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authorkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2006-01-24 22:42:43 +0000
committerkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2006-01-24 22:42:43 +0000
commit85f4432d674857e6a3b9530cbd4b914f95b8eaf8 (patch)
treeb014de9950559748eb5a63fecbb5c5b170b6f8c0 /ChangeLog
parent429f10b01a816a89bbe0773a3d6ea52848cb9118 (diff)
Merged revisions 8173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r8173 | russell | 2006-01-17 20:49:21 -0600 (Tue, 17 Jan 2006) | 2 lines remove ChangeLog from the 1.2 branch. It will only be present in the tags. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8595 f38db490-d61c-443f-a65b-d21fe96a405b
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-2005-11-29 Josh Roberson <josh@asteriasgi.com>
-
- * apps/app_voicemail.c: Only look in 'default' context when no context defined to VoiceMailMain(). (issue #5887)
-
-2005-11-25 Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c: Properly duplicate the string for ANI (issue #5850)
-
-2005-11-23 Russell Bryant <russell@digium.com>
-
- * configs/voicemail.conf.sample: Add note to indicate that #include should not be used for this file. (issue #5828)
-
-2005-11-23 Josh Roberson <josh@asteriasgi.com>
-
- * configs/indications.conf.sample: Fix misspellings of cadence (issue #5826)
- * include/asterisk/indications.h: Fix misspellings of cadence (issue #5826)
- * res/res_indications.c: Fix misspellings of cadence (issue #5826)
- * indications.c: Fix misspellings of cadence (issue #5826)
-
-2005-11-22 Josh Roberson <josh@asteriasgi.com>
-
- * apps/app_voicemail.c: Remove "yay!" message. (issue #5829)
-
-2005-11-21 Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_cut.c: remove unnecessary include that causes spurious rebuilding
-
- * channels/chan_sip.c (build_peer): ensure that case changes made to peer names are not ignored during reload operations
- (build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified
-
- * channels/chan_iax2.c (build_peer and build_user): ensure that case changes made to peer/user names are not ignored during reload operations
- (build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified
-
-2005-11-21 Josh Roberson <josh@asteriasgi.com>
-
- * Makefile: Re-fix Darwin poll issues.
-
-2005-11-21 Russell Bryant <russell@digium.com>
-
- * apps/app_osplookup.c: Properly populate the number of results. (issue #5789)
-
- * Makefile: Don't hard-code that poll functionality needs to be provided on Darwin.
- * apps/Makefile: Fix incorrect portion of the patch to fix 'make install' on Solaris.
-
- * channels/chan_iax2.c (iax2_getpeername): Return non-zero to indicate that a peer was found when using realtime (issue #5815)
-
-2005-11-20 Russell Bryant <russell@digium.com>
-
- * Makefile apps/Makefile: Fix 'make install' for Solaris. (issue #5775)
-
- * apps/app_record.c: Don't leak a frame if writing it to the file fails. (issue #5787)
-
- * Makefile: Create the monitor spool directory when the other spool directories are created.
-
- * pbx.c.c: Remove some useless checks and unnecessary calls to ast_strlen_zero(). (issue #5805)
-
- * cli.c: Remove some unnecessary calls to ast_strlen_zero(). (issue #5804)
-
- * channels/chan_oss.c configs/oss.conf.sample: Add the ability to set callerid in oss.conf.
-
- * channels/chan_sip.c channels/chan_iax2.c: Change warning messages about the number of scheduled events happening all at once to debug messages. (issue #5794)
-
-2005-11-20 Josh Roberson <josh@asteriasgi.com>
-
- * pbx/pbx_spool.c: Fix crash in spooler if set/setvar declared incorrectly. (issue #5806)
-
- * apps/app_meetme.c: fix 'X' option in MeetMe, with slight modification. (issue #5773)
-
- * apps/app_voicemail.c: Make sure we're copying the read digits when calling voicemail without a box. (issue #5774)
-
- * apps/app_md5.c: Fix conditional jump option.
-
- * apps/app_controlplayback.c: Fix conditional jump option.
-
- * apps/app_hasnewvoicemail.c: Fix conditional jump option to jump properly, also correct a small typo in the description. (issue #5795)
-
- * channels/chan_iax2.c: Fix output of iax2 show peer <peer> (issue #5792)
-
- * UPGRADE.txt: Adjust note for naming conventions of iax2 channels. (issue #5792)
-
- * res/res_musiconhold.c: Correct typo in ast_copy_string() for class->mode. (issue #5803)
-
-2005-11-19 Josh Roberson <josh@asteriasgi.com>
-
- * channels/Makefile: Put chan_oss back into the default build. (issue #5799)
-
- * funcs/func_enum.c: Fix long text description causing cosmetic defect on module load. (issue #5791)
-
-2005-11-19 Russell Bryant <russell@digium.com>
-
- * app/app_echo.c: Update application description to be a bit more accurate, and clean up a little bit of code formatting
-
-2005-11-16 Russell Bryant <russell@digium.com>
-
- * Makefile: Fix the output of Makefile generated variables to doxygen
-
- * channels/chan_sip.c: Add missing carriage return and line feed to the SDP line indicating that we don't support VAD (issue #5780)
-
-2005-11-16 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.2.0 released.
-
-2005-11-16 Jeremy McNamara <jj@nufone.net>
-
- * apps/app_voicemail.c (load_config): do not terminate asterisk if no voicemail config file
- * channels/chan_skinny: Don't register channel type until ready, code formatting updates
-
-2005-11-16 Josh Roberson <josh@asteriasgi.com>
-
- * Makefile: Update to fix non-responsive remote console on Darwin (OSX)(issue #5757)
-
-2005-11-16 Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/Makefile: don't build chan_modem and sub-modules by default
- * configs/modules.conf.sample: explicitly 'noload' chan_modem.so and submodules, in case old versions exist
-
- * res/Makefile: issue mpg123 not-installed warning at 'make install' time, not 'make'
-
- * apps/app_forkcdr.c (forkcdr_exec): issue warning (and don't segfault) if ForkCDR is called on a channel that doesn't have a CDR (issue #5763)
-
- * channel.c (ast_queue_hangup): ensure that the channel lock is held before changing its fields... (issue #5770)
-
- * res/res_musiconhold.c: don't spit out incorrect log messages (and leak memory) during reload (issue #5766)
-
- * channels/chan_sip.c (process_sdp): don't pass video codec number into ast_getformatname(), it is not valid input for that function (issue #5764)
-
- * pbx/pbx_ael.c (match_assignment): properly parse equal signs surrounded by whitespace (issue #5761)
-
- * doc/README.realtime: document the limitations of using FreeTDS with Realtime (issue #5767)
-
-2005-11-15 Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile: use -g3 for compiler to include macro information for debugger
-
- * astmm.c (__ast_vasprintf): don't re-use the ap list without copying it; that's not safe on some platforms (issue #5035)
-
- * doc/README.backtrace: add note about properly building Asterisk to be able to produce backtraces; wrap text and remove DOS line endings
-
- * channels/chan_sip.c (add_codec_to_sdp): add 'annexb=no' to G.729A SDP (issue #5539)
-
- * channels/chan_alsa.c (alsa_hangup): handle autohangup properly (issue #5672)
-
- * channels/chan_misdn.c (and other files): various fixes (issue #5739)
-
- * channels/chan_sip.c (handle_request_info): properly forward 'flash' events received via SIP INFO (issue #5751, different patch)
-
- * apps/app_disa.c (disa_exec): don't duplicate constant strings when not needed
-
- * apps/app_playback.c (playback_exec): use correct logic tests for options (issue #5752)
-
- * apps/app_disa.c (disa_exec): use standard arg parsing routines (issue #5736)
-
-2005-11-15 Russell Bryant <russell@digium.com>
-
- * manager.c: Don't crash on a SetVar action if the channel name is not set, or variable's value is not set (issue #5760)
-
- * doc/README.variables: Add application exit status variables
-
-2005-11-14 Josh Roberson <josh@asteriasgi.com>
-
- * manager.c: Fix crash on variable passing from AMI originate (issue #5737)
-
-2005-11-14 Russell Bryant <russell@digium.com>
-
- * many files: Merge doxygen documentation updates. (issue #5605)
-
- * apps/app_dial.c: Fix typo in RetryDial description.
-
-2005-11-12 Russell Bryant <russell@digium.com>
-
- * channels/chan_oss.c: Fix a typo in an error message.
-
-2005-11-11 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.2.0-rc2 released.
-
-2005-11-11 Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c (thread_safe_rand): ensure that threads don't get the same random number (issue #5712)
-
- * apps/app_voicemail.c (forward_message): correct bugs in message forwarding (issue #5718)
- (copy_message): use correct path for locking (issue #5704)
-
- * apps/app_dial.c (wait_for_answer): correct flag copying for automon feature (issue #5720)
-
- * channels/chan_iax2.c: correct comment
-
- * apps/app_voicemail.c (close_mailbox): correct previous commit (issue #5663)
- (vm_change_password): fix password change writing (issue #5721)
-
- * channels/chan_sip.c (transmit_invite): remove useless debug message; don't try to add OSP tokens to OPTIONS pings
-
- * apps/app_voicemail.c (close_mailbox): properly remove deleted messages at mailbox close time (issue #5663)
-
-2005-11-11 Mark Spencer <markster@digium.com>
-
- * channels/chan_zap.c (zt_bridge): only enable/disable DTMF detection on SUB_REAL channels
-
-2005-11-10 Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: ensure that system headers that provide basic types are included first (issue #5713)
-
-2005-11-11 Russell Bryant <russell@digium.com>
-
- * many files in apps/: Clean up application descriptions. Clarify some wording and make sure they wrap at 80 characters.
-
-2005-11-10 Mark Spencer <markster@digium.com>
-
- * rtp.c (ast_rtp_raw_write): use unsigned int for return value from calc_txstamp() (issue #5595)
- (calc_txstamp): never return a value that was less than zero before being turned into 'unsigned int' (issue #5595)
-
-2005-11-10 Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/chanspy.h: move spy-related stuff into separate header so chan_h323 can build (issue #5590)
-
- * include/asterisk/linkedlists.h (AST_LIST_HEAD_SET_NOLOCK): properly initialize tail pointer when list head is directly set (issue #5669)
-
- * app.c (ast_app_parse_options): ok, so we aren't all perfect... let's make the while loop actually work properly here (issue #5684)
-
- * apps/app_disa.c (disa_exec): correct password file parsing (issue #5676)
-
- * apps/app_meetme.c (conf_run): don't restrict admin users from joining a locked conference (issue #5680)
-
- * channels/chan_misdn.c: include stdio.h (issue #5671)
- * channels/chan_misdn_config.c: fix prototype warning (issue #5671)
-
- * pbx.c: remove apps that were deprecated before 1.0 was released (issue #5673)
-
- * apps/app_striplsd.c, apps/app_substring.c: remove apps that were deprecated before 1.0 was released (issue #5673)
-
- * include/asterisk/lock.h (PTHREAD_MUTEX_RECURSIVE_NP): work around header problems on Cygwin (issue #5668)
-
- * pbx/pbx_ael.c: handle switch default cases inside macros properly (issue #5354)
-
- * configs/voicemail.conf.sample (format): add strong warning about changing format list when mailboxes contain messages (issue #5689)
-
- * many files: ensure that system headers are included before Asterisk headers (issue #5693)
-
- * channels/chan_iax2.c (complete_iax2_show_peer): don't return from function without releasing lock (issue #5685)
-
- * channels/iax2-provision.c (iax_provision_reload): don't leak memory (issue #5700)
-
- * pbx/pbx_ael.c (handle_macro): don't leak memory (issue #5701)
- (handle_context): ditto
-
- * res/res_features.c (load_config): properly initialize referenced variable (issue #5703)
-
- * apps/app_queue.c (rqm_exec): correct segfault problem (issue #5705)
- (aqm_exec): ditto
-
- * app.c (ast_app_parse_options): don't increment 's' until after checking for NULL (related to issue #5630)
-
- * apps/app_rpt.c: solve a memory leak (config structure was not freed) (issue #5706)
-
-2005-11-10 Russell Bryant <russell@digium.com>
-
- * app.c (ast_app_separate_args): Don't consider the open parenthesis as part of the arguments to an option. (issue #5630)
-
- * many files: Change all references to ast_separate_app_args to ast_app_separate_args
-
- * many files in apps/: Clean up some application descriptions. Make sure all descriptions in changed files are wrapped at 80 characters.
-
-2005-11-09 Russell Bryant <russell@digium.com>
-
- * pbx.c: Clean up descriptions of built-in dialplan applications. Changes include clearer wording and not referring to return values.
-
-2005-11-09 Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c (update_registry): don't complain about unspecifed registration expiration intervals, just use the minimum
-
-2005-11-08 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.2.0-rc1 released.
-
- * include/asterisk/file.h: add test to ensure that stdio.h is included before this file (issue #5658)
-
- * res/res_odbc.c (odbc_prepare_and_execute): add new API call for use to properly handle prepared statements across server disconnects (issue #5563)
-
- * pbx.c (pbx_substitute_variables_helper_full): use already-substituted buffer for parsing variable name (issue #5664)
-
- * channels/chan_zap.c (zt_request): return AST_CAUSE_CONGESTION when a group-channel is requested and the group exists but all channels are busy (issue #3360, related fix)
- * channels/chan_iax2.c (create_addr): treat UNREACHABLE as AST_CAUSE_UNREGISTERED so that it will generate CHANUNAVAIL from app_dial (issue #3360)
-
- * res/res_features.c (ast_bridge_call_thread_launch): set SCHED_RR separately from thread creation, so it won't fail when running as non-root (issue #5601, different fix)
-
- * pbx.c (pbx_builtin_pushvar_helper): add new API function for setting variables that can exist multiple times (issue #2720)
- * apps/Makefile (APPS): add app_stack (issue #2720)
- * apps/app_stack.c: new applications (issue #2720)
-
- * apps/app_meetme.c: fix two audio delay problems related to using non-Zap channels in conferences (issues #3599 and #4252)
- * configs/meetme.conf.sample: add documentation of new 'audiobuffers' setting to control buffering on incoming audio from non-Zap channels
-
- * channels/chan_local.c (local_call): move channel variables from incoming to outgoing instead of inheriting them (issue #5604)
-
- * many files: add explicit include of stdio.h (issue #5650)
-
-2005-11-07 Kevin P. Fleming <kpfleming@digium.com>
-
- * UPGRADE.txt (Parking): add note about new parking behavior (issue #5532)
-
- * many files: more Cygwin compatibility, and proper getloadavg() prototype/macro (issue #5569)
-
- * include/asterisk/lock.h (__ast_pthread_mutex_lock): correct build with DETECT_DEADLOCKS defined (issue #5570)
-
-2005-11-07 Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: upgrade to new arg/option API and implement priority jumping control (issue #5580)
- * many files: Add missing include of stdio.h, and remove some duplicate and unused header includes
-
- * include/asterisk/app.h: Increment the arg_index in the options structure to fix applicaiton options that have arguments to them
-
-2005-11-07 Kevin P. Fleming <kpfleming@digium.com>
-
- * cryptostub.c: include necessary headers
- * include/asterisk/crypto.h: don't include unnecessary headers
-
- * manager.c (action_setvar): add support for setting global variables (issue #5571)
-
- * Makefile: correct cross-compilation issue introduced in Cygwin patches (issue #5572)
-
- * apps/app_voicemail.c: upgrade to new arg/option API and implement priority jumping control (issue #5649)
-
- * asterisk.c (main): setpriority() failure is not a reason to stop the process (issue #5581)
-
- * say.c (ast_say_date_with_format_da): say hours properly (issue #5576)
-
- * manager.c (astman_get_variables): restore old multiple-variable behavior for "Variable" header (issue #5585)
-
- * many files: don't check for NULL before calling ast_strlen_zero, it can do it itself (issue #5648)
-
- * pbx.c (handle_show_hints): use proper state-to-string function for hint state (issue #5583)
-
- * rtp.c: use unsigned format for debug packet output (issue #5595)
-
- * asterisk.c (main): force a dnsmgr background refresh after all other modules are initialized (issue #5599)
- * dnsmgr.c: add ability to start a background refresh on demand (issue #5599)
-
- * apps/app_dial.c (HANDLE_CAUSE): set CDR disposition to match cause code (issue #5602)
-
- * asterisk.c: support 'runuser' and 'rungroup' options in asterisk.conf (issue #5621)
-
- * res/Makefile, apps/Makefile, channels/Makefile, Makefile: support WITHOUT_ZAPTEL define to forcibly avoid building Zaptel support (issue #5634)
-
- * Makefile: various fixes (issue #5633)
-
- * apps/app_osplookup.c: upgrade to new arg/option API and implement priority jumping control
-
- * channels/chan_misdn.c: various fixes (issue #5639)
- * channels/misdn/isdn_lib.c: various fixes (issue #5639)
-
- * apps/app_playback.c: upgrade to new arg/option API and implement priority jumping control
-
- * apps/app_privacy.c: upgrade to new arg/option API and implement priority jumping control
-
- * apps/app_sendtext.c: upgrade to new arg/option API and implement priority jumping control
-
- * apps/app_transfer.c: upgrade to new arg/option API and implement priority jumping control
-
- * apps/app_txtcidname.c: upgrade to new arg/option API and implement priority jumping control
-
- * Makefile: restore function of 'dont-optimize'
-
- * config.c (config_text_file_load): don't generate log message when stat() fails
-
- * many files: clean up application documentation to not refer to return values, since they cannot be used in the dialplan (work done by Neil Lewis)
-
-2005-11-06 Russell Bryant <russell@digium.com>
-
- * many files: alphabetize options in applicaiton descriptions
-
- * channels/chan_iax2.c: Use an enum to define iax peer/user flags as well as the pvt structure state. Use the ast_flags macros for checking or setting the state.
-
- * sounds.txt: Add missing words from the description of the vm-opts prompt
-
- * apps/app_externalivr.c: Add a space that fixes building on older versions of gcc
-
- * many files: Add doxygen updates to categorize modules into groups. Convert a lot of comments over to doxygen style. Add some text giving a basic overview of channels.
-
- * many files: Update applications to add an exit status variable, make priority jumping optional, and use new args parsing macros
-
- * pbx.c cdr.c res/res_features.c apps/app_dial.c include/asterisk/cdr.h: Convert some built-in applications to use new args parsing macros. Change ast_cdr_reset to take a pointer to an ast_flags structure instead of an integer for flags.
-
- * channels/chan_agent.c: Don't loop forever on an invalid options string
-
- * apps/app_disa.c apps/app_forkcdr.c: Fix to use correct arguments to ast_cdr_reset
-
-2005-11-05 Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile: don't rebuild asterisk/build.h unless the asterisk binary is going to be relinked for some other reason (stops spurious recompile/link every time 'make' is issued); clean up variable substitutions to use consistent syntax
- * asterisk.c: don't include asterisk/build.h (it's unnecessary)
- * cli.c: don't include asterisk/build.h, use extern references to buildinfo.c
- * buildinfo.c: new file to hold version info strings
-
-2005-11-04 Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_mixmonitor.c (mixmonitor_exec): correct app name in an error message
-
-2005-11-04 Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Create a function that stores a peer's status in a given buffer. Use this function in "iax2 show peers" and "iax2 show peer <peername>". Also, add the peer's status as an option to the IAXPEER dialplan function.
-
-2005-11-04 Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/compiler.h: don't try to use always_inline on old compilers
-
-2005-11-03 Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: initialize buffer for result so that the contents are always valid in the response to GET FULL VARIABLE
-
-2005-11-03 Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/README.variables: document DYNAMIC_FEATURES
-
- * res/res_features.c (ast_bridge_call): remove unused variables
-
- * apps/app_dial.c (dial_exec_full): simplify options and flag usage
-
- * include/asterisk/app.h: re-work application arg/option parsing APIs for consistent naming, add doxygen docs for option API
- * many files: update to new APIs
-
-2005-11-02 Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_dial.c (dial_exec_full): convert to use API calls for argument/option parsing
-
- * include/asterisk/channel.h: add doxygen docs for silence generator APIs
-
- * channel.c (ast_channel_bridge): simplify native-bridge return logic, remove 'unsuccessful' message since it causes too many questions :-)
-
-2005-11-01 Kevin P. Fleming <kpfleming@digium.com>
-
- * stdtime/localtime.c: fix build failure on uClibc systems (issue #5558)
- * devicestate.c: same
-
- * many files: make chan_misdn actually build (issue #5566)
-
- * many files: more Cygwin build system support (issue #4678)
-
- * apps/app_parkandannounce.c (parkandannounce_exec): supply parent channel to ast_request_and_dial so channel variables can be inherited (issue #5564)
- * include/asterisk/channel.h: add parent_channel field
- * channel.c (__ast_request_and_dial): use parent_channel field to inherit variables into new channel
-
- * apps/app_cut.c (cut_internal): use ast_app_separate_args() instead of open code (issue #5560)
-
- * apps/app_mixmonitor.c (launch_monitor_thread): ast_strlen_zero can handle NULL input (issue #5561)
- (mixmonitor_exec): same
-
- * res/res_features.c (ast_feature_request_and_dial): ensure that channel variables are inherited from the channel placing the call (issue #5499)
-
- * utils.c (getloadavg): change to using _BSD_SOURCE as the indicator for whether this function is present or not (issue #5549)
-
- * include/asterisk/utils.h (ast_slinear_saturated_add): force to be inlined whenever possible
- (ast_slinear_saturated_multiply): same
- (ast_slinear_saturated_divide): same
- (inaddrcmp): same
- * include/asterisk/strings.h (ast_strlen_zero): force to be inlined whenever possible
- * include/asterisk/compiler.h (force_inline): add macro to force inlining of functions
-
- * app.c (ast_play_and_record): use ast_silence_generator during recording if requested
- * asterisk.c: add global option to enable silence-during-record (issue #5135)
- * channel.c (silence_generator_alloc): new
- (silence_generator_release): new
- (silence_generator_generate): new
- (ast_channel_start_silence_generator): new API call to start generating silence on a channel
- (ast_channel_stop_silence_generator): parallel call to stop silence generation
- * apps/app_record.c (record_exec): use ast_silence_generator during recording if requested
-
-2005-11-01 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.2.0-beta2 released.
-