asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	a3a58d9d44	Prevent res_timing_pthread from blocking callers There were several reports of deadlock when using res_timing_pthread. Backtraces indicated that one thread was blocked waiting for the write to the pipe to complete and this thread held the container lock for the timers. Therefore any thread that wanted to create a new timer or read an existing timer would block waiting for either the timer lock or the container lock and deadlock ensued. This patch changes the way the pipe is used to eliminate this source of deadlocks: 1) The pipe is placed in non-blocking mode so that it would never block even if the following changes someone fail... 2) Instead of writing bytes into the pipe for each "tick" that's fired the pipe now has two states--signaled and unsignaled. If signaled, the pipe is hot and any pollers of the read side filedescriptor will be woken up. If unsigned the pipe is idle. This eliminates even the chance of filling up the pipe and reduces the potential overhead of calling unnecessary writes. 3) Since we're tracking the signaled / unsignaled state, we can eliminate the exta poll system call for every firing because we know that there is data to be read. (closes issue ASTERISK-21389) Reported by: Matt Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches: 0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417) (closes issue ASTERISK-19754) Reported by: Nikola Ciprich (closes issue ASTERISK-20577) Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported by: Henry Fernandes (closes issue ASTERISK-17467) Reported by: isrl (closes issue ASTERISK-17458) Reported by: isrl Review: https://reviewboard.asterisk.org/r/2441/ ........ Merged revisions 386109 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	f49c09b8e5	Distributed Device State broken at sites using res_xmpp or res_jabber where Secuity Advisory AST-2012-015 is inplace res_xmpp was not adding AST_EVENT_IE_CACHABLE to the event as each message came in, then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2452/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	2814d40134	Distributed Device State broken at sites using res_xmpp or res_jabber where Secuity Advisory AST-2012-015 is inplace res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the event as each message came in, then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2452/ ........ Merged revisions 385916 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	918addee55	Fix the svn:keywords property on several files. Normally I think keyword expansion is silly, but the one time it would have been good, it didn't work because the property had quotes in it. This patch fixes obviously busted svn:keywords properties. ........ Merged revisions 385683 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	70c792d035	Calculate the timestamp for outbound RTP if we don't have timing information This patch calculates the timestamp for outbound RTP when we don't have timing information. This uses the same approach in res_rtp_asterisk. Thanks to both Pietro and Tzafrir for providing patches. (closes issue ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035) rtp-timestamp.patch uploaded by pbertera (License 5943) ........ Merged revisions 385636 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	d8216bd9ee	Add dependency on libuuid, for res_rtp_asterisk pjproject is what actually requires libuuid. (closes issue ASTERISK-21125) reported by Private Name (Ed. note: Really? Private Name? I am rolling my eyes so hard right now.) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3a73c367f9	Use LDAP memory management functions instead of Asterisk's When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk: invalid pointer errors) can occur as the memory is being allocated with Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the LDAP library's wrappers. This patch uses the LDAP library's wrappers where appropriate, so that compiling with MALLOC_DEBUG doesn't cause more problems than it solves. Note that the patch listed below was modified slightly for this commit to account for some additional memory allocation/deallocations. (closes issue ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham patches: issue18789-1.8-r316873.patch uploaded by seanbright (License 5060) ........ Merged revisions 385190 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	d014e51a18	Fix white noise on SRTP decryption When res_rtp_asterisk.c was altered to avoid attempting to apply unprotect algorithms to non-audio RTP packets, the test used was incorrect. This caused the audio packets to not be decrypted and resulted in loud white noise on the other endpoint (or both endpoints depending on the call legs involved). The test now properly checks the version field in the RTP header to ensure that RTP and RTCP are decrypted while other types of packets are not. (closes issue ASTERISK-21323) Reported by: andrea Tested by: Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff uploaded by Kinsey Moore ........ Merged revisions 384048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	916a397fc3	AST-2013-001: Prevent buffer overflow through H.264 format negotiation The format attribute resource for H.264 video performs an unsafe read against a media attribute when parsing the SDP. The value passed in with the format attribute is not checked for its length when parsed into a fixed length buffer. This patch resolves the vulnerability by only reading as many characters from the SDP value as will fit into the buffer. (closes issue ASTERISK-20901) Reported by: Ulf Harnhammar patches: h264_overflow_security_patch.diff uploaded by jrose (License 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	a52a841d8e	Properly delimit post data in res_config_curl. ........ Merged revisions 383667 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	c379172cae	Fix a bug where resources were not found due to hashing on the priority itself. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	7031ad62e8	Fix a crash when res_xmpp is configured using a username without a domain. (closes issue ASTERISK-21156) Reported by: amsoft2001 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5b1533ca7b	Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source Often, Asterisk may realize that a change in the source of an RTP stream is about to occur and ask that the RTP engine reset it's lock on the current RTP source. In certain scenarios, it may take awhile for the new remote system to send RTP packets, while the old remote system may continue providing RTP during that time period. This causes Asterisk to re-lock onto the old source, thereby rejecting the new source when the old source stops sending RTP and the new source begins. This patch prevents that by having a constant secondary, 'secret' probation mode enabled when an RTP source has been chosen. RTP packets from other sources are always considered, but never chosen unless the current RTP source stops sending RTP. Review: https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124) Reported by: John Bigelow Tested by: John Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	a78bb96d94	While the ICE negotiation is occurring leave strictrtp in an open state, media can and will come from different places. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f6b368216a	Fix a bug with ICE and strictrtp where media could get dropped. If the end result of the ICE negotiation resulted in the path for media changing it was possible for the strictrtp code to discard the RTP packets. This change causes strictrtp to enter learning mode once again when the ICE negotiation has completed successfully. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	2d451c64f7	Fix FastAGI To Properly Check For A Connection When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	cb3dd02781	Fix crash in res_xmpp when deleting pubsub node from CLI An error existed in res_xmpp where it would attempt to delete attributes from a node that itself was also deleted. Per the iksemel documentation, attributes added using iks_insert are copied to the parent node's stack, and will be reclaimed when that node is itself destroyed. (closes issue ASTERISK-20982) Reported by: marcelloceschia patches: delete-node-fix.diff uploaded by marcelloceschia (License 6036) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	e7e730f973	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	d9d5028b01	Ignore warnings caused by PJ_TODO()s in pjproject. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	47f8394517	Fix a few compiler warnings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	d865240168	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	ff0d016390	Always check for libm, regardless of configure options. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	e02189ffa1	Remove a cross-compile workaround. ar and ranlib can be easily detected with autoconf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	2a458061b1	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c7d22a6af6	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fb72636fd8	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	28bdf7d292	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	af1666f1cf	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	cef6475eb0	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	d566b201ae	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	8653377baf	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	c4587d7170	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	17467a6926	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3778ed7fc1	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	eda6664de0	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5ebec60090	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	bdd5921950	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b2c2a84e17	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	e875e4b2c0	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	4c924bdce1	Added missing newlines to websocket ast_logs. Without these newlines, log messages just continue tacking onto the same line, and do not flush immediately. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c0c552de2c	monitor: prevent attempts to move/remove recordings skipped with 'i' and 'o'. The i and o options for monitor skip the input and output sides of a recording respectively. This patch addresses a problem in those options when monitor is called without specifying a specific filename where monitor will try to move the recording that was skipped. Since this usually doesn't exist when these options are used, it would produce a warning when it does this in most cases, but it is conceivable that there are use cases where this could result in moving/removing a file unintentionally. (closes issue ASTERISK-20641) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2190/ ........ Merged revisions 376389 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376390 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	fb74294b92	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	0e7a64e55e	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a65fbf8012	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f0cd27e027	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e4afa2c0e2	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	05be2e8bee	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7cedfef4fd	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	59d02d37de	Remove code that should not have gotten in. (issue ASTERISK-20554) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	385b30fbc6	Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures. This change removes the requirement for ufrag and pwd in the transport stanza and also makes us the controlling agent. (closes issue ASTERISK-20554) Reported by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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1906 Commits (aa676fbb84cfa50c2c67c6c83e6478b585057bc0)