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2013-09-17Fix doxygen to use correct units of features.conf options.Richard Mudgett
........ Merged revisions 399257 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-17Fix other timeouts (atxferloopdelay and atxfernoanswertimeout) to use ↵Mark Michelson
seconds instead of milliseconds. Thanks to Richard Mudgett for pointing this out. ........ Merged revisions 399247 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-17Switch transferdigittimeout to be configured as seconds instead of milliseconds.Mark Michelson
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. ........ Merged revisions 399237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-17Confbridge: empty conference not being torn downKevin Harwell
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 ........ Merged revisions 399225 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-16Fix module load errors for test_ari_model.so.Richard Mudgett
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 ........ Merged revisions 399207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-16app_speech_utils: Fix unresolved symbol ast_speech_get_setting().Richard Mudgett
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 ........ Merged revisions 399197 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-16chan_iax2: Fix saving the wrong expiry time in astdb.Richard Mudgett
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 ........ Merged revisions 399160 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-16Filter internal channels out of bridge enter/leave message handlingMatthew Jordan
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. ........ Merged revisions 399146 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Restore Dial, Queue, and FollowMe 'I' option support.Richard Mudgett
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/ ........ Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Don't write to /tmp/refs when REF_DEBUG is not defined.David M. Lee
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 ........ Merged revisions 399100 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Create more accurate Contact headers for dialogs when we are the UAS.Mark Michelson
(closes issue AST-1207) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2842 ........ Merged revisions 399083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Change how realms are handled for outbound authentication.Mark Michelson
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) ........ Merged revisions 399059 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Recorded merge of revisions 399035,399049 from ↵David M. Lee
http://svn.asterisk.org/svn/asterisk/branches/12 These were lost in r399071 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Put merge tracking for r399039 back.David M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Broke the build! Forgot para tags within my description.Rusty Newton
https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304 ........ Merged revisions 399064 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13res_pjsip: Forward PJSIP logging to Asterisk loggingDavid M. Lee
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/ ........ Merged revisions 399049 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13ARI: Fix WebSocket response when subprotocol isn't specifiedDavid M. Lee
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/ ........ Merged revisions 399039 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Fix several crashes in MeetMeAdminKinsey Moore
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 ........ Merged revisions 399035 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13'identify' configObject doesn't have a synopsisRusty Newton
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 ........ Merged revisions 399031 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12CLI bridge: Fix "bridge destroy <id>" and "bridge kick <id> <chan>" tab ↵Richard Mudgett
completion. These two commands must deal with the live bridges container for tab completion and not the stasis cache. ........ Merged revisions 399021 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12astobj2: Register the bridges container for debug inspection.Richard Mudgett
........ Merged revisions 399019 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12Documentation fix and improvements to XML configuration help res_pjsip_aclRusty Newton
* 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/ ........ Merged revisions 399017 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12chan_sip: Revert r398835 due to failing tests involving originateJonathan Rose
(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 ........ Merged revisions 398991 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12core_local: Fix memory corruption race condition.Richard Mudgett
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/ ........ Merged revisions 398938 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12Fix symbol collision with pjsua.David M. Lee
We shouldn't be exporting any symbols that start with pjsip_. ........ Merged revisions 398927 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12'queue add member' help text correctionRusty Newton
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 ........ Merged revisions 398886 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11Documentation fix - waitfordialtone is not boolean, it's time in millisecondsRusty Newton
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 ........ Merged revisions 398882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11chan_sip: Reject calls without prior SDP on 200 OKJonathan Rose
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 ........ Merged revisions 398837 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11Fix typo in confbridge.conf.sampleRussell Bryant
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 ........ Merged revisions 398821 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11pjsip: reinvite for connected line updates occurs when it should notKevin Harwell
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/ ........ Merged revisions 398806 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-10Fix incorrect usages of ast_realloc().Richard Mudgett
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 ........ Merged revisions 398759 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-10Fixed utils directory breakage from r398748, this time with extra hate.David M. Lee
........ 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 ........ Merged revisions 398754 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-10Fixed utils directory breakage from r398648David M. Lee
........ 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 ........ Merged revisions 398750 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09MALLOC_DEBUG: Change fence magic number to be completely different from the ↵Richard Mudgett
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 ........ Merged revisions 398726 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09Add extra debugging to res_pjsip_endpoint_identifier_ipMark Michelson
........ Merged revisions 398694 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09Fix DEBUG_THREADS when lock is acquired in __constructor__David M. Lee
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 ........ Merged revisions 398651 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09Multiple revisions 398638-398639David M. Lee
........ r398638 | dlee | 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line Added note about expected behavior of originate ........ r398639 | dlee | 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line Added note about expected behavior of originate (the rest of the commit) ........ Merged revisions 398638-398639 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09Blocked revisions 398559,398578David M. Lee
........ Multiple revisions 398559,398578 ........ r398559 | kmoore | 2013-09-06 14:32:03 -0500 (Fri, 06 Sep 2013) | 20 lines 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 ........ r398578 | kmoore | 2013-09-06 16:03:45 -0500 (Fri, 06 Sep 2013) | 1 line Unblock r398558 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-08Update CDR Unit tests to reflect container changes in r398579Matthew Jordan
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. ........ Merged revisions 398628 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-07Prevent XMPP timeout on blank responsesKinsey Moore
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 ........ Merged revisions 398619 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06Multiple revisions 398558,398577Kinsey Moore
........ 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 ........ Merged revisions 398580 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06cdr: Change the number of container buckets to be similar to the channels ↵Richard Mudgett
container. * Fix the temporary cdr candidate containers to use a prime number of buckets. ........ Merged revisions 398579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06core_local: Fix LocalOptimizationBegin AMI event missing Source channel ↵Richard Mudgett
snapshot. * Fix the LocalOptimizationBegin AMI event by eliminating an artificial buffer size limitation that is too small anyway. ........ Merged revisions 398572 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06cdr: Fix some ref leaks.Richard Mudgett
* 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(). ........ Merged revisions 398562 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06astobj2: Add warn unused attribute to some functions.Richard Mudgett
* Fixed resulting warnings with improper use of ao2_global_obj_replace(). * Made a couple uses of ao2_global_obj_replace_unref(x, NULL) into the equivalent and more appropriate ao2_global_obj_release() call. ........ Merged revisions 398533 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06Fix build warningsKinsey Moore
When AST_DEVMODE is not defined, ast_asserts are not compiled into the binary. In some cases, this means variables are not referenced or are set but unused which causes warnings to show up. (closes issue ASTERISK-22446) Reported by: Jason Parker (qwell) ........ Merged revisions 398521 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06Fix chan_h323 compilationKinsey Moore
This fixes the things in chan_h323 that were missed or ignored in the great channel opaquification and gets chan_h323 back into a compiling state. (closes issue ASTERISK-22365) Reported by: Dmitry Melekhov Patches: chan_h323.patch uploaded by Dmitry Melekhov ........ Merged revisions 398510 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398511 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-05astobj2: Only define ao2_bt() once.Richard Mudgett
* Make ao2_bt() not use single char variable names. * Fix ao2_bt() formatting. ........ Merged revisions 398498 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-05chan_iax2: Reduce indentation in __attempt_transmit().Richard Mudgett
* Reduce indentation in __attempt_transmit(). * Don't update the static last error time variable every time in __schedule_action() and socket_read(). ........ Merged revisions 398456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398457 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398458 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-05chan_iax2: Fix stray reference to worker thread idle_list.Richard Mudgett
* Fix stray reference to idle_list in cleanup_thread_list(). This may be the reason for the note in iax2_process_thread() about threads not being removed from the task lists. * Move cleanup_thread_list(&idle_list) to after the other lists are cleaned up. ........ Merged revisions 398416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398417 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398418 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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