21150 Commits (0219829eef97b7faadd9b73b20a5278ac6ce9dd1)

 

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Author	SHA1	Message	Date
Terry Wilson	0219829eef	Unbreak the storing of registrations for restart ... The fix for issue 18882 broke retrieving non-realtime peers from the ast_db on restart/reload. This patch tries to unbreak things while leaving the intent of the original fix intact. (closes issue #19318) Reported by: remiq Patches: diff.txt uploaded by twilson (license 396) Tested by: lmadsen, remiq git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	249f4b9022	Merged revisions 319528 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	789411102a	Merged revision 319468 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on dialed channels. The use of mISDN HDLC mode is prevented if the mISDN dial technology option 'h1' is used when config option astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC mode. Instead of setting the channel to HDLC mode it is set to transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the logging message is correct, but the if condition is not. Make check the nodsp and hdlc flags. JIRA ABE-2787 JIRA SWP-3437 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	c23377d8f2	Don't create [general] voicemail context when using users.conf ... Prior to this patch, app_voicemail would create a [general] context when parsing users.conf. (closes issue #18891) Reported by: pdugas Patches: app_voicemail-ignore-general.patch uploaded by pdugas (license 1222) app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71) Tested by: pdugas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	e93fa0bba4	Make Debian init script lsb compliant ... (closes issue #18896) Reported by: manwe Patches: debian_init_lsb.patch uploaded by manwe (license 1223) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	81ee872a32	Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	933cf293cd	Deadlock between generic CCSS agent and native ISDN CCSS. ... Deadlock can occur when the generic CCSS agent is deleting duplicate CC offers and the native ISDN CC driver is processing an incoming CC message. The cc_core_instances container lock cannot be held when an agent or monitor callback is invoked without the possibility of a deadlock. * Make kill_duplicate_offers() remove the reference in cc_core_instances outside of the container lock. JIRA AST-566 JIRA SWP-3469 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	ac0cc37ab5	Merged revisions 319202 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) | 4 lines Unlink a peer from peers_by_ip when expiring a registration Review: https://reviewboard.asterisk.org/r/1218/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	215638e661	Merged revisions 319144 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) | 2 lines Fixes issue with peer ref-counting during handle_request_subscribe. (closes issue #19293) Reported by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	6f625f139a	Make sure tcptls_session exists before dereferencing it. ... (closes issue #19192) Reported by: stknob Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723) Tested by: vois, Chainsaw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	e8935ca8e4	Support gmime-2.4 ... (closes issue #18863) Reported by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir (license 46) Tested by: tzafrir Review: https://reviewboard.asterisk.org/r/1213/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d75f80fe08	Fixes Big Endian build issue. ... (closes issue #19298) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Brett Bryant	7c38148a7d	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/branches/1.8@318921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Brett Bryant	7f7f233e8f	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/branches/1.8@318919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Brett Bryant	6ddb2e9ee0	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/branches/1.8@318917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7a0766e4ad	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/branches/1.8@318868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	209d8d3c15	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/branches/1.8@318783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1b1961f73f	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/branches/1.8@318720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	87d80af96c	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/branches/1.8@318671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	84b9092e03	Comment out the REF_DEBUG that slipped in during debugging ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5badb39856	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/branches/1.8@318549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0ec0f72506	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/branches/1.8@318499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	acb1bb3026	Filter out blacklisted manager events when using eventfilter. ... Merging change from trunk in revision 306432. (closes issue #19260) Reported by: dhubbard Tested by: dhubbard git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	5578557df1	chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	eab9b5992d	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/branches/1.8@318351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f96cf88212	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/branches/1.8@318337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	607164ad91	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/branches/1.8@318282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	a0d4192e2d	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/branches/1.8@318233 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	90c544c0e1	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/branches/1.8@318231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	2f5ff86317	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/branches/1.8@318148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	9bfd90d329	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/branches/1.8@318142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	6313aeb1fd	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/branches/1.8@318057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	1c168cd613	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/branches/1.8@318055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	10e40660c7	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/branches/1.8@317969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	91c8e4d297	Fix some more "set but unused" compiler warnings. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	46a4825fcf	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/branches/1.8@317918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	df0cc7f905	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/branches/1.8@317917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	7c6e763258	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/branches/1.8@317867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d5a0fb899a	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/branches/1.8@317865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	25ccc62ee9	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/branches/1.8@317861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	8ed15a49a1	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	5caa350fab	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/branches/1.8@317837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	65f01fd713	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/branches/1.8@317805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	900e3e5d03	Fix SIP connected line updates. ... This patch fixes a couple SIP connected line update problems: 1) The connected line needs to be updated when the initial INVITE is sent if there is a peer callerid configured. Previously, the connected line information did not get reported until the call was connected so SIP could not report connected line information in ringing or progress messages. 2) The connected line should not be updated on initial connect if there is no connected line information. Previously, all it did was wipe out any default preset CONNECTEDLINE information set by the dialplan with empty strings. (closes issue #18367) Reported by: GeorgeKonopacki Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6cf3280dd6	Merged revisions 317575 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	0388304195	If the configure script runs, force a rebuild of menuselect-tree. ... Some contents in the menuselect tree are dependent on configure script parameters, namely --enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis Patches: issue_17219.rev1.txt uploaded by russell (license 2) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	a5e0730e6a	Fix some more realtime MySQL schema issues. ... (closes issue #18537) Reported by: denzs Patches: sipfriends.sql.svndiff uploaded by denzs (license 1182) queue_log.sql.svndiff uploaded by denzs (license 1182) meetme.sql.svndiff uploaded by denzs (license 1182) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	809c22fb36	Fix some errors in sample MySQL realtime schema files. ... (closes issue #18915) Reported by: Dovid Patches: sipfriends.patch uploaded by Dovid (license 652) meetme.patch uploaded by Dovid (license 652) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	448ceb5291	Don't lose cdr_syslog config on a reload. ... (closes issue #18679) Reported by: enegaard Patches: issue18679_seanbright.patch uploaded by seanbright (license 71) Tested by: enegaard git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	6446ba738b	Fix some consistency issues with jitterbuffer config. ... Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago