asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	19af5d6103	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/branches/10@351183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b003707140	Ensure ACK retransmit & hangup on non-200 response to INVITE 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@351131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	75f8a7940a	Don't prematurely stop SIP session timer 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@351081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	76c9a985a4	Create and initialize udptl only when dialog negotiates for image media 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@351028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	e91ada0ae4	Sort the output of 'database showkey' as well. You can pass wildcards (%) to the database CLI commands, so this will sort the returned list of matches. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	ca3a020b0a	Add missing code to set direct RTP setup information during dialing. ........ Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	0e41fe0f2f	Sort the output of 'database show' by key. This more closely mimics the behavior of 'database show' before the conversion to sqlite3. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	dd6c7a95c3	Allow only one thread at a time to do asterisk cleanup/shutdown. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	63919d6167	Fix -Werror=unused-but-set-variable compile error in utils/extconf.c. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	0a83562d7a	Ensure that all AC_LANG_PROGRAM calls in the configure script are properly 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	bbf49866fd	Multiple revisions 350788-350789 ........ 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	bb11601d06	Run bootstrap.sh for the for the ASTERISK-18929 fix 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350737 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	bb4c8dc0a4	Correct eventtype names in cel_odbc and cel_pgsql sample files ........ Merged revisions 350733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	491d50fdcb	Make sure asterisk builds on OpenBSD 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	c456ccca60	Set port to a default sane value if a bogus one is provided when parsing hostnames. ........ Merged revisions 350679 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4ed91e3f7e	Add missing CEL logging fields to various CEL backends. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	fa56c109a3	Realtime queues failed to load queue information without queue member table 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	d3fbfc50f4	Fix crash from bridge channel hangup race condition in ConfBridge 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	4bbda2752c	Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_END (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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	05e229b4e0	Remove extraneous BRIDGEPEER AMI VarSet event on a CEL dummy channel. (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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	031a95b7ec	Make FollowMe optionally update connected line information when the 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0de8f20ce3	Fix absolute/relative time mismatch in LOCK function. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ab8a7f838a	Fix joinable thread terminating without joiner memory leak in chan_iax.c. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	c5f91df206	Fix shutdown handling of sqlite3 astdb. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	fdf0fd973b	Multiple revisions 350127-350128 ........ 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0265617b20	Make Asterisk -x command line parameter imply -r parameter presence. The Asterisk -x command line parameter is documented inconsistently. * Made the -x documentation and behavior consistent. * Since this is also a new year, updated the copyright notices while here. (closes issue ASTERISK-19094) Reported by: Eugene Patches: issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified) Tested by: Eugene ........ Merged revisions 350075 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	2d27c6c255	Prevent SLA settings from getting wiped out on reload If SLA was reloaded without the config file being changed, current settings got wiped out before the SLA reload code decided it wasn't going to reload the file since nothing was changed. Moving the settings reset later in the reload process fixes this. (closes issue AST-744) ........ Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3ca369ec95	Don't leak CID in From header when presentation=unavailable When someone does Set(CALLERPRES()=unavailable) (or Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows "Anonymous" <anonymous@anonymous.invalid>. When sendrpid=yes/pai, the From header will still display the callerid info, even though we supply an rpid header with the anonymous info. It seems like we shouldn't leak that info in any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems to indicate that one shouldn't send identifying info in the From in this case. This patch anonymizes the From header as well even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review: https://reviewboard.asterisk.org/r/1649/ ........ Merged revisions 349968 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	28d121076d	Fix lua goto detection to prevent unexpected behavior with confbridge A bug in the pbx_lua goto detection was causing the dialplan to hangup unexpectedly after confbridge exited if it had called lua dialplan code during execution. Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue ASTERISK-18976) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c5c52328c4	Fix memory leaks in app_followme find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt Jordan ........ Merged revisions 349872 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	5f8e471f1c	Fix premature free'ing of the frame committed in r349608 Even though we set the frame to the ast_null_frame and return that, the caller of the frame hook may still need the frame. This now is a bit more careful about when it frees the frame, i.e., only under the same conditions that applied when we duplicated it in the first place. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9b53ea3943	Make not assume that the cel_sqlite3_custom SQL table primary key is AcctId. If a table is created by some other application and the primary key is not named "AcctId", cel/cel_sqlite3_custom.c will always try to create the table and fail because it already exists. * Change the SQL table query to not require AcctId as the primary key. (closes issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch (license #6337) patch uploaded by socketpair ........ Merged revisions 349819 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	41b01e009f	Allow playback of formats that don't support seeking ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. This regression was introduced in r158062. (closes issue ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	fbcc23d61e	Fix an issue where dsp.c would interpret multiple dtmf events from a single key press. When receiving calls from a mobile phone into a DISA system on a connection with significant interference, the reporter's Asterisk system would interpret DTMF incorrectly and replicate digits received. This patch resolves that by increasing the number of frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and adjusts dtmf_detect function to reset hits and misses only when an edge is detected. (closes issue ASTERISK-17493) Reported by: Alec Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546) Review: https://reviewboard.asterisk.org/r/1130/ ........ Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	3fe594bf03	Ensures Asterisk closes when receiving terminal signals in 'no fork' mode. When catching a signal, in no fork mode the console thread is identical to the thread responsible for catching the signal and closing Asterisk, which requires it to first dispense with the console thread. Prior to this patch, if these threads were identical, upon receiving a killing signal, the thread will send an URG signal to itself, which we also catch and then promptly do nothing with. Obviously this isn't useful behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 349672 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	12d2720aa4	Fix for ConfBridge config parser unlocking channel mutex too many times When looking up a ConfBridge profile, the config parser would, if it found a channel datastore on the channel requesting the bridge profile, unlock the channel mutex twice. Since that's a little aggressive, it now only unlocks it once. (closes issue ASTERISK-19042) Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042 uploaded by David Vossel (license 5628) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	e8d238e896	Free successfully translated frame in fax_gateway_framehook A frame that is translated via ast_translate is also duplicated via ast_frdup. This will allocate a new frame on the heap, which needs to be free'd at the appropriate time. This issue reporter used valgrind to find that this occurred in res_fax's fax_gateway_framehook; a quick search through the code showed that only place this was currently not handling the translatted frame properly. (closes issue ASTERISK-19133) Reported by: Sylvain Rochet git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2defa0b88e	Fix segfault in chan_dahdi for CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private pointer checks in the following chan_dahdi channel callbacks: dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by: Diego Aguirre Tested by: rmudgett ........ Merged revisions 349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	51620e6c29	Make debian init script conform to the LSB standard Previously, this init script would return 1 if Asterisk was already running. This is incorrect behavior according to the LSB standard and has been fixed by returning 0 instead. (closes issue ASTERISK-17958) Reported-by: johnc ........ Merged revisions 349529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	269519b731	Update autosupport script and man page Added information collection from the output of the utilities: top, free, uptime, ifconfig Added information collection from the output of the Asterisk command 'dahdi show status' Added option / flag '-n, --non-interactive' Updated man page to reflect new option / flag '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes issue AST-749) ........ Merged revisions 349504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	480e7da3c0	Adds Subscription-State header to notify with call completion. per RFC3265 (Closes issue ASTERISK-17953) Reported by: George Konopacki Patches: 19400.patch uploaded by mmichelson (license 5049) ........ Merged revisions 349482 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	9c7126146d	Fix documentation for SayNumber to reflect the fact that language is changed in CHANNEL() (closes issue ASTERISK-18962) reported by: Nir Simionovich ........ Merged revisions 349450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	631cdbab31	Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (issue ASTERISK-19040) (issue ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/ ........ Merged revisions 349339 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	bad3035124	Use ast_audiohook_write_list_empty to determine if our lists are empty instead of duplicating that logic. ........ Merged revisions 349289 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	5b4fd1ca77	Tell Subversion to gnore the 'astdb2bdb' binary file if it exists. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	a881e1740d	Improve T.38 gateway V.21 preamble detection. This commit removes the V.21 preamble detection code previously added to the generic DSP implementation in Asterisk, and instead enhances the res_fax module to be able to utilize V.21 preamble detection functionality made available by FAX technology modules. This commit also adds such support to res_fax_spandsp, which uses the Spandsp modem tone detection code to do the V.21 preamble detection. There should be no functional change here, other than much more reliable V.21 preamble detection (and thus T.38 gateway initiation). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	580c6f9a48	Fix timing source dependency issues with MOH Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patches: asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026) Review: https://reviewboard.asterisk.org/r/1578/ ........ Merged revisions 349194 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	5fa6d211f3	Once an audiohook is attached to a channel, we continue to transcode all of the frames, even after all of the hooks are detached. This patch short-cicuits us out before we transcode unnecessarily. ........ Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	36410bc60a	Merged revisions 349044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r349044 \| seanbright \| 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) \| 18 lines In ChanSpy, don't create audiohooks that will never be used. When ChanSpy is initialized it creates and attaches 3 audiohooks: 1) Read audio off of the channel that we are spying on 2) Write audio to the channel that we are spying on 3) Write audio to the channel that is bridged to the channel that we are spying on. The first is always necessary, but the others are used only when specific options are passed to the ChanSpy application (B, d, w, and W to be specific). When those flags are not passed, neither of those audiohooks are ever sent frames, but we still try to process the hooks for each voice frame that we recieve on the channel. So in short - only create and attach audiohooks that we actually need. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	0c81db6d3d	Fix missing doc tags found while fixing ASTERISK-18689 Add missing <variable></variable> tags in app_dial documentation. ........ Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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