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changelog 

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9164 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Anthony Minessale 2008-07-24 11:08:03 +00:00
parent 7ebef5b4ad
commit b4ba1a2255
1 changed files with 127 additions and 0 deletions
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freeswitch (1.0.1-1) unstable; urgency=low
* 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)
* ADD: Linksys P-RTP-Stat SIP header values (SFSIP-66)
* FIX: small leak in core
* ADD: progress_timeout var to originate
* UPDATE: portaudio library
* FIX: added timeout to iax read
* ADD: 'pa rescan' to portaudio to look for new devices
* FIX: wait for broadcast to start when starting async hold to avoid race
* FIX: mod_rss, don't always play the first news feed
* FIX: mod_rss inverval to use the session inteval (audio problems on 30ms channels)
* ADD: Path: support in mod_sofia on register
* FIX: mod_shout record stream
* ENHANCEMENT: mod_voicemail support for effective_caller_id_name/number
* ADD: url encode/decode api calls
* FIX: "nua()" in debug information in sofia instead of the real function name
* FIX: better handling of sips: uris
* FIX: don't seg when using more than SWITCH_MAX_CODECS and bump SWITCH_MAX_CODECS to 50 (we have more than 30 in tree) (MODFORM-10)
* ADD: mod_yaml
* FIX: segfault on freeswitch startup if installed directories are removed
* FIX: segfault when intercept with inbound_late_negotiation=true set
* FIX: dont flood logs with eavesdrop messages (MODAPP-101)
* FIX: don't destroy a codec that has not been created (MODAPP-101)
* ENHANCEMENT: allows the "eavesdrop_group" variable to contain several groups, comma separated. (MODAPP-101)
* FIX: cross compile (FSBUILD-53)
* FIX: add header that Nuaunce considers mandatory (MODASRTTS-5)
* ADD: write locks to the core and a function to unregister event bindings (adds better ability to unload modules)
* ENHANCEMENT: make modules unbind events and un-reserve subclasses on module unload
* ADD: removable xml hook bindings
* ADD: EventConsumer object to embedded languages so you can make event handlers
* FIX: sending CN with supress-cng true
* FIX: segfault in the event system when trying to remove NULL event
* ADD: flags to turn off srtp auth and rtp auto adj (FSCORE-149 && MODENDP-115)
* FIX: use lighter math and avoid infinite loop in port allocator (FSCORE-148)
* ENHANCEMENT: let conference pin entry start during prompt (MODAPP-111)
* ADD: mod_pocketsphinx
* FIX: Misuse of SQLRowCount, issues with MSSQL (MODAPP-105)
* FIX: segfaults in mod_python with dtmf callback
* ENHANCEMENT: mod_conference auto-record parameter (MODAPP-112)
* ENHANCEMENT: reload support to many modules
* FIX: mod_sofia add replaces to supported header
* ENHANCEMENT: add args callback to sleep so you can process dtmf and events while "sleeping"
* ADD: mod_flite
* ENHANCEMENT: switch_xml converted back to c code and support double globs on windows
* ENHANCEMENT: mod_sofia support for adding and removing gateways without restarting profiles
* ADD: extract contact header info into A channel when unhandled 3xx response is received (MODENDP-116)
* FIX: outbound event_socket + late negotiation
* ADD: copy_xml_cdr variable
* ADD: silence_stream (like tone_stream but silent)
* ADD: module_exists api call
* ADD: emailer implementation for windows
* ADD: wait_for_silence application
* FIX: no error message generated if OS is unable to load a module ( due to dependency/installation issues )
* FIX: segfault in media bugs
* FIX: acl lists not correctly matching all ip adresses
* FIX: mod_spidermonkey exit() does not stop script when called from the hangup callback (return "exit" from the callback)
* FIX: mod_syslog works again
* FIX: crash on terminal resize
* FIX: audio problems on big endian
* ENHANCEMENT: Disable multiple registrations on a per-device basis (MODENDP-117)
* ADD: fifo_consumer_exit_key variable (MODAPP-100)
* ADD: cidr based user auth in mod_sofia
* ADD: uuid_send_dtmf fsapi command (MODAPP-114)
* ADD: server registration fiels to sip_registration database (MODENDP-118)
* FIX: use a variable, realm or to host to find gateway when it's not obvious (handles challenged REFER)
* ADD: timeout to curl run in javascript
* ADD: voicemail_inject fsapi command
* ADD: reboot option for sip phones to flush_inboud_reg sofia profile api command
* FIX: add small padding to end of mp3 to avoid cut off mp3 recording
* FIX: patch multiple SDP connection lines in sdp for proxy media mode (MODENDP-109)
* FIX: don't parse ringback varable in proxy situations
* ADD: per call vm recording ext with vm_message_ext variable
* ADD: sip_bye_h prefix to add headers to bye
* ENHANCEMENT: more interfaces available in show fsapi command
* FIX: don't leak in buffers on realloc fail
* FIX: fail out of a conference call if write fails
* ADD: auto ip-change detection
* ADD: mod_snom
* FIX: mod_sofia don't send sipfrag on transfer to cisco so they don't hang up the call
-- Mike Jerris <mike@jerris.com> Thu, 24 Jul 2008 07:00:00 -0500
freeswitch (1.0.1~trunk) unstable; urgency=low
* Updated revision number