74 Commits (c46372c67f1661f80fc4695148fc190d97b681ef)

Author	SHA1	Message	Date
George Joseph	bcdb008c0c	Alternate Channel Storage Backends ... Full details: http://s.asterisk.net/dc679ec3 The previous proof-of-concept showed that the cpp_map_name_id alternate storage backed performed better than all the others so this final PR adds only that option. You still need to enable it in menuselect under the "Alternate Channel Storage Backends" category. To select which one is used at runtime, set the "channel_storage_backend" option in asterisk.conf to one of the values described in asterisk.conf.sample. The default remains "ao2_legacy". UpgradeNote: With this release, you can now select an alternate channel storage backend based on C++ Maps. Using the new backend may increase performance and reduce the chances of deadlocks on heavily loaded systems. For more information, see http://s.asterisk.net/dc679ec3	5 months ago
George Joseph	67fd66e2fe	file.c: Add ability to search custom dir for sounds ... To better co-exist with sounds files that may be managed by packages, custom sound files may now be placed in AST_DATA_DIR/sounds/custom instead of the standard AST_DATA_DIR/sounds/<lang> directory. If the new "sounds_search_custom_dir" option in asterisk.conf is set to "true", asterisk will search the custom directory for sounds files before searching the standard directory. For performance reasons, the "sounds_search_custom_dir" defaults to "false". Resolves: #315 UserNote: A new option "sounds_search_custom_dir" has been added to asterisk.conf that allows asterisk to search AST_DATA_DIR/sounds/custom for sounds files before searching the standard AST_DATA_DIR/sounds/<lang> directory.	2 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. ... For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	3 years ago
George Joseph	ca3c22c5f1	Scope Tracing: A new facility for tracing scope enter/exit ... What's wrong with ast_debug? ast_debug is fine for general purpose debug output but it's not really geared for scope tracing since it doesn't present its output in a way that makes capturing and analyzing flow through Asterisk easy. How is scope tracing better? Scope tracing uses the same "cleanup" attribute that RAII_VAR uses to print messages to a separate "trace" log level. Even better, the messages are indented and unindented based on a thread-local call depth counter. When output to a separate log file, the output is uncluttered and easy to follow. Here's an example of the output. The leading timestamps and thread ids are removed and the output cut off at 68 columns for commit message restrictions but you get the idea. --> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 --> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 --> res_pjsip_session.c:3669 handle_incoming_response PJSIP/ --> chan_pjsip.c:3265 chan_pjsip_incoming_response_after --> chan_pjsip.c:3194 chan_pjsip_incoming_response P chan_pjsip.c:3245 chan_pjsip_incoming_respon <-- chan_pjsip.c:3194 chan_pjsip_incoming_response P <-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after <-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/ <-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 <-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 The messages with the "-->" or "<--" were produced by including the following at the top of each function: SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session)); Scope isn't limited to functions any more than RAII_VAR is. You can also see entry and exit from "if", "for", "while", etc blocks. There is also an ast_trace() macro that doesn't track entry or exit but simply outputs a message to the trace log using the current indent level. The deepest message in the sample (chan_pjsip.c:3245) was used to indicate which "case" in a "select" was executed. How do you use it? More documentation is available in logger.h but here's an overview: * Configure with --enable-dev-mode. Like debug, scope tracing is #ifdef'd out if devmode isn't enabled. * Add a SCOPE_TRACE() call to the top of your function. * Set a logger channel in logger.conf to output the "trace" level. * Use the CLI (or cli.conf) to set a trace level similar to setting debug level... CLI> core set trace 2 res_pjsip.so Summary Of Changes: * Added LOG_TRACE logger level. Actually it occupies the slot formerly occupied by the now defunct "event" level. * Added core asterisk option "trace" similar to debug. Includes ability to specify global trace level in asterisk.conf and CLI commands to turn on/off and set levels. Levels can be set globally (probably not a good idea), or by module/source file. * Updated sample asterisk.conf and logger.conf. Tracing is disabled by default in both. * Added __ast_trace() to logger.c which keeps track of the indent level using TLS. It's #ifdef'd out if devmode isn't enabled. * Added ast_trace() and SCOPE_TRACE() macros to logger.h. These are all #ifdef'd out if devmode isn't enabled. Why not use gcc's -finstrument-functions capability? gcc's facility doesn't allow access to local data and doesn't operate on non-function scopes. Known Issues: The only know issue is that we currently don't know the line number where the scope exited. It's reported as the same place the scope was entered. There's probably a way to get around it but it might involve looking at the stack and doing an 'addr2line' to get the line number. Kind of like ast_backtrace() does. Not sure if it's worth it. Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027	5 years ago
George Joseph	b76ab5e5c9	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	6 years ago
Joshua Colp	f03b984724	Merge "core: Remove additional symbols."	8 years ago
Jenkins2	60a06fbd1d	Merge "core: Remove dead symbols from asterisk.exports.in."	8 years ago
Corey Farrell	040bb21771	core: Remove additional symbols. ... Remove symbols that are depreacated and replaced: * ast_channel_datastore_alloc * ast_channel_datastore_free * ast_channel_cmpwhentohangup * ast_channel_setwhentohangup * config_text_file_save * devstate2str * ast_device_state_changed * ast_device_state_changed_literal * ast_verbose_get_by_module Remove unused symbols: * channelreloadreason2txt (last used in Asterisk 12). Remove unused ast_options flags: * AST_OPT_FLAG_END_CDR_BEFORE_H_EXTEN / ast_opt_end_cdr_before_h_exten * AST_OPT_FLAG_VERBOSE_MODULE / ast_opt_verb_module * AST_OPT_FLAG_INITIATED_SECONDS Change-Id: I841255995d195f8efc1ed47af9c7a2f131c08645	8 years ago
Corey Farrell	de77cf8698	core: Remove dead symbols from asterisk.exports.in. ... * dahdi_chan_name * dahdi_chan_name_len * dahdi_chan_mode * __manager_event * dialed_interface_info Added comment about __progname and environ being needed for FreeBSD to prevent accidental removal in the future. Change-Id: I3ae026bc541cd9cb572be2ffa95fc359547642b5	8 years ago
George Joseph	5d097f8236	channel.c: Allow generic plc then channel formats are equal ... If the two formats on a channel are equal, we don't transcode and since the generic plc needs slin to work, it doesn't get invoked. * A new configuration option "genericplc_on_equal_codecs" was added to the "plc" section of codecs.conf to allow generic packet loss concealment even if no transcoding was originally needed. Transcoding via SLIN is forced in this case. ASTERISK-27743 Change-Id: I0577026a179dea34232e63123254b4e0508378f4	8 years ago
Richard Mudgett	1a36a452bd	pjproject: Add cache_pools debugging option. ... The pool cache gets in the way of finding use after free errors of memory pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a pool is released because it gets put into the cache instead of being freed. * Added the "cache_pools" option to pjproject.conf. Disabling the option helps track down pool content mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the pool contents are used after free and who freed it. To disable the pool caching simply disable the cache_pools option in pjproject.conf and restart Asterisk. Sample pjproject.conf setting: [startup] cache_pools=no * Made current users of the caching pool factory initialization and destruction calls call common routines to create and destroy cached pools. ASTERISK-27704 Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828	8 years ago
Richard Mudgett	90bb0a3e10	core: Add cache_media_frames debugging option. ... The media frame cache gets in the way of finding use after free errors of media frames. Tools like valgrind and MALLOC_DEBUG don't know when a frame is released because it gets put into the cache instead of being freed. * Added the "cache_media_frames" option to asterisk.conf. Disabling the option helps track down media frame mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the frame is used after free and who freed it. NOTE: This option has no effect when Asterisk is compiled with the LOW_MEMORY compile time option enabled because the cache code does not exist. To disable the media frame cache simply disable the cache_media_frames option in asterisk.conf and restart Asterisk. Sample asterisk.conf setting: [options] cache_media_frames=no ASTERISK-27413 Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00	8 years ago
Kevin Harwell	9b103e7bea	rtp_engine: allocate RTP dynamic payloads per session ... Dynamic payload types were statically defined in Asterisk. This unfortunately limited the number of dynamic payloads that could be registered. With this patch dynamic payload type numbers are now assigned dynamically and per RTP instance. However, in order to limit any issues where some clients expect the old statically defined value this patch makes it so the value Asterisk used to pre- designate is used for the dynamic assignment if available. An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf) that turns the new dynamic behavior on or off. When off it reverts back to using statically defined payload values. This option defaults to "yes" in Asterisk 15. ASTERISK-26515 #close patches: ASTERISK-26515.diff submitted by jcolp (license 5000 Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc	9 years ago
Richard Mudgett	6f3e8c8e01	PJPROJECT logging: Fix detection of max supported log level. ... The mechanism used for detecting the maximum log level compiled into the linked pjproject did not work. The API call simply stores the requested level into an integer and does no range checking. Asterisk was assuming that there was range checking and limited the new value to the allowable range. To get the actual maximum log level compiled into the linked pjproject we need to get and save off the initial set log level from pjproject. This is the maximum log level supported. * Get and save off the initial log level setting before altering it to the desired level on startup. This has to be done by a macro rather than calling a core function to avoid incorrectly linking pjproject. * Split the initial log level warning messages to warn if the linked pjproject cannot support the requested startup level and if it is too low to get the pjproject buildopts for "pjproject show buildopts". * Adjust the CLI "pjproject set log level" to check the saved max log level and to generate normal output messages instead of a warning message. ASTERISK-26743 #close Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4	9 years ago
Richard Mudgett	1dfa11b65c	PJPROJECT logging: Made easier to get available logging levels. ... Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	9 years ago
Alexander Traud	9ac53877f6	rtp_engine: Allow more than 32 dynamic payload types. ... Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK (Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, this change utilizes payload types in the range between 35 and 63 giving room for another 29 payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964	9 years ago
Corey Farrell	57386dcb67	Allow command-line options to override asterisk.conf. ... Previous versions of Asterisk processed command-line options before processing asterisk.conf. This meant that if an option was set in asterisk.conf, it could not be overridden with the equivelent command line option. This change causes Asterisk to process the command-line twice. First it processes options that are needed to load asterisk.conf, then it processes the remaining options after the config is read. This changes the function of -X slightly. Previously using -X without disabling execincludes in asterisk.conf caused #exec to be usable in any config. Now -X only enables #exec for the load of asterisk.conf, if it is wanted in the rest of the system it must be enabled with execincludes in asterisk.conf. Updated 'asterisk -h' and 'man asterisk' to reflect the limited function of -X. ASTERISK-25042 #close Reported by: Corey Farrell Change-Id: I1450d45c15b4467274b871914d893ed4f6564cd7	11 years ago
Corey Farrell	5c1d07baf0	Astobj2: Allow reference debugging to be enabled/disabled by config. ... * The REF_DEBUG compiler flag no longer has any effect on code that uses Astobj2. It is used to determine if reference debugging is enabled by default. Reference debugging can be enabled or disabled in asterisk.conf. * Caller information is provided in logger errors for ao2 bad magic numbers. * Optimizes AO2 by merging internal functions with the public counterpart. This was possible now that we no longer require a dual ABI. ASTERISK-24974 #close Reported by: Corey Farrell Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1	11 years ago
Matthew Jordan	97834718c2	Remove many deprecated modules ... Billing records are fair, To get paid is quite bright, You should really use ODBC; Good-bye cdr_sqlite. Microsoft did once push H.323, Hell, we all remember NetMeeting. But try to compile chan_h323 now And you will take quite a beating. The XMPP and SIP war was fierce, And in the distant fray Was birthed res_jabber/chan_jingle; But neither to stay. For everyone did care and chase what Google professed. "Free Internet Calling" was what devotees cried, But Google did change the specs so often That the developers were happy the day chan_gtalk died. And then there was that odd application Dedicated to the Polish tongue. app_saycountpl was subsumed by Say; One could say its bell was rung. To read and parse a file from the dialplan You could (I guess) use an application. app_readfile did fill that purpose, but I think A function is perhaps better in its creation. Barging is rude, I'm not sure why we do it. Inwardly, the caller will probably sigh. But if you really must do it, Don't use app_dahdibarge, use ChanSpy. We all despise the sound of tinny robots It makes our queues so cold. To control such an abomination It's better to not use Wait/SetMusicOnHold. It's often nice to know properties of a channel It makes our calls right We have a nice function called CHANNEL And so SIPCHANINFO is sent off into the night. And now things get odd; Apparently one could delimit with a colon Properties from the SIPPEER function! Commas are in; all others are done. Finally, a word on pipes and commas. We're sorry. We can't say it enough. But those compatibility options in asterisk.conf; To maintain them forever was just too tough. This patch removes: * cdr_sqlite * chan_gtalk * chan_jingle * chan_h323 * res_jabber * app_saycountpl * app_readfile * app_dahdibarge It removes the following applications/functions: * WaitMusicOnHold * SetMusicOnHold * SIPCHANINFO It removes the colon delimiter from the SIPPEER function. Finally, it also removes all compatibility options that were configurable from asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems. Review: https://reviewboard.asterisk.org/r/3698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	03beadb6e9	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(). * Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	357b275239	Fix res_ari_asterisk load issue ... The new res_ari_asterisk.so module presents several config options from asterisk main. Unfortunately, they aren't exported, so the module won't load on Linux. This patch renames the variables, adding the ast_ prefix so they will be exported. Review: https://reviewboard.asterisk.org/r/2737 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Olle Johansson	7aa0c3c64b	Make it possible to change the minimum DTMF duration in asterisk.conf ... Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a35c7ba8e7	Add option to invoke the extensions.conf stdexten using the legacy macro method. ... ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in favor of the Gosub method without a means of backwards compatibility. (issue ASTERISK-18809) (closes issue ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1855/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	6bb45831eb	Fix transcode_via_sln option with SIP calls and improve PLC usage. ... From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b5a629624a	Merged revisions 264248 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) | 17 lines Internal timing is now on by default, if you're using DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is that this version ensures that a timer is always available, whereas in previous versions, it was possible for DAHDI to be loaded, but have no drivers to actually generate timing. If internal_timing was turned on in this circumstance, a complete lack of audio would result. This is the reason why internal_timing was not on by default. However, now that DAHDI ensures the availability of a timer, there is no reason for this setting to be off (and in fact, it solves a great many initial user problems). (closes issue #15932) Reported by: dimas Patches: 20100519__issue15932.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ae6008ef3a	Change per-file debug and verbose levels to be per-module, the way ... users expect them to work. 'core set debug' and 'core set verbose' can optionally change the level for a specific filename; however, this is actually for a specific source file name, not the module that source file is included in. With examples like chan_sip, chan_iax2, chan_misdn and others consisting of multiple source files, this will not lead to the behavior that users expect. If they want to set the debug level for chan_sip, they want it set for all of chan_sip, and not to have to also set it for reqresp_parser and other files that comprise the chan_sip module. This patch changes this functionality to be module-name based instead of file-name based. To make this work, some Makefile modifications were required to ensure that the AST_MODULE definition is present in each object file produced for each module as well. Review: https://reviewboard.asterisk.org/r/574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	a170cd28e0	Add new option to asterisk.conf (lockconfdir) to protect conf dir during reloads ... (closes issue #16358) Reported by: raarts Patches: lockconfdir.diff uploaded by raarts (license 937) modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	7c6d9c7235	Add option to hide console connect messages ... (closes issue #14222) Reported by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176) Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9335b3ad34	Allow people to select the old console behavior of white text on a black ... background, by using the startup flag '-B'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	8953b0f359	(closes issue #13366) ... Reported by: erousseau This was a reasonable enhancement request, which was easy to implement. Since it's an enhancement, it could only be applied to trunk. Basically, for accounting where "initiated" seconds are billed for, if the microseconds field on the end time is greater than the microseconds field for the answer time, add one second to the billsec field. The implementation was requested by erousseau, and I've implemented it as requested. I've updated the CHANGES, the cdr.conf.sample, and the .h files accordingly, to accept and set a flag for the corresponding new option. cdr.c adds in the extra second based on the usec fields if the option is set. Tested, seems to be working fine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6dd5b8609f	Optional light colored background, for those who use black on white terminals. ... (closes issue #13306) Reported by: Corydon76 Patches: 20080814__bug13306__3.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ed6323cb73	Merged revisions 133169 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	ef3b214728	Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	76506b7baa	Move compatibility options into asterisk.conf, default them to on for upgrades, ... and off for new installations. This includes the translation from pipes to commas for pbx_realtime and the EXEC command for AGI, as well as the change to the Set application not to support multiple variables at once. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	358ac2f76a	Merged revisions 110628 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f1f72312bb	(closes issue #10192) ... Reported by: bbryant Patches: 20070720__core_debug_by_file.patch uploaded by bbryant (license 36) (with some modifications by me) Tested by: russell, bbryant This set of changes introduces the ability to set the core debug or verbose levels on a per-file basis. Interestingly enough, in 1.4, you have the ability to set core debug for a single file, but that functionality was accidentally lost in the conversion of the CLI commands to the new format. This patch improves upon what was in 1.4 by letting you set it for more than 1 file, and by also supporting verbose. *** Janitor Project *** This patch also introduces a new macro, ast_verb(), which is similar to ast_debug(). Setting the per file verbose value only works for messages that use this macro. Converting existing uses of ast_verbose() can be done like: if (option_debug > 2) ast_verbose(VERBOSE_PREFIX_3 "Something useful\n"); ... ast_verb(3, "Something useful\n"); git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	81bc1d7af5	Merge in ast_strftime branch, which changes timestamps to be accurate to the microsecond, instead of only to the second ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	9e0458e9f1	Completely remove all of the code related to jumping to priority n + 101. yay! ... (issue #9926, caio1982) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	51f99c5265	- Add manager command CoreSettings ... - Add missing option to options.h - Add missing variables to asterisk.h - Move manager version to manager.h include file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Dwayne M. Hubbard	2151e532fe	changed #if HAVE_SYSINFO to #if defined(HAVE_SYSINFO) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Dwayne M. Hubbard	6a5f3599bb	added HAVE_SYSINFO preprocessor directives for portability and general happiness ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Dwayne M. Hubbard	62256ee410	added option_minmemfree for use in asterisk.conf to specify the amount of minimum free memory prior to accepting calls. added CLI 'core show sysinfo' to display system information ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	16b09ac48c	Merged revisions 48998 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006) | 3 lines move extern declaration for this option to a header file where it belongs provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	b04f059b4a	As discussed and decided on the asterisk-dev mailing list ... ... - Fix some breakage I introduced a while ago that made the timestamps option not functional for CLI verbose output. - Remove the use of the timestamps option for log output, since it was not functional. - clarify text referring to the timestamps option so that it is clear that it only applies to CLI verbose output git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	0f912e5926	Fix various problems in the addition of the ability to mute log/verbose ... output to remove consoles. The prototypes added to logger.h still need doxygen documentation, as well. - Add the new command line option to the man page - make the mute option a flag instead of an int since it is only a binary option - remove useless extern keywords for prototypes added to logger.h - rename ast_console_mute() to ast_console_toggle_mute() since that is what it actually does - actually apply the mute option to newly created remote consoles instead of only working when the CLI command is used - don't imply the NO_FORK option if the mute command line option is provided - place the new CLI command in the correct place in the list which has to be in alphabetical order - Finally, clean up a few spacing issues to conform to the coding guidelines git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
BJ Weschke	569ecaf5a2	use pid_t instead of long for pid variables. #7099 (casper) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	eb38f13a2e	add a command-line flag and option to force forking, even with -v or -d ... rename a flag to have the proper name git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Russell Bryant	03ce34e1aa	convert internal timing to be stored as a flag in the ast_options flags ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Olle Johansson	50f0b12880	Issue #5374 - Enable internal timing of generators (cmantunes) ... Thanks everyone involved for hard work, testing and testing! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Russell Bryant	4279bf18f8	clarify which global options are enabled by default ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago