asterisk

Commit Graph

Author	SHA1	Message	Date
Brett Bryant	475ef22b20	Merged revisions 318921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318921 \| bbryant \| 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) \| 8 lines Fixes a segmentation fault in dynamic hints when a channel technology isn't loaded for a hint. (closes issue #18495) Reported by: bertrand Tested by: bertrand ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	085b7b212a	Merged revisions 318919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318919 \| bbryant \| 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) \| 10 lines This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too much time has passed between sending audio. (closes issue #18206) Reported by: bernhardsi Patches: res_srtp_unhold.patch uploaded by bernhards (license 1138) Tested by: bernhards, notthematrix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	547490144c	Merged revisions 318917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318917 \| bbryant \| 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) \| 11 lines This patch allows TCP peers into the ast_db where they were previously restricted. (closes issue #18882) Reported by: cmaj Patches: patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt uploaded by cmaj (license 830) Tested by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	db89abf0bd	Merged revisions 318868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318868 \| rmudgett \| 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) \| 19 lines CDR's are being written immediately on caller hangup. CDR's are being written immediately on caller hangup. The dialplan is not able to modify it in the h exten. The h exten in the initial context is not run before closing CDR's when the bridge is unlinked if a macro is active and does not have an h exten. * Make ast_bridge_call() check for an h exten in the current context and if a macro is active then the initial context. The first h exten found is then run before closing the CDR. (closes issue #18212) Reported by: leearcher Patches: issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	54bb8a0ca8	Move exten used for dialing from device to subchannel. There were some issues where if a simple switch was cancelled and a new switch started before the first had timed out where the d->exten would be used for both subchannels. This was bad leading to possible invalid extensions if some digits had been entered in the abandoned simple switch and the second one was completed before the first timed out, or the second would be cancelled because d->exten would be set to nothing on the time out of the first. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	9066db4329	Merged revisions 318720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318720 \| mnicholson \| 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) \| 4 lines Handle ipv6 addresses in the sent-by Via: field. This change fixes a regression in via header parsing and ipv6 handling. (closes issue #18951) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1ad49f46ce	Merged revisions 318783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318783 \| rmudgett \| 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) \| 14 lines PRI early media won't ring. And another way to pass early media. Don't indicate that there is inband information present, just assume that the B channel is connected. * Restore clearing the dialing flag Rx squelch unconditionally when a PROCEEDING message comes in. (closes issue #19268) Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett (license 664) Tested by: tbsky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	892b7a2efd	Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318671 \| alecdavis \| 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) \| 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	c37c017781	Consolidate setsubstate_* into setsubstate and use a switch. Consolidate the functions and add some debugging info. Allows to be able to set a substate without explicitly knowing what the state is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	bdbb3a506f	Add setsubstate_onhook. Add the setsubstate_onhook to complete the initial substate handling procedures. Added dumpsub(sub, forcehangup) which is the common way of calling setsubstate_onhook. Dumpsub attempts to activate another sub after setting the current one onhook. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	475c264bd2	Merged revisions 318550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318550 \| twilson \| 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) \| 2 lines Comment out the REF_DEBUG that slipped in during debugging ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	da4016544e	Merged revisions 318549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318549 \| twilson \| 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) \| 27 lines Merged revisions 318548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318548 \| twilson \| 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) \| 19 lines Clean up several chan_sip reference leaks Several situations in the code could lead to peers or sip_pvt references being leaked. This would cause RTP ports to never be destroyed (leading to exhaustion of all available RTP ports) and memory leaks. The original patch for this issue from rgagnon was the result of an obscene amount of testing and hard work, for which I am very grateful. I did some cleanup and added a few additional refcount fixes that I found. (closes issue #17255) Reported by: kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ Review: https://reviewboard.asterisk.org/r/1207/ Review: https://reviewboard.asterisk.org/r/1210/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d1e27b1026	Merged revisions 318499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318499 \| rmudgett \| 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) \| 15 lines Unable to pickup DAHDI/PRI call because call state is reported as DIALING. The channel state is not updated to RINGING when an ALERTING message is received. Regression caused when sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added missing channel state update to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes issue #19257) Reported by: alecdavis Patches: issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	0ccfc8609a	Merged revisions 318436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318436 \| russell \| 2011-05-10 10:13:16 -0500 (Tue, 10 May 2011) \| 2 lines chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	07b3742ad2	Merged revisions 318337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318337 \| twilson \| 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) \| 18 lines Merged revisions 318331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318331 \| twilson \| 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) \| 12 lines Don't offer video to directmedia callee unless caller offered it as well Make sure that when directmedia is enabled, that video is not offered to the callee even if it supports it. p->vrtp will not exist since the caller didn't offer video. (closes issue #19195) Reported by: one47 Patches: sip_cant_add_video_rtp uploaded by one47 (license 23) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	0886204011	Merged revisions 318351 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318351 \| rmudgett \| 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) \| 6 lines Remove references to res_features and its export file. The contents of res/res_features.c was moved to into main/features.c awhile ago. There is no longer any need for the res/Makefile to reference res_features or the res_features linker exports file to exist. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bf57bb3c89	Merged revisions 318282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318282 \| rmudgett \| 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) \| 24 lines Hangup extension executed twice. When a user hangs up a call, in certain circumstances, the hangup extension can end up being executed twice: 1) If a call is bridged and the 'h' extension executes the Hangup application, then the 'h' extension will be executed twice. 2) If a call is bridged within a macro (Dial or Queue), it has its own 'h' extension, the main context also has an 'h' extension, and the macro 'h' extension executes the Hangup application, then both 'h' extensions will be executed. * Revert originally commited fix for #16106 and just set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call(). The bridge code just executed an 'h' extension so the main PBX loop does not need to execute one as well. (issue #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4c35291c6b	Merged revisions 318233 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318233 \| dvossel \| 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) \| 14 lines Merged revisions 318230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318230 \| dvossel \| 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) \| 7 lines Fixes cases where sip_set_rtp_peer can return too early during media path reset. (closes issue #19225) Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d7c94e1e04	Merged revisions 318231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318231 \| rmudgett \| 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) \| 41 lines Don't get early media for ISDN on outgoing calls. It looks to be a long-standing misinterpretation of the progress indicator ie values: 1 - Call is not end-to-end ISDN; further call progress information may be available in-band. 8 - In-band information or an appropriate pattern is now available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled as early media probably because the meaning of the second half of it's description was overlooked. * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. (closes issue #18868) Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: satish_lx .......... No inband progress on PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider sends an ALERTING with "Inband information or appropriate pattern now available", but Asterisk only generates and passes the RING to the SIP extension, not the inband message. Unfortunately, the inband message is not a ringback tone but a prompt that says the number is not in service. The SIP extension then hears two rings and the call is hungup which confuses the caller. * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband audio is indicated with an ALERTING message. (closes issue #19246) Reported by: cristiandimache Patches: issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: cristiandimache ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	f2df0ed9f1	Increase prepend filename length. (closes issue #19238) Reported by: byronclark Patches: increase_prepend_filename_length.patch uploaded by byronclark (license 1200) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	ff4c7d46c0	Minor change to 318141 to improve parsing behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	6eb9d7e1b5	Merged revisions 318148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318148 \| jrose \| 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) \| 4 lines Documenting an observed behavior of features in features.conf. Since parkinglots use an integer for the parkinglot extensions, leading zeros specified in the configuration file are ignored. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	5b77bb5060	Merged revisions 318142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318142 \| mnicholson \| 2011-05-09 09:09:38 -0500 (Mon, 09 May 2011) \| 9 lines Make indicate/control frames WRITE events on framehooks. Also, if a framehook returns a non-control frame, don't forward it to the channel. (closes issue #19251) Reported by: irroot Patches: (modified) framehook_indicate.patch2 uploaded by irroot (license 52) Tested by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	229e066dcb	Allows ParkedCall application to specify a parkinglot. When invoking the app parkedcall, the argument can now include '@parkinglot' after the extension. (closes issue #18777) Reported by: cartama Patches: 0018777.diff uploaded by cartama (license 1157) Review: https://reviewboard.asterisk.org/r/1209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	7002adcb3e	Add setsubstate_callwait. If a call is made to a line that already has a call and the device is offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	7cccaf93b2	Merged revisions 318057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318057 \| russell \| 2011-05-07 18:35:37 -0500 (Sat, 07 May 2011) \| 8 lines res_config_curl: fix a crash with static realtime. (closes issue #18413) Reported by: jmls Patches: 20101202__issue18413.diff.txt uploaded by tilghman (license 14) Tested by: jmls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3736b02d97	Merged revisions 318055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318055 \| russell \| 2011-05-07 18:24:18 -0500 (Sat, 07 May 2011) \| 7 lines chan_iax2: Don't overwrite port found with an SRV lookup. (closes issue #17291) Reported by: jcovert Patches: chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	8c0b1115cd	Only allow voicemail if substate is OFFHOOK or no channel active (UNSET). (closes issue #17901) Reported by: salecha git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	a9beb8323e	Rename sub->parent to sub->line. Improve readability of code, eg, (sub->parent == d->activeline) becomes (sub->line == d->activeline). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	bc61836c1b	Move the hookstate from line to device. Long time coming, finally moving the hookstate from line to device. This may fix some issues where a device has multiple lines. Previously we had to run through all lines on a device to see if it was actually onhook or not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	6df3b851e3	Merged revisions 317969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317969 \| russell \| 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) \| 10 lines Use the right variable to print the time in a debug message. The original patch also increased some buffer sizes, but that was already done in this version. (closes issue #17034) Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded by sysreq (license 1009) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	d05e5281da	Merged revisions 317967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317967 \| russell \| 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) \| 2 lines Fix some more "set but unused" compiler warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d2f16ce587	Merged revisions 317918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317918 \| dvossel \| 2011-05-06 16:06:55 -0500 (Fri, 06 May 2011) \| 7 lines Fixes missing colon from To/From headers in RTCP manager events. (closes issue #18221) Reported by: clegall_proformatique Patches: 18221_1.patch uploaded by ebroad (license 878) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c73ea18012	Merged revisions 317917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317917 \| russell \| 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) \| 7 lines Fix calculation of free RAM to make minmemfree option work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff uploaded by loic (license 1020) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	04b653358e	Add a cdr_csv to MySQL import script to contrib/scripts. (closes issue #17036) Reported by: precisenetworks Patches: import-cdr-csv-mysql.pl uploaded by precisenetworks (license 1010) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	4fc020c965	Add the Uniqueid header to Userevent. (closes issue #16962) Reported by: jlpedrosa Patches: patch.diff uploaded by jlpedrosa (license 1002) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	33b7cc2ef6	Merged revisions 317867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317867 \| russell \| 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) \| 10 lines chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer. Don't duplicate variables on the sip_pvt. Just reset the variable list each time. (closes issue #19202) Reported by: wdoekes Patches: issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	ae8dbde4a8	Merged revisions 317865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317865 \| russell \| 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) \| 11 lines chan_sip: fix a deadlock in check_rtp_timeout. Don't block doing silly deadlock avoidance. Just return and try again later. The funciton gets called often enough that it's fine. Also, this change was already made in trunk. (closes issue #18791) Reported by: irroot Patches: chan_sip.rtptimeout.patch uploaded by irroot (license 52) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	426aa5e09b	Blocked revisions 317861 via svnmerge ........ r317861 \| russell \| 2011-05-06 14:35:00 -0500 (Fri, 06 May 2011) \| 11 lines URI encode less characters in the RPID and Contact headers. If this change causes any problems, we will need to backport the more extensive uri encoding and decoding handling changes that are in trunk/1.10. (closes issue #18686) Reported by: wolfgang Patches: quick-and-dirty.patch uploaded by wdoekes (license 717) Tested by: wdoekes, devellow, wolfgang, mav3rick ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	94b3848878	Blocked revisions 317858 via svnmerge ........ r317858 \| mnicholson \| 2011-05-06 14:31:50 -0500 (Fri, 06 May 2011) \| 6 lines pbx_lua autoservice fixes Don't start an autoservice in pbx_lua if pbx_lua already started one and don't stop one if we didn't start one. Also start and stop the autoservice when transferring control from and to the pbx. ........ This change is already implemented in trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	a5e6e75b02	Merged revisions 317837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317837 \| russell \| 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) \| 11 lines Fix a crash in the MySQL() application. This code was not handling channel datastores safely. The channel must be locked. (closes issue #17964) Reported by: wuwu Patches: issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71) Tested by: wuwu ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	669f49b384	Updated CHANGES to note the autoservice changes for pbx_lua git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	07ba8b1474	Updated the sample pbx_lua config file to reflect autoservice changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	001d6c5c00	Merged revisions 317805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317805 \| russell \| 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) \| 7 lines Add a new sipfriends.sql for MySQL that has more fields in it. (closes issue #16399) Reported by: pabelanger Patches: sipfriends.sql.v3 uploaded by pabelanger (license 224) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	7a1204d129	Default to starting an autoservice in pbx_lua. The autoservice is automatically stopped when applications are executed, so this shouldn't cause any problems. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	d5e9ce9ab1	Make pbx_lua handle managing the autoservice better. Make autoservice_start() and autoservice_stop() return nothing. Also check if the autoservice flag is set before starting or stopping the autoservice and stop and start the autoservice when returning control to and getting control from the pbx engine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	6c38322870	Added note about changes in pbx_lua's behavior when applications do dialplan jumps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	6d04d190dc	Use two spaces after periods for the recent pbx_lua change descriptions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	f005c153f8	Updated CHANGES for hints support in pbx_lua git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	bccba53bcf	Detect Goto in pbx_lua. This code will actually detect any dialplan jump from any application that calls ast_explicit_goto(). This change is only being done in trunk as it may change the way some dialplans execute. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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21274 Commits (475ef22b205a5e228e796e7758149b953e949049) All Branches Search

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