asterisk

Commit Graph

Author	SHA1	Message	Date
Jason Parker	eb61bb96b7	Fix how we build pjproject. Allow parallel builds, better tolerate failures, build faster. This also stops running dependencies before top-level configure has been run. (closes issue ASTERISK-20815) Review: https://reviewboard.asterisk.org/r/2292/ ........ Merged revisions 380816 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	5b41fbfe8b	Multiple revisions 380735-380736 ........ r380735 \| qwell \| 2013-01-31 15:40:09 -0600 (Thu, 31 Jan 2013) \| 1 line Fix a few compiler warnings. ........ r380736 \| qwell \| 2013-01-31 15:42:34 -0600 (Thu, 31 Jan 2013) \| 1 line Ignore warnings caused by PJ_TODO()s in pjproject. ........ Merged revisions 380735-380736 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	c1b4a93f49	Multiple revisions 380671-380673 ........ r380671 \| qwell \| 2013-01-31 12:59:28 -0600 (Thu, 31 Jan 2013) \| 4 lines Remove a cross-compile workaround. ar and ranlib can be easily detected with autoconf. ........ r380672 \| qwell \| 2013-01-31 13:00:38 -0600 (Thu, 31 Jan 2013) \| 2 lines Always check for libm, regardless of configure options. ........ r380673 \| qwell \| 2013-01-31 13:03:03 -0600 (Thu, 31 Jan 2013) \| 7 lines Add support for parallel builds of pjproject. Also adds proper dependency checking, and direct .a file targets. We don't take advantage of this currently, but we will soon. (issue ASTERISK-20815) ........ Merged revisions 380671-380673 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0728c6d7ae	Fix memory leak in res_calendar_icalendar The ICalendar module had a systemic memory leak on each fetch of data from the ICalendar source. The previous fetched data was not being properly disposed. This patch makes it so that before each fetch of data, we dispose of the previously fetched data. (closes issue ASTERISK-21012) Reported by: Joel Vandal Tested by: Joel Vandal ........ Merged revisions 380451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380452 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	9d623f3a73	Make sorcery modules global, since they are required by other modules that are global. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	6300c37152	Add a missing '\' to a log message. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	3fa4278a31	Merge the sorcery data access layer API. Sorcery is a unifying data access layer which provides a pluggable mechanism to allow object creation, retrieval, updating, and deletion using different backends (or wizards). This is a fancy way of saying "one interface to rule them all" where them is configuration, realtime, and anything else that comes along. Review: https://reviewboard.asterisk.org/r/2259/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	945fc670f9	res_fax_spandsp: fix t38 transmission bug caused by not returning success This patch fixes the problem, but the issue includes a test which is still being considered for the automated test suite. (issue ASTERISK-20919) Reported by: NITESH BANSAL Patches: patch_ast_fax_spandsp.patch uploaded by NITESH BANSAL (license 6418) ........ Merged revisions 379949 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7d9871b394	Add ControlPlayback manager action This patch adds the capability for asynchronous manipulation of audio being played back to a channel though a new AMI action "ControlPlayback". The ControlPlayback action supports a number of operations, the availability of which depend on the application being used to send audio to the channel. When the audio playback was initiated using the ControlPlayback application or CONTROL STREAM FILE AGI command, the audio can be paused, stopped, restarted, reversed, or skipped forward. When initiated by other mechanisms (such as the Playback application), the audio can be stopped, reversed, or skipped forward. Review: https://reviewboard.asterisk.org/r/2265/ (closes issue ASTERISK-20882) Reported by: mjordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	472e29df62	Let documentation reference links specify which module they're linking to Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref links have to specify which reference they're referring to. The various documentation parsers can interpret the module attribute however they want in order to construct the appropriate links. ........ Merged revisions 379228 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b84d37a711	Multiple revisions 379209-379210 ........ r379209 \| mjordan \| 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) \| 8 lines Add module tags to documentation for res_jabber/res_xmpp Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.), the XML documentation for each needs to call out which module is providing the documentation. The module attribute has been added to the various XML fragments for this purpose. ........ r379210 \| mjordan \| 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) \| 4 lines Update the dtd to actually support the module attribute in all elements Mea culpa. ........ Merged revisions 379209-379210 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	9b475cd3ef	Reset RTP timestamp; sequence number on SSRC change In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to better account for out of order RTP packets. This was accomplished by using the RTP timestamp and sequence number to check for out of order packets. However, when a SSRC change occurs, the timestamp and sequence number will no longer have any relation to the previously received packets. The variables tracking the timestamp and sequence number therefore have to be reset. (closes issue ASTERISK-20906) Reported by: Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442) ........ Merged revisions 378967 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378984 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	c5ec471766	Retain XMPP filters across reconnections so external modules continue to function as expected. Previously if an XMPP client reconnected any filters added by an external module were lost. This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling. (closes issue ASTERISK-20916) Reported by: kuj ........ Merged revisions 378917 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	7695ea2643	Add JSON API for Asterisk. This provides a JSON API by pulling in and wrapping the Jansson JSON library[1]. The Asterisk API basically mirrors the Jansson functionality, with a few minor tweaks. * Some names have been asteriskified to protect the innocent. * Jansson provides both reference-stealing and reference-borrowing versions of several API's. The Asterisk API is exclusively reference-stealing for operations that put elements into arrays and objects. * No support for doubles, since we usually don't need that. * Coming along for the ride is the ast_test_validate macro, which made the unit tests much easier to write. [1]: http://www.digip.org/jansson/ (issue ASTERISK-20887) (closes issue ASTERISK-20888) Review: https://reviewboard.asterisk.org/r/2264/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	9d5f6e050e	res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create Under some circumstances, libsrtp's srtp_create function deallocates memory that it wasn't initially responsible for allocating. Because we weren't initially aware of this behavior, this memory was still used in spite of being unallocated during the course of the srtp_unprotect function. A while back I made a patch which would set this value to NULL, but that exposed a possible condition where we would then try to check a member of the struct which would cause a segfault. In order to address these problems, ast_srtp_unprotect will now set an error value when it ends without a valid SRTP session which will result in the caller of srtp_unprotect observing this error and hanging up the relevant channel instead of trying to keep using the invalid session address. (closes issue ASTERISK-20499) Reported by: Tootai Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header ........ Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378592 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	9e814816cb	Fix pjproject compilation in certain circumstances On a fresh checkout of Asterisk 11, running make before ./configure could cause the pjproject subdirectory to get in an odd state that would prevent compilation. This patch by Tilghman prevents that from occurring. (closes issue ASTERISK-20681) Reported by: Dinesh Ramjuttun Tested by: danilo borges, Steve Lang patches: 20121208__ccar_solved.diff.txt uploaded by Tilghman Lesher (license 5003) ........ Merged revisions 378582 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	4838d6ff68	Don't pass STUN packets through the SRTP unprotect function. (closes issue AST-1036) Reported by: jbigelow ........ Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378555 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	a30e74da4f	Doxygen Cleanups Baseline clean up of formating to make room for extended documentation (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	099cd07887	Prevent exhaustion of system resources through exploitation of event cache This patch changes res_xmpp to no longer cache events under certain circumstances. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore ........ Merged revisions 378411 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8bf1f1745b	Prevent crashes in res_xmpp when receiving large messages Similar to r378287, res_xmpp was marshaling data read from an external source onto the stack. For a sufficiently large message, this could cause a stack overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by removing the stack allocation, as it was unnecessary. (issue ASTERISK-20658) Reported by: wdoekes ........ Merged revisions 378409 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8fb5bdce9a	Prevent exhaustion of system resources through exploitation of event cache Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) ........ Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378321 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1fb06fde95	Resolve crashes due to large stack allocations when using TCP Asterisk had several places where messages received over various network transports may be copied in a single stack allocation. In the case of TCP, since multiple packets in a stream may be concatenated together, this can lead to large allocations that overflow the stack. This patch modifies those portions of Asterisk using TCP to either favor heap allocations or use an upper bound to ensure that the stack will not overflow: * For SIP, the allocation now has an upper limit * For HTTP, the allocation is now a heap allocation instead of a stack allocation * For XMPP (in res_jabber), the allocation has been eliminated since it was unnecesary. Note that the HTTP portion of this issue was independently found by Brandon Edwards of Exodus Intelligence. (issue ASTERISK-20658) Reported by: wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches: ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378287 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	1e51b9eaa1	Make generate_exchange_uuid() always return the passed ast_str pointer. I changed this code earlier to return NULL if it wasn't able to generate a UUID, whereas the earlier code would always return the ast_str that was passed in. Switch back to returning the ast_str, only set it to the empty string instead if UUID generation fails. We still do a validity check later which will catch this and blow up if necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	d33e46a47f	Use the UUID API to generate and validate UUIDs for res_calendar_exchange. Currently the res_calendar_exchange module uses its own method of generating UUIDs using ast_random(). Now that we have a UUID API we should use that instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	609ffb429a	The UUID commit removed changes made in res_clialiases.c This puts back in the changes that are designed to work around a memory leak fix in the CLI code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	8cb156bfc1	Add UUID support to Asterisk. This provides a common API for dealing with unique identifiers. The API provides methods to create, parse, copy, and stringify UUIDs. An accompanying unit test is provided that tests all operations. (closes issue ASTERISK-20726) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2217 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	619de7f012	Fix crash that can occur if CLI registration fails for an aliased command. A recent memory leak fix in main/cli.c causes an ast_cli_entry's command field to be freed and NULLed if ast_cli_register() fails. res_clialiases was ignoring the return value of ast_cli_register() and was then passing the NULL command off to a a hash function. This resulted in a crash. The fix is not to ignore the erroneous return value. If ast_cli_register() fails, then we do not continue trying to process the current alias. ........ Merged revisions 377840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377842 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377843 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	34cbefe62f	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 ........ Merged revisions 377656 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377657 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	d7372766dc	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 ........ Merged revisions 377261 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377262 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Olle Johansson	e3faeb67e8	Formatting fixes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Olle Johansson	1b47dbe991	Formatting changes Found a large amount of missing {} in the code before patching in another branch git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	cc01a79463	Added missing newlines to websocket ast_logs. Without these newlines, log messages just continue tacking onto the same line, and do not flush immediately. ........ Merged revisions 376561 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	866d968149	Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default. With ICE support enabled in chan_sip and a large number of interfaces on the system it was possible for the produced SDP to be truncated due to some fixed size buffers. These buffers have now been changed so they will dynamically grow as needed. ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is no longer enabled by default there. (closes issue ASTERISK-20643) Reported by: coopvr ........ Merged revisions 376130 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e773bbdd10	Fix a "set but not used" warning on newer gccs. Turns out the "helpful" setting of ms and res in this macro is completely useless after the timeout antipattern fix. If you're a new guy looking to write code, don't write a macro like this one. ........ Merged revisions 376087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376088 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376089 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f2bb9afe17	Multiple revisions 375993-375994 ........ r375993 \| mmichelson \| 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) \| 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 \| mmichelson \| 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) \| 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a0c363e227	Refactor ast_timer_ack to return an error and handle the error in timer users Currently, if an acknowledgement of a timer fails Asterisk will not realize that a serious error occurred and will continue attempting to use the timer's file descriptor. This can lead to situations where errors stream to the CLI/log file. This consumes significant resources, masks the actual problem that occurred (whatever caused the timer to fail in the first place), and can leave channels in odd states. This patch propagates the errors in the timing resource modules up through the timer core, and makes users of these timers handle acknowledgement failures. It also adds some defensive coding around the use of timers to prevent using bad file descriptors in off nominal code paths. Note that the patch created by the issue reporter was modified slightly for this commit and backported to 1.8, as it was originally written for Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches: jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 375893 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375894 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375895 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	069f5f8b93	Only deref a reserved gateway session if we actually reserved one Its perfectly acceptable to have a gateway session unreserved when we go to first allocate one. Unreffing the reserved gateway session - when its NULL - will result in an assertion error. This problem was caught by the Asterisk Test Suite (once we had enough of the debugging flags enabled) ........ Merged revisions 375797 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375798 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	6de0b18b3b	Fix an issue with res_http_websocket where the chan_sip WebSocket handler could not be registered. On some systems the optional API support uses the GCC compiler attribute "weakref" to provide its functionality. This code changes the function names and prefixes "__" to the front. The res_http_websocket exports file did not take this into account, thereby not allowing those functions to be global and ultimately found. (closes issue ASTERISK-20631) Reported by: danjenkins ........ Merged revisions 375559 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	05cee7b717	Properly extract the Body information of an EWS calendar item Unlike all other calendar modules, res_calendar_ews fails to extract the Body information for a calendar item. This is due, in part, to a quirk in the schema in the XML - not only does a CalendarItem contain a Body element, but the CalendarItem exists as a descendant of a different Body element. The neon parser was erroneously skipping all Body elements. This patch fixes that by bypassing Body elements that are not a child of CalendarItem, and parsing the Body element out if it is a child. Note that the original patch by Terry Wilson only needed slight modifications to make it properly pull the Body information out; as such, while I've linked to the patch that I uploaded for Dmitry, I've attributed the patch to Terry. (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff uploaded by Terry Wilson (license 6283) ........ Merged revisions 375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375531 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375532 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	6d57ecd48c	Change a few warnings to debug and the inverse. Remove the "RTP Read too short" warning for RTP keepalives. Remove the the warning about the application delimiter switch from pipe to comma. (You should've done this by now.) Make cdr_odbc report more when an insert fails. Make chan_sip warn less when the peer wants SRTP (and we don't) or sends a zero port to disable a media type. Review: https://reviewboard.asterisk.org/r/2167 (closes issue ASTERISK-20538) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	c7857504df	Doxygen Updates - Title update Update and extend the configuration_file group and enable linking to the resource. Update title that was left behind many years ago. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3620fcff36	Disable ICE support by default Since there are a number of legacy devices out there that fail to handle ICE candidates properly (which is a nice way of saying something much uglier), disable it by default. Support for ICE candidates can be enabled in rtp.conf using the icesupport setting. ........ Merged revisions 374676 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	35b12af8b6	pjproject: Fix for Solaris builds. Do not undef s_addr. pjproject, in order to solve build problems on Windows [1], undefines s_addr in one of it's headers that is included in res_rtp_asterisk.c. On Solaris s_addr is not a structure member, but defined to map to the real strucuture member, therefore when building on Solaris it's possible to get build errors like: [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o In file included from /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29, from res_rtp_asterisk.c:51: /export/home/admin/asterisk-11-svn/include/asterisk/network.h: In function `inaddrcmp': /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr' /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr' res_rtp_asterisk.c: In function `ast_rtp_on_ice_tx_pkt': res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer will break strict-aliasing rules res_rtp_asterisk.c:710: warning: dereferencing type-punned pointer will break strict-aliasing rules res_rtp_asterisk.c: In function `rtp_add_candidates_to_ice': res_rtp_asterisk.c:1085: error: structure has no member named `s_addr' make[2]: * [res_rtp_asterisk.o] Error 1 make[1]: * [res] Error 2 make[1]: Leaving directory `/export/home/admin/asterisk-11-svn' gmake: *** [_cleantest_all] Error 2 Unfortunately, in order to make this work, I also had to make sure pjproject only used the typdef pj_in_addr and not the struct pj_in_addr so that when building Asterisk I could "typedef struct in_addr pj_in_addr". It's possible then that the library and users of those interfaces in Asterisk have a different idea about the type of the argument, while on the surface it looks like they are all 32 bit big endian values. [1] http://trac.pjsip.org/repos/changeset/484 (issues ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang, mjordan patches: 0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch uploaded by Shaun Ruffell (license 5417) ........ Merged revisions 374642 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	bd36827e98	Handle capability stanzas that fail to provide node or version information While XEP-0115 states that the node and ver attributes are both required, some devices fail to provide either field. Prior to this patch, failure to provide the node or ver attribute would cause a crash in res_xmpp. While failing to provide the node or ver attribute is technically invalid, since this information is not utilized by Asterisk except for reporting purposes, for interoperability reasons, we continue to process the capability stanza anyways. (closes issue ASTERISK-20495) Reported by: Martin W Tested by: Martin W patches: 20495.patch uploaded by Martin W (license #6434) ........ Merged revisions 374622 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	15b35972ff	Update documentation for MessageSend application/command's From field for XMPP When using the channel technology agnostic application/AMI command MessageSend, the "From" field is technically optional for the SIP channel driver. However, if being sent by the XMPP resource module (either res_xmpp or res_jabber), the "From" field is necessary, and must correspond to a defined account. This patch updates the documentation for this application/AMI command to reflect this. (closes issue ASTERISK-20405) Reported by: Leif Madsen ........ Merged revisions 374611 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	c5acf22cec	Fix DBDelTree error codes for AMI, CLI and AGI The AMI DBDelTree command will return Success/Key tree deleted successfully even if the given key does not exist. The CLI command 'database deltree' had a similar problem, but was saved because it actually responded with '0 database entries removed'. AGI had a slightly different error, where it would return success if the database was unavailable. This came from confusion about the ast_db_deltree retval, which is -1 in the event of a database error, or number of entries deleted (including 0 for deleting nothing). * Changed some poorly named res variables to num_deleted * Specified specific errors when calling ast_db_deltree (database unavailable vs. entry not found vs. success) * Fixed similar bug in AGI database deltree, where 'Database unavailable' results in successful result (closes issue AST-967) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2138/ ........ Merged revisions 374426 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374427 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374428 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	481df22eac	Check for presence of buddy in info/dinfo handlers The res_jabber resource module uses the ASTOBJ library for managing its ref counted objects. After calling ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to the object has to be checked to see if the buddy existed. Prior to this patch, the buddy object was not checked for NULL; with this patch in both aji_client_info_handler and aji_dinfo_handler the pointer is checked before used and, if no buddy object was found, the handlers return an error code. This patch does not take the approach that our JID can be used to log in from another resource. If that approach is desired, an improvement could be made to this patch to create the buddy on the fly. This patch seeks only to prevent Asterisk from crashing. FYI: In Asterisk 11+, you really should be using res_xmpp. It does not have this problem, as it moved to the astobj2 library. Note that multiple people have proposed patches for this issue; the patch being committed here is based on those. (closes issue ASTERISK-19532) Reported by: Karsten Wemheuer Tested by: Byron Clark patches: fix-jabber uploaded by Karsten Wemheuer (license #5930) xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark (license #6157) (closes issue ASTERISK-19557) Reported by: ulugutz ........ Merged revisions 374335 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374336 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374337 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a094707d51	Fix a variety of ref counting issues This patch resolves a number of ref leaks that occur primarily on Asterisk shutdown. It adds a variety of shutdown routines to core portions of Asterisk such that they can reclaim resources allocate duringd initialization. Review: https://reviewboard.asterisk.org/r/2137 ........ Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374178 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374196 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	e11cc29360	Doxygen Cleanup Start adding configuration file linking and pages. Add module loading doxygen block. Breaking up commits to keep it easy to track (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	0fc114dc65	Add support for retrieving engine specific settings using the speech API and from dialplan. (closes issue ASTERISK-17136) Reported by: kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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1901 Commits (657aa491f00b7a3693ab8ddfb4623f516fda33b0)