22737 Commits (b718f91873e39e75aee91bd73d95ffb62ed38f1e)

 

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Author	SHA1	Message	Date
Igor Goncharovskiy	b718f91873	We have faced situation when using CDR and CEL by sqlite3 modules. With system having high load (~100 concurrent calls created by sipp) we found many cdr and cel records missed. There is special finction in sqlite3, that make able to fix this situation - sqlite3_wait_timeout, that also can replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be used for aastdb and res_config_sqlite3 to avoid missed writes to sqlite db. ...	11 years ago
Matthew Jordan	6b782f84b8	MoH: Undo commit r416150 (1.8) ... This patch reverts r416150. When the comparison between mohclass->name and state->class->name is made, you are not guaranteed that (a) state->class is non-NULL or that state or state->class are in a safe state. Crashes caught by the bridges/transfer_capabilities test. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4707dd1ce5	MoH: Don't restart stream on repeated start calls ... Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. Review: https://reviewboard.asterisk.org/r/3615/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416150 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	1509ae5779	AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. ... ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	989b441f6c	main/pbx - documentation - enhance 'core show hints' and 'core show hint' help text ... Adds descriptive help text to 'core show hints' and 'core show hint'. The text describes the various columns for the sake of clarity. ASTERISK-23764 Review: https://reviewboard.asterisk.org/r/3610/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	883d2e6924	chan_sip: DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero ... Change debug level for messages in sdp_crypto.c from zero to one. This ensures the messages are not displayed when debugging is disabled. Change does not apply to 12+ as it was already fixed in those versions. ASTERISK-23246 #close Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3605/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7550fe7efb	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	855fceccac	app_queue: delayed state can cause early leavewhenempty ringing ... In app_queue, device state changes arrive in event messages and update the queue member status value. That value is checked in get_member_status() to decide that the caller should leave when there are no available members. Although event messages can be delayed by other activity, there is no adverse affect by lagged status except in one specific case: there is only one available member, it was just rung, and leavewhenempty is enabled set for ringing members. This change adds a direct check of the device state only under this condition where the caller may be dropped incorrectly, resolving this issue without affecting performance of app_queue normally. AST-1248 #close Review: https://reviewboard.asterisk.org/r/3595/ Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	9972136127	chan_ooh323: fix loading module failure if there no accessible h323_log or ooh323 config file ... change return 1 to return AST_MODULE_LOAD_FAILURE ASTERISK-23814 #close (closes issue ASTERISK-23814) Reported by: Igor Goncharovsky Patches: ASTERISK-23814-ast18.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	ef59e04d6c	safe_asterisk: Cleanup additions to r415132. ... Replaced a stray echo that should've been a message call in safe_asterisk. I'm using the contents of the old message inside the if $NOTIFY so peoples log parsing scripts won't get confused by new messages. I'll clean that up in trunk. (Note that a 'make install' still won't overwrite your old safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	1dff17f051	autoservice: stop thread on graceful shutdown ... This change adds thread shutdown to autoservice for graceful shutdowns only. ast_register_cleanup is backported to 1.8 to allow this. The logger callid is also released on shutdown in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3594/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	560380a018	chan_sip: Fix order of variables specified in SIPNotify action ... Prior to this patch, sequential variables would be ordered in reverse from the order specified in the manager action. Review: https://reviewboard.asterisk.org/r/3588/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ae71ca1d2f	config: Fix config files not reloading when only an included file changes. ... The twisted logic determining if a config file should be reloaded was mostly broken and disabled. The incorrect test that ASTERISK-23383 fixed actually reenabled the broken logic. The incorrect test was causing the timestamp to always be cleared which caused config files with includes to always be reloaded. * Made wildcard includes always cause a reload. Determining if a file was deleted cannot be determined without restructuring the cache to determine if any files are missing from the last files actually loaded. Also without refactoring config_text_file_load(), the glob loop couldn't check more than one file for changes anyway. * Made remove the cache entry if the file no longer exists when trying to get its timestamp or it is no longer a regular file. This fixes the corner case where the file was loaded, then deleted, then the config reloaded, then the file restored with the same timestamp, and then the config reloaded again. * Made remove the cache entry include list when actually loading the file. This gets rid of any stale includes the file had from the last time the file was loaded. ASTERISK-23683 #close Reported by: tootai Review: https://reviewboard.asterisk.org/r/3575/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	14cfff4a02	safe_asterisk: Cleanup and debian compatibility. ... Cleans up the safe_asterisk script and adds the ASTSAFE_FOREGROUND option that allows the debian asterisk init script to capture the right pid. * Drop the vim #modeline which wasn't used. Use test consistently without the odd configure xno syntax. Double quote all paths. General cleanup. * Don't output message()s to the console but only to TTY if set. * Allow TTY to be "no" as well as empty (debian compatibility with debian/patches/safe_asterisk-config). * Add option to export ASTSAFE_FOREGROUND=1 from the init script that calls this to disable backgrounding. Debian uses a similar method in debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review: https://reviewboard.asterisk.org/r/3574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	45c62c1733	app_confbridge: Correct verification of conference name length ... Conference names were not checked for maximum length, allowing unexpected behaviour. This change adds checking to ensure the maximum length is not exceeded. The maximum length is also changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches: confbridge-enforce_max-1.8.patch uploaded by coreyfarrell (license 5909) confbridge-enforce_max-11up.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	6175902998	func_odbc: Fix fixed size buffers fix (r414968). ... The change that removed the fixed size buffers in odbc-related code -- removing arbitrary column width limits -- was incomplete. This change adds: no segfault on writesql without insertsql and return value checks after strdup. While I was in the vicinity I cleaned up the linefeeds in the odbc function descriptions, moved some code for clarity, removed some blobs and noted (but didn't fix) that the 'odbc write ... exec' CLI command doesn't behave as the dialplan equivalent when insertsql= is used. ASTERISK-23582 #close Review: https://reviewboard.asterisk.org/r/3579/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c1be1f43cb	main/config.c: AMI action UpdateConfig EmptyCat clears all categories ... When invoking UpdateConfig AMI action with Action set to EmptyCat, Asterisk will make all categories empty in the config but the one requested with a Cat variable. This is due to a bug in ast_category_empty (main/config.c) that makes an incorrect comparison for a category name. This patch corrects the comparison such that only the requested category is cleared. Review: https://reviewboard.asterisk.org/r/3573/ ASTERISK-23803 #close Reported by: zvision patches: manager.c.diff uploaded by zvision (License 5755) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	27d80767db	PBX: Prevent incorrect hint parsing ... Dynamic and pattern matching hints should not be checked for their last known state until they are instantiated by subscribers. (closes issue AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted by Matt Jordan (license 6283) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	8a52013dca	res_config_odbc: Use dynamically sized buffers to store row data so values do not get truncated. ... ASTERISK-23582 #close ASTERISk-23582 #comment Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/3557/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	bf94ff5763	chan_sip: Start session timer at 200, not at INVITE. ... Asterisk started counting the session timer at INVITE while the other end correctly started at 200. This meant that for short session-expiries (90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk would wrongly assume that the timer was hit before the other end thought it was time to send a session refresh. This resulted in prematurely ended calls. This changes the session timer to start counting first at 200 like RFC says it should. (Also removed a few excess NULL checks that would never hit, because if they did, asterisk would have crashed already.) ASTERISK-22551 #close Reported by: i2045 Review: https://reviewboard.asterisk.org/r/3562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	81b4fbb578	res_config_odbc: Fix old and new ast_string_field memory leaks. ... The ODBC realtime driver uses ^NN parameter encoding to cope with the special meaning of the semi-colon. A semi-colon in a field is interpreted as if the key was supplied twice, something which isn't otherwise possible with fixed database columns. E.g. allow=alaw;ulaw is parsed as allow=alaw and allow=ulaw. A literal semi-colon is rewritten to ^3B when stored in the database. The module uses a stringfield to efficiently store the encoded parameters. However, this stringfield wasn't always freed in some off-nominal cases. Commit r413241 fixed initialization so the encoding for INSERT and DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked apparently.) But that commit forgot the frees. This change cleans that up. Review: https://reviewboard.asterisk.org/r/3555/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	71df5a4a63	Backport Asterisk 11 r413876 to 1.8 ... ........ r413876 | jrose | 2014-05-13 12:40:00 -0500 (Tue, 13 May 2014) | 6 lines chan_sip: Add TLS and SRTP status to CLI command 'sip show channel' ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2beabaabeb	app_meetme: Don't interrupt MOH for waitmarked users. ... Occasionally, when the last marked user leaves the conference, waitmarked users don't get MOH if MOH is supposed to be played while a waitmarked user is waiting for another marked user. * Made not interrupt MOH when the user is a waitmarked user. The waitmarked user doesn't need to hear any leave announcements from the conference as the user would have already heard different leave announcements if they were enabled. Apparently DAHDI occasionally sends unending non-silent streams to these users or a normal user still in the conference has continuous high background noise. These non-silent streams cause MOH to be suspended while the never ending "announcement" is played. Issue caused by ASTERISK-13680. AST-1349 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3543/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4f78ecf9b5	UPGRADE: Add note for REF_DEBUG flag ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	46aff9f0f8	chan_local: Only block media frames when a generator is on both ends of a local channel. ... The fix for ASTERISK-12292 was a bit too aggressive. You could have generators pointed at each other on local channels but need to get other kinds of frames such as DTMF or CONNECTED_LINE frames accross. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	1035ef957c	pbx.c: prevent potential crash from recursive replace() ... Recurisve usage of replace() resulted in corruption of the temporary string storage and potential crash. By changing the string to be allocated separtely per instance, this is eliminated. ASTERISK-23650 #comment Reported by: Roel van Meer ASTERISK-23650 #close Review: https://reviewboard.asterisk.org/r/3539/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	7fa61bee7d	chan_ooh323: fix h323_log full path name ... * fix to use astlogdir option for h323_log file instead of hardcoded ASTERISK-23754 #close Reported by: Igor Goncharovsky Patches: ooh323_logger_patch.diff git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ffe97a9c8f	chan_dahdi: Fix analog dialtone detection. ... * Check if waitingfordt (waitfordialtone) is enabled in dahdi_read() to allow the DSP to operate early enough to detect dialtone. * Made use the correct variable in my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve Davies Patches: dialtone_detect_fix (license #5012) patch uploaded by Steve Davies Review: https://reviewboard.asterisk.org/r/3534/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e97c2eccc0	app_meetme: Fix overwrite of DAHDI conference data structure. ... Starting a conference recording using the admin menu overwrites the DAHDI conference data structure used to modify the admin user's conference mute mode. * Made no longer pass the user's DAHDI conference data structure into the menu functions. The menu now uses its own DAHDI conference data structure to start the recording channel. * Moved the unlock conf->playlock to before playing the conf-full message. No sense keeping the lock while that prompt is playing. The user is never going to get into the conference at that point. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	8bf18ed4d3	chan_local+app_dial: Propagagate call answered elsewhere over local channels. ... AST_FLAG_ANSWERED_ELSEWHERE was not propagated back from local channels. It is now. That means that when a call is picked up from a callgroup of local channels, the other channels will now properly see it as "picked up". This occurs when you use a construct like Dial(Local/a@context&Local/b@context) where a@context and b@context dial two chan_sip devices respectively. If one device picks up, the other will not see "1 missed call" anymore. In this respect, it now behaves the same as when doing Dial(SIP/a&SIP/b). Review: https://reviewboard.asterisk.org/r/3540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	76a57fcbc4	res_musiconhold: Minor cleanup. ... Fix a few free()'s that should be ast_free()'s. Reverted an old workaround that isn't necessary. Reorder a tiny bit of code. Remove a bit of commented-out code. Review: https://reviewboard.asterisk.org/r/3536/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	adb0ee2dcd	chan_sip+CEL: Add missing ANSWER and PICKUP events to INVITE/w/replaces pickup. ... When doing a "BLF-style call pickup" -- an INVITE with Replaces: header -- the CEL log would lack the ANSWER and PICKUP events. This patch adds the two missing events to the handle_invite_replaces() function. ASTERISK-22977 #close Review: https://reviewboard.asterisk.org/r/3073/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	228d5c9052	rtp: Fix case typo in H263+ mime. ... http://tools.ietf.org/html/rfc3555#section-4.2.6 says the canonical mime subtype is "H263-1998", not "h263-1998". Original code was added in r183101 on 2009-03-19 02:26:50 +0100. This fixes issues with Polycom phones. ASTERISK-23665 #close ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume Maudoux, backported by me. Review: https://reviewboard.asterisk.org/r/3529/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	3dace58a49	chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled. ... When overlap dialing is enabled, the lack of inband audio available information in the SETUP_ACKNOWLEDGE events causes an interoperability problem with SIP. sig_pri doesn't know if there is dialtone present when a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183 Session Progress and blocks the desired 180 Ringing message when the ALERTING message comes in. * Made the configure script detect if the installed version of libpri supports the SETUP_ACKNOWLEDGE enhancements. * Using the new API, made generate an AST_CONTROL_PROGRESS frame on an incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio is present instead of assuming that dialtone is present. * Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio available indication only if dialtone is expected. The change also makes the fallback behaviour of sending the PROGRESS message better by sending it only if dialtone is expected. * Changed receiving a PROCEEDING message to not generate an AST_CONTROL_PROGRESS frame if the progress indication ie indicates non-end-to-end-ISDN. This helps interoperability with SIP. * Changed sending a PROCEEDING message in response to an AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It was silly to do so anyway because the channel driver doesn't know if inband audio is even available. This helps interoperability with SIP. This patch and a corresponding change in libpri work together to allow Asterisk to control the inband audio available progress indication ie on the SETUP_ACKNOWLEDGE message when dialtone is present. AST-1338 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b2f263b374	Fix 32bit build for func_env ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	650b948d21	Fix 32bit build for chan_sip ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	3e9a54d857	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	cb8877e720	app_queue: Extend documentation for various Manager actions and events. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	afecc456ad	Fix encoding of custom prepare extra data. ... Patches: res_config_odbc-take2.patch by John Hardin (License #6512) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	b7b11ef2f0	Ensure that all parts of SQL UPDATEs and DELETEs are encoded. ... Patches: res_config_odbc.patch by John Hardin (License #6512) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1a60296a80	Prevent crashes in res_config_odbc due to uninitialized string fields. ... Patches: odbc-crash.patch by John Hardin (License #6512) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	c299732b72	Return the number of rows affected by a SQL insert, rather than an object ID. ... The realtime API specifies that the store callback is supposed to return the number of rows affected. res_config_pgsql was instead returning an Oid cast as an int, which during any nominal execution would be cast to 0. Returning 0 when more than 0 rows were inserted causes problems to the function's callers. To give an idea of how strange code can be, this is the necessary code change to fix a device state issue reported against chan_pjsip in Asterisk 12+. The issue was that the registrar would attempt to insert contacts into the database. Because of the 0 return from res_config_pgsql, the registrar would think that the contact was not successfully inserted, even though it actually was. As such, even though the contact was query-able and it was possible to call the endpoint, Asterisk would "think" the endpoint was unregistered, meaning it would report the device state as UNAVAILABLE instead of NOT_INUSE. The necessary fix applies to all versions of Asterisk, so even though the bug reported only applies to Asterisk 12+, the code correction is being inserted into 1.8+. Closes issue ASTERISK-23707 Reported by Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	8ccad4ce21	http: Fix spurious ERROR message in responses with no content. ... Backport -r411687 and fix the fix because content_length is the length of out plus the length of the file controlled by fd. When a response has an out content length of 0, fwrite would be called to write a buffer with no data in it. This resulted in the following classic error message: [Apr 3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success This patch makes it so that we only attempt to write the content of out if the out string is non-zero. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	500ce9b5f4	chan_sip: trust_id_outbound CHANGES message improvement ... (closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	4100002e21	Typo in CHANGES ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	24f054f497	HTTP: Add TCP_NODELAY to accepted connections ... This adds the TCP_NODELAY option to accepted connections on the HTTP server built into Asterisk. This option disables the Nagle algorithm which controls queueing of outbound data and in some cases can cause delays on receipt of response by the client due to how the Nagle algorithm interacts with TCP delayed ACK. This option is already set on all non-HTTP AMI connections and this change would cover standard HTTP requests, manager HTTP connections, and ARI HTTP requests and websockets in Asterisk 12+ along with any future use of the HTTP server. Review: https://reviewboard.asterisk.org/r/3466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	5d924bc0fc	chan_sip: Add sendrpid trust options ... In r411189, some behavior was changed which made sendrpid behavior act in a more trusting manner by sending full user data for peers set with private caller presence in P-Asserted-Identity headers. Since this changed long time expected behaviors, we decided to pull that patch when that was pointed out by the community. Instead, this patch provides a trust_id_outbound setting which will expose the data per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers at all if set to 'no'. By default trust_id_outbound will be set to 'legacy' which will preserve the behavior prior to these patches. Extra special thanks to Walter Doekes for providing advice and feedback. (closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski Review: https://reviewboard.asterisk.org/r/3447/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	e1eaa7dcd3	app_sms: Fix uninitialized values; hangup channel when REL is sent successfully ... This patch fixes two issues in app_sms: (1) Firstly, the 'flags' field on the stack in sms_exec() is uninitialised, causing it to use the wrong protocol in some cases. This patch correctly initializes the flags fields. (2) Secondly, when disconnect supervision is not working or inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was failing to terminate the call after it sent the REL(ease) message and the peer stopped talking to it. This patch fixes the code to handle the 'bad stop bit' message more gracefully in that case, and hang up the call. Review: https://reviewboard.asterisk.org/r/1392/ ASTERISK-18331 #close Reported by: David Woodhouse patches: asterisk-fix-sms.patch uploaded by David Woodhouse (License 5754) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	4a68620cfb	sounds: Fix Sounds Makefile and XML that didn't support new sound prompt sets ... In sounds/Makefile 1 Adds and moves some lines necessary for the en_GB core set. I'm just following how the other sets are defined here. 2 removes the ES extra sounds related lines as we don't have ES extra sound sets. In sounds/sounds.xml 3 Adds <support_level> definitions to all the sound sets as we have these defined in 11,12,Trunk, but not in 1.8 4 Adds member definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in extra sound sets ASTERISK-23550 Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3464/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	03ed8d30ba	channels/chan_oss: Fix compilation problem on SmartOS/Illumos/SunOS ... THis patch fixes an issue in chan_oss when building on certain platforms. It ensures that soundcard.h is found. Review: https://reviewboard.asterisk.org/r/3426 Note that this patch is a part of the patch on ASTERISK-23576; the Makefile portion only applies to Asterisk 11+. (issue ASTERISK-23576) Reported by: Sebastian Wiedenroth patches: fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago