asterisk

Commit Graph

Author	SHA1	Message	Date
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
Jonathan Rose	445b1ced10	Reverting r411189 so that it can be put up for public review --- r411189 \| jrose \| 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) \| 12 lines chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior to this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) --- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ee72a7a7d1	app_stack: Add missing unlock in off-nominal path of STACK_PEEK function. ASTERISK-23620 #close Reported by: Bradley Watkins Patches: ASTERISK-23620_unlock_oldlist.patch (license #5021) patch uploaded by Bradley Watkins git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	751dc2b7da	main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output This patch does the following: (1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables REF_DEBUG globally throughout Asterisk. (2) The ref debug log file is now created in the AST_LOG_DIR directory. Every run will now blow away the previous run (as large ref files sometimes caused issues). We now also no longer open/close the file on each write, instead relying on fflush to make sure data gets written to the file (in case the ao2 call being performed is about to cause a crash) (3) It goes with a comma delineated format for the ref debug file. This makes parsing much easier. This also now includes the thread ID of the thread that caused ref change. (4) A new python script instead for refcounting has been added in the contrib/scripts folder. Review: https://reviewboard.asterisk.org/r/3377/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	85a69184c7	Internal timing: Add notice that the -I and internal_timing option are no longer needed. Add notice messages during execution that the -I command line option and the astersik.conf internal_timing option are no longer needed. The internal timing functionality is now always enabled if there is a timing module loaded. NOTE: Since the command line options and the asterisk.conf config file are processed before the logging system is initialized, the messages are output to stderr. Change requested as a result of asterisk-dev list comments about the commit for ASTERISK-22846 that removed the -I and internal_timing options. Review: https://reviewboard.asterisk.org/r/3423/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	70cdee628c	config: Fix CB_ADD_LEN() to work as originally intended. Fix a long standing bug in CB_ADD_LEN() behaving like CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	7dc87fa2b9	configs: Clean up long line and typo in res_odbc.conf.sample. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a997ddc829	internal_timing: Remove the option and always make it enabled if a timing module is loaded. The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	6f85424652	app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected. This patch causes realtime queue members to remain in queues during the reload process. Previously these members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY". ASTERISK-23547 #close ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409) Review: https://reviewboard.asterisk.org/r/3404/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	dd5c0ffd4d	http: response body often missing after specific request This patch works around a problem with the HTTP body being dropped from the response to a specific client and under specific circumstances: a) Client request comes from node.js user agent "Shred" via use of swagger-client library. b) Asterisk and Client are not on the same host or TCP/IP stack In testing this problem, it has been determined that the write of the HTTP body is lost, even if the data is written using low level write function. The only solution found is to instruct the TCP stack with the shutdown function to flush the last write and finish the transmission. See review for more details. ASTERISK-23548 #close (closes issue ASTERISK-23548) Reported by: Sam Galarneau Review: https://reviewboard.asterisk.org/r/3402/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9e5d570c46	UPGRADE: Note IAX2 compatibility issue between 1.4 and 1.8+ systems. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f222e30437	res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values. This patch fixes setting nullable integer columns to NULL instead of an empty string, which fails for PostgreSQL, for example. The current code is supposed to do so, but the check is broken. The patch also allows the first column in the list to be a nullable integer. This patch also adds a compatibility setting in res_odbc.conf, allow_empty_string_in_nontext. It is enabled by default. It should be disabled for database backends (such as PostgreSQL) that require NULL instead of an empty string for Integer columns. Review: https://reviewboard.asterisk.org/r/3375 (issue ASTERISK-23459) Reported by: zvision patches: res_config_odbc.diff uploaded by zvision (License 5755) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	588550c950	chan_sip: Add MESSAGE request to allowed methods The allowed methods advertised by chan_sip did not previously note the MESSAGE request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we should advertise that we support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504 #comment Reported by: Martin Kontsek ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587) Review: https://reviewboard.asterisk.org/r/3396/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	824c8d4b6b	Fix dialplan function NULL channel safety issues (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e86e40892c	say: Fix a bug where SayNumber in Polish tries to play incorrect sound. This change fixes a bug where calling SayNumber with a number divisible by 100 using the Polish language would cause the code to attempt to play a sound file with an empty name. (closes issue ASTERISK-23509) Reported by: zvision Review: https://reviewboard.asterisk.org/r/3378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	e2708b03b6	chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7ae2541b15	chan_sip: Fix incorrect use of timers If update_provisional_keepalive() is called while send_provisional_keepalive_full() is waiting on the PVT lock, then pvt->provisional_keepalive_sched_id will be changed to a new sched_id value by update_provisional_keepalive(), but that new sched_id then may be overwritten with -1 by send_provisional_keepalive_full(), killing the pvt's reference to a schedule and "leaking" the reference. (closes issue ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies Patches: provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	9c7ed786e5	chan_sip: Always use fromdomain if set for domain, even if callerid is set to restricted. (closes issue ASTERISK-20841) Reported by: Kelly Goedert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russ Meyerriecks	e62a5f78e9	!fixup: callerid: Logic error in checksum processing Fixes syntax error in previous commit :-( git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russ Meyerriecks	2478dc0606	callerid: Logic error in checksum processing Callerid checksum-ing was being handled incorrectly here. When the checksum is calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This value will then fail the otherwise correct callerid message. This patch changes the logic to simply add the calculated checksum to the transmitted 2's compliment checksum. Review: https://reviewboard.asterisk.org/r/3356/ (closes issue ASTERISK-23488) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	1d8a661b30	AST-2014-001: Stack overflow in HTTP processing of Cookie headers. Sending a HTTP request that is handled by Asterisk with a large number of Cookie headers could overflow the stack. Another vulnerability along similar lines is any HTTP request with a ridiculous number of headers in the request could exhaust system memory. (closes issue ASTERISK-23340) Reported by: Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. Manuel Sadosky, Buenos Aires, Argentina git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b0df15e1bf	AST-2014-002: chan_sip: Exit early on bad session timers request This change allows chan_sip to avoid creation of the channel and consumption of associated file descriptors altogether if the inbound request is going to be rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	dbfbcbb1c7	chan_sip: Fix deadlock of monlock between unload_module and do_monitor Release monlock before calling pthread_join. This ensures do_monitor cannot freeze by locking monlock during module unload. (closes issue ASTERISK-21406) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3284/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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