summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2012-01-20SIP session timeout AMI eventKinsey Moore
Add an AMI event in the Call category that is issued when a call is terminated due to either RTP stream inactivity or SIP session timer expiration. Event description: Event: SessionTimeout Source: source Channel: channel-name Uniqueid: channel-unique-id `source` can be either RTPTimeout or SIPSessionTimer (closes issue ASTERISK-16467) Patch-by: Kirill Katsnelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Various parking improvements.Mark Michelson
* Adds per-parking lot options comebackcontext and comebackdialtime * Makes comebacktoorigin settable per parking lot * Sets a PARKER channel variable when comebacktoorigin is disabled (closes issue ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches: asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231 with updates by me. Review: https://reviewboard.asterisk.org/r/1674 Review: https://reviewboard.asterisk.org/r/963 Reviewed by Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Prevent potential buffer overflow on AMI MixMonitor command.Mark Michelson
Don't be alarmed. This only affected trunk, and it would have required manager access to your system. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20More corrections for the ilbc codeKinsey Moore
These changes are in a file that is not compiled by default, and so were missed on earlier checks. ........ Merged revisions 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351861 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Restore LSF_check function calls from set/unused variable removalKinsey Moore
These functions are not noops and modify the array that is passed in. Thanks for the catch Richard. ........ Merged revisions 351818 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Remove more set, but unused variables in the ilbc codecKinsey Moore
GCC 4.6.3 caught these in dev mode as well. ........ Merged revisions 351816 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Adds setting of mwi_from field to check_auth_result check_peer_okJonathan Rose
(closes ASTERISK-19057) Reported By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license 5242) ........ Merged revisions 351759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351762 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20Remove unused variable 'tmp' from helpfun in ilbc codecMatthew Jordan
gcc version 4.6.2 caught an unused variable in the ilbc codec library. This would prevent compilation with --enable-dev-mode; variable removed. ........ Merged revisions 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351761 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20enable doxygen build for files in the channels/sip folder like reqresp_parser.cStefan Schmidt
........ Merged revisions 351707 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351708 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Misc minor fixes in reqresp_parser.c and chan_sip.c.Richard Mudgett
* Fix corner cases in get_calleridname() parsing and ensure that the output buffer is nul terminated. * Make get_calleridname() truncate the name it parses if the given buffer is too small rather than abandoning the parse and not returning anything for the name. Adjusted get_calleridname_test() unit test to handle the truncation change. * Fix get_in_brackets_test() unit test to check the results of get_in_brackets() correctly. * Fix parse_name_andor_addr() to not return the address of a local buffer. This function is currently not used. * Fix potential NULL pointer dereference in sip_sendtext(). * No need to memset(calleridname) in check_user_full() or tmp_name in get_name_and_number() because get_calleridname() ensures that it is nul terminated. * Reply with an accurate response if get_msg_text() fails in receive_message(). This is academic in v1.8 because get_msg_text() can never fail. ........ Merged revisions 351618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351646 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Correct output of RTCP jitter statistics in SR and RR reportsKinsey Moore
Change the RTCP RR and SR generation code to convert Asterisk's internal jitter statistics to be represented in RTP timestamp units based on the rate of the codec in use instead of in seconds. (closes issue ASTERISK-14530) ........ Merged revisions 351611 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351612 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Eliminates doubling the :port part of SIP Notify Message-Account headers.Jonathan Rose
This patch prevents the domain string from getting mangled during the initreqprep step by moving the initialization to before its immediate use. It also documents this pitfall for the ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported by: Yuri Review: https://reviewboard.asterisk.org/r/1678/ ........ Merged revisions 351559 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351560 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Prevent crash when an SDP offer is received with an encrypted video stream ↵Joshua Colp
when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) Reported by: Catalin Sanda ........ Merged revisions 351504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351505 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-18Include iLBC source code for distribution with AsteriskMatthew Jordan
This patch includes the iLBC source code for distribution with Asterisk. Clarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder. Review: https://reviewboard.asterisk.org/r/1675 Review: https://reviewboard.asterisk.org/r/1649 (closes issue: ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan ........ Merged revisions 351450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351451 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-18The get_pai function in chan_sip.c didn't recognized a proper callerid name andStefan Schmidt
number from a P-Asserted-Identity cause the header parsing logic was wrong. Changing the parsing functions to the sip header parsing APIs in reqresp_parser.h solves this problem. Review: https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and Mark Michelson ........ Merged revisions 351396 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351408 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Fix support for parallel building with make (-j).Walter Doekes
Previously make -j <N> would cause a race between doing cleanup of certain files (defaults.h, menuselect, ...) and creating them anew. Add a new target that depends on cleanup only and has a submake doing the rest as command string. This way the cleanup goes first. (closes issue ASTERISK-18751) Tested by: Jeremy Kister Reviewed by: Paul Belanger Review: https://reviewboard.asterisk.org/r/1660 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Eliminate odd initialization of probation variable.Mark Michelson
........ Merged revisions 351306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351308 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Adds pjmedia probation concepts to res_rtp_asterisk's learning mode.Jonathan Rose
In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ ........ Merged revisions 351287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351289 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Use built-in parsing functions for Contact and Record-Route headers.Mark Michelson
If a Contact or a Record-Route header had a quoted string with an item in angle brackets, then we would mis-parse it. For instance, "Bob <1234>" <1234@example.org> would be misparsed as having the URI "1234" The fix for this is to use parsing functions from reqresp_parser.h since they are heavily tested and are awesome. (issue ASTERISK-18990) ........ Merged revisions 351284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351286 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Fix udptl issue with initial INVITE introduced by r351027Matthew Jordan
When an inital INVITE occurs that contains image media, a channel is not yet associated with the SIP dialog. The file descriptor associated with the udptl session needs to be set in initialize_udptl or in sip_new to account for this scenario. ........ Merged revisions 351233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351234 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-17Merged revisions 351183 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r351183 | russell | 2012-01-16 20:43:19 -0500 (Mon, 16 Jan 2012) | 29 lines Merged revisions 351182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines Add some missing locking in chan_sip. This patch adds some missing locking to the function send_provisional_keepalive_full(). This function is called from the scheduler, which is processed in the SIP monitor thread. The associated channel (or pbx) thread will also be using the same sip_pvt and ast_channel so locking must be used. The sip_pvt_lock_full() function is used to ensure proper locking order in a safe manner. In passing, document a suspected reference counting error in this function. The "fix" is left commented out because when the "fix" is present, crashes occur. My theory is that fixing it is exposing a reference counting error elsewhere, but I don't know where. (Or my analysis of this being a problem could have been completely wrong in the first place). Leave the comment in the code for so that someone may investigate it again in the future. Also add a bit of doxygen to transmit_provisional_response(). (closes issue ASTERISK-18979) Review: https://reviewboard.asterisk.org/r/1648 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Ensure ACK retransmit & hangup on non-200 response to INVITETerry Wilson
When handling a non-2xx final response on an INVITE transaction, we have to keep the transaction around after we send an ACK in case we receive a retransmission of the response so we can re-transmit the ACK, but also tear down the ast_channel as soon as we transmit the ACK. Before this patch, we could fail at both of these things. Calling sip_alreadygone/needdestroy prevented us from keeping the transaction up and retransmitting the ACK, and queueing CONGESTION was not sufficient to cause the channel to be torn down when originating calls via the CLI, for example. This patch queues a hangup with CONGESTION instead of just queueing CONGESTION for these responses and removes the sip_alreadygone and sip_needdestroy calls from handle_response_invite on non-2xx responses. It relies on the hangup calling sip_scheddestroy. For more information, see section 17.1.1.1 of RFC 3261. (closes issue ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/ ........ Merged revisions 351130 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351131 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Don't prematurely stop SIP session timerTerry Wilson
When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry. (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: session_timer_fix.diff by Terry Wilson (License #5357) based on session_timer.patch by Thomas Arimont (License #5525) ........ Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351081 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Add ABS() absolute value function to the expression parser.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Create and initialize udptl only when dialog negotiates for image mediaMatthew Jordan
Prior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or what an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This occurred even in non-INVITE dialogs that would never send image media. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates through the appropriate control frame that a dialog is to support T.38. (closes issue ASTERISK-16698) Reported by: under Tested by: Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar Broad Tested by: Stefan Schmidt review: https://reviewboard.asterisk.org/r/1668/ ........ Merged revisions 351027 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351028 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Sort the output of 'database showkey' as well.Sean Bright
You can pass wildcards (%) to the database CLI commands, so this will sort the returned list of matches. ........ Merged revisions 350978 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Add missing code to set direct RTP setup information during dialing.Joshua Colp
........ Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350976 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-16Sort the output of 'database show' by key.Sean Bright
This more closely mimics the behavior of 'database show' before the conversion to sqlite3. ........ Merged revisions 350938 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-15Allow only one thread at a time to do asterisk cleanup/shutdown.Walter Doekes
Add locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1662/ Review: https://reviewboard.asterisk.org/r/1658/ ........ Merged revisions 350888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350889 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-15Fix -Werror=unused-but-set-variable compile error in utils/extconf.c.Walter Doekes
Note that I'm not confirming legitimacy of having that file in tree at all. Is anyone using aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged revisions 350885 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350886 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-14Ensure that all AC_LANG_PROGRAM calls in the configure script are properly ↵Kevin P. Fleming
quoted. Recent versions of autoconf (2.68 on my system) won't properly process the configure script unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in the script were, but many were not. This patch corrects the unquoted calls. ........ Merged revisions 350837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350838 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-14Multiple revisions 350788-350789Kevin P. Fleming
........ r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two prerequisites are properly installed on Debian-style distributions. * Don't specify a specific version of libgmime; newer versions are available now and acceptable. * Install libsrtp so that res_srtp can be built. ........ r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines Correct some 'set-but-not-used' variable warnings. ........ Merged revisions 350788-350789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350790 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Run bootstrap.sh for the for the ASTERISK-18929 fixKinsey Moore
configure and autoconfig.h.in were not regenerated when the fix was committed. ........ Merged revisions 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350737 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Correct eventtype names in cel_odbc and cel_pgsql sample filesRichard Mudgett
........ Merged revisions 350733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350734 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Make sure asterisk builds on OpenBSDKinsey Moore
OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not 'struct ucred', which causes compilation of main/asterisk.c to fail in read_credentials(). This allows configure to check for sockpeercred and asterisk to deal with it properly. (closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller ........ Merged revisions 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350731 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Set port to a default sane value if a bogus one is provided when parsing ↵Mark Michelson
hostnames. ........ Merged revisions 350679 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350680 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Remove some dead code in ast_bridge_call().Richard Mudgett
None of the parameters to ast_bridge_call() can be NULL for the bridge to work so no need to check for it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Add missing CEL logging fields to various CEL backends.Richard Mudgett
Multiple revisions 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51 -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging fields to various CEL backends. * Add missing eventextra to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and EventExtra to cel_manager.c. * Add missing userdeftype support for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 Jan 2012) | 8 lines Use compatible names for event extra data for various CEL backends. * Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190) ........ Merged revisions 350555,350571 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350585 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Realtime queues failed to load queue information without queue member tableMatthew Jordan
Previously, realtime queues could be loaded without defining the queue member table. This allowed for queue members to be dynamic, while the realtime queue definitions could exist in some backing storage. Revision 342223 broke this when it changed the return value for realtime_multientry to return NULL when no results are returned. Previously, an empty ast_config object was expected. (closes issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 350552 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350553 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Fix crash from bridge channel hangup race condition in ConfBridgeMatthew Jordan
This patch addresses two issues in ConfBridge and the channel bridge layer: 1. It fixes a race condition wherein the bridge channel could be hung up 2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object Patch by David Vossel (mjordan was merely the commit monkey) (issue ASTERISK-18988) (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628) (closes issue ASTERISK-19100) Reported by: Matt Jordan Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1654/ ........ Merged revisions 350550 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-12Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_ENDJonathan Rose
(closes issue ASTERISK-17940) Reporter: Nic Colledge Patches: features_18.patch uploaded by Nic Colledge (license 6245) ........ Merged revisions 350501 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350502 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-11Remove extraneous BRIDGEPEER AMI VarSet event on a CEL dummy channel.Richard Mudgett
(closes issue ASTERISK-19180) Reported by: Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350453 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-11Make FollowMe optionally update connected line information when the ↵Richard Mudgett
accepting endpoint is bridged. Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. * Made 'N' option ignored if the call is already answered. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ ........ Merged revisions 350364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350415 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-11Always treat arguments to get_by_name_cb as stringsTerry Wilson
Initially, support was left in for the old style of searching, even though it wasn't actually used. In the case of name_len != 0, the OBJ_KEY flag isn't passed because we aren't matching on a full key and therefor can't use the hash function to optimize. The code left in to support the old way of searching unfortunately treated a prefix search like this as though an ast_channel struct was passed as an arg and caused a crash. This patch also adds needed parentheses around some matching conditions. (closes issue ASTERISK-19182) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-10Fix absolute/relative time mismatch in LOCK function.Richard Mudgett
The time passed by the LOCK function to an internal function was relative time when the function expected absolute time. * Don't use C++ keywords in get_lock(). (closes issue ASTERISK-16868) Reported by: Andrey Solovyev Patches: 20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified) ........ Merged revisions 350311 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350312 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Fix compiler warnings reported by gcc v4.2.4.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Replace direct access to channel name with accessor functionsTerry Wilson
There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Fix joinable thread terminating without joiner memory leak in chan_iax.c.Richard Mudgett
The iax2_process_thread() can exit without anyone waiting to join the thread. If noone is waiting to join the thread then a large memory leak occurs. * Made iax2_process_thread() deatach itself if nobody is waiting to join the thread. (closes issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified) (closes issue ASTERISK-17825) Reported by: wangjin ........ Merged revisions 350220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350221 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Fix shutdown handling of sqlite3 astdb.Walter Doekes
If a db_sync was scheduled just before shutdown, the atexit code calling db_sync would have no effect, causing the astdb commit thread to stay alive. This caused the SIP/realtime_sipregs test to fail. (The fallback kill would run the atexit code again and that would wreak havoc.) This fixes that the atexit kill condition is picked up properly. (closes issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1659 ........ Merged revisions 350180 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Multiple revisions 350127-350128Richard Mudgett
........ r350127 | rmudgett | 2012-01-09 12:40:33 -0600 (Mon, 09 Jan 2012) | 12 lines Update contrib script live_ast to invoke Asterisk with valgrind and suppression file. * Added valgrind_compare script to compare two valgrind log files for differences. (issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: valgrind_compare (license #5035) script uploaded by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger ........ r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines live_ast: valgrind: run asterisk under valgrind Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under valgrind. The extra command-line parameters are passed to Asterisk as usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 350127-350128 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350129 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-12 09:50:50 +0000