core: Fix incorrect variable assignment in switch_channel_set_timestamp (r:8a7f38c6/FSCORE-636)
core: add api_reporting_hook (like api_hangup_hook but after reporting state) both honor session_in_hangup_hook (r:ed7ccc14)
core: only let force_transfer_* vars work when an explicit value was not supplied (r:91a87e9d)
core: Make switch_ivr_session_echo stop when CF_BREAK is set (r:2b120311/DP-19)
core: change channel app_flags to be realm specific and default old version to use __FILE__ as the realm name to avoid cross fire between apps using app flags (r:09c1815c)
core: preanswer before getting variables to avoid crash (r:25fe16df)
core: Windows: Don't report "unknown command" on console when empty command has been given (r:c8f9fb56/FSCORE-641)
core: Fix SQLLEN to prevent queue buffer overrun (r:68d1c32a/FS-2149)
core: add origination_caller_profile to log all attempted calls for a paticular leg (r:977a8ad7)
core: Add attribute "path" to autoload_configs/modules.conf.xml <load> entry. (r:1a75821d)
core: add tone2wav (r:6f2c455f)
core: add speed boost to sql thread (r:ef79535c)
core: reverse the linked list in ivr menus to support accidental feature of multiple entries for the same keys (r:d4a01324)
core: Add time of day string compare function switch_tod_cmp. It usable in XML dialplan with time-of-day. String format is hh:mm:ss you can define a range like this : 09:00-17:00 (Second are not optional) (r:4ab8fa13)
core: Add date time range string compare function switch_fulldate_cmp. It usable in XML dialplan with date-time. String format example : 2009-10-10 14:33:22~2009-11-10 17:32:31. (r:c9fcce08)
core: Add day of week 3 letter initial usage in "wday" field in the dialplan. Example : mon-fri. Using number as before is still supported. Several public switch function are available. (r:59ec8ced)
core: set conditionals to only fire when the mutex can be obtained (r:07ec7867)
core: avoid segfault when sofia tries to update the callee id at the same time as the outbound call is transferred (r:df63657e)
core: make code more automagic to shut up the dude on the list (r:d093a4a4)
core: Fix memory leak if we fail to enqueue new event to EVENT_QUEUE in switch_event.c (r:ef773e07/FS-2148)
core: Fix timeout while bridge is waiting for CF_BRIDGED flag (r:2572621b/FS-2368)
core: don't parse events for b legs from a leg thread in case they are using a monolothic python script as a group_confirm exec over socket to send it messages while the call is ringing (r:ed5266d3)
core: add new function to check when messages need parsing to improve performance on parsing messages during originate (r:8b0421ff)
core: run execute_on_answer on_media _on_ring apps async (r:ef4a4ed0)
core: add switch_ivr_insert_file to insert one file into another at an arbitrary sample point (r:82394b37)
core: Slow reload cause calls to hang (r:1ba98b02/FS-2852)
core: Application intercept causes FS to stop processing calls (r:12fc65f7/FS-2872)
core: fix edge cases for endless loop in sql thread (r:5d7c09ed)
core: prevent race while changing codecs mid call (r:7aa72b67)
core: prevent race on execute_on_answer called from the B-leg of a call (r:751e0291)
core: drop rtp frame that was already replaced with a cng frame (r:34a0ca50)
core: fix partial match counting as exact match in dmachine (r:5eb951aa)
core: try to adjust the timer to be ok with the horrible 10000 microsecond kernel resolution on amazon ec3 but that doesn't mean it's not horribly wrong to run the kernel that slow (r:903b2901)
core: make exact matches return sooner in dmachine (r:e897646e)
core: don't let inherit_codec work when we have ep_codec_string set and the B-leg codec is not in that list since it will lead to failure (r:f79f9766)
core: set maximum query run time to 30 seconds at least on drivers that support SQL_ATTR_QUERY_TIMEOUT (r:5bb525e1)
core: dd switch_cache_db_affected_rows() to switch_core_sqldb (and switch_odbc) and expose it through Lua dbh:affected_rows() (r:d79cf484/FS-2962)
core: add bind meta on A-D and refactor (r:27869d7a)
core: add temp_hold_music var that is only valid until you transfer the call and finishing touches on bind meta to A-D (r:b262f44c)
core: add function to help set session read codec to slinear (r:1a08df9b)
core: add rtp_bug IGNORE_DTMF_DURATION to speed up dtmf detection of RFC2833 on strange carriers (r:b3fc001e)
core: Fix crash when re-connecting to non-working database server (r:29daaa07/FS-2960)
core: treat EINTR returns as a BREAK (now mapped to SWITCH_STATUS_INTR), we appriciate the interrupted syscalls but we would like to continue working properly (r:316963c5)
core: eat rtp frames with the wrong payload type number (r:fe1711fd)
core: up assert vaule on header loop detection to 1 meeeeelyonne for hmmhesegs (r:d9c56345)
core: Fix race condition in originate where USER_BUSY is reported as a no answer (r:cc06fdb5/FS-2992)
core: Allow check ip change to be optional (r:1cf79386/FS-2917)
core: handle 2833 in do_flush instead of dropping valid dtmf (r:3fa3e11c/FS-3002)
core: add record_restart_time_limit_on_dtmf var (r:7a1dcb69)
core: fix unreachable condition with a null args to make any key stop playback/record etc without dequing and remove hard-coded flush digits in play_and_get_digits be sure to flush it yourself before using (r:976859bb)
core: Fix a lock on reloadxml when stderr write is blocked. Also remove an error parsing print since reason generated were wrong and duplicate. (r:2d6161e8)
core: fix || where it should be or in sql stmt that may cause stray records in the calls table
core: Add CS_NONE and correct variable name (r:3fd7b8f2)
core: Fix SQL issue (r:04bb74fc/FS-3050,FS-3051)
core: fix race with media bug exiting from write frame while read frame is trying to use it (r:1341a75a)
core: fix regression in rtp stack trying to avoid broken clients who send the wrong payload type, we were eating the stun packets in jingle calls (r:0bce777a)
core: Add capability to specify core-db-name in switch.conf.xml to have sqlite in a different location. This is important for everyone with relatively 'high' sip registration since the addition of sip registration to the core require sqlite db to be moved to a faster location (Ramdisk for example). Useful for everyone who moved their sqlite db for sofia to ramdisk because of performance issue. (r:500e9acd)
core: Index column name wrong on table registrations. (This wont create the index for people who already have the table) (r:1096e673)
core: allow uuid bridge on unaswered channels as long as there is media available on at least one (r:4f93ea25)
core: add multiple-registrations flag to the core similar to mod_sofia (r:b36a7c0b)
core: tolerate dtmf where there is no start packet and only end packets (r:097caed4)
core: Fix RTP auto flush during bridge w/Windows causing packet loss (r:f8d326de/FS-3057)
core: possible ill placed assert() before a NULL check in soa_static.c (r:91a5e776/FS-2803)
core: prevent crash on double call to asr_close (r:4f5ca9e8/FS-3077)
core: fix bug in switch_itodtmf (r:b53a6848)
core: use strdup instead of core_session_strdup in hangup hook (r:3a10d6a1)
core: fix jb + no timer situations (r:61d3c56f)
core: Add events PLAYBACK_START and PLAYBACK_STOP, plus some minor improvments for RECORD_STOP (r:bc397ab6/FS-2971)
core: Fix Freeswitch crash on Debian ARM (r:a80fae92/FS-3126)
core: switch_xml: reloadxml will(should) never lock again. It will load the XML structure into a new XML structure, and just replace the currently available ROOT XML. It then the job of the last user of the switch_xml structure to free it. (r:471bd6df)
core: switch_xml: Remove commented out mmap. With the changes in the past 2 year, mmap can't really be put back in it current state. (r:34bd0e5e)
core: Fix jitterbuffer with SRTP enabled (r:069f5f7d/FS-3075)
core: this will remove the reported symptom but does not change the fact that 1khz resolution is ideal for proper performance (r:5f18ec94/FS-3168)
core: this was specific to the user channel which is not a real channel in every sense of the word as it has no running thread or any usable state changes so this new line of code in 233d3164be4412aaaf8f9f42d8042e48279a018a to wait for the state machine to stabilize before returning from originate caused an issue with user/ channels (r:88a6ac2f/FS-3170)
core: this also fixes the incorrect usage of L16 on payload 10 which may or may not break interop with other sip devices if we do it right. also added rtp_disable_byteswap variable that can be set to false to disable byteswap when a device is encountered that is incompat (including all previous version of FS up till now) (r:e657e32f/FS-3172)
core: dont calibrate clock when timerfd enabled (r:26f5ebd4)
core: fix DTMF in SRTP/ZRTP (r:fd608901/FS-3165)
core: add switch_atomic_* type and functions switch_apr.c and switch_apr.h (r:3b56c119/FS-3173)
core: improve some defaults to tune performance if you use -heavy_timer, try not using it (r:5d783134)
core: Fix api_hangup_hook with no args (r:484a397d/FS-3194)
core: allow 100 microsecond tolerance on timer loop (r:6388e03d)
core: Fix X-PREPROCESS exec to wait pid (r:dae2cb4a)
core: Ability to use mod_say with native files; native is a special case so use the extension native e.g. en.native (r:3a2e1d03/FS-3176)
core: Fix: Bridging a call to multiple legs and using leg_delay_start, legs that lost the race before the leg_delay_start time is up still get originated for a brief moment (r:c5daf80e/FS-3218)
core: add switch_clean_name_string util function to strip out caller id name chars that can cause issues (r:244048f8)
core: switch_core_sqldb - clear pointer on release (r:aaef33cc)
core: all [] {} and <> can be stacked and override the delim per set <><^^:>{}{^^:}{^^;}[][^^:] (r:4c4bf59e/FS-3246)
core: fix default tipping point it was too low (r:e4eade33)
core: enable optimal defaults on linux kernels that can support newer features. (r:0b51aca3)
core: Lower NAT port mapping disabled log msg from WARNING to INFO (r:973a850d)
core: Change the structure of the phrases/language system. Previously it was fxml->phrases->macros->language->macro. Changed it so fxml->languages->language->phrases->macros->macro You can have sub macros <macros name="voicemail"><macro ...> and allow you to call it login@voicemail. Change the sound-path to sound-prefix to make it constistant with the rest of freeswitch. Also allow to set a sound-prefix to a macros, so you can override it for a specific file set. You can set say-modules="en" or whatever in the <language section to define that say module to use. (r:4137b360)
core: add bridged timestamp and hangup_complete_with_xml=true to add xml_cdr to the body of hangup_complete events (r:bd471fc6)
core: Modify freeswitch to use a configurable switchname instead of a hostname (r:00b53a91/FS-3277)
core: add option to disable srtp with --disable-srtp (r:a6b336e4)
core: record_session: Will auto create recursive destination folder if it doesn't already exist (Doesn't create folder when used with local cache feature) (r:c4b78a49)
core: fix edge case between fail_on_single_reject and group_confirm (r:fae95433/FS-3303)
core: add prefix chars to playback_terminators + means include the term in the string and x means include the char and return SWITCH_STATUS_RESTART eg #+* only includes the * if you type it but not the # (r:38b3f43d)
core: add additional format YYYYMMDDHHMMSS to strepoch (r:38f06a3b)
core: fire SWITCH_EVENT_RECORD_STOP after closing file (r:94e9957e/FS-3311)
core: add arrays to event headers and chanvars (r:c1c75952)
core: allow -1 as silence generation divisor to specify only zeroes silence (r:294a57fb)
core: Don't send silence frames for parked calls until media is ready. (r:dc028b36/FS-3046)
core: add code to pass recording bugs on to other legs when executing an attended transfer, needs testing and possible follup commits before using (r:e2da3bea)
core: flip_record_on_hold to make the recording flip to the other leg on hold, record_check_bridge to make recording the same file on the opposite leg of a bridge considered a duplicate attempt and record_toggle_on_repeat to make repeat recording the same file toggle the recording off (r:7bbbb9cc)
core: lower log-level of failed ivr_originate for mundane conditions like no answer and attended transfer (r:f25085e0)
core: add scoped channel variables (%[var=val,var2=val2] blocks valid in any app data field and will only last for that one app execution) (r:b2c3199f)
core: enable recursion for scoped variables so applications that exec more apps will preserve the scope, the most recent app will mask variables just during the duration of that app (r:c6268da5)
core: only clear scope vars when they were set (r:d4fcba74,r:77688084)
core: Add the ability to issue a break to switch_ivr_sleep when media is not ready, allowing continuation of processing of the dialplan. (r:dfc30b2e/FS-3373)
core: parse events and messages in channel_ready (r:94148095)
core: add last_hold_time and hold_accum vars for cdr data (r:676ef808)
core: avoid recursion loop in parse_all_events vs channel_ready (r:22d89943)
core: auto populate global origination_caller_id_name/number from effective_caller_id_name/number in enterprise originate (r:f8c029a1)
core: add --enable-timerfd-wrapper to wrap timefd syscalls for platforms with the right kernel and wrong libc (r:306b332d)
core: don't parse events in channel_ready during hold (r:cad68d53)
core: only parse messages from channel_ready when its a session calling channel ready on itself not when another thread calls it (r:1d12519d)
core: Fix single quote stripping and add %y to turn ' into \' (r:3b5a0ae5/FS-3359)
core: push out signal data into its own queue system (r:f1ee225c)
core: When in a dialplan hunt and we have a custom caller_profile, ${destination_number} and other variable kept the previous value of the original dialplan parsing. This correct this so it take the custom created caller_profile for that hunt (r:b0e0dd22)
core: pause traffic if sql_queue gets to big (r:2939262e)
core: fix detection of tones in monitor_early_media_fail (r:3cbae3fb/FS-3413)
core: use rwlock for global vars to reduce contention (r:0521886d)
core: Fix separate_string_blank_delim to handle strings with '&' (r:f3a42258/FS-3099)
core: Fix setting display on wrong channel on eavesdrop (r:3dc4b530)
core: add new detailed_calls view a version of the channels table that shows only one legged calls or bridged calls (r:beecd937)
libesl: added python eslmod installation to esl Makefiles (r:b83a30ca)
libesl: null terminate buffer after reading from the socket to prevent cross-over to old data that confuses the parser and throws off framing (r:e8a10558/ESL-56)
libesl: add optional job-uuid param to bgapi in oop mod (r:e96acac3)
libesl: fix linger support in esl client lib (r:0444626b)
libsofiasip: set minimum initital sip t1 timer to 1000ms to work around race condition on retry timer firing before all the things that are supposed to be handled by the timer are set. The base resolution on this timer is 500ms, so doubling up makes sure we always hit the initial retry timer on the next run, where everything should be set. The side effect was, 1/2 the time on a request that did not get immediate response, the timer would be fired and cleared, but the action (sending retry) was never done, and a new timer was not set, causing the request to just sit zombied and never retry. A better solution would be to find and correct the race condition so the timer is never set to early and we never hit this condition. (r:20c2740c)
libsofiasip: Fix for issue reported on the mailing list with a Chinese locale and windows. This commit removes a hidden char that should not have been there anyway. (r:7adaceb8)
libsofiasip: resolve edge case in the 3rd party sofia sip stack library when dealing with a malformed contact and missing ack. Will push upstream to sofia devs (r:d68605f5/FS-3394)
libsofiasip: use individual pools instead of sub-pools for nua handles to avoid pool swell (r:f7612413)
libsofiasip: Fix segfault in sofia's stun code (r:7403db70)
libsofiasip: add homer capture hooks to libsofia (r:3e029f0d)
libspandsp: Fixed a typo in spandsp's msvc/inttypes.h Updated sig_tone processing in spandsp to the latest, to allow moy to proceed with his signaling work.
libspandsp: removed a saturate16 from spandsp that was causing problems fixed a typo in the MSVC inttypes.h file for spandsp
libspandsp: Changes to the signaling tone detector to detect concurrent 2400Hz + 2600Hz tones. This passes voice immunity and other key tests, but it bounces a bit when transitions like 2400 -> 2400+2600 -> 2600 occur. Transitions between tone off and tone on are clean. (r:bc13e944)
libspandsp: Fix for T.30 processing of operator interrupts, to improve compatibility with some machines, which seem to send them when no operator is around. (r:84ee0ae6)
libspandsp: Fixed a vulnerability in T.4 and T.6 processing which is similar to http://bugzilla.maptools.org/show_bug.cgi?id=2297 in libtiff. A really screwed up 2D T.4 image, or a maliciously constructed T.4 2D or T.6 image should potential run off the end of an image decoder buffer. (r:c6f67322)
libspandsp: Changed T.38 terminal handling, so errors from the user's packet transmit routine properly filter up the chain, cause termination of the FAX session, and are reported to the caller. (r:c890fbfa)
libspandsp: Numerous little changes to spandsp that haven't been pushed to Freeswitch for a while. The only big changes are a majorly rewritten V.42 and V.42bis which are now basically functional. (r:d30e82e2)
libspandsp: Another round of tweaks for spandsp. There should be no functional changes, although quite a few things have changed in the test suite (r:4a7bbf4e)
mod_callcenter: Initial commit of the mod_callcenter application. This module is in it early state of developpement. You can see documentation on the wiki at : <a href="http://wiki.freeswitch.org/wiki/Mod_callcenter">http://wiki.freeswitch.org/wiki/Mod_callcenter</a> For support/comments, please use <a href="http://jira.freeswitch.org/">http://jira.freeswitch.org/</a> and select the MOD CALLCENTER module. (r:ba09b96d)
mod_callcenter: Add ability to unload/reload/load a queue setting (You still need to reloadxml before). Note that joining a queue will check for it in the config and load it on the fly... I've used the same system as in mod_voicemail. Not sure if we should allow this, but just comment it out of the config before unload and it wont be available anymore (r:3eafca60)
mod_callcenter: Try to fix the ring-all, also add cli auto complete done in previous commit (r:1666783c)
mod_callcenter: Add missing odbc db support (Not tested, please someone test this) (r:42436e27)
mod_callcenter: More ODBC changes. It is not a global settings value. Cannot be changed in runtime. (r:6980305f)
mod_callcenter: Added value busy_delay_time and reject_delay_time so we can wait if those 2 occur (Un registred phone are considered as busy). Add a ready_time epoch time when we can contact an again again, fix ring-all (good this time I hope). (r:8082aa98)
mod_callcenter: Add tiers rules before jumping to a different level. Also added support for dial-in agent. (r:86c9bed7)
mod_callcenter: Default the level to 0 since the new tier system will wait x second at level 1... just level 0 that will ring agent right away (if set to do so) (r:6558276a)
mod_callcenter: You can now allow caller that have hangup before agent answer to call back and resume their previous position. (r:ab2529d4)
mod_callcenter: correct multiple little things following the recent tiers and join back features (r:9b33bd1c)
mod_callcenter: Add more channel variable and event and fix a mem leak (r:2d3d8c8d)
mod_callcenter: Make more sence to bridge the caller to the agent. Before, in the xml_cdr you saw it it like the agent initiated the call to the member (r:0be95658)
mod_callcenter: Added max-wait-time and max-wait-time-with-no-agent param to a queue. (r:3482f95e)
mod_callcenter: Make callcenter_config agent get return the value of the item requested. Also added queue param max-wait-time-with-no-agent-time-reached: If the max-wai-time-with-no-agent is already reached for the queue, then new caller can wait for x amount of second before it kicked out of the queue rather than get rejected automatically. (r:81a03869)
mod_callcenter: Add new event socket agent-offering. Plus some documentation and better handling of bad agent type -- FS-2869 (r:80174cf3/FS-2869)
mod_callcenter: IMPORTANT UPDATE, DTMF during moh created an loop to reactivate MOH but got canceled right away because of pending DTMF in the queue never been cleaned. Could cause masive disk write of debug, and can cause problem to the rest of FS stability. This patch also include basic fundation for DTMF capture support for member waiting. (r:cd1982ce)
mod_callcenter: force loopback_bowout=false on originate. This will need to be reworked, but should fix basic issues call to an agent using loopback (r:2e399b0b)
mod_callcenter: segfault using busy-delay-time parameter (r:c6f044d5/FS-3067)
mod_callcenter: Fix a bug when an caller leave the queue from a BREAK Call (Transfer...), it doesn't think an agent answered. (r:51a531aa)
mod_callcenter: Add new CLI cmd and change some to be more standard. Patch from Francois Delawarde, thanks. (r:30dd1774)
mod_callcenter: >WARNING, some event value got removed< Adding new event time value that can then be used to calculate the Wait;Talk;Total duration of a member were on call. CC-Wait-Time CC-Talk-Time and CC-Total-Time are no longer returned. Visit the code or check the wiki for the updated variable. (r:5f233785)
mod_callcenter: Add better support when agent doesn't answer, including creating a new variable for the delay that is different than reject or busy. Thanks to Francois Delawarde (r:26303c5c)
mod_callcenter: Add better handle of failed agent, member channel getting a break, and debuging info upon leaving. Thanks to Fran?ois Delawarde (with some changes) (r:25cee255)
mod_callcenter: New Agent order Possibility: Agent order by Level and Position by agents.last_offered_call. Change the default and sequentially-by-agent-order strategy to include the longest-idle-agent. This should offer a default consistant way to go through all the agent within the same tier/position. (Before, it was left to the DB to return the order of the result) (r:dcafff20/FS-3158)
mod_callcenter: Generate per member uuid different from the member session uuid. Might fix transfer between queue. More changes are coming (r:b63a72f8)
mod_callcenter: Remove the concept of Caller for Members. Event Socket event have been changed (CC-Caller.* to CC-Member.*) Also CC-Caller-UUID is renamed to CC-Member-Session-UUID. The reason for this is you could actually put people to be call in the queue. So they are not caller per say. But they are a member of a queue. (r:40a134bd)
mod_callcenter: Reload a queue wont delete all the currently waiting members. Only a reload of the module will. (r:c5ae5de0/FS-3250)
mod_callcenter: Add a very prototype (and maybe not functional) strategy called : sequentially-by-next-agent-order. It will try to find the last agent we tried to reach, and start calling more agent after that one based on position. It will use the level for the next agent, but once that level is done, it start back at the lowest level (r:bef6f0f4)
mod_callcenter: New strategies: round-robin, random, and 'top-down' (r:2b4b23aa,r:bee247ca)
mod_cdr_sqlite: Drop transaction BEGIN/END around INSERT. We're only executing one command, and autocommit will automatically rollback if the INSERT fails. Sync state_handlers with mod_cdr_csv. Minor libpq fixups. (r:0f95b870)
mod_celt: Bump celt to 0.10.0 (r:231fbe5e)
mod_celt: update code in mod_celt to match API of 0.10.0 (r:6e4c30ea)
mod_commands: add uuid_fileman <uuid> <cmd>:<val> <-- same vals as the callbacks in js and lua to control the currently playing file of a channel from the cli or ESL (for the people who were ignoring me on the conference call so I decided to implement it instead of try to explain it ) (r:c4369fc8)
mod_commands: FS-2210 Add support for auto completion for uuid_simplify (r:72bcc01b/FS-2210)
mod_commands: add uuid_buglist to fetch the current media-bugs attached to a given session uuid (r:f6eab64c)
mod_commands: add recovery_refresh app and api and use it in mod_conference to send a message to the channel telling it to sync its recovery snapshot (r:650393fb)
mod_commands: fix crash when uuid_break all cannot find bonded uuid channel (r:69e61f76/FS-3468)
mod_commands: fix uuid_dual_transfer for inline dialplan (r:5d84efc3/FS-3403)
mod_commands: update show calls to show both 1 legged calls and bridged calls, also show bridged_calls for previous behaviour of show calls (r:c16c74d9)
mod_commands: Add 'presence_data' field to 'show channels like xxx' list of fields. This makes anthm's trick mentioned on the mailing list even more handy. (r:4872e6ff)
mod_commands: Update tab-complete for show cmd to include bridged_calls, detailed_calls, detailed_bridged_calls and removed distinct_channels (r:2fa8f110)
mod_conference: refactor conference to use switch_ivr_dmachine for the digit parsing controls are now bound to each member separately based on conference_controls channel var, then the caller-controls param in the profile or a default to "default" (r:ac19f73c)
mod_conference: Fix crash on dtmf action (r:4d5389bd/FS-2781)
mod_conference: Add conf_uuid chan var for djbinter (Thanks Math) (r:3c9ee25a)
mod_conference: removes the existing conference transfer function and replaces it using the core transfer it also introduces a new tracking method where the same conference id is reserved for a particular member for the lifetime of the call allowing a user to transfer in and out of conferences and ivr and bridges etc and retain the same member id for the duration of that call (r:246b2195/FS-3095)
mod_conference: prevent race condition on conference join/exit (r:1552ecf5)
mod_conference: I finally tracked this down to the actual recordings generated by mod_conference. This patch delays the recording slightly to allow time for the buffer to fill up, we were riding it so closely that sometimes we would come up short and inject silence into the file to preserve time passing (r:3253bcb3/FS-3147)
mod_conference: wait for channels to come up in paging mode (r:b8063c3d)
mod_dptools: transfer_on_fail note I changed the variable name to auto_cause (r:45edec4c/FS-3193)
mod_dptools: merge file_string into dptools (r:eefdb764)
mod_dptools: change mod_dptools to use the better method of fetching user xml that does not hang onto the xml root (r:e52e44e3)
mod_dptools: the intent for having the module and lang separate is for things where the same module can use different sets of sounds like en module and en-male or en-female lang (sound dirs) there was indeed a disconnect in the dialplan version of this app. Originally say was only available in phrase macros so I change the syntax of the say app so you can specify both the module and the lang absolte from the dp with something like he:he as the module name. (r:44304f49)
mod_dptools: add digit_action_set_target app that can set the target (direction of the dtmf flow and subsequent channel who gets the events) to self or peer (bridged channel when possible) (r:cf9859ea)
mod_dptools: get rid of digit_action_set target and add target,bind_target params to bind_digit_action (r:42b64ccd)
mod_hash: avoid scheduler calling a function on null hash during shutdown (r:8458adeb)
mod_hash: add realm filter to hash_dump db command so that you can quickly dump all entries that belong only to a specific realm without getting the whole db (r:81347126)
mod_h323: Add mod_h323 to windows (r:015bcaf6/MODENDP-301)
mod_h323: move PTrace level set to FSH323EndPoint::Initialise. partially apply patch from from Peter Olsson, Remove UnLock() when TryLock() failed and DEBUG_RTP_PACKETS directiv e. (r:7b5803f7)
mod_khomp: Removed alternative contexts / extensions - New struct for matchs - On calls originated from an FXS branch, the Endpoint searches for a valid extension (digits sent) after the DTMF '#' or after the timeout (option fxs-digit-timeout). That search is done in the context defined in section <fxs-options>, or if no context configured, the search is done in context defined in context-fxs. - Added "dialplan" configuration: Name of the dialplan module in use (default XML) - Group context enabled. If set, the search for a valid extension is done only in that context. - Updated documentation (r:1ef3fc9a)
mod_protovm: This is a very early new prototype voicemail ivr system. You need to copy the sounds.xml and make it loadale in the language folder and protovm.conf.xml inside the autoload_configs folder. Configs file will most definitly change. Once stabilized, we make it install those file by default. (r:fb549777)
mod_sangoma_codec: do not return 0 len frames and return silence instead when there is no transcoding output update stats only when we really receive a frame (r:dc4d19e9)
mod_sangoma_codec: flush sockets on first use (r:bbba1148)
mod_say_en: introduce new say_string method of doing say and use it in mod_say_en as an example. try: eval ${say_string en.gsm en current_date_time pronounced ${strepoch()}} from the cli with this patch. We can do more to centralize the say things and go back and apply it to other langs, using this method you can set the desired file ext as well which I think is a bounty.... (r:d5ef86d7)
mod_say_en: If you only tell SAY CURRENCY to say 100 it should only say 100 dollars without the "0 cents" (r:426a4e76/FS-2922)
mod_say_ru: now support say_string like mod_say_en. Now support channel variables gender,cases can be set in english and russian for example: <action application="set" data="cases=nominativus/> <action application="set" data="gender=male_h"/> <action application="say" data="ru NUMBER PRONOUNCED 1001"/> (r:8b5ecd2f)
mod_skypopen: now answer a call only when directed to do it (before was trying to answer any incoming call). Lot of changes to a messy part, so maybe some problem will come out... (r:45c6c4d3)
mod_skypopen: refining oss driver, removing audio sync during call (was each 20 secs), audio sync at the tcp interfacing with the skype client (reading more than 20ms worth) (r:891015e6)
mod_skypopen: fixed a demented bug (incrementing a variable zeroed in the same loop) maybe responsible for moh sputtering under load on virtual machines (r:43eeeb82)
mod_sofia: allow video negotiation on re-invite (r:be92e5d/SFSIP-211)
mod_sofia: use rfc recommended default session timeout of 30 min according to RFC 4028 4.2 (r:52cd8cdd/MODSOFIA-76)
mod_sofia: add sip_force_audio_fmtp (r:6360264f)
mod_sofia: add sip_copy_multipart to work like sip_copy_custom_headers (r:a291af57)
mod_sofia: Rename sofia_glue_get_user_host to switch_split_user_domain and move to switch_utils. To allow use by other modules. (r:3f7cafd7)
mod_sofia: allow the profile gateway config to set sip_cid_type for each gateway (r:0152706f/BOUNTY-19)
mod_sofia: Adding subject to SEND_MESSAGE (r:2e347c93)
mod_sofia: add multiple rtp-ip support to sofia profiles add extra rtp-ip params to a profile to add more ip which will be used round-robin as new calls progress. (r:22569d4a)
mod_sofia: Add openssl build support to windows - no external build support needed (step 1 - not hooked up yet) vs2008 pro+ only (r:b0de3585/MODSOFIA-92)
mod_sofia: REFER: to-tag and from-tag should be set other way around when other (bridged) channel is incoming. (r:92d324d3/MODSOFIA-91)
mod_sofia: fix 302 to hangup in the two cases where switch_ivr_transfer is used and not in the case when it should carry on and follow the redirect (r:00b51403)
mod_sofia: Remove OPENSSL_USE_APPLINK - not needed (r:437c7805/MODSOFIA-92)
mod_sofia: Send Instant Messages To All Endpoints Registered to Targeted Extension (r:96b790fa/BOUNTY-20)
mod_sofia: increase sps during recovery (r:f1aead31)
mod_sofia: speed up db action in sofia recover (r:8114b3f1)
mod_sofia: Support display updates for Cisco SIP endpoints (tested on SPA series) (r:ac205288/FS-884)
mod_sofia: dont put an rpid in 183 or 200 if pass-callee-id is false (r:86de47ff)
mod_sofia: improve sofia recover in some nat cases (r:4526ba30)
mod_sofia: edge cases for sofia recover (r:646a5609)
mod_sofia: Correct the order what param and variables are overriding them self in user/group/domain (r:5a6f0f5c)
mod_sofia: include accumulated stats from rtcp into vars (r:d5ff3e04)
mod_sofia: make sure hold-related code is skipped 100% with disable-hold set (r:403bf6af)
mod_sofia: make force-subscription-expires only work on nonzero expire deltas, 0 means unscubscribe (r:b7751868)
mod_sofia: presence tweaks and addition of all-reg-options-ping which is like nat-options-ping only for every registered host (r:04b52156)
mod_sofia: If sip_invite_domain is used lets use it for rpid_domain no matter what because I know best if I set it (r:8726104a)
mod_sofia: add inline lists for tab complete db using ::[a:b syntax (r:445731ee)
mod_sofia: add sofia profile <profile> gwlist up|down to list up or downed profiles for feeding into mod distributor to exclude dead gateways (r:0477cb67)
mod_sofia: add 'sofia global siptrace on' so we don't have to always teach people to enable sip trace on each profile (r:09fa6678)
mod_sofia: Support display updates for Cisco SIP endpoints (tested on SPA series) (r:6937ca39/FS-884)
mod_sofia: BLF compliance with RFC-4235: dialog-info 'version=' field is reset to 0 on every new call instead of being incremented (r:589502d3/FS-2747)
mod_sofia: fix parsing of sofia tracelevel param, moved param from profile params to global_settings as its global, and it only worked on reparse before anyways. Please correct any documentation on this issue on the wiki (r:82c4c4cc/FS-523)
mod_sofia: fix nat acl count check to check against the number of nat acls (r:e11550e7/FS-502)
mod_sofia: add sofia_glue_find_parameter_value function to get a specific value from a url params string (r:c701d41c)
mod_sofia: adjust sql stmts in presence to allow even non-registered entities to be tracked (r:4e0399d0)
mod_sofia: dont update display to ring when call is hungup in pidf presence (r:36851a90)
mod_sofia: 1) Add force-publish-expires to set custom presence update expires delta (-1 means endless) 2) Check how many users are registered when receiving a PUBLISH AND Multiple Registrations is enabled: if there is more than just 1 AND you are sending a offline message: skip publishing it to everyone to prevent clients from thinking themselves has gone offline. (r:fd1736b3)
mod_sofia: fix race in codec failure condition, then fix bug in sdp parsing (likely a regression from recent codec changes) to never have the problem in the first place so you are double-protected (r:19325c43)
mod_sofia: fix mem leak (r:1970ec1d/FS-2810)
mod_sofia: parse static route in sip uri in notify by event (r:35676e7e)
mod_sofia: add support for NDLB-force-rport=safe param that does force-rport behavior only on endpoints we know are safe to do so on. This is a dirty hack to try to work with certain endpoints behind sonicwall which does not use the same port when it does nat, when the devices do not support rport, while not breaking devices that acutally use different ports that force-rport will break (r:fc4d290c)
mod_sofia: add separate reg timeout from retry sec (r:e5b891ee)
mod_sofia: fix display of timeout (r:2043d5a)
mod_sofia: fix missing name and potential segfault in gateway status (r:40ac860a)
mod_sofia: Add missing RTP info for early SDP in bypass media (r:10119e9e/FS-2824)
mod_sofia: add manual_rtp_bugs to profile and chan var and 3 new RTP bugs SEND_LINEAR_TIMESTAMPS|START_SEQ_AT_ZERO|NEVER_SEND_MARKER (r:b278dd23)
mod_sofia: apparently some sip device vendors did not read the RFC (who knew?) adding verbose_sdp=true var to add needless a= lines for standard iana codecs that explicitly do not require them (r:6c4f49a8)
mod_sofia: Fix registering a gateway, sofia always places a Via header with ext-sip-ip, even if this gateway is local (r:cf398e1a/FS-535)
mod_sofia: add presence-probe-on-register sofia param to send a probe on register instead of presence to deal with some broken phones and add some general improvements to allow multi homed presence (r:14394994)
mod_sofia: Fix issue when fs_path is used so we pick the correct media IP in our outbound invite this was soemthing that wouldn't work correctly over ATT on the iphone. (r:a669f76f)
mod_sofia: Add reuse-connections sofia profile param to allow users to turn off TPTAG_REUSE, thus not re-using TCP connections (r:98ed05cc)
mod_sofia: Make sofia recover also work on custom uuid (r:3a645dee/FS-2913)
mod_sofia: remove check for va_list completely in sofia since i don't even think it happens ever (r:dfecc914)
mod_sofia: have mod_sofia always elect to be the session refresher so we know it will work, also make the session-expires set to 0 imply 100% disabled session timers (r:321013ef)
mod_sofia: Do not set nat mode when the device's network_ip is within the acl also so if your FS is behind nat and your phone is too then it will still make the right decisions (r:6c6eab8c)
mod_sofia: refactor sofia_contact to try the profile_name first then the domain to resolve the profile then fall back to querying every profile to reduce confusion with multi-homers (d'oh) also special profile name * will force a search-all situation (r:81608da0)
mod_sofia: support allowing pidf-ful presence clients to share the same account and 'appear offline' without influencing each other =/ also refactor the contact generation string based on nat into a helper function (r:97a68c50)
mod_sofia: gateway not identified when extension-in-contact is set (r:7b289941/FS-502)
mod_sofia: Fix erroneous error log on SQL statement (r:2c595a6c/FS-2961)
mod_sofia: Fix routing behavior of inbound calls from gateways that only match gateway based on the gw request uri param (r:0132cd3f)
mod_sofia: don't say we are not for polycom phones (safe rport) when its not really nat (r:9462f53c)
mod_sofia: Set route header as a var on channel like Diversion header (r:d41e6498)
mod_sofia: fix seg related to ptime mismatch + CNG + PLC (if you ever get purple ptime mismatch warnings you want this patch) (r:54de293b)
mod_sofia: be more iOS friendly when using TCP or TLS because the phone never sleeps thus drains the battery (r:159ae989)
mod_sofia: add send-presence-on-register (true|false|first-only) param to sofia and api command sofia global debug [presence|sla|none]
mod_sofia: disable media timeout when encountering a recvonly stream (r:029d68ce)
mod_sofia: fix sofia flush_inbound_reg to work when @domain is given (r:68bf642c)
mod_sofia: fix session timer failure when freeswitch is generating the sdp and there are enough dynamic codecs enabled to conflict with the 2833 pt (4 by default) (r:018a3800)
mod_sofia: Places ;fs_path= within the contact string <...> when using NDLB-connectile-dysfunction-2.0, instead of just appending to the end of the contact string. (r:afc02747/FS-2989)
mod_sofia: Fix handling SUBSCRIBE to "park+" fifo, the NOTIFY data was not being generated from mod_fifo data. (r:3dd9d5c0/FS-3007)
mod_sofia: only pass publish on when you have a subscription (r:85913b70)
mod_sofia: sip_codec_negotiation to override inbound-codec-negotiation setting (r:74a0cfd1/FS-3027)
mod_sofia: fix uuid_jitterbuffer edge case debugging a non-existant jb causing a seg (r:88d410d3)
mod_sofia: tell rtp stack about what remote payload type to expect when the receiving end follows the stupid SHOULD as WONT and sends a different dynamic payload number than the one in the offer (r:c565501f)
mod_sofia: rip off the fs_ args on message like we do in SEND_MESSAGE (r:b7fd81de)
mod_sofia: use the correct URI on endpoints behind nat (r:5f2857b8)
mod_sofia: put transport in the request uri on outbound registers if register_transport is set and proxy does not already contain a transport param (r:4b62ff79)
mod_sofia: pass custom headers backwards over sofia (r:13dc6058)
mod_sofia: fix profile SIP INFO dtmf not working (r:4c4ca08d)
mod_sofia: Fix SIP INFO DTMF (r:39ff78bf/FS-3078)
mod_sofia: contact-params should not be set if the string is empty (r:06988e1a/FS-3084)
mod_sofia: segfault with sofia_contact when invalid parameters are given (r:4e60f14a/FS-3072)
mod_sofia: Fix minupnpd nat_map updated IP not getting set in SIP profiles (r:e7acd4d1/FS-3054)
mod_sofia: add sip_execute_on_image variable similar to execute_on_answer etc so you can run t38_gateway or rxfax etc when you get a T.38 re-invite but no CNG tone or you want to ignore the tone and only react when getting a T.38 re-invite (r:53fc3f7f)
mod_sofia: add sip_jitter_buffer_during_bridge which you can set to true to keep a jitter buffer on both ends of the call when you are NormT (r:01073a79)
mod_sofia: fix race condition in sofia recover for atom processors (r:3eeb4995/FS-3117)
mod_sofia: improve codec ordering in ep_codec_string (r:8fe24a29/FS-3121)
mod_sofia: Send BYE to endpoints that lose race even if they answered (r:8c3651fa/FS-640)
mod_sofia: do not renegotiate codecs on hold re-invites (r:bfd0ba97)
mod_sofia: add rtp-notimer-during-bridge (alternative to rtp-autoflush-during-bridge (r:2a35dfb5)
mod_sofia: send another presence event on calls that were cancelled from LOSE_RACE to fix winnable race in Broadsoft SCA (r:59f6654e)
mod_sofia: pass header in X-FS headers on attended transfer CID update to indicate specific situation to flip callee/caller id when targeting a 1 legged call (r:24a97292)
mod_sofia: change text of error message to be more descriptive (r:4c435ec5)
mod_sofia: Correct a problem where restarting profile would cause some profile hash entry to remain. (r:81bfe435)
mod_sofia: New Sofia API to look up the username of a given user (r:7556ec57/FS-3187)
mod_sofia: sip_authentication was not cleared after nonce expired -caused sofia_reg_internal.db grow bigger and bigger with time (r:c735e28a/FS-3190)
mod_sofia: pass failure across in T.38 passthru mode (r:31273b42)
mod_sofia: auto-aleg-full and auto-aleg-domain for from_domain field in gateway (r:fda2283b)
mod_sofia: After further review I can concede the point that we should always say partial considering how we do things. With this commit we should at least be sending separate partial updates for each existing dialog to everyone with a subscription. If we need to introduce more data, consolidate them etc. We need to do it in small chunks and keep things sane. (r:7eae7f37/FS-2877)
mod_sofia: Fix:Attended transfer with bypass media fails in various ways (r:4b706dac/FS-3227)
mod_sofia: SO, If the RFC told you to jump off a cliff......? (r:07b9186d/FS-3226)
mod_sofia: Fix TLS crash when NAT configured w/o actual external IP addr (r:64f8ad3f/FS-3324)
mod_sofia: only accept info dtmf when its configured to (r:51c21580)
mod_sofia: add support for 3pcc-proxy when in bypass media. (r:68c389df/FS-3326)
mod_sofia: release rwlock on error (r:0675b59b/FS-3321)
mod_sofia: Mask remote party identity in SIP presence if channel var presence_privacy=true (r:8d8e5a23)
mod_sofia: add check_sync to sofia cli (like flush_inbound_reg without the unreg) (r:079f4845)
mod_sofia: pop ::<profile_name> off the domain name in mwi events to hint at the profile (r:e2ed8c08)
mod_sofia: dig into the database to figure out what profile to send mwi on when they are not willing to alias the domain to the profile =/ (r:b14340a5)
mod_sofia: add mutex around gateway access on per-profile basis and token based access to global profiles to prevent hanging on to the hash mutex while doing sql stmts which may cause issues/slowdowns (r:9df8169d)
mod_sofia: add parallelism to sofia by offsetting sip messages to the concerned sessions and using multiple queue threads for message handling (r:fb68746e)
mod_sofia: removed the vid refresh thing (r:49e52b4c/FS-3362)
mod_sofia: add sip_liberal_dtmf chanvar and liberal-dtmf profile param to use the maximum methods of DTMF avoiding sticking to the spec which leads to incompatability (r:bc7cb400)
mod_sofia: support final response in response header passing (r:acd0898e)
mod_sofia: Fix failure to fall back to g.711 when t.38 attempt fails (r:07a79752/FS-3214)
mod_sofia: pop ::<profile_name> off the domain name in mwi events to hint at the profile (r:e2ed8c08)
mod_sofia: dig into the database to figure out what profile to send mwi on when they are not willing to alais the domain to the profile =/ (r:b14340a5)
mod_sofia: Fix 3pcc codec negotiation issue with bypass_media (r:c5a2275f/FS-3340)
mod_sofia: add mutex around gateway access on per-profile basis and token based access to global profiles to prevent hanging on to the hash mutex while doing sql stmts which may cause issues/slowdowns (r:9df8169d)
mod_sofia: add parallelism to sofia by offsetting sip messages to the concerned sessions and using multiple queue threads for message handling (r:fb68746e)
mod_sofia: Fix sofia hang on shutdown (r:3be64cbf/FS-3354)
mod_sofia: remove vid refresh from SDP on declined video connection (r:49e52b4c/FS-3362)
mod_sofia: fix small mem leak in sofia (r:6f62f391/FS-3386)
mod_sofia: add proxy tag to UPDATE packets if it was set by INVITE (r:e6605139)
mod_sofia: resolve attended transfers, it fails to parse the Replaces when encoded (r:d9bbf129/FS-3304)
mod_sofia: if user has set presence_id, don't override it (r:7cdc8342)
mod_sofia: only list real profiles not aliases in presence code (r:f9969f38)
mod_sofia: Fix 200 OK not passed for Callee-Initiated ReInvite for T.38 (r:b2299035/FS-3421)
mod_sofia: destroy nh if SIP transaction terminated by a 488 (r:a0cec8ab/FS-3444)
mod_sofia: use register contact to determine proper contact in 200 ok response to register (r:f9612fec)
mod_sofia: add NDLB-allow-nondup-sdp to indicate you want to parse a differnt sdp in 200 ok from 1xx (previous default) this is a RFC violation so I decided not to support it by default anymore. Enable this if you want that broken behaviour (r:3f489a2a)
mod_sofia: add homer capture hooks to mod_sofia (r:98473085)
mod_sofia: sdp_m_per_ptime is now implied to be true, if you don't like this set it to false but its going to be undefined behaviour. This basically means if you call in with ptime 30 then you have a bunch of ptime 20 codecs in your outbound list that there will be one m= line with 30 and the original inbound codec and more m= lines for each discinct ptime in your list. This is, of course, will depend on disable_trancoding or absolute_codec_string as well (r:56d67ead)
mod_sofia: filter re-transmission of extra SIP headers (r:9e399c19/FS-3439)
mod_sofia: Fix caller ID name on bridged appearance (r:7efa4fb2/FS-3532)
mod_sofia: speed up restart speed of profiles (r:fb5f29c2)
mod_sofia: this is actually compliant when mixing ptimes in the same sdp but since iLBC uses its own fmtp for ptime I will add this patch to make it beleive its 20 for the sake of arguement. If you have any other problems with this, set the channel or global variable sdp_m_per_ptime=false to completely disable the default correct behaviour (r:247537a9/FS-3545)
mod_sofia: I missed a few more spots to hack in the exception for iLBC, (thanks for marring my code iLBC ppl) it should work as expected now even with the m_per_ptime on (r:83a78fbf/FS-3545)
mod_sofia: don't turn X-FS- headers into variables, they are reserved for FS specific communication and should not be passed on (r:7d399cce)
mod_sofia: This patch will probably make it work but the bug is actually in the phone, the patch is simply tolerating the bad behaviour. You are correct about the a=sendonly missing, this was fixed in a later revision of the polycom firmware. I suggest that even if this patch works, that you update your phones to a newer firmware, preferably the most recent. (r:7acddfac/FS-3549)
mod_sofia: add auth username to unreg event (r:1b9b3456)
mod_sofia: fix wrong media ip in recover data issue (r:5154b881)
mod_spandsp: initial checkin of mod_fax/mod_voipcodecs merge into mod_spandsp (r:fa9a59a8)
mod_spandsp: rework of new mod_spandsp to have functions broken up into different c files (r:65400642)
mod_spandsp: improve duplicate digit detection and add 'min_dup_digit_spacing_ms' channel variable for use with the dtmf detector (r:eab4f246/FSMOD-45)
mod_spandsp: add start_tone_detect/stop_tone_detect app and api commands for tone and cadence detection (r:a6e65147/MODAPP-378)
mod_spandsp: Fix mod_spandsp receive t38 fax error in windows7 (r:fca93f29/MODAPP-443)
mod_spandsp: Moved spandsp to a more recent version. A huge number of little changes occur here, as recently spandsp lost all the $Id$ entries the source files had for the dark old days of CVS (r:f029f7ef)
mod_spandsp: new option to set sip_execute_on_image to 't38_gateway self nocng' this should skip the tone detection adn go right into the gateway mode so you should be able to do only this and have it work based on remote re-invite (r:9227b538/FS-3252)
mod_spandsp: additional fix to this bug and add better fax detect code to mod_spandsp (r:7fe313cf/FS-3252)
mod_timer_fd: external timerfd module by Timo Ter?s (r:48b11935)
mod_timer_fd: add timerfd support to the core for now you must enable it in switch.conf.xml with the param enable-softtimer-timerfd=true later if it proves to work well we can make it on by default, please test if you have a new kernel that supports this option kernel >= 2.6.25 and libc >= 2.8 (r:10174ea6)
mod_voicemail: Allow to forward a message or send it via email key during the playback of the recording, not just when the menu is playing. (r:83aeda79)
mod_voicemail: fix vm_inject to a group and change syntax for sending to a whole domain to domain= for clarity sake (r:f30a1cc6)
mod_voicemail: add quotes to vm_cc command generated internally to escape spaces in the caller id name (r:5f012813)
mod_voicemail: Implement 10 new standard api function call that allow you to control fs voicemail storage system. The goal is to have a standard API set for any additional storage system we wish the voicemail to run off. Current list of added api name are : vm_fsdb_msg_count, vm_fsdb_msg_list, vm_fsdb_msg_get, vm_fsdb_msg_delete, vm_fsdb_msg_undelete, vm_fsdb_msg_purge, vm_fsdb_msg_save, vm_fsdb_pref_greeting_set, vm_fsdb_pref_recname_set, vm_fsdb_pref_password_set. (r:1f4cb488)
mod_voicemail: Adding a new voicemail fsdb api vm_fsdb_auth_login that does basic login authentication for a user (r:bfdfac5e)
mod_voicemail: Fix vm to email dial 8 option (r:8592b6d9/FS-3382)
mod_voicemail: Add 2 new profile settings, db-password-override and allow-empty-password-auth. By default, they have value of their previous behavior. If db-password-override=true, the db password will only be used if present, if not present fallback to the xml config file vm-password. If allow-empty-password-auth=false, it will disable login via a authentication method if there is no password set in the user account (This wont affect voicemail_authorize=true login). (r:a9db642a)
mod_voicemail: use vm_email as notification address if vm_notify_email isn't set (that behavior was in voicemail_leave_main but not in deliver_vm) (r:8974f9d6)
mod_voicemail: better fix for voicemail email key match (r:aff4bcbe/FS-3080)
sofia-sip: extend timeout for session expires on short timeouts to be 90% of timeout instead of 1/3 to handle devices that do not refresh in time such as polycom (r:a7f48928/SFSIP-212)
core: re-factor mod_event_socket so it uses switch_log functions to duplicate and free log nodes while it uses them internally (fix bad ptr for userdata on event socket listeners) (r:14598)
core: add recursive flags and workaround for nested broadcast in controlled situations (r:14644)
core: add origination_cancel_key variable for a dtmf key that can abort an originating call (r:14645)
core: add state change hooks for destroy (r:14647)
core: fix reaction in att_xfer when call is cancelled or times out (r:14650)
core: fix timeout while bridge is waiting for CF_BRIDGED flag (FSCORE-424/r:14702)
core: set the hostname core variable in switch_core_init (r:14726)
core: add more stuff to event in hangup hook api (r:14740)
core: add graceful zrtp failure (r:14745)
core: reduce poll timeout when dtmf is present (r:14749)
core: fix start_dtmf muting input from portaudio (FSCORE-426/r:14755)
core: fix missing reset causing the same timestamp forever on perfect storm of conditions involving transcoding and ptime combo (gotta love it) (r:14771)
core: CoreSession::originate: set the uuid on success (r:14773)
core: add optional prefix arg to set_user (FSCORE-429/r:14789)
core: try to improve autoflush and other silly audio glitches from edge cases and help (FSCORE-416/r:14800)
core: Add min_dtmf_duration setting to FS core and modified default_dtmf_duration in rfc2833 queuing function to not be used as min (FSCORE-442/r:14893)
core: improve behavior of leg_delay_start, leg_timeout, leg_progress_timeout (FSCORE-439/r:15027)
core: Fix spinning threads that receive 183 on bridge/originate with bypass_media set (FSCORE-452/r:15028)
core: Check handling of SIGINT (FSCORE-456/r:15051)
core: add bridge_answer_timeout variable, a timeout in seconds how long to tolerate a bridge that is in early media without being answered (can be set on either leg) (r:15057)
core: Enable auto update displays in more places (r:15110)
core: fix file handle mem leak (thanks pressuerman) (r:15114)
core: re-factor how record_answer_req=true works, add media_bug_answer_req=true variable and backport record_answer_req=true to use it (IRQ-00/r:15235)
core: send callee id info as caller id info when an outbound call becomes the a leg of another outbound call (r:15390)
core: fix issue with global bypass_media_after_bridge messing up callflow (r:15404)
core: add DUMP_EVENT macro (r:15439)
core: Add Record-File-Path to SWITCH_EVENT_RECORD_* fired by switch_ivr_record_file() (FSCORE-486/r:15446)
core: expand per-thread db caching to odbc (r:15453)
core: add switch_ivr_enterprise_originate optional new dimension to originate strings every element in :_: separated list in originate strings will fire in a dedicated thread and can contain their own {} and , and | lists (r:15455)
core: add read_frame_callback to gentones (r:15459)
core: add ringback to enterprise_originate (r:15461)
core: tolerate MySQL ODBC limitation (FSCORE-487/r:15465)
core: add <> var container to enterprise_originate syntax to set variables to pass to every thread (r:15469)
core: add cache_db handle api (odbc/sqlite abstraction) (r:15473)
core: Add support for coma-delimited lists in a user's auth-acl param (MODENDP-224/r:15760)
core: allow listeners to pre-bind to subclasses before the main subscriber is loaded and fix off by one issue in unbinding event handler functions (r:15790)
core: switch_ivr_originate: fix windows compiler warning (r:16320)
core: few new command line opts -vm for conditonal timer, -nocal to skip timer calibration and -nort to turn off clock_realtime fam of functions (r:16324)
core: fix failed_xml_cdr prefix is not respected (FSCORE-529/r:16401)
core: allow alias expansion from fs_cli (r:16416)
core: allow double escape in parser for \ (r:16440)
core: Add a -version flag to freeswitch binary (FSCORE-534/r:16482)
core: Delay windows service startup so other command line options are processed (r:16558)
core: ZRTP Video works with ZFone (but you have to set your endpoint to use 99 for the payload for video or 125 in mod_h26x.c for H264 becuase the payloads MUST match or it can't figure it out (r:16563)
core: disable cpu timer affinity by default but make it still possible via config and fix stray constant in tipping_point (r:16679)
core: rename switch_socket_create_pollfd to switch_socket_create_pollset, add switch_socket_create_pollfd that really creates a pollfd out of a socket, expose switch_pollset_poll and switch_pollset_remove (r:16683)
core: Fix play_and_get_digits not honoring max_tries param (FSCORE-554/r:16706)
core: Fix bridge_answer_timeout dropping calls even after being answered (FSCORE-556/r:16737)
core: add force_local_ip_v4 and force_local_ip_v6 global vars to override the core funcs to always discover the same ip (r:16801)
core: compromising on timing code, remove -vm and make it default, make new -heavy-timing for previous default, change tipping-point to work of count of active timers rather than sessions, this should statisfy the droves of 'I wish it worked like 1.0.4 people' (r:16853)
mod_cidlookup: fix minor bugs, make whitepages work if regular url not set (oops), channel var set for area, ignore "UNKNOWN" and "UNAVAILABLE" from API (r:15331)
mod_cidlookup: more cleanup, always try to get a location, add verbose option to api call, cache area and src so the cache is full rep of cid data (r:15333)
mod_cidlookup: proper fix for working in npanxx only mode (thanks for the heads up bkw) (r:15341)
mod_conference: mod_conference add conference_max_members channel variable that can be set on the first channel calling a conference to override the profiles max-members param (r:16597)
mod_dingaling: Fix mod_dingaling does not reads profile information from configuration file at runtime till whole module is reloaded (LBDING-15/r:14917)
mod_dptools: att_xfer sometimes doesn't detect B leg hanging up on PSTN - this is a patch to hangup B and initiate the transfer of A to C by pressing * (DP-8/r:15013)
mod_event_socket: add userauth <user>@<domain>:<pass> to event_socket to auth against user directory uses esl-password esl-allowed-api esl-allowed-events and esl-allowed-log to control resource access (r:16160)
mod_limit: remove memcache version, it never worked right and is unworkable w/out a lot of effort. will revisit when pluggable limit implemented (r:15908)
mod_limit: mod_limit: use = instead of like (r:16020)
mod_opal: Added setting of outgoing number and display name from extension number so H.323/Q.931 fields are set correctly in ALERTING and CONNECT messages sent for incoming calls.(r:14900)
mod_skypiax: now it accepts a max of 1 call from the same skypename to the same skypename (multiple instances of user A calling multiple instances of user B) each 1.5 seconds (r:14834)
mod_skypiax: fires a custom event when an incoming CHATMESSAGE arrives. (r:14845)
mod_skypiax: added skypiax_chat command.(r:14857)
mod_skypiax: now outbound chatmessages follow the standard path, you can send them with: chat,<proto>|<from>|<to>|<message> or with: skypiax_chat,<interfacename> <to> <text>. (r:14884)
mod_skypiax: when sending a chatmessage, do not report it as an incoming message (eg: do not report the messages sent by myself) (r:15327)
mod_skypiax: added one more 'PROTOCOL 7' command, because windows was not getting it. Now CHATMESSAGES will probably work on windows too (r:15342)
mod_skypiax: added 'report_incoming_chatmessages' configuration file per_interface setting. To activate it, put its value to '1' or 'true', any other value will be 'false'. Defualts to 'true' (r:15343)
mod_skypiax: commented out 'XCloseDisplay' lines, seems to cause crashes... (r:15344)
mod_skypiax: no more pipes for audio samples exchange between threads.(r:15541)
mod_skypiax: do not add delay when a sleep() or play() are executed by a script after answer() and before sending audio. (MODSKYPIAX-29, MODSKYPIAX-58/r:15585)
mod_sofia: ATTENTION BEHAVIOR CHANGE... you now have to explicitly set sip_invite_to_params to add params to the to field we will NO longer fall back to sip_invite_params in this case. (FSCORE-433/r:14849)
mod_sofia: Fix missing NOTIFY MWI when registering via proxy (MODSOFIA-26/r:14851)
mod_sofia: add set funcs for impls (r:14881)
mod_sofia: Segfault when receiving 30x response without contact header (MODENDP-247/r:14953)
mod_sofia: doing caret dialing to a location without a gateway will result in the host being null in some cases this will not allow that if you specify the full sip:dest@host it works fine but if you only specify the number then host will be null (r:15332)
mod_sofia: solve problem from MODENDP-258 with an alternate approach, and make send-message-query-on-register=first-only and sql-in-transactions=true the defaults (MODENDP-258/r:15441)
mod_sofia: lookup gateway by realm when not found by name (r:15475)
mod_sofia: Fix sofia statistics (MODSOFIA-38/r:15491)
mod_sofia: add content-disposition to SEND_INFO (r:15495)
mod_sofia: video improvements (r:15524)
mod_sofia: always check for callee update not just because of pai (r:15589)
mod_sofia: add ani and aniii to caller profile (MODSOFIA-34/r:15596)
mod_sofia: reset timestamp counter when we get new sdp etc because sonus likes to say ptime 20 and send 30ms timestamps in the 183 then once they say 200 ok with the same sdp they decide to actually send 20 for real this time (FSRTP-8/r:15597)
mod_sofia: convert sofia to use new core version of cache db handles (r:15600)
mod_sofia: Fix crash where client register request does not contain "nc" parameter (MODSOFIA-39/r:15609)
mod_sofia: Add ping hysteresis for SIP peer qualification (MODSOFIA-40/r:15628)
mod_sofia: add sip_profile_name go go with sip_gateway_name on outgoing calls (r:15629)
mod_sofia: fail2ban support in mod_sofia thanks jay binks. (MODSOFIA-41/r:15654)
mod_sofia: Revert part of MODSOFIA-41 r15654 that deals with phone reboot - breaks Aastra/Polycom (r:15656)
mod_sofia: Fix call not hanging up after hold (FSCORE-481/r:15657)
mod_sofia: add separate inbound/outound codec prefs params to sofia profile original codec-prefs sets both to the same value for back-compat (r:15658)
mod_sofia: reset the remote_media ip/port vars more often (r:15676)
mod_sofia: add event header to gateway registration events (MODENDP-281/r:16319)
mod_sofia: fix PUBLISH/200 response sent without ETag (FSCORE-262/r:16388)
mod_sofia: set chanvars on both directions in sip and introduce sip_to_tag and sip_from_tag vars (r:16395)
mod_sofia: fix rpid and from correctly when auto-nat is enabled so polycom won't display the flipping URI and freak some poor souls out :P (r:16473)
mod_sofia: sip is stupid MODSOFIA-51, the 202 has to have the correct contact or the phone comes back and subscribes to the contact and not the actual extension (MODSOFIA-51/r:16500)
mod_sofia: fix it so the Cisco phones will update their display properly in these cases and not show Private for all calls (r:17066)
mod_sofia: only allow barge-in on SCA mode when calls are from their own line (r:17067)
mod_sofia: add sip_local_network_addr var to see the ip of the sip profile a call came in or went out on for cdr (FSCORE-562/r:17073)
mod_sofia: mod_sofia: Add url encode to a var in the xml output to be valid xml. Also changed switch_url_encode to return the pointer of the string rather than the length, same as switch_amp_encode() (r:17087)
mod_sofia: prevent race in killgw followed by an immediate rescan with the same gateway name (r:17096)
mod_sofia: fix telephone-event negotiation with devices that don't do what the rfc says they SHOULD do (r:17097)
mod_sofia: Double @ in To header (MODENDP-300/r:17098)
mod_unimrcp: Do not allow speech_channel_destroy() to return unless MRCP session has been terminated. Do not explicitly destroy mutexes, buffers, and condvars that are allocated off of pool. (r:16938)
mod_voicemail: decrease sql queries for message counts (MODAPP-359/r:15636)
mod_voicemail: Fix r15636 so New are New, and Old are Old messages (r:15639)
mod_voicemail: mod_voicemail: Change so total_new_messages and total_saved_messages include the count of new/saved urgent messages (Issue caused by MODAPP-359) (r:15670)
mod_voicemail: Allow param for voicemail to allow message file type to be changed (MODAPP-394/r:16576)
mod_voicemail: copy user xml to a dynamic xml obj so you do not hold exclusive lock on global xml registry the whole time you are leaving a vm (r:16577)
mod_voicemail: fix vm to inherit params from domain/group (r:16644)
mod_voicemail: that shouldve been wrlock, and its missing an unlock (r:16647)
mod_voipcodecs: rearrange codecs so we don't have crazy overlap and remove G723-32 and move it up to dynamic cuz people have nothing better to do then write stupid RFC's (r:16251)
mod_xml_cdr: Allow specifying auth-scheme in config (MDXMLINT-56/r:16247)
build: fix msvc configuration manager for x64 targets in both 2008 sln files (r:12227)
build: use compiler intrinsics for windows x64 build (FSBUILD-131/r:12245)
build: fix version generator on windows (FSBUILD-115/FSBUILD-69/r:12247)
build: fix various msvc build issues (r:12272-12276)
build: silence nuisance sun cc warnings (r:12462)
build: updated windows projects for flite (FSBUILD-128/FSBUILD-133/r:12477)
build: download flite from right tarball location (FSBUILD-135/r:12482)
build: rework Windows build for pocketsphinx 5.1 (MODASRTTS-13/r:12483,12541)
build: fix automake 1.7 build (r:12487)
build: build path cleanups (FSBUILD-130/r:12490)
build: fix rebuild every time on msvc 2008 non team editions (FSBUILD-132/r:12492)
build: add skypiax to sln file and fix some warnings (r:12502)
build: fixed openzap.conf issue in deb package (r:12506)
build: fix rlim_t build on Mac (FSCORE-325/r:12533)
build: Solaris doesn't define RLIMIT_NPROC (FSCORE-326/r:12534)
build: removal of file reference for flite (FSBUILD-139/r:12540)
build: integrate mod_erlang_event into the buildsystem (FSBUILD-142/r:12587)
build: Fix FreeBSD build (r:12678)
build: fix libsoundtouch build dependencies after a configure failure (MODAPP-243/r:12783)
build: fix odbc detection to not try to use odbc when no headers are installed (r:12788)
build: add build of fs_ivrd (r:12843)
build: sphinx downloads for windows (FSBUILD-146/r:12853)
build: Build fails at bootstrap with libtool 2.2.6a (FSBUILD-82/r:12899)
build: add python build dependency to debian package build (FSBUILD-145/r:12920)
build: add libtool major version detection to configure in prep for supporting both libtool 2.x and 1.5.x at the same time (r:12922,12926-27)
build: fix older gcc build (FSBUILD-150/r:12924)
build: Add a fallback check in case libtool is not yet available in the builddir (get the version from build/config/ltmain.sh instead). print an error message and exit configure if that fails too (r:12933)
build: Add mod_memcache (commented) to modules.conf.in (r:12955)
build: make libtool version detection more robust (r:12979)
build: we need DYNAMIC_LIB_EXTEN for mod_perl and others (r:13023)
build: add mod_say_ru to modules.conf.in (r:13037)
build: bump sounds version (r:13040)
build: use different version file for moh version (r:13093)
build: de-couple version numbers and builds of sound files and moh files (FSBUILD-153/r:13096)
build: use sound_version.txt and moh_version.txt to determine sound file version on windows (FSBUILD-152/r:13097)
config: small regex adjustment so we can dial PSTN numbers that happen to have 4000 in them (r:13409)
config: you can call 9999 from a zrtp endpoint and enroll this should be replaced with an IVR to explain it a bit but most of the work happens client side at this point (r:13445)
config: add mod_nibblebill (r:13458)
config: clear up docs (r:13468)
config: clarify zrtp docs in vars.xml and give link on where to get more info (r:13469)
config: tweaks to codecs in default config (r:13485)
core: add sessions since start up to heartbeat event (r:12472)
core: increase buffer size for regex api (MODAPP-228/r:12480)
core: fix FreeSWITCH start failure - "Error: stacksize 240 is too large" (FSCORE-323/r:12509)
core: unloading endpoint module now removes from endpoint list (MODENDP-196/r:12535)
core: fix deadlock in fsctl hupall (r:12544)
core: add origination_pricavy var to originate api (r:12546)
core: increase stack limit for switch_system (FSCORE-328/r:12569)
core: fix uuid_originate param not assigning uuid properly (FSCORE-322/r:12591)
core: add check for a and b leg no answer on bridge (r:12595)
core: fix origination_privacy var (r:12603)
core: fix group_confirm regression from svn r12403 (r:12616)
core: fix caller ID values not being set in CHANNEL_OUTGOING (FSCORE-336/r:12643)
core: make -vg imply -waste so valgrind runs won't re-exec (r:12670)
core: add apr_pool_mutex_set() to our apr to fix thread-saftey issue (r:12672)
core: make port allocator more random (r:12673,12675)
core: make switch_channel_get_variable strdup so the pointer returned is safe and clean up the state locking (r:12674)
core: Empty audio files should not be created when RECORD_ANSWER_REQ is set to true and a call is not bridged (r:12682)
core: add transfer_after_bridge var (r:12691)
core: other_leg_unique_id incorrectly set when briding with using ',' (FSCORE-331/r:12704)
core: make gaussian noise less noisy (FSCORE-340/r:12720)
core: add import vars to FIFO (r:12722)
core: fix switch_core_file_write method not writing the entire buffer to the file (FSCORE-341/r:12728)
core: fix hanguphook order of operations vs destroy method issue in c++ code (r:12730)
core: rearrange hangup callflow to do more work in the session's own thread (r:12784)
core: Allow variables containing variables in set and export (MODAPP-241/r:12804)
core: add read_terminator_used var (r:12840)
core: change blocking rtp to psuedo-blocking to avoid endlessly blocking reads and refactor jitter buffer (r:12846)
core: add rtp-autoflush profile param and rtp_autoflush var (r:12854)
core: fix various small leaks (FSCORE-347/r:12873)
core: add a couple, two tree stats (r:12913)
core: fix play_and_get_digits to unset the var if the regex didn't match (r:12921)
core: drop divide by 2 from normalize func (r12935)
core: add missing begin/end, allow threads to read and play_and_get_digits methods (r:12958)
core: fix windows calling conventions for modules with sub-modules broken in svn r12919 (r:12960)
core: add flag to denote if a codec is init or not (FSCORE-349/r:12961)
core: add more specific checks to new macro just to be safe (r:12973)
core: change CS_DONE to CS_DESTROY and add state handler for destroy and tear down critical parts of the channel from this method which is not called until it's impossible for anything to be referencing the channel (after final write lock and before destroying the pool) (r:12986)
core: fix regression from earlier commit (FSCORE-352/r:12987)
core: expand channel variables for sound files in IVRs (MODAPP-257/r:13005)
core: clone frames in loopback so we can smooth it out better, now with more memory usage (tm) maybe this will curb pepople from using it like candy (r:13011)
core: run expand_vars if input contains a special escaped character not just when it contains a variable to eat up back slashes (r:13015)
core: change names to be more explicit (r:13028)
core: xml_config: Fix issue where default NULL strings were ignored on reload (r:13052)
core: autoflush on bridge and add bridge_hangup_cause variable to indicate the hangup cause of the last bridged channel (r:13065)
core: add record_ms, record_samples, playback_ms and playback_samples chanvars (r:13105)
core: make state_handler macros not let you install the same one more than once (r:13111)
core: Do not use struct fd_set uninitialized (always FD_ZERO() them, before using FD_SET() on a new one, or reusing them after select()). Fixes a segfault on solaris (r:13125)
core: Fix missing UNPROTECT_INTERFACE in case pre_answer fails (r:13130)
core: fix segfault on out of memory situation. (r:13398/FSCORE-366)
core: treat app::arg syntax in execute_on_answer as a broadcast request (r:13400)
core: use our own handler so it won't get overriden by anyone if zrtp is on (r:13411)
core: fix backslash getting removed from regex when using mod_xml_curl (r:13414/FSCORE-370)
core: fix handle leak in switch_thread_self on windows (r:13421/FSCORE-371)
core: allow switching from secure to clear and back (r:13422)
core: add more options to zrtp (r:13424)
core: add buffer flush (r:13425)
core: add some stuff for zrtp (r:13426)
core: can't flush because you have a chance of dumping zrtp control frames (r:13427)
core: add zrtp_sas1_string and zrtp_sas2_string variables (r:13429)
core: zrtp - this should make sure the secure mitm has a chance over latent connections (r:13436)
core: properly detect unterminated (r:13438)
core: zrtp - fix mitm to be more reliable (r:13443)
core: zrtp - mark verified (r:13444)
core: zrtp_secure_media=true will have to be set to true in order for your zrtp to work moving forward similar to how srtp_secure_media works.(r:13461)
core: prevent zrtp and srtp at the same time (r:13486)
core: add show secure channel status in "show channels" output (r:13502/FSRTP-2)
core: add CHANNEL_HANGUP_COMPLETE event to take the place of CHANNEL_HANGUP, CHANNEL_HANGUP now fires as soon as the channel is hungup but you must wait for CHANNEL_HANGUP_COMPLETE for the CDR data (r:13505)
core: fix core dump when calling session.execute (r:13508/FSCORE-373)
core: add netmask detection for nat discovery work (r:13549)
core: fix order of ops to enable logging sooner (r:13556)
core: add hunt_caller_profile (r:13625,13626)
core: add usec delta to log (r:13647)
core: add padding to cycles on session_record (r:13648)
mod_cidlookup: add new module, mod_cidlookup (r:12990)
mod_cidlookup: switch to using CURL instead of mod_http (r:12992)
mod_cidlookup: add config file (r:12994)
mod_cidlookup: set initial value for status (r:13029)
mod_clue_choo: add new module, mod_clue_choo (r:13068,13069,13070)
mod_commands: Add show api [name] and show application [name] (r:12296)
mod_commands: fix header_string "key" from being set twice from SWITCH_STANDARD_API(user_data_function) (MODAPP-242/r:12786)
mod_commands: add show modules functionality (MODAPP-227/r:12806)
mod_commands: add echo api command (echo back exact input) and add expand api command (executes another api command with variable expansion) (r:13101,13102)
mod_conference: add conference wait-mod flags and member moderator flag to delay starting a conference until someone with a moderator flag has joined (r:13442)
mod_conference: add member-type header to relevant events (r:13471)
mod_dahdi_codec: delay init of resources until the first time they are actually used to avoid unnecessary waste of resources in hardware codec (r:12962)
mod_dahdi_codec: set mod_dahdi_codec dahdi transcoding device sockets to non-blocking to avoid hanging when there is no data and just return 0 bytes frame (MODCODEC-8/r:13257)
mod_dahdi_codec: added proper waiting (up to 10ms) for the DAHDI transcoder output frame (r:13262)
mod_dahdi_codec: return silence frame in dahdi codec when there is no output from the decoder (r:13265)
mod_dptools: add group_recurse_variables and user_recurse_variables to {} vars (default is true, set to false to not pass vars down to user or group channels) (r:13241,13246)
mod_dptools: add param dial-timeout to directory (MODAPP-271/r:13247)
mod_erlang_event: Bind to 0.0.0.0 instead of 127.0.0.1 by default; like most erlang nodes do. (r:12249)
mod_erlang_event: Reply appropriately to net_adm:ping() (r:13066)
mod_erlang_event: snprintf needs a format string too, and write has the warn_unused_result attribute set, so store the return value somewhere (r:13090)
mod_nibblebill: Added feature mod_nibblebill to check balance and transfer the caller to an alternate dialplan context & extension if they deplete their funds. (r:13432)
mod_say_en: fix missing "at" in time readback, change from cardinal to ordinal numbers on dates, e.g. "January 20th" vs. "January 20" (MODAPP-263/r:13099)
mod_skypiax: when repeatedly you try to connect to non-existing Skype account in a short period, the Skype client send you back the two halves of the message 'ERROR 92 CALL: Unrecognised identity' inverted in a way that breaks the flux of the API messages. Maybe an anti-spam feature? Anyway, let's try to work around it and restore sanity with a 1 second delay (r:13663)
mod_skypiax: the Skype client sends us BOTH inband and out_of_band DTMFs, no way to shut the inbands. Let's intercept the out_of_bands ONLY if we are not bridged (eg: IVR, so not to waste CPU in detecting inband), but not propagate the out_of_band DTMFs if we are bridged (inband ones will be propagated) (r:13664)
mod_skypiax: added directory 'kernel', contains .config file for compilation of kernel 2.6.24.7 (64bit) tickless and 100HZ. (r:14336)
mod_skypiax: added directory 'configs/multiple-instance-same-skype-username', contains configuration file and script to launch Skype clients with multiple instances of the same skype user (r:14337)
mod_skypiax: added configs/windows-service directory, contains batch files needed to start skype instances as service, and then start FreeSWITCH (r:14359)
mod_skypiax: patch from Muhammad Shahzad for adding and removing interfaces on the fly (r:14367)
mod_skypiax: manage the 'BUSY' call status message (r:14371)
mod_skypiax: manage the 'WAITING_REDIAL_COMMAND' call status message (r:14372)
mod_skypiax: patch from Seven Du for hunting IDLE channels in a round-robin way (RR interface vs ANY interface). patch from Seven Du for removing interface as #'interface_id' and #'interface_name'. gmaruzz (meh) patch interface_remove() not to alter the global 'running' variable (it would cause all running signaling and API thread to exit) but to use a newly added tech_pvt->running variable. Also, changed behavior of interface_exists() for correct identification when using #interface_name and #interface_id. PLEASE TEST IT HEAVILY (r:14410)
mod_skypiax: fixed crash when you set an interface as 'sk' console, then remove the interface, then use the 'sk' command to send a message to the API (it try to use a non-existing interface) (r:14411)
mod_skypiax: now interprets correctly the_interface whichever it is: interface_id, interface_name, skype_username. And reload works too :-) (r:14414)
mod_skypiax: fix segfault when used with record_session (MODSKYPIAX-35/r:14444)
mod_sofia: Update CID on Polycom when doing an Attended transfer, Make send_display work with Polycom and others, add patch with mods from SFSIP-111 (r:13492/SFSIP-111)
mod_sofia: handle vegastream broken sip info packets (r:13506)
mod_sofia: this one was my fault, it shouldn't set them as sip_h just in case. Moved it to set the full header into sip_HEADERNAME so you can use it as you see fit or re-export it to a sip_h on the b-leg if needed unchanged. (r:14081)
mod_sofia: manually handle bye to delay the 200 OK till after the call is torn down (reversible with a define) (r:14121)
mod_sofia: Have I said how dumb sip is sometimes? (r:14142)
mod_voicemail: Flush DTMF Before Tone (MODAPP-312/r:14380)
mod_voicemail: add option to move to next and previous message(MODAPP-313/r:14386)
mod_voicemail: allow Voicemail FF REW key to have configurable millisec interval AND allow REW to limit at the begining of the recording. (MODAPP-311/r:14389)
mod_voicemail: allow to skip the info section of a message by pressing a key (MODAPP-314/r:14392)
mod_unimrcp: make mod_unimrcp compile with gcc 4.3 (MODUNIMRCP-1/r:13833)
mod_unimrcp: mrcp_profile for unimrcpserver 0.6.0 (MODASRTTS-18/r:13835)
mod_unimrcp: use paths referenced to the project file dir (r:13840)
mod_unimrcp: Added LPCM (16-bit linear PCM) codec, which is used internally in host order; while L16 is RFC3551 defined 16-bit linear PCM codec in network order. (r:13859)
mod_event_socket: inbound connection to event_socket can now take over an existing channel with 'myevents <uuid>' to take on the behaviour of an outbound socket
mod_event_socket: let any channel get messages
mod_event_socket: make event socket wait for hangup on outbound mode and send disconnect message
mod_expr: fix endless loop
mod_fax: new module
mod_fifo: add fifo_consumer_wrapup_time var (MODAPP-117)
mod_fifo: added callback agents
mod_fifo: honor keyword silence (MODAPP-118)
mod_flite: added windows build
mod_fsv: fix in a windows enviroment opening the record file in text mode. (MODAPP-169)
mod_http: added new module
mod_java: updated to new module api to support read/write locks on interface
mod_limit: accept dialplan context for transfer (MODAPP-161)
mod_spidermonkey_odbc: fix numRows and add numCols
mod_spidermonkey_odbc: fix segfault (MODLANG-75)
mod_stress: new module for voice stress analysis
mod_syslog: don't log blank lines (FSCORE-163)
mod_tone_stream: let silence_stream://0 indicate perpetual silence
mod_vmd: add new module to detect voicemail "beep"
mod_voicemail: Add vm_alternate_greet_id param to directory entry (MODAPP-174)
mod_voicemail: Patch to add voicemail preference controlling date announcement new param 'play-date-announcement' to values 'first' 'last' or 'never' defaults to first to retain previous behavior (MODAPP-121)
mod_voicemail: Update mwi light after delete vm via web. (MODAPP-124)
mod_voicemail: add ability to get to options without listening to every saved message (MODAPP-115)
mod_voicemail: add ability to skip greeting when leaving a voicemail. (MODAPP-181)
FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel
FIX: NULL dereference detected by klockwork (www.klockwork.com)
FIX: don't open failed local stream (MODFORM-9)
FIX: instability in mod_local_stream in failure scenarios
FIX: xmlrpc-c build on OS X 10.4 (FSBUILD-47)
ENHANCEMENT: Added tab completion on many api commands in console
ENHANCEMENT: polycom BLF support
FIX: many sip NAT related fixes in mod_sofia
FIX: support sip unregister with Contact: *
FIX: multiple segfaults in xmlrpc-c
FIX: sip unregister event being skipped
FIX: hangup properly on malformed sip 3pcc calls being used as a way to ping
ADD: enable-3pcc sofia profile param, it is now disabled by default.
ADD: presence events to sip proxy mode
ADD: legs param to cdr_csv
ADD: support for perl as an embedded lanugage
ENHANCEMENT: many new api's and functions to the embedded languages including api support, xml interface support, auto start scripts, and many new objects
CHANGE: python embedded language api changed to match perl, lua, java
FIX: many stability fixes in embedded langauges perl, lua, java, python
ADD: failed_xml_cdr magic channel variable
FIX: access free memory error in mod_sofia when using respond app
ENHNACEMENT: make global_setvar only have 2 fields so you can set foo=bar=blah w/o quotes
FIX: mod_spidermonkey keep hangup hook in the session thread
ENHANCEMENT: mod_ldap added sasl support and search filters
ADD: answered, waitForAnswer and mediaReady methods to embedded language Session object
ENHANCEMENT: mod_voicemail param change to allow notification emails using templates
ADD: per user acl in sofia
FIX: deadlock in mod_portaudio
ENHANCEMENT: blank username in sip will trigger a lookup for the user "nobody"
ADD: import variable to import variables from a peer channel at time of originate
FIX: api type fix for c++ modules when incorrectly using enums
FIX: eliminate need for escaped , in [] on originate
ADD: NDLB-force-rport option to force behavior as if rport was sent in the via
ENHANCEMENT: honor execute_on_answer on outbound legs too
ADD: execute_on_ring variable
FIX: Seg fault in CoreSession() class destructor
ADD: per channel caller id in originate
ADD: sip_outgoing_call_id variable
FIX: multiple memory leaks in mod_sofia
FIX: find_local_ip IPv6 support
ADD: variable expansion to on execute vars.(FSCORE-114)
ADD: count optional arg to show calls and show channels (MODAPP-103)
FIX: MODEVENT-25 (WSAWOULDBLOCK error on socket send in windows) in event socket
FIX: multiple fixes to the logic in mod_say_zh
ADD: inter digit timeout to swigged embedded languages getDigits method. (MODLANG-65)
