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Author	SHA1	Message	Date
Matthew Jordan	daa51f9845	Add a WARNING in bridge_softmix when a timing module isn't loaded ... If bridge_softmix fails to be created because no timing source is present in Asterisk, this will currently fail gracefully but with (most likely) a generic error message by whatever module tried to create the softmix bridge. This patch adds a more explicit warning so you can actually diagnose and fix the problem. Review: https://reviewboard.asterisk.org/r/2857/ ........ Merged revisions 399353 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	6c016dc642	res_pjsip_messaging: Register message technology as pjsip ... pjsip's message technology was being registered as 'sip', which was causing it to not load due it conflicting with chan_sip's registered 'sip' technology for messaging. It now registers as 'pjsip'. However, due to this change the "to" field for outgoing pjsip messages need to be prefixed with 'pjsip:' instead of 'sip:'. Incoming messages to res_pjsip_messaging will automatically have their "to" fields altered in order to accommodate the change. Outgoing messages also handle changing it back to 'sip' before being sent so the pjsip library will properly handle it. (closes issue ASTERISK-22445) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2833/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	6935184b56	Blocked revisions 399305 ... ........ Fix Segfault When Syntax Of A Line Under [applicationmap] Is Invalid When processing the lines under the [applicationmap] context in features.conf, a segfault occurs from attempting to process a line with an invalid syntax (basically missing most of the arguments). Example: [applicationmap] automon=*6 * This patch moves the checking for empty arguments to before they are accessed. * Also, checked the "todo" comment and removed it. Some applications do not require arguments. (closes issue ASTERISK-22416) Reported by: CGI.NET Tested by: CGI.NET Patches: asterisk-22416-check-syntax-first_v2.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2803 ........ Merged revisions 399304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	474c5df62f	Fix Segfault In features-config.c When Application Has No Arguments ... Some applications do not require arguments. Therefore, when parsing application maps in features.conf, it is possible that app_data will be set to NULL. * This patch sets app_data to "" if it is NULL. Review: https://reviewboard.asterisk.org/r/2804 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	c4fdbd079c	Change the "external_media_address" PJSIP endpoint option to "media_address". ... The endpoint option does not apply to communication with external entities. Rather, the option is applied to all communications with the endpoint. The external_media_address transport configuration option may override the endpoint option if it turns out that we are going to be communicating with an external entity. Two things of note: 1) I have not updated the XML documentation. This is being taken care of by Rusty as part of his work on issue ASTERISK-22405 2) This commit is likely to cause testsuite failures since there are tests that use the external_media_address endpoint option, and they will need to be changed over. Well, I'm planning to get that updated ASAP after this commit. (closes issue ASTERISK-22528) reported by Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	3f668d3a19	Remote console: more output discrepancies ... The remote console continued to have issues with its output. In this case CLI command output would either not show up (if verbose level = 0) or would contain verbose prefixes (if verbose level > 0) once log messages were sent to the remote console. The fix now now adds verbose prefix data to all new lines contained in a verbose log string. (closes issue ASTERISK-22450) Reported by: David Brillert (closes issue AST-1193) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2825/ ........ Merged revisions 399267 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	2e17814c28	Fix doxygen to use correct units of features.conf options. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9bdbccac14	Fix other timeouts (atxferloopdelay and atxfernoanswertimeout) to use seconds instead of milliseconds. ... Thanks to Richard Mudgett for pointing this out. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	edb351ebf6	Switch transferdigittimeout to be configured as seconds instead of milliseconds. ... This was an unintentional consequence of the update of features.conf to use the config framework in Asterisk 12. Thanks to Marco Signorini on the Asterisk developers list for pointing out the problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	a7527fc783	Confbridge: empty conference not being torn down ... Confbridge would not properly tear down an empty conference bridge when all users were kicked via end_marked=yes and at least one user was also set to wait_marked. This occurred because while end_marked users were being kicked and at least one was also set to wait_marked then the leave wait_marked handler would be called on that user, but there would be no waiting user (still considered active). The waiting users would decrement and now be negative. The conference would remain, but be put into an inactive state. The solution was to move from the active list to the wait list, those users with wait_marked set right before kicking. This allows both the active and wait users to decrement correctly and the confbridge to tear down properly. A crashed also occurred when trying to list the specific conference from the CLI. This happened because the conference specified was invalid. Since the conference properly tears down now there is no way to reference it thus alleviating the crash as well. (closes issue ASTERISK-21859) Reported by: Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/ ........ Merged revisions 399222 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fe904eefbe	Fix module load errors for test_ari_model.so. ... You cannot use a function pointer variable with an external function from another dynamically loaded module because data variables are always resolved even with RTLD_LAZY. * Added wrapper functions for ast_ari_validate_int() and ast_ari_validate_string() to use instead for the function pointer variable. (closes issue ASTERISK-22457) Reported by: David M. Lee git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	44f24f6c0f	app_speech_utils: Fix unresolved symbol ast_speech_get_setting(). ... Fixes regression introduced by -r374096. * Made res_speech.export.in export ast_* symbols instead of specific functions. * Made app_speech_utils.c declare that it is dependent upon res_speech. (issue ASTERISK-17136) Reported by: Richard Kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8eb165d2da	chan_iax2: Fix saving the wrong expiry time in astdb. ... When a new IAX2 client registers, the astdb database is updated with the value of minregexpire defined in iax.conf instead of using the expiry time that is provided by the client. The provided expiry time of the client is updated after inserting the astdb entry. As a consequence, restarting or reloading asterisk creates clients whose registration may expire before they reregister. The clients are therefore unavailable after minregexpire seconds until they reregister. * Move updating of the expiry time to before inserting into the astdb. (closes issue ASTERISK-22504) Reported by: Stefan Wachtler Patches: chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler ........ Merged revisions 399158 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399159 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5cf1cae533	Filter internal channels out of bridge enter/leave message handling ... Some channels exist merely as an implementation detail in Asterisk, such as ConfBridge's announcer/recorder channels. These channels should never be exposed to the outside world, or to interfaces that report on Asterisk. We already filter out such channels in snapshot processing; however, we failed to filter out bridge related messages that involved these channels. This patch filters out bridge related messages that are for such channels. This prevents a spurious WARNING message from being displayed when those channels move in and out of bridges. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	74c9781273	Restore Dial, Queue, and FollowMe 'I' option support. ... The Dial, Queue, and FollowMe applications need to inhibit the bridging initial connected line exchange in order to support the 'I' option. * Replaced the pass_reference flag on ast_bridge_join() with a flags parameter to pass other flags defined by enum ast_bridge_join_flags. * Replaced the independent flag on ast_bridge_impart() with a flags parameter to pass other flags defined by enum ast_bridge_impart_flags. * Since the Dial, Queue, and FollowMe applications are now the only callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the calling contract to require the initial COLP exchange to already have been done by the caller. * Made all callers of ast_bridge_impart() check the return value. It is important. As a precaution, I also made the compiler complain now if it is not checked. * Did some cleanup in parking_tests.c as a result of checking the ast_bridge_impart() return value. An independent, but associated change is: * Reduce stack usage in ast_indicate_data() and add a dropping redundant connected line verbose message. (closes issue ASTERISK-22072) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	fcde3a5320	Don't write to /tmp/refs when REF_DEBUG is not defined. ... If MALLOC_DEBUG is enabled, then the debug destructor for the container is used, which would erroneously write to /tmp/refs. This patch only uses the debug destructor if ref_debug is used. (closes issue ASTERISK-22536) ........ Merged revisions 399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399099 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	66834ff561	Create more accurate Contact headers for dialogs when we are the UAS. ... (closes issue AST-1207) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2842 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	b45699daaa	Broke the build! Forgot para tags within my description. ... https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a0f8a2e4ce	Change how realms are handled for outbound authentication. ... With this change, if no realm is specified in an outbound auth section, then we will simply match the realm that was present in the 401/407 challenge. (closes issue ASTERISK-22471) Reported by George Joseph (closes issue ASTERISK-22386) Reported by Rusty Newton Patches: outbound_auth_realm_v4.patch uploaded by George Joseph (License #6322) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	491188ab75	res_pjsip: Forward PJSIP logging to Asterisk logging ... This patch uses PJSIP's pj_log_set_log_func() to forward PJSIP's log messages to Asterisk's logger. This is done in a new module: res_pjsip_log_forwarder.so. This patch sets defaultenabled on the existing res_pjsip_logger.so to no, since logging every SIP packet seems a bit odd to do by default, and is (hopefully) less necessary with regular PJSIP logging. It also removes res_rtp_asterisk's disabling of PJSIP logging. (closes issue ASTERISK-22360) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2830/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	ae292e7e54	ARI: Fix WebSocket response when subprotocol isn't specified ... When I moved the ARI WebSocket from /ws to /ari/events, I added code to allow a WebSocket to connect without specifying the subprotocol if there's only one subprotocol handler registered for the WebSocket. Naively, I coded it to always respond with the subprotocol in use. Unfortunately, according to RFC 6455, if the server's response includes a subprotocol header field that "indicates the use of a subprotocol that was not present in the client's handshake [...], the client MUST _Fail the WebSocket Connection_.", emphasis theirs. This patch correctly omits the Sec-WebSocket-Protocol if one is not specified by the client. (closes issue ASTERISK-22441) Review: https://reviewboard.asterisk.org/r/2828/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	58f4d05287	Fix several crashes in MeetMeAdmin ... This change ensures that MeetMeAdmin commands requiring a user actually get a user and fixes another issue where an extra dereference could occur for a last-entered user being ejected if a user identifier was also provided. (closes issue ASTERISK-21907) Reported by: Alex Epshteyn Review: https://reviewboard.asterisk.org/r/2844/ ........ Merged revisions 399033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399034 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	4748a7c584	'identify' configObject doesn't have a synopsis ... Add a straightforward synopsis and description to the identify config object in XML documentation. (issue ASTERISK-22311) (closes issue ASTERISK-22311) Reported By: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	2b9f4fe644	CLI bridge: Fix "bridge destroy <id>" and "bridge kick <id> <chan>" tab completion. ... These two commands must deal with the live bridges container for tab completion and not the stasis cache. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9efc5a8a3f	astobj2: Register the bridges container for debug inspection. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	a4d929c226	Documentation fix and improvements to XML configuration help res_pjsip_acl ... * One bug fix. Made the synopsis for "type" to accurate. * changing the usage of "IP-domains" to "IP addresses" * clarifying the usage for the options, by adding a relevant description for each * modified other areas of the XML help for clarity, such as the module description and a few synopsis changes here and there. See the patch. (issue ASTERISK-22458) (closes issue ASTERISK-22458) Reported By: Rusty Newton Review: https://reviewboard.asterisk.org/r/2823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c1ffcb84b0	chan_sip: Revert r398835 due to failing tests involving originate ... (issue ASTERISK-22424) Reported by: Jonathan Rose ........ Merged revisions 398977 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398986 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	be81695a85	core_local: Fix memory corruption race condition. ... The masquerade super test is failing on v12 with high fence violations and crashing. The fence violations are showing that party id allocated memory strings are somehow getting corrupted in the bridge_reconfigured_connected_line_update() function. The invalid string values happen to be the freed memory fill pattern. After much puzzling, I deduced that the bridge_reconfigured_connected_line_update() is copying a string out of the source channel's caller party id struct just as another thread is updating it with a new value. The copying thread is using the old string pointer being freed by the updating thread. A search of the code found the unreal_colp_redirect_indicate() routine updating the caller party id's without holding the channel lock. A latent bug in v1.8 and v11 hatched in v12 because of the bridging and connected line changes. :) (issue ASTERISK-22221) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2839/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	9757d56a67	Fix symbol collision with pjsua. ... We shouldn't be exporting any symbols that start with pjsip_. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	f2f8770494	'queue add member' help text correction ... You are adding dial strings to the queue, not channels. An aribitrary string could be used, but you are typically referencing a channel. Correcting the command help text. (issue ASTERISK-22263) (closes issue ASTERISK-22263) Reported By: Rusty Newton ........ Merged revisions 398884 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398885 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	cc92876275	Documentation fix - waitfordialtone is not boolean, it's time in milliseconds ... Changing text in chan_dahdi.conf sample to be accurate. (issue ASTERISK-22308) (closes issue ASTERISK-22308) Reported By: Malcolm Davenport ........ Merged revisions 398880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398881 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	1324ab39af	chan_sip: Reject calls without prior SDP on 200 OK ... If we receive a 200 OK without SDP, we will now check to see if the remote address has been established for that channel's RTP session and if the to tag for that channel has changed from the most recent to tag in a response less than 200. If either a change has been made since the last to-tag was received or the remote address is unset, then we will drop the call. (closes issue ASTERISK-22424) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header ........ Merged revisions 398835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398836 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	627ef51096	Fix typo in confbridge.conf.sample ... The denoise filter requires func_speex, not codec_speex. Fix this in the description of the denoise=yes option in confbridge.conf. ........ Merged revisions 398820 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	cd8720b3ec	pjsip: reinvite for connected line updates occurs when it should not ... Connected line updates are now only sent out if an actual update needs to occur. This happens under the following conditions: 1. The endpoint we are sending to is trusted. 2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent. 3. The connected id's number and name are valid. Also added an SDP when an update is sent out. (closes issue AST-1212) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2831/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b9f8a74838	Fix incorrect usages of ast_realloc(). ... There are several locations in the code base where this is done: buf = ast_realloc(buf, new_size); This is going to leak the original buf contents if the realloc fails. Review: https://reviewboard.asterisk.org/r/2832/ ........ Merged revisions 398757 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398758 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	98ecd8e64b	Fixed utils directory breakage from r398748, this time with extra hate. ... ........ Merged revisions 398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398753 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398754 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	dce5ab61e9	Fixed utils directory breakage from r398648 ... ........ Merged revisions 398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398749 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	cca2273fd2	MALLOC_DEBUG: Change fence magic number to be completely different from the freed magic number. ... Race conditions between freeing a nul terminated string and ast_strdup()'ing it are more likely to be detected if the fence and freed magic numbers are completely different. ........ Merged revisions 398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398721 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9e7efc9560	Add extra debugging to res_pjsip_endpoint_identifier_ip ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	ffd49f6b2d	Fix DEBUG_THREADS when lock is acquired in __constructor__ ... This patch fixes some long-standing bugs in debug threads that were exacerbated with recent Optional API work in Asterisk 12. With debug threads enabled, on some systems, there's a lock ordering problem between our mutex and glibc's mutex protecting its module list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module list will be locked before acquiring our mutex. In another thread, our mutex will be locked before locking the module list (which happens in the depths of calling backtrace()). This patch fixes this issue by moving backtrace() calls outside of critical sections that have the mutex acquired. The bigger change was to reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed that waiting on the mutex was equivalent to a single unlock (it actually suspends all recursive locks on the mutex). (closes issue ASTERISK-22455) Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged revisions 398648 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398649 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	8aabad21c0	Added note about expected behavior of originate (the rest of the commit) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	730ad84481	Added note about expected behavior of originate ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8e5157d869	Update CDR Unit tests to reflect container changes in r398579 ... When a channel joins a multi-party bridge, the ordering of the CDRs that is created is determined by the ordering of the channels who happen to be in that bridge. When r398579 changed the number of buckets in the container to something sensible, it changed the ordering that the CDRs was created in, causing one of the multiparty tests to fail. This fixes the test with the now expected ordering. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	68cc3b279e	Prevent XMPP timeout on blank responses ... Sometimes the Google Voice servers have a bad habit of sending out 1 byte replies to the xmpp resource. When a blank 1 byte reply is received from the socket the buffer attempts to wait (endlessly) for the rest of the reply from google which effectively blocks the socket and google voice calls will no longer come into the server. This patch allows the xmpp module to correctly detect empty packets and send out ping replies to google. It also sets a socket timeout on the default socket which prevents the xmpp socket from closing and preventing future google voice calls from coming into the server. Furthermore instead of sending an empty reply back to google we send a proper xmpp ping reply back. This also adds several more socket messages. (closes issue ASTERISK-22347) Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/2771 Patches: xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524) ........ Merged revisions 398618 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	f2837725b3	Multiple revisions 398558,398577 ... ........ r398558 | kmoore | 2013-09-06 14:28:16 -0500 (Fri, 06 Sep 2013) | 17 lines Fix Jabber/XMPP distributed MWI The mailbox and context are swapped on the receiving end for all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent versions. This swaps those values to be correct when publishing to the internal event system from Jabber/XMPP distributed MWI state. (closes issue ASTERISK-22435) Reported by: abelbeck Tested by: Michael Keuter Patches: asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck ........ Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) | 10 lines Commit the remainder of r398523 This is a missing part of the commit in revision 398523 that corrects the name of a variable. (issue ASTERISK-22435) ........ Merged revisions 398576 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398558,398577 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	83d741f85e	cdr: Change the number of container buckets to be similar to the channels container. ... * Fix the temporary cdr candidate containers to use a prime number of buckets. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	374c3f4336	Unblock r398558 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ad42934fbc	core_local: Fix LocalOptimizationBegin AMI event missing Source channel snapshot. ... * Fix the LocalOptimizationBegin AMI event by eliminating an artificial buffer size limitation that is too small anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398572 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	334329de07	cdr: Fix some ref leaks. ... * Added missing unregister of the cdr container in cdr_engine_shutdown(). * Fixed ref leak in off nominal path of cdr_object_alloc(). * Removed some unnecessary NULL checks in cdr_object_dtor(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	90982cbcf0	Blocked revisions 398558 ... ........ Fix Jabber/XMPP distributed MWI The mailbox and context are swapped on the receiving end for all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent versions. This swaps those values to be correct when publishing to the internal event system from Jabber/XMPP distributed MWI state. (closes issue ASTERISK-22435) Reported by: abelbeck Tested by: Michael Keuter Patches: asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck ........ Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago