diff --git a/docs/ChangeLog b/docs/ChangeLog
index c928c5c9dc..3093f6cf07 100644
--- a/docs/ChangeLog
+++ b/docs/ChangeLog
@@ -1,143 +1,270 @@
-freeswitch (1.0.0)
-
-	Enhanced sofia sip nat handling
-	Many fixes found by Klockwork (www.klocwork.com)
-	Added disable_app_log variable
-	Fixed mod_local_stream with rates on windows
-	Fixed finding of files in rate dirs on windows
-	Fixed memory corruption from sofia_contact function
-	Added sofia profile param NDLB-received-in-nat-reg-contact
-	Added sofia profile param aggressive-nat-detection
-	Fixed video sip calls in proxy media mode
-	Added bridge_terminate_key var
-	Update xmlrpc-c lib to trunk revision from upstream, fix windows xmlrpc
-	Enhanced nat handling in proxy media mode in sip
-	Add progress media to timetable so you can calculate pdd
-	Fixed seg when using unicast on socket when call has no read_codec
-	Fixed missed log events on busy box
-	Added -bleg to intercept
-	Enhance configure detection of python
-	Fixed build on solaris and freebsd for several modules
-	Added param "vm-email-only" to make voicemail sent by email only (previously default behavior)
-	Added param "vm-mailto-notify" to allow sending a notification email
-	Fixed mod_java build
-	Fixed mwi failures for some devices that don't subscribe
-	Removed fsapi functions (killchan, transfer, session_displace, reject)
-	Removed fsapi functions (session_record, broadcast, hold, media)
-	Many updates to sofia-sip library including over 100 fixes
-	
-freeswitch (1.0.rc6)
-
-	Changed to not allow pass_2833 on transcoded calls (it never worked, now it will tell you)
-	Enhanced sofia sip nat handling
-	Fix libedit build on solaris
-	Fix session timers in mod_sofia
-	Fix conference fire-call
-	Change: add var_event down into the endpoints so chans with no parents can still pass options
-	Added enable-post-var param to xml_rpc
-	Fix mod_lua build on solaris
-	Many fixes found by Klockwork (www.klocwork.com)
-	Add unregister event in mod_sofia
-	Enhance python configure detection
-	Add vm_boxcount api func
-	Fixed att_xfer issue
-	Fix sip now includes the Allow-Events header in more places
-
-freeswitch (1.0.rc5)
-
-	Changed internal state names to avoid confusion
-	Fixed video negotiation
-	Enhanced accuracy of windows timer
-	Fixed mod_ldap build
-	Added dialplan and context to sql table for channels
-	Multiple fixes to mod_lua and mod_perl
-	Fixed logic bug in fifo causing segfault
-	internal changes to sip stack so we can remove a hash redundant to the stack
-	Fixed multiple memory leaks in mod_sofia
-	Fixed event fetch segfault on sip subscribe
-	Fixed segfault on timer rollover in sofia on 64bit
-	Fixed audio timing issues in mod_portaudio
-	Changed names of sip profiles in default config to avoid confusion
-	Fixed memory usage leak-like behavior when playing files requiring resampling
-	Removed some unused api's
-	Fix rtp timeout when playing moh
-	Removed some un-needed libraries and files from tree
-	Fixed multiple issues in sip stack including multiple segfaults
-	Added support for sip transfers on bypass_media and proxy_media calls
-	Added say application
-	Fixed --disable-debug configure option
-	Enhanced switch_cpp wrapper (and perl, python, lua, java)
-	Fixed segfault on inavalid stun response
-	Fixed configure help output
-	Fixed segfault on mp3 playback
-	Fixed assert on invalid sdp (missing m= line)
-	Added configurable windows service name
-	Fixed proxy mode call transition to non proxy call
-	Fixed solaris build of voipcodecs
-	Fixed sofia seg when call failure edge case
-
-freeswitch (1.0.rc4)
-
-	Add tab completion in cli
-	Add "inline" dialplan
-	Fixed segfault in enum
-	Enhance enum to fork dial equal priority entries
-	Added auto-reload to enum
-	Fixed odbc bug is mod_sofia presence handling
-	Add presence for conference and dial an eavesdrop
-	Fix stack overflow segfault when recursively parking calls
-	Fixed race is sofia registration handling
-	Enhance sofia registration, unregister on keep-alive OPTIONS failure
-	Added internal routing loop detection/avoidance
-	Fixed race in bgapi in event socket
-	Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"
-	Fixed re-setting sound prefix to no prefix after a pharse
-	Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event
-	Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia
-	Add originate_timeout to originate vars
-	Fixed hanging channels in mod_portaudio
-	Added auto time sync on vps migration to different hardware
-	Fixed seg on transfer when both legs are not sip
-	Added configurable dtmf duration defaults
-	Enhanced voicemail, allow interruption of hello message
-	Fixed voicemail to not light up light on saved messages
-	Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)
-	Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed
-	Fixed MODLANG-56 (bad audio on originate and javascript streamFile)
-	Added hold/unhold dialplan apps
-	Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params
-	Backport fixes from sofia-sip tree
-	Fixed MSVC build
-	Fixed segfault on sip SUBSCRIBE with Expires: 0
-	Added mod_say_zh
-	Added --with-pyton and --with-pyton-config configure options
-	Added mod_lua
-	Enhanced switch_cpp wrapper in core and normalized interfaces for perl, python, lua, and java
-	Fixed multiple issues in cpp wrapper and the languages perl, python, lua and java
-	Added back mod_perl
-	Added sofia gateway option ping to adjust options ping frequency
-	Added .net event socket lib to contrib
-	Fixed passing of exact response codes of sip across a bridge
-	Added mod_reference, reference endpoint module
-	Enhanced build so you can now make commented out modules using "make mod_name"
-	
-freeswitch (1.0.rc3)
-
-        Enhance xml menu system
-        Fixes upstream from sofia-sip library
-        Enhance mod_fifo
-        Added close method to ODBC spidermonkey class
-        Fix multiple bugs in the cpp wrapper used in mod_java and mod_python
-        Fix hung sip channel issue using respond app or on re-invite with bypass media after 1xx or 2xx responses
-
-freeswitch (1.0.rc2)
-
-	Fixed speex protocol negotiation issues (8k vs 16k)
-	Fixed mod_iax race conditions
-	Fixed ptime negotiation issues when re-packetizing
-	Added ip based acl lists
-
-freeswitch (1.0.rc1)
-
-
-	
+freeswitch (1.0.1)
+
+	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
+
+
+
+freeswitch (1.0.0)
+
+	Enhanced sofia sip nat handling
+	Many fixes found by Klockwork (www.klocwork.com)
+	Added disable_app_log variable
+	Fixed mod_local_stream with rates on windows
+	Fixed finding of files in rate dirs on windows
+	Fixed memory corruption from sofia_contact function
+	Added sofia profile param NDLB-received-in-nat-reg-contact
+	Added sofia profile param aggressive-nat-detection
+	Fixed video sip calls in proxy media mode
+	Added bridge_terminate_key var
+	Update xmlrpc-c lib to trunk revision from upstream, fix windows xmlrpc
+	Enhanced nat handling in proxy media mode in sip
+	Add progress media to timetable so you can calculate pdd
+	Fixed seg when using unicast on socket when call has no read_codec
+	Fixed missed log events on busy box
+	Added -bleg to intercept
+	Enhance configure detection of python
+	Fixed build on solaris and freebsd for several modules
+	Added param "vm-email-only" to make voicemail sent by email only (previously default behavior)
+	Added param "vm-mailto-notify" to allow sending a notification email
+	Fixed mod_java build
+	Fixed mwi failures for some devices that don't subscribe
+	Removed fsapi functions (killchan, transfer, session_displace, reject)
+	Removed fsapi functions (session_record, broadcast, hold, media)
+	Many updates to sofia-sip library including over 100 fixes
+	
+freeswitch (1.0.rc6)
+
+	Changed to not allow pass_2833 on transcoded calls (it never worked, now it will tell you)
+	Enhanced sofia sip nat handling
+	Fix libedit build on solaris
+	Fix session timers in mod_sofia
+	Fix conference fire-call
+	Change: add var_event down into the endpoints so chans with no parents can still pass options
+	Added enable-post-var param to xml_rpc
+	Fix mod_lua build on solaris
+	Many fixes found by Klockwork (www.klocwork.com)
+	Add unregister event in mod_sofia
+	Enhance python configure detection
+	Add vm_boxcount api func
+	Fixed att_xfer issue
+	Fix sip now includes the Allow-Events header in more places
+
+freeswitch (1.0.rc5)
+
+	Changed internal state names to avoid confusion
+	Fixed video negotiation
+	Enhanced accuracy of windows timer
+	Fixed mod_ldap build
+	Added dialplan and context to sql table for channels
+	Multiple fixes to mod_lua and mod_perl
+	Fixed logic bug in fifo causing segfault
+	internal changes to sip stack so we can remove a hash redundant to the stack
+	Fixed multiple memory leaks in mod_sofia
+	Fixed event fetch segfault on sip subscribe
+	Fixed segfault on timer rollover in sofia on 64bit
+	Fixed audio timing issues in mod_portaudio
+	Changed names of sip profiles in default config to avoid confusion
+	Fixed memory usage leak-like behavior when playing files requiring resampling
+	Removed some unused api's
+	Fix rtp timeout when playing moh
+	Removed some un-needed libraries and files from tree
+	Fixed multiple issues in sip stack including multiple segfaults
+	Added support for sip transfers on bypass_media and proxy_media calls
+	Added say application
+	Fixed --disable-debug configure option
+	Enhanced switch_cpp wrapper (and perl, python, lua, java)
+	Fixed segfault on inavalid stun response
+	Fixed configure help output
+	Fixed segfault on mp3 playback
+	Fixed assert on invalid sdp (missing m= line)
+	Added configurable windows service name
+	Fixed proxy mode call transition to non proxy call
+	Fixed solaris build of voipcodecs
+	Fixed sofia seg when call failure edge case
+
+freeswitch (1.0.rc4)
+
+	Add tab completion in cli
+	Add "inline" dialplan
+	Fixed segfault in enum
+	Enhance enum to fork dial equal priority entries
+	Added auto-reload to enum
+	Fixed odbc bug is mod_sofia presence handling
+	Add presence for conference and dial an eavesdrop
+	Fix stack overflow segfault when recursively parking calls
+	Fixed race is sofia registration handling
+	Enhance sofia registration, unregister on keep-alive OPTIONS failure
+	Added internal routing loop detection/avoidance
+	Fixed race in bgapi in event socket
+	Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"
+	Fixed re-setting sound prefix to no prefix after a pharse
+	Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event
+	Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia
+	Add originate_timeout to originate vars
+	Fixed hanging channels in mod_portaudio
+	Added auto time sync on vps migration to different hardware
+	Fixed seg on transfer when both legs are not sip
+	Added configurable dtmf duration defaults
+	Enhanced voicemail, allow interruption of hello message
+	Fixed voicemail to not light up light on saved messages
+	Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)
+	Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed
+	Fixed MODLANG-56 (bad audio on originate and javascript streamFile)
+	Added hold/unhold dialplan apps
+	Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params
+	Backport fixes from sofia-sip tree
+	Fixed MSVC build
+	Fixed segfault on sip SUBSCRIBE with Expires: 0
+	Added mod_say_zh
+	Added --with-pyton and --with-pyton-config configure options
+	Added mod_lua
+	Enhanced switch_cpp wrapper in core and normalized interfaces for perl, python, lua, and java
+	Fixed multiple issues in cpp wrapper and the languages perl, python, lua and java
+	Added back mod_perl
+	Added sofia gateway option ping to adjust options ping frequency
+	Added .net event socket lib to contrib
+	Fixed passing of exact response codes of sip across a bridge
+	Added mod_reference, reference endpoint module
+	Enhanced build so you can now make commented out modules using "make mod_name"
+	
+freeswitch (1.0.rc3)
+
+        Enhance xml menu system
+        Fixes upstream from sofia-sip library
+        Enhance mod_fifo
+        Added close method to ODBC spidermonkey class
+        Fix multiple bugs in the cpp wrapper used in mod_java and mod_python
+        Fix hung sip channel issue using respond app or on re-invite with bypass media after 1xx or 2xx responses
+
+freeswitch (1.0.rc2)
+
+	Fixed speex protocol negotiation issues (8k vs 16k)
+	Fixed mod_iax race conditions
+	Fixed ptime negotiation issues when re-packetizing
+	Added ip based acl lists
+
+freeswitch (1.0.rc1)
+
+
+