 361803 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361855 | rmudgett | 2012-04-10 16:47:42 -0500 (Tue, 10 Apr 2012) | 19 lines Prevent invalid access of free'd memory if DAHDI channel during an MWI event In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. * Rework the -r361705 patch to better manage the cs and mtd allocated resources. * Fixed use of mwimonitoractive flag to be correct if the mwi_thread() fails to start. ........ Merged revisions 361854 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361907 | jrose | 2012-04-11 11:07:50 -0500 (Wed, 11 Apr 2012) | 10 lines Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8 Prior to this patch, in order to restore that behavior, a function would have to be used on the QueueMember to make the ringinuse option do anything, which is pretty unreasonable. (closes issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1860/ ........ r361956 | kmoore | 2012-04-12 10:01:13 -0500 (Thu, 12 Apr 2012) | 13 lines Simplify build system architecture optimization This change to the build system rips out any usage of PROC along with architecture-specific optimizations in favor of using -march=native where it is supported. This fixes broken builds on 64bit Intel systems and results in better optimized code on systems running GCC 4.2+. Review: https://reviewboard.asterisk.org/r/1852/ (closes issue ASTERISK-19462) ........ Merged revisions 361955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r361981 | kmoore | 2012-04-12 11:22:28 -0500 (Thu, 12 Apr 2012) | 12 lines Make trunkfreq take effect when set Previously, setting trunkfreq had no effect on initial load or on reload and only ever used the default value. This causes trunkfreq to be used appropriately on initial load and reload. (closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........ Merged revisions 361972 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362080 | jrose | 2012-04-13 10:30:22 -0500 (Fri, 13 Apr 2012) | 10 lines Send relative path named recordings to the meetme directory instead of sounds Prior to this patch, no effort was made to parse the path name to determine a proper destination for recordings of MeetMe's r option. This fixes that. Review: https://reviewboard.asterisk.org/r/1846/ ........ Merged revisions 362079 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362084 | jrose | 2012-04-13 11:04:22 -0500 (Fri, 13 Apr 2012) | 15 lines Make ForkCDR e option not set end time of the newly forked CDR log Prior to this patch, ForkCDR's e option would immediately set the end time of the forked CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time being roughly the same as it's beginning time (which is in turn roughly the same as the original's end time). (closes issue ASTERISK-19164) Reported by: Steve Davies Patches: cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012) ........ Merged revisions 362082 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362152 | mjordan | 2012-04-16 14:39:32 -0500 (Mon, 16 Apr 2012) | 19 lines Check for IO stream failures in various format's truncate/seek operations For the formats that support seek and/or truncate operations, many of the C library calls used to determine or set the current position indicator in the file stream were not being checked. In some situations, if an error occurred, a negative value would be returned from the library call. This could then be interpreted inappropriately as positional data. This patch checks the return values from these library calls before using them in subsequent operations. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362151 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362202 | mjordan | 2012-04-16 16:40:29 -0500 (Mon, 16 Apr 2012) | 18 lines Fix handling of negative return code when storing voicemails in ODBC storage When storing a voicemail message using an ODBC connection to a database, the voicemail message is first stored on disk. The sound file associated with the message is read into memory before being transmitted to the database. When this occurs, a failure in the C library's lseek function would cause a negative value to be passed to the mmap as the size of the memory map to create. This would almost certainly cause the creation of the memory map to fail, resulting in the message being lost. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863 ........ Merged revisions 362201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362205 | mjordan | 2012-04-16 16:57:19 -0500 (Mon, 16 Apr 2012) | 25 lines Fix negative return handling in channel drivers In chan_agent, while handling a channel indicate, the agent channel driver must obtain a lock on both the agent channel, as well as the channel the agent channel is using. To do so, it attempts to lock the other channel first, then unlock the agent channel which is locked prior to entry into the indicate handler. If this unlock fails with a negative return value, which can occur if the object passed to agent_indicate is an invalid ao2 object or is NULL, the return value is passed directly to strerror, which can only accept positive integer values. In chan_dahdi, the return value of dahdi_get_index is used to directly index into the sub-channel array. If dahd_get_index returns a negative value, it would use that value to index into the array, which could cause an invalid memory access. If dahdi_get_index returns a negative number, we now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362204 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362264 | elguero | 2012-04-17 09:53:04 -0500 (Tue, 17 Apr 2012) | 23 lines Turn off warning message when bind address is set to any. When a bind address is set to an ANY address (udpbindport=::), a warning message is displayed stating that "Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove 'localnet' and/or 'externaddr' settings." But if one is running dual stack, we shouldn't be told to turn those settings off. This patch checks if the bind address is an ANY address or not. The warning message will now only be displayed if the bind address is NOT an ANY address and IPv6 is being used. Also, updated the copyright year. (closes issue ASTERISK-19456) Reported by: Michael L. Young Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 362253 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362305 | mjordan | 2012-04-17 13:27:44 -0500 (Tue, 17 Apr 2012) | 15 lines Fix error that caused seek format operations to set max file size to '1' or '0' A very inappropriate placement of a ')' (introduced in r362151) caused the maximum size of a file to be set as the result of a comparison operation, as opposed to the result of the ftello operation. This resulted in seeking being restricted to the beginning of the file, or 1 byte into the file. Thanks to the Asterisk Test Suite for properly freaking out about this on at least one test. (issue ASTERISK-19655) Reported by: Matt Jordan ........ Merged revisions 362304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362356 | mjordan | 2012-04-17 15:56:05 -0500 (Tue, 17 Apr 2012) | 17 lines Fix places where a negative return from ftello could be used as invalid input In a variety of locations in both reading and writing a file, the result from the C library function ftello is used as input to other functions. For the parameters and functions in question, a negative value is invalid input. This patch checks the return value from the ftello function to determine if we were able to determine the current position in the file stream and, if not, fail gracefully. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362355 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362357 | jrose | 2012-04-17 15:57:36 -0500 (Tue, 17 Apr 2012) | 12 lines Make use of va_args more appropriate to form in various res_config modules plus utils. A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad. va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy. The invokers of those functions are responsible for calling va_end on them. (issue ASTERISK-19451) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1848/ ........ Merged revisions 362354 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362360 | mjordan | 2012-04-17 16:07:29 -0500 (Tue, 17 Apr 2012) | 24 lines Fix places in main where a negative return value could impact execution This patch addresses a number of modules in main that did not handle the negative return value from function calls adequately, or were not sufficiently clear that the conditions leading to improper handling of the return values could not occur. This includes: * asterisk.c: A negative return value from the read function would be used directly as an index into a buffer. We now check for success of the read function prior to using its result as an index. * manager.c: Check for failures in mkstemp and lseek when handling the temporary file created for processing data returned from a CLI command in action_command. Also check that the result of an lseek is sanitized prior to using it as the size of a memory map to allocate. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362364 | mjordan | 2012-04-17 16:11:25 -0500 (Tue, 17 Apr 2012) | 29 lines Fix places in resources where a negative return value could impact execution This patch addresses a number of modules in resources that did not handle the negative return value from function calls adequately. This includes: * res_agi.c: if the result of the read function is a negative number, indicating some failure, the result would instead be treated as the number of bytes read. This patch now treats negative results in the same manner as an end of file condition, with the exception that it also logs the error code indicated by the return. * res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd, and instead assigns a negative value, that file descriptor could later be passed to functions that require a valid file descriptor. If spawn_mp3 fails, we now immediately retry instead of continuing in the logic. * res_rtp_asterisk.c: if no codec can be matched between two RTP instances in a peer to peer bridge, we immediately return instead of attempting to use the codec payload type as an index to determine the appropriate negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362377 | mjordan | 2012-04-17 16:22:37 -0500 (Tue, 17 Apr 2012) | 13 lines Handle case where an unknown format is used to get the preferred codec size In ast_codec_pref_getsize, if an unknown format is passed to the method, no preferred codec will be selected and a negative number will be used to index into the format list. The method now logs an unknown format as a warning, and returns an empty format list. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ r362429 | rmudgett | 2012-04-18 11:27:51 -0500 (Wed, 18 Apr 2012) | 19 lines Add ability to ignore layer 1 alarms for BRI PTMP lines. Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362496 | mjordan | 2012-04-18 21:27:08 -0500 (Wed, 18 Apr 2012) | 50 lines Fix a variety of potential buffer overflows * chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * format_pref: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan ........ Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362537 | twilson | 2012-04-19 09:31:59 -0500 (Thu, 19 Apr 2012) | 14 lines Handle multiple commands per connection via netconsole Asterisk would accept multiple NULL-delimited CLI commands via the netconsole socket, but would occasionally miss a command due to the command not being completely read into the buffer. This patch ensures that any partial commands get moved to the front of the read buffer, appended to, and properly sent. (closes issue ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 362536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362587 | seanbright | 2012-04-19 11:04:21 -0500 (Thu, 19 Apr 2012) | 12 lines Prevent a crash in ExternalIVR when the 'S' command is sent first. If the first command sent from an ExternalIVR client is an 'S' command, we were blindly removing the first element from the play list and deferencing it, even if it was NULL. This corrects that and also locks appropriately in one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski ........ Merged revisions 362586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362678 | rmudgett | 2012-04-19 16:00:21 -0500 (Thu, 19 Apr 2012) | 5 lines Update membermacro and membergosub documentation in queues.conf.sample. ........ Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362681 | elguero | 2012-04-19 16:11:35 -0500 (Thu, 19 Apr 2012) | 9 lines Add leading and trailing backslashes A couple of unit tests did not have have leading or trailing backslashes when setting their test category resulting in a warning message being displayed. Added the backslash where needed. ........ Merged revisions 362680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362730 | wdoekes | 2012-04-19 16:59:43 -0500 (Thu, 19 Apr 2012) | 5 lines Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362816 | twilson | 2012-04-20 09:49:42 -0500 (Fri, 20 Apr 2012) | 13 lines Document Speech* apps hangup on failure and suggest TryExec The Speech API apps return -1 on failure, which will hang up the channel. This may not be desirable behavior for some, but it isn't something that can be changed without breaking people's dialplans or writing an option to all of the Speech apps that does what TryExec already does. This patch documents the hangup behavior of the apps, and suggests TryExec as the solution. (closes issue AST-813) ........ Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362869 | twilson | 2012-04-20 11:12:34 -0500 (Fri, 20 Apr 2012) | 11 lines OpenBSD doesn't have rawmemchr, use strchr (closes issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller (license 5434) ........ Merged revisions 362868 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r362918 | elguero | 2012-04-20 11:47:51 -0500 (Fri, 20 Apr 2012) | 11 lines Add missing payload type to events API The Security Events Framework API was changed while adding the generation of security events in chan_sip. A payload type and name was missed from being added to struct ie_maps. (closes issue ASTERISK-19759) Reported by: Michael L. Young Patches: issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026) ........ r362998 | rmudgett | 2012-04-20 20:45:13 -0500 (Fri, 20 Apr 2012) | 5 lines Update app_dial M and U option GOTO return value documentation. ........ Merged revisions 362997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363212 | tilghman | 2012-04-23 11:06:53 -0500 (Mon, 23 Apr 2012) | 8 lines On some platforms, O_RDONLY is not a flag to be checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX specification does not mandate how these 3 flags must be specified, only that one of the three must be specified in every call. ........ Merged revisions 363209 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363376 | rmudgett | 2012-04-24 19:01:21 -0500 (Tue, 24 Apr 2012) | 5 lines Hangup affected channel in error paths of bridge_call_thread(). ........ Merged revisions 363375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363429 | rmudgett | 2012-04-24 20:23:08 -0500 (Tue, 24 Apr 2012) | 27 lines Fix recalled party B feature flags for a failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to C 3) B hangs up 4) C does not answer 5) B is called back 6) B answers 7) B cannot initiate transfers anymore * Add dial features datastore to recalled party B channel that is a copy of the original party B channel's dial features datastore. * Extracted add_features_datastore() from add_features_datastores(). * Renamed struct ast_dial_features features_caller and features_callee members to my_features and peer_features respectively. These better names eliminate the need for some explanatory comments. * Simplified code accessing the struct ast_dial_features datastore. (closes issue ASTERISK-19383) Reported by: lgfsantos ........ Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363688 | rmudgett | 2012-04-25 14:47:44 -0500 (Wed, 25 Apr 2012) | 19 lines Clear ISDN channel resetting state if the peer continues to use it. Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in response to a RESTART request. * Made the second SETUP received after sending a RESTART request clear the channel resetting state as if the peer had sent the expected RESTART ACKNOWLEDGE before continuing to process the SETUP. The peer may not be sending the expected RESTART ACKNOWLEDGE. (issue ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified) ........ Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363734 | rmudgett | 2012-04-25 15:48:22 -0500 (Wed, 25 Apr 2012) | 18 lines Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call. Some switches may not handle the call-deflection/call-rerouting message if the call is disconnected too soon after being sent. Asteisk was not waiting for any reply before disconnecting the call. * Added a 5 second delay before disconnecting the call to wait for a potential response if the peer does not disconnect first. (closes issue ASTERISK-19708) Reported by: mehdi Shirazi Patches: jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363789 | rmudgett | 2012-04-25 17:59:46 -0500 (Wed, 25 Apr 2012) | 5 lines Update Pickup application documentation. ........ Merged revisions 363788 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363876 | rmudgett | 2012-04-25 22:11:45 -0500 (Wed, 25 Apr 2012) | 5 lines Update Pickup application documentation. (Even better) ........ Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363935 | alecdavis | 2012-04-26 04:46:38 -0500 (Thu, 26 Apr 2012) | 14 lines chan_sip: [general] maxforwards, not checked for a value greater than 255 The peer maxforwards is checked for both '< 1' and '> 255', but the default 'maxforwards' in the [general] section is only checked for '< 1' alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1888/ ........ Merged revisions 363934 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r363987 | kmoore | 2012-04-26 08:27:34 -0500 (Thu, 26 Apr 2012) | 15 lines Fix reference leaks involving SIP Replaces transfers The reference held for SIP blind transfers using the Replaces header in an INVITE was never freed on success and also failed to be freed in some error conditions. This caused a file descriptor leak since the RTP structures in use at the time of the transfer were never freed. This reference leak and another relating to subscriptions in the same code path have now been corrected. (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski Tested by: Maciej Karjewski ........ Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364047 | twilson | 2012-04-26 14:30:55 -0500 (Thu, 26 Apr 2012) | 8 lines Add more constness to the end_buf pointer in the netconsole issue ASTERISK-18308 Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 364046 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364065 | rmudgett | 2012-04-26 15:25:05 -0500 (Thu, 26 Apr 2012) | 24 lines Fix DTMF atxfer running h exten after the wrong bridge ends. When party B does an attended transfer of party A to party C, the attending bridge between party B and C should not be running an h exten when the bridge ends. Running an h exten now sets a softhangup flag to ensure that an AGI will run in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the attending bridge between party B and C. (closes issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario ........ Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364109 | rmudgett | 2012-04-26 16:10:46 -0500 (Thu, 26 Apr 2012) | 5 lines Update Pickup application documentation. (With feeling this time.) ........ Merged revisions 364108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364163 | schmidts | 2012-04-27 07:54:19 -0500 (Fri, 27 Apr 2012) | 3 lines fix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt use utc. This change uses the same timezone from the start time. ........ r364204 | mjordan | 2012-04-27 09:44:13 -0500 (Fri, 27 Apr 2012) | 23 lines Allow for reloading SRTP crypto keys within the same SIP dialog As a continuation of the patch in r356604, which allowed for the reloading of SRTP keys in re-INVITE transfer scenarios, this patch addresses the more common case where a new key is requested within the context of a current SIP dialog. This can occur, for example, when certain phones request a SIP hold. Previously, once a dialog was associated with an SRTP object, any subsequent attempt to process crypto keys in any SDP offer - either the current one or a new offer in a new SIP request - were ignored. This patch changes this behavior to only ignore subsequent crypto keys within the current SDP offer, but allows future SDP offers to change the keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Review: https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364259 | kmoore | 2012-04-27 13:58:34 -0500 (Fri, 27 Apr 2012) | 14 lines Allow SIP pvts involved in Replaces transfers to fall out of reference sooner Unref the SIP pvt stored in the refer structure as soon as it is no longer needed so that the pvt and associated file descriptors can be freed sooner. This change makes a reference decrement unnecessary in code that handles SIP BYE/Also transfers which should not touch the reference anyway. (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski Tested by: Maciej Krajewski ........ Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364285 | mjordan | 2012-04-27 14:30:19 -0500 (Fri, 27 Apr 2012) | 43 lines Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds, between two timeval structs, and return the difference in a 64-bit integer. Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval struct are large enough to hold the calculated values before it returns. On 64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit machines, however, a long may be less (32-bits), in which case, the calculation can overflow. This overflow caused significant problems in MixMonitor, which uses the method to determine if an audio factory, which has not presented audio to an audiohook, is merely late in providing said audio or will never provide audio. In an overflow situation, the audiohook would incorrectly determine that an audio factory that will never provide audio is merely late instead. This led to situations where a MixMonitor never recorded any audio. Note that this happened most frequently when that MixMonitor was started by the ConfBridge application itself, or when the MixMonitor was attached to a Local channel. (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski Tested by: Michael L. Young Patches: 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) (closes issue ASTERISK-19471) Reported by: feyfre Tested by: feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1889/ ........ Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364342 | mmichelson | 2012-04-27 16:58:06 -0500 (Fri, 27 Apr 2012) | 10 lines Don't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails. (closes issue ASTERISK-18321) Reported by Dan Lukes Patches: ASTERISK-18321.patch by Mark Michelson (license #5049) ........ Merged revisions 364341 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) | 11 lines Fix ast_parse_arg numeric type range checking and add tests ast_parse_arg wasn't checking for strto* parse errors or limiting the results by the actual range of the numeric types. This patch fixes that and adds unit tests as well. Review: https://reviewboard.asterisk.org/r/1879/ ........ Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines Add missing test_config.c ........ r364536 | elguero | 2012-04-28 21:21:10 -0500 (Sat, 28 Apr 2012) | 13 lines Fix configuring custom sound_leader_has_left in confbridge.conf The configuration option to specify a custom sound_leader_has_left file for a conference bridge was not being parsed. This patch fixes it so that a custom sound file will now be used. (closes issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380) Review: https://reviewboard.asterisk.org/r/1884/ ........ r364579 | mjordan | 2012-04-29 14:43:53 -0500 (Sun, 29 Apr 2012) | 15 lines Fix error that caused truncate operations to fail Another very inappropriate placement of a ')' (again introduced in r362151) caused the various truncate operations to attempt to truncate the sound file at a position of '0'. (issue ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec ........ Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364650 | markm | 2012-04-30 11:43:11 -0500 (Mon, 30 Apr 2012) | 15 lines Merged revisions 364635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | 10 lines Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs (closes issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark Murawski ........ ........ r364651 | may | 2012-04-30 11:48:57 -0500 (Mon, 30 Apr 2012) | 10 lines Fix use freed pointer in return value from call thread (issue ASTERISK-19663) Reported by: Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415) ........ Merged revisions 364649 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364777 | jrose | 2012-05-01 13:23:08 -0500 (Tue, 01 May 2012) | 13 lines Fix bad check in voicemail functions for ast_inboxcount2_func Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes issue ASTERISK-19718) Reported by: Corey Farrell Patches: ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364787 | kmoore | 2012-05-01 14:07:09 -0500 (Tue, 01 May 2012) | 12 lines Play conf-placeintoconf message to the correct channel Correct the code in app_confbridge to play the conf-placeintoconf message to the marked user entering the bridge instead of to the conference while the marked user hears silence. (closes issue ASTERISK-19641) Reported-by: Mark A Walters ........ Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364845 | rmudgett | 2012-05-01 16:50:32 -0500 (Tue, 01 May 2012) | 7 lines * Fix error path resouce leak in local_request(). * Restructure local_request() to reduce indentation. ........ Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364900 | mmichelson | 2012-05-01 18:10:16 -0500 (Tue, 01 May 2012) | 16 lines Fix Coverity-reported ARRAY_VS_SINGLETON error. As it turned out, this wasn't a huge deal. We were calling ast_app_parse_options() for a set of options of which none took arguments. The proper thing to do for this case is to pass NULL for the "args" parameter here. We were instead passing a seemingly-randomly chosen char * from the function. While this would never get written to, you can rest assured things would have gotten bad had new options (which took arguments) been added to func_volume. (closes issue ASTERISK-19656) ........ Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364903 | rmudgett | 2012-05-01 18:14:12 -0500 (Tue, 01 May 2012) | 10 lines Fixed __ao2_ref() validating user_data twice. (closes issue ASTERISK-19755) Reported by: Gunther Kelleter Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter ........ Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364965 | mjordan | 2012-05-01 21:44:15 -0500 (Tue, 01 May 2012) | 11 lines Only log a failure to get read/write samples from factories if it didn't happen In audiohook_read_frame_both, anytime samples are obtained from the read/write factories a debug statement is logged stating that samples were not obtained from the factories. This statement used to only occur if option_debug was turned on and no samples were obtained; in some refactoring when the option_debug statement was removed, the "else" clause was removed as well. This patch makes it so that those debug log statements only occur if the condition leading up to them actually happened. ........ r365014 | elguero | 2012-05-02 11:16:03 -0500 (Wed, 02 May 2012) | 18 lines Update security events unit tests The security events framework API was changed in Asterisk 10 but the unit tests were not updated at the same time. This patch does the following: * Adds two more security events that were added to the API * Add challenge, received_challenge and received_hash in the inval_password security event unit test (issue ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael L. Young Patches: issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1877/ ........ r365083 | twilson | 2012-05-02 12:29:54 -0500 (Wed, 02 May 2012) | 33 lines Multiple revisions 365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race and local channel linkedids This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes the race condition by no longer scanning the channel list for "other" channels with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings and uses the refcount of the string as a counter of how many channels with the linkedid exist. Not only does this eliminate the race condition, but it also allows us to look up the linkedid by the hashed key instead of traversing the entire channel list. Review: https://reviewboard.asterisk.org/r/1895/ ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines Don't leak a ref if out of memory and can't link the linkedid If the ao2_link fails, we are most likely out of memory and bad things are going to happen. Before those bad things happen, make sure to clean up the linkedid references. This patch also adds a comment explaining why linkedid can't be passed to both local channel allocations and combines two ao2_ref calls into 1. Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@365264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jeff Peeler	e0ac582b5e	Merged revisions 291791 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291791 | jpeeler | 2010-10-14 13:45:02 -0500 (Thu, 14 Oct 2010) | 10 lines Add missing ifdefs for test framework and new locale code. (closes issue #18137) Reported by: ovi Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes (license 717) 18137_localelist_warning.patch uploaded by wdoekes (license 717) Tested by: ovi ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6d0e383321	Merged revisions 289543,289581 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289543 | tilghman | 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010) | 2 lines More Solaris compatibility fixes ........ r289581 | tilghman | 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) | 2 lines Solaris fixes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b0357dcc3e	Support setting locale per-mailbox (changes date/time languages for email, pager messages). ... (closes issue #14333) Reported by: klaus3000 Patches: 20090515__issue14333.diff.txt uploaded by tilghman (license 14) app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	420acb8f0a	Fix build on linux. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8d6ee962c7	Add kqueue(2) implementation to Asterisk in various places. ... This will save a considerable amount of CPU on the BSDs, including Mac OS X, as it eliminates several places in the code that we previously used a busy loop. Additionally, this adds a res_timing interface, using kqueue timers. Review: https://reviewboard.asterisk.org/r/543/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 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
Tilghman Lesher	0c0bf14d99	Just in case of a race, send the signal on interrupt. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	385a40226a	Fix test_time on Mac OS X (and other platforms without inotify) ... Reviewboard: https://reviewboard.asterisk.org/r/554/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	da6ba8e60e	Remove portions that weren't meant to be committed for the OS X compat fix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e58fc610ae	Change needed to make Mac OS X 10.6 happy ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8e39288f61	Use nanosleep instead of poll. ... This is not just because mmichelson suggested it, but also because Mac OS X puked on my poll(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4aeb36b1c5	Compatibility fix for glibc 2.4 ... (Closes issue #14820) Reported by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	baf144c655	Picky, picky buildbots ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4ac2fd4cde	Use notification when timezone files change and re-scan then. ... (closes issue #14300) Reported by: jamessan Patches: 20090127__bug14300.diff.txt uploaded by tilghman (license 14) 20090224__bug14300.diff uploaded by jamessan (license 246) Tested by: jamessan Review: http://reviewboard.digium.com/r/136/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	8d5deb312b	Merged revisions 157859 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems. with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course). while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	529874de7b	Add schedule extensions to app_meetme. In addition, the reporter found a ... problem within strptime(3), which we are correcting here with ast_strptime(). (closes issue #11040) Reported by: DEA Patches: 20080910__bug11040.diff.txt uploaded by Corydon76 (license 14) Tested by: DEA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	629861a705	Merged revisions 144924-144925 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines 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 ........ r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines fix some minor issues with rev 144924 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	3dd4296de7	Merged revisions 103845 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20 Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022) Reported by: asgaroth Patches: 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14) Tested by: asgaroth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	3570ad103d	Added a new module, res_phoneprov, which allows auto-provisioning of phones ... based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect ... build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	bd27085c00	Merged revisions 85921 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85921 | tilghman | 2007-10-16 14:41:40 -0500 (Tue, 16 Oct 2007) | 4 lines Also set up gmtoff (this is used in the %z gnu extension to strftime) Reported and fixed by jcmoore Closes issue #11002 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	96d9f820e9	Merged revisions 82676 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82676 | russell | 2007-09-17 15:16:25 -0500 (Mon, 17 Sep 2007) | 4 lines Put a memset in ast_localtime() instead of a couple places in app_voicemail to prevent the problem everywhere instead of just a couple of places. (related to issue #10746) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	22436131f7	Merged revisions 82291 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82291 | tilghman | 2007-09-12 16:28:33 -0500 (Wed, 12 Sep 2007) | 2 lines Oops, wrong location for FreeBSD zone files ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	82cf384e03	Merged revisions 82285 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82285 | tilghman | 2007-09-12 15:12:06 -0500 (Wed, 12 Sep 2007) | 4 lines Working on issue #10531 exposed a rather nasty 64-bit issue on ast_mktime, so we updated the localtime.c file from source. Next we'll have to write ast_strptime to match. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	81bc1d7af5	Merge in ast_strftime branch, which changes timestamps to be accurate to the microsecond, instead of only to the second ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	5fdba27ea2	Merged revisions 69392 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69392 | kpfleming | 2007-06-14 16:50:40 -0500 (Thu, 14 Jun 2007) | 2 lines use ast_localtime() in every place localtime_r() was being used ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	9d74ea97c0	Merged revisions 51256 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51256 | tilghman | 2007-01-18 15:14:24 -0600 (Thu, 18 Jan 2007) | 10 lines Merged revisions 51255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 Jan 2007) | 2 lines If a timezone is not specified, assume localtime (instead of gmtime) (Issue #7748) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	05d6c4a2d0	Merged revisions 48906 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48906 | qwell | 2006-12-22 16:33:46 -0600 (Fri, 22 Dec 2006) | 2 lines Minor fixes for Solaris. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	018a9c86e4	Merged revisions 48521 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48521 | kpfleming | 2006-12-16 14:12:41 -0600 (Sat, 16 Dec 2006) | 2 lines since we really, really have to have autoconfig.h included before all other headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	0a27d8bfe5	merge new_loader_completion branch, including (at least): ... - restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago