asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	38c0a62413	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 ........ Merged revisions 361754 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	97f813f3a4	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 ........ Merged revisions 361706 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f4fd1b2fb0	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 ........ Merged revisions 361658 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	90226b6fd7	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 ........ Merged revisions 361607 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	aa21d4fc6b	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 ........ Merged revisions 361560 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4665986fb1	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. ........ Merged revisions 361522 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	a485f44022	Add missing newlines to CLI logging ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361472 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	fcb7eb3c59	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 ........ Merged revisions 361422 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	b2f7b0c649	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 ........ Merged revisions 361381 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a2e127a651	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 ........ Merged revisions 361333 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	51f0e5c53d	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 ........ Merged revisions 361330 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	fc45af331b	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 ........ Merged revisions 361270 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	8d3855229a	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 ........ Merged revisions 361211 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	6cd462cca8	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 ........ Merged revisions 361208 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	e96a59acfd	Replace GNU old-style field designator extensions to fix clang warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches: clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) ........ Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8 Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue ASTERISK-19540) ........ Merged revisions 361143 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	97b2fa8de1	Make the MeetMeAdmin N command (mute all nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/ ........ Merged revisions 361090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361091 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	93781fa161	Fix the display of documentation for Transfer This came up while fixing documentation generation for many other cases where the argument separator was not being displayed properly. Now that it is displayed properly, it shows up in the wrong place for Transfer since the '/' is only required if Tech is present. (related to issue ASTERISK-18168) ........ Merged revisions 361040 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361041 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	745fcdbffe	Fix dev-mode compiler warning about gnu_printf (related to ASTERISK-19575) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	e4252eac10	Allow the Hangup manager action to match channels by regex * Hangup now can take a regular expression as the Channel option. If you want to hangup multiple channels, use /regex/ as the Channel option. Existing behavior to hanging up a single channel is unchanged, but if you pass a regex, the manager will send you a list of channels back that were hung up. (closes issue ASTERISK-19575) Reported by: Mark Murawski Tested by: Mark Murawski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	9cc6f2c59e	Stop sending out RTCP if RTP is inactive This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) ........ Merged revisions 360987 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360993 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6a540e9087	Fix logger deadlock on Asterisk shutdown. The logger_thread() had an exit path that failed to release the logmsgs list lock. * Make logger_thread() exit path unlock the logmsgs list lock. * Made ast_log() not queue any messages to the logmsgs list if the close_logger_thread flag is set. (issue ASTERISK-19463) Reported by: Matt Jordan ........ Merged revisions 360933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360934 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	314d459317	Fix potential race condition during call pickup. Prior to this patch, a connected line update was queued during call pickup and then an answer frame was queued. The original caller would presumably then have his connected line updated and then the call would be answered. In actuality, the answer frame was not how the call ended up being answered. Rather, an odd section in app_dial that checks if the called channel's state is up. The result is that the order of the connected line update and the answer were variable. In most cases, this wasn't actually a bad thing. However, if the 'I' option was passed to dial, the connected line update would be inhibited. The fix is to queued the connected line after the answer frame is queued. This way the race in app_dial is between two conditions resulting in an answer. This way the connected line update occurs after the answer every time. (closes issue ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by Mark Michelson (license 5049) ........ Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360885 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	cc2366bca0	Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests. This change makes use of connected party information in addition to caller ID in order to populate local and remote XML elements in the dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej Krajewski Tested by: Maciej Krajewski Patches: local_remote_hint2.diff uploaded by Mark Michelson (license 5049) ........ Merged revisions 360862 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360863 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	fb796aac06	Misc changes to make astobj2 enhancement diffs easier to follow. * Rename astobj2 API parameter funcname to func. * Rename astobj2 API iterator parameter to iter. * Update some documentation for OBJ_MULTIPLE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	655a8d4420	Introducing the log message unique call identifiers feature Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	d501c2ea2d	undoing 360785 due to merging mistake git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	bf994f0e04	Introducing the log message unique call identifiers feature Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	dd9405db05	Fix setting CDR variables in the hangup extension A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ ........ Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358989 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	01cc64585e	Make a debug message regarding subscription changes more accurate. I was getting confused during some testing why Asterisk was saying that a subscription was being added when it was clearly being removed. This fixes that confusion. ........ Merged revisions 360625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360672 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	38e892b370	Add global ao2 array container. Global ao2 objects must always exist after initialization because there is no access control to obtain another reference to the global object. It is expected that module configuration could use these new API calls to replace an active configuration parameter object with an updated configuration parameter object. With these new API calls, the global object could be replaced, removed, or referenced without the risk of someone using a stale global object pointer. Review: https://reviewboard.asterisk.org/r/1824/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8611bea122	Attempt to be more helpful when using a bad ao2 object pointer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	65f56cda7a	Updates config with bootstrap where I changed configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon Clark ........ Merged revisions 360574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360575 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	8331a89892	Blocked revisions 360476 ........ Update CHANGES for r360471 ........ Merged revisions 360474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	dea8936f89	Convert ast_verb() to ast_debug() and increase log level Rather then flood the CLI with verbose messages, we've changed the level to debug. This will help keep the CLI clean. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	337f933977	Fix BETTER_BACKTRACES library detection for Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360489 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	5affceaa15	func_curl: Fix leak of an ast_str in error handling code path. ........ Merged revisions 360413 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360414 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d6d7f51476	chan_iax2: Use OBJ_NODATA to be a bit more explicit. This is just a minor code cleanup change. These uses of ao2_callback() would never return anything since the callbacks always returned 0. However, be more explicit that no returned results are wanted by specifying OBJ_NODATA. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	0ec73946fa	app_page: Fix a memory leak on every Page(). dial_list is a dynamically allocated array that is allocated at the beginning of Page() based on how many devices will be dialed. This was never being freed. ........ Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360364 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	71a1541b0c	app_jack: fix datastore memory leak in error handling path. ........ Merged revisions 360360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360361 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	cad07b3800	Multiple revisions 360356-360357 ........ r360356 \| russell \| 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) \| 6 lines expression parser: Fix (theoretical) memory leak. Fix a memory leak that is very unlikely to actually happen. If a malloc() succeeded, but the following strdup() failed, the memory from the original malloc() would be leaked. ........ r360357 \| russell \| 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) \| 6 lines Rebuild parsers. This is needed to include the last fix to main/ast_expr2.y. The changes look much bigger as this regeneration of the code was done with newer versions of flex and bison. ........ Merged revisions 360356-360357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360358 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	721f92058f	Make number not available presentation also set screening to network provided. Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. ........ Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	df16bd973e	Add missing initialization of update_redirecting in chan_sip.c ........ Merged revisions 360262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360263 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	c6979ff581	Adds F option to Bridge application Similar to dial and queue F option. (Closes issue ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff uploaded by To (license 6347) Review: https://reviewboard.asterisk.org/r/1825/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c5b3db1956	Kill off red blobs in most of main/* Everything still compiled after making these changes, so I assume these whitespace-only changes didn't break anything (and shouldn't have). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1d1c28ac4b	Update install_prereq script to include missing GSM library for debian amd move SQLite3. (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches: debian_install_prereq.diff uploaded by Andrew Latham (license 5985) ........ Merged revisions 360138 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360139 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	ab6f40bd12	Also detect gmime 2.6 Also detect gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com> ........ Merged revisions 360087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360098 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	c88d1c8337	Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE When Asterisk detects a hangup and cannot send a BYE due to a pending INVITE, it sets the pendingbye flag and waits for the final response to that INVITE. When the response is received, it transmits the BYE. If, however, that INVITE request is a pending re-INVITE, it needs to first send a CANCEL request to terminate the pending re-INVITE. In that circumstance, Asterisk was, in some scenarios, clearing the pendingbye flag after processing the CANCEL request and not checking for a pending BYE when receiving the final 487 response to the INVITE. This patch ensures that if the pendingbye flag is set, it is honored regardless of the nature of the INVITE request currently in flight. (closes issue ASTERISK-19365) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283) Review: https://reviewboard.asterisk.org/r/1807 ........ Merged revisions 360086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360088 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	6ff8f14865	Prevent Echo() from relaying control, null, and modem frames Echo()'s description states that it echoes audio, video, and DTMF except for # while it actually echoes any frame that it receives other than DTMF #. This was causing frame storms in the test suite in some circumstances where Echo() was attached to both ends of a pair of local channels and control frames were being periodically generated. Echo()'s behavior and description have been modifed so that it only echoes media and non-# DTMF frames. ........ Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360034 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	3a231e090f	chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus AMI Events The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. Post was removed and the AMI version has been updated to 1.3. ........ Merged revisions 359982 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	334f13d8b8	Allow AMI action callback to be reentrant. Fix AMI module reload deadlock regression from ASTERISK-18479 when it tried to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The patch stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ Review: https://reviewboard.asterisk.org/r/1820/ ........ Merged revisions 359979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359980 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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