1797 Commits (4863b465af2d833720ee3dc26652f57f05d8b4da)

Author	SHA1	Message	Date
Richard Mudgett	6eff978ff7	res_musiconhold: Fix MOH restarting where it left off from the last hold. ... Restore code removed by https://reviewboard.asterisk.org/r/3536/ that introduced a regression that prevents MOH from restarting were it left off the last time. ASTERISK-24019 #close Reported by: Jason Richards Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3928/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@421976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f565f9a708	res_musiconhold.c: Remove obsolete REF_DEBUG code. ... Remove unneeded code that writes to the wrong file location in an obsolete format. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@421799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	21c4871730	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. This includes an extra check to prevent the errors previously experienced in the testsuite and has 100+ test runs behind it. Review: https://reviewboard.asterisk.org/r/3615/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 years ago
Russell Bryant	ad637d43d5	moh: fix a refcount error with realtime MOH ... I observed a crash in res_musiconhold on an Asterisk 11 system using realtime MOH. Investigation of the backtrace showed a corrupt mohclass, implying that it got destroyed before the code expected it to. I went looking for reference counting errors that could have caused this crash and this patch this result. It contains 2 changes. 1) Remove a usless block of code that was impossible to reach. There was even a comment indicating that it was impossible to reach. The conditional includes "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's inside of an if block with the opposite check "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no good reason to keep it around. 2) A similar block to #1 contained a reference counting error. It stores state->class in the local variable mohclass without increasing its reference count. The reference count on mohclass is decremented at the end of the function. This block of code probably very rarely runs, which would help explain why this system was working fine for many months before experiencing a crash. Review: https://reviewboard.asterisk.org/r/3282/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	a6a92ffe3d	res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load. ... Change minrate from 2400 to 4800 on config reload in response to changes from ASTERISK-22790 only. Any config with minrate of 2400 that would fail before r405693 will still fail. Comment out many settings in res_fax.conf.sample. The defaults are set in res_fax.c, so setting the same value in sample config does nothing but make the sample config more fragile. (closes issue ASTERISK-23231) Reported by: David Brillert Review: https://reviewboard.asterisk.org/r/3261/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	b78f5a2d30	Remove extra defines of AST_PBX_MAX_STACK. ... * Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix incorrect function parameters in utils/extconf.c. (closes issue ASTERISK-23141) Reported by: Maxim Review: https://reviewboard.asterisk.org/r/3241/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	f6aed66f77	res_config_pgsql: Fix ast_update2_realtime calls. ... Fix so multiple updates from a single call works (add missing ','). Remove bogus ast_free's that weren't supposed to be there. Moved a few spaces for readability. Review: https://reviewboard.asterisk.org/r/3194/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3e1ec590f3	res_clialiases: Fix crash when reloading and re-aliasing an alias that is in use. ... The code assumed that unregistering the alias would always succeed while in practice this is not actually true. A common case is the "reload" command itself. If the cli_aliases.conf configuration file was changed and reload executed the command would fail to unregister and ultimately point to freed memory. The reload process now checks whether unregistering succeeded or not and if not the old CLI alias is retained. (closes issue ASTERISK-19773) Reported by: Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth Blades git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	ea65e4d44c	res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions ... ast_bind to a port reserved for another program by SELinux causes errno == EACCES. This caused random failures when binding rtp or udptl sockets. Treat EACCES as a non-fatal error, try next port. (closes issue ASTERISK-23134) Reported by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	8c63b6d468	res_config_pgsql: Fix a memory leak and use RAII_VAR for cleanup when practical ... Review: https://reviewboard.asterisk.org/r/3141/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4d70fa4750	chan_mgcp: Enforce locking for oseq ... This restricts direct usage of global oseq so that all accesses are locked and threads are not racing to get oseq values that they did not claim. This also fixes a build error in res_pktccops under dev mode. (closes issue ASTERISK-23100) Reported by: adomjan Patch by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	affe507d5b	res_fax: check_modem_rate() returned incorrect rate for V.27 ... According to the new standard for V.27 and V.32 they are able to transmit at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be updated to reflect this. Also, because of this change the default 'minrate' value was updated to be 4800. (closes issue ASTERISK-22790) Reported by: Paolo Compagnini Patches: res_fax.txt uploaded by looserouting (license 6548) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	4a289473f1	res_calendar: Protect channel when adding datastore. ... This change adds a missing channel lock when adding a datastore to a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@404135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	72641b7711	res_fax_spandsp: Always init T.38 session to avoid crashes during state change ... Prior to this patch, res_fax_spandsp was conservative with how it initialized the spandsp T.38 context. It would only initialize it if the driver thought the current state was a T.38 fax. While this works fine in nominal situations, in certain off nominal situations, res_fax_spandsp can believe that a T.38 fax will not occur when in fact one has started. In particular, this was discovered when res_fax would fall back to audio after timing out on a T.38 upgrade. The SIP channel driver would continue to retry the re-INVITE and - if the remote end responded after res_fax timed out with a 200 OK - a T.38 frame would be delivered to the res_fax stack when it no longer expected it. As it turns out, there does not appear to be any downside to always initializing the T.38 context, other than the actual memory allocation. Since that avoids this off nominal situation (and others which are equally likely hard to predict), this is the safest way to avoid this problem. Much thanks to Torrey as well for providing a scenario that reproduces this issue. (closes issue ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey Searle patches: always-init-t38.patch uploaded by awinters (License 6477) A_PARTY.xml uploaded by tsearle (License 5334) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@403449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	8c07e036b2	res_rtp_asterisk: Address jittery DTMF events in RTP streams ... (closes issue ASTERISK-21170) Reported by: NITESH BANSAL Patches: dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418) Review: https://reviewboard.asterisk.org/r/2938/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	58542a7c27	res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change ... In r400089, a patch was put in to correct erroneous RTCP statistic resets. Unfortunately, ast_rtp_read can be called on an RTP instance that does not have RTCP information. This patch prevents that crash by only resetting the statistics if we do actually have an RTCP instance. (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	cd1d5ea15e	Reduce log level of a non-pubsub error message ... Drop an error log message to debug level 1 since distributed device state functions correctly when receiving this message and it spams the logs. (closes issue ASTERISK-22410) Reported by: abelbeck Patches: asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4393ef861c	res_rtp_multicast: Ensure SSRC is set properly ... This fixes a bug where the SSRC field on multicast RTP can be stuck at 0 which can cause problems for endpoints trying to make sense of incoming streams. (closes issue ASTERISK-22567) Reported by: Simone Camporeale Patches: 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	068e1928cb	res_rtp_asterisk: Correct erroneous lost packet information in RTCP reports ... RTCP's calculation of the number of lost packets in an RTP stream is based on that stream's sequence number count, the number of received packets, and how many packets we expect to receive. When the SSRC for an RTP stream changes, there can - and almost always will be - a large jump in the next packet's timestamp and sequence number. If we don't reset the number of received packets, sequence number count, and other metrics used by RTCP, the next RR/SR report will use the previous SSRC's values to calculate the lost packet count for the new SSRC - resulting in a very large number of lost packets. This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it will reset the various values used by the RTCP calculations. From the perspective of RTCP, this appears as a new media stream - which is what it is. Review: https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	be50098c82	res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set ... Due to a faulty function for debugging reference decrementing, it was possible to reduce the refcount on the wrong object if two moh classes of the same name were in the moh class container. (closes issue ASTERISK-22252) Reported by: Walter Doekes Patches: 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	303d6c56f9	Fix incorrect usages of ast_realloc(). ... There are several locations in the code base where this is done: buf = ast_realloc(buf, new_size); This is going to leak the original buf contents if the realloc fails. Review: https://reviewboard.asterisk.org/r/2832/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	f09236541e	Commit the remainder of r398523 ... This is a missing part of the commit in revision 398523 that corrects the name of a variable. (issue ASTERISK-22435) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	05a7d1998d	Fix Jabber/XMPP distributed MWI ... The mailbox and context are swapped on the receiving end for all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent versions. This swaps those values to be correct when publishing to the internal event system from Jabber/XMPP distributed MWI state. (closes issue ASTERISK-22435) Reported by: abelbeck Tested by: Michael Keuter Patches: asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	e179b6e56e	Fix various memory leaks ... main/config.c - cleanup cache fie includes res/res_security_log.c - unregister logger level channesl/chan_sip.c - cleanup io context and notify_types (closes issues ASTERISK-22378) Reported by: Corey Farrell Patches: config_shutdown.patch uploaded by coreyfarrell (license 5909) res_security_log.patch uploaded by coreyfarrell (license 5909) chan_sip-1.8.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	3c840c1224	Memory leak fix ... ast_xmldoc_printable returns an allocated block that must be freed by the caller. Fixed manager.c and res_agi.c to stop leaking these results. (closes issue ASTERISK-22395) Reported by: Corey Farrell Patches: manager-leaks-1.8.patch uploaded by coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	6d49e9827b	Properly indicate failure to open an audio stream in res_agi ... If there is an error streaming an audio file, the current return status makes it difficult for an AGI script to determine that there was an error with the audio file. This patches changes the result to return -1 and the function returns RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other parts of res_agi, this would appear to be the proper way to handle an error. (closes issue ASTERISK-21903) Reported by: Ariel Wainer Tested by: Ariel Wainer Patches: asterisk-21903-return-stream-res_1.8.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2625/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@394640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	f0a8ba971b	Use srtp_shutdown when available ... This allows the SRTP library to be shut down properly when the functionality is offered by libsrtp. Review: https://reviewboard.asterisk.org/r/2538/ (closes issue ASTERISK-21719) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	556e0a26ff	Revert r388529 for now ... Adding the cleanup function needs some deeper thought since it apparently doesn't exist for all variants of libsrtp. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	95ad8ea190	Close libsrtp properly ... Ensure that libsrtp is shutdown properly when res_srtp is unloaded. (closes issue ASTERISK-21719) Reported by: Corey Farrell Patches: res_srtp-library-shutdown.patch uploaded by Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	276c5e6d45	Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit ... When we send out a CN packet (for instance, in the case of using rtpkeepalives), we are not setting the payload code properly. Also, we are setting the marker bit when we shouldn't be according to RFC 3389, section 4. AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we should be using ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup(). 11 and trunk already use the appropriate function. * In 1.8, use ast_rtp_codecs_payload_code() * Remove the setting of the marker bit * Fix the debug message by incrementing the seqno after the debug message is set in order to display the correct seqno that was sent out (closes issue ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter Katzmann, Michael L. Young Patches: asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2500/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@388111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a7732dfa30	Clear the DTMF sending digit tracking on off nominal paths ... In certain situations, when the RTP engine goes to send a DTMF end digit it may be in a situation where the remote address is no longer available, or the digit that was supposed to be sent is invalid. In such cases, we need to clear the RTP counters appropriately. Otherwise, when the RTP source is set again, we'll continue to think that we're in the middle of sending a DTMF digit, which can confuse the remote party (signficantly). (closes issue ASTERISK-21522) Reported by: Corey Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	6f8468d863	Prevent res_timing_pthread from blocking callers ... There were several reports of deadlock when using res_timing_pthread. Backtraces indicated that one thread was blocked waiting for the write to the pipe to complete and this thread held the container lock for the timers. Therefore any thread that wanted to create a new timer or read an existing timer would block waiting for either the timer lock or the container lock and deadlock ensued. This patch changes the way the pipe is used to eliminate this source of deadlocks: 1) The pipe is placed in non-blocking mode so that it would never block even if the following changes someone fail... 2) Instead of writing bytes into the pipe for each "tick" that's fired the pipe now has two states--signaled and unsignaled. If signaled, the pipe is hot and any pollers of the read side filedescriptor will be woken up. If unsigned the pipe is idle. This eliminates even the chance of filling up the pipe and reduces the potential overhead of calling unnecessary writes. 3) Since we're tracking the signaled / unsignaled state, we can eliminate the exta poll system call for every firing because we know that there is data to be read. (closes issue ASTERISK-21389) Reported by: Matt Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches: 0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417) (closes issue ASTERISK-19754) Reported by: Nikola Ciprich (closes issue ASTERISK-20577) Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported by: Henry Fernandes (closes issue ASTERISK-17467) Reported by: isrl (closes issue ASTERISK-17458) Reported by: isrl Review: https://reviewboard.asterisk.org/r/2441/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	e4854dd1b2	Distributed Device State broken at sites using res_xmpp or res_jabber where Secuity Advisory AST-2012-015 is inplace ... res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the event as each message came in, then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2452/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	a6d0dfc3d1	Fix the svn:keywords property on several files. ... Normally I think keyword expansion is silly, but the one time it would have been good, it didn't work because the property had quotes in it. This patch fixes obviously busted svn:keywords properties. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5a756f15f8	Calculate the timestamp for outbound RTP if we don't have timing information ... This patch calculates the timestamp for outbound RTP when we don't have timing information. This uses the same approach in res_rtp_asterisk. Thanks to both Pietro and Tzafrir for providing patches. (closes issue ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035) rtp-timestamp.patch uploaded by pbertera (License 5943) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	6166563bb3	Use LDAP memory management functions instead of Asterisk's ... When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk: invalid pointer errors) can occur as the memory is being allocated with Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the LDAP library's wrappers. This patch uses the LDAP library's wrappers where appropriate, so that compiling with MALLOC_DEBUG doesn't cause more problems than it solves. Note that the patch listed below was modified slightly for this commit to account for some additional memory allocation/deallocations. (closes issue ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham patches: issue18789-1.8-r316873.patch uploaded by seanbright (License 5060) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	36706e0a93	Fix white noise on SRTP decryption ... When res_rtp_asterisk.c was altered to avoid attempting to apply unprotect algorithms to non-audio RTP packets, the test used was incorrect. This caused the audio packets to not be decrypted and resulted in loud white noise on the other endpoint (or both endpoints depending on the call legs involved). The test now properly checks the version field in the RTP header to ensure that RTP and RTCP are decrypted while other types of packets are not. (closes issue ASTERISK-21323) Reported by: andrea Tested by: Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff uploaded by Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	8c4732636f	Properly delimit post data in res_config_curl. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago