February 2012 Archives by thread
Starting: Wed Feb 1 01:01:26 CST 2012
Ending: Wed Feb 29 20:46:30 CST 2012
Messages: 587
- [asterisk-dev] const char * to AST_STRING_FIELD
Szasz Tamas
- [asterisk-dev] [Code Review] Fix ExtenSpy on trunk.
Mark Michelson
- [asterisk-dev] [Code Review] Fix broken SIP realtime update of lastms (and a couple other SIP realtime issues)
Mark Michelson
- [asterisk-dev] [Code Review] Fix deadlock between AMI action Agents and writing of frames to Agent channel
a_villacis
- [asterisk-dev] [Code Review] Constify some more channel driver technology callback parameters.
rmudgett
- [asterisk-dev] [Code Review] Fix a deadlock in agents occuring due to trying to lock a channel while having a lock on the pvt.
jrose
- [asterisk-dev] [Code Review]: SIP: peer matching by callbackextension
Terry Wilson
- [asterisk-dev] [Code Review] Fix TLS port settings for Manager and HTTP
Mark Michelson
- [asterisk-dev] [Code Review]: Remove some dead code found in _sip_show_peers()
Paul Belanger
- [asterisk-dev] [Code Review]: Remove some dead code found in _sip_show_peers()
Paul Belanger
- [asterisk-dev] [Code Review]: Remove some dead code found in _sip_show_peers()
Terry Wilson
- [asterisk-dev] [Code Review]: Remove some dead code found in _sip_show_peers()
jrose
- [asterisk-dev] [Code Review]: Remove some dead code found in _sip_show_peers()
jrose
- [asterisk-dev] [Code Review] Make reloading cdr_pgsql.so not screw up catastrophically.
jrose
- [asterisk-dev] [Code Review] Replace res_ais with a new module, res_corosync.
Russell Bryant
- [asterisk-dev] [Code Review] Remove some unnecessary locking from ast_hangup()
Russell Bryant
- [asterisk-dev] [Code Review] Add reload support to res_fax
Mark Michelson
- [asterisk-dev] [Code Review] Replace res_ais with a new module, res_corosync.
Mark Michelson
- [asterisk-dev] Higher scheduling priority for voice handling
J.A. Bezemer
- [asterisk-dev] [Code Review] Resource leak in SIP/TCP
rmudgett
- [asterisk-dev] [Code Review] Update to chan_unistim functionality
IgorG
- [asterisk-dev] [Code Review] avoid cppcheck warnings
wdoekes
- [asterisk-dev] [Code Review] Don't take computational cost into account when creating multistep translations
Mark Michelson
- [asterisk-dev] kindly asking for a bit of testing for a DTMF issue
Alex Hermann
- [asterisk-dev] Need help getting Resfax F option up to current 1.8 trunk
Bryant Zimmerman
- [asterisk-dev] [Code Review] Make callbackextension work for realtime and also match on callbackextension when multiple peers with the same host have differing callbackextensions
Terry Wilson
- [asterisk-dev] Asterisk 10.1.1 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.8.9.1 Now Available
Asterisk Development Team
- [asterisk-dev] [Code Review]: adding shared locks around usage of odbc handle in res_odbc
jrose
- [asterisk-dev] [Code Review] Resolve odbc segfaults by adding shared locks around usage of odbc handle in res_odbc
jrose
- [asterisk-dev] [Code Review] Document that CDRs cannot be modified after the bridge between two channels is broken
Terry Wilson
- [asterisk-dev] [Code Review] If a CDR variable is set during a bridged call, make sure the values get copied over to the bridge CDR
Terry Wilson
- [asterisk-dev] [Code Review] avoid cppcheck warnings
junky
- [asterisk-dev] t.38 passthrough filters redudant packages ?
Kristijan Vrban
- [asterisk-dev] [Code Review] Send udptl packets directly between both call legs without extrac them
vrban
- [asterisk-dev] [asterisk-commits] kmoore: trunk r354165 - in /trunk/channels: chan_dahdi.c sig_pri.h
Kevin P. Fleming
- [asterisk-dev] [Code Review]: Add a SIP nat=auto setting
Kevin P. Fleming
- [asterisk-dev] Could pri-123 be reopened as this is still an issue?
Steve Hanselman
- [asterisk-dev] (last call for comments) Proposed changes to Asterisk release and support cycles
Kevin P. Fleming
- [asterisk-dev] [Code Review] iLBC codec can be added to asterisk again as it is available under a new license now
Paul Belanger
- [asterisk-dev] [Code Review] SIP INFO DTMF non-numeric codes treated as '1'
Matt Jordan
- [asterisk-dev] [Code Review] Add SIP INFO DTMF test to the Asterisk Test Suite
Matt Jordan
- [asterisk-dev] [Code Review] Add a SIP nat=auto setting
Terry Wilson
- [asterisk-dev] [Code Review] Make the config parser remove escaping backslashes
opticron
- [asterisk-dev] [Code Review] Add test for LEN function which includes testing for config escapes
opticron
- [asterisk-dev] [Code Review] Add tests to check parsing of compact and non-compact headers
opticron
- [asterisk-dev] [Code Review] Fix parsing of SIP headers where compact and non-compact headers are mixed
opticron
- [asterisk-dev] app_voicemail fails to comp with imap storage
Taylor, Jonn
- [asterisk-dev] [Code Review] Remove some dead code found in _sip_show_peers()
Mark Michelson
- [asterisk-dev] [Code Review] Fix SIP blind transfer playing parking slot to caller.
rmudgett
- [asterisk-dev] Is externalivr really ready for "real" usage?
Steve Murphy
- [asterisk-dev] [Code Review] Fix reconnecting to pgsql database after connection loss.
rmudgett
- [asterisk-dev] [Code Review] Convert two classes of strncpy usage to ast_copy_string
Terry Wilson
- [asterisk-dev] [Code Review] Opaquify const char * and char[] in the ast_channel
Terry Wilson
- [asterisk-dev] [Code Review] Add AGIEXITONHANGUP variable.
Russell Bryant
- [asterisk-dev] New agents does not get noted for waiting callers in the queue
Kristijan Vrban
- [asterisk-dev] [Code Review] chan_dahdi: Add dialtonedetect option
Jeremy Pepper
- [asterisk-dev] [Code Review]: Bring the (missing) changes from Mantis ID 13495 in trunk.
KNK
- [asterisk-dev] [Code Review] Changes from Mantis ID 13495 in trunk.
KNK
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
Matt Jordan
- [asterisk-dev] [Code Review]: Bring the (missing) changes from Mantis ID 13495 in trunk.
rmudgett
- [asterisk-dev] [Code Review] Fix unlocking typo in cel_sqlite3_custom module reload
a_villacis
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
Matt Jordan
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
Kevin Fleming
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
Kevin Fleming
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
Matt Jordan
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
Mark Michelson
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
rmudgett
- [asterisk-dev] [Code Review] Add ability to reload SRTP policies
Matt Jordan
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
rmudgett
- [asterisk-dev] [Code Review] avoid many cppcheck (#2)
junky
- [asterisk-dev] [Code Review] Bring the (missing) changes from Mantis ID 13495 in trunk.
KNK
- [asterisk-dev] [Code Review]: Bring the (missing) changes from Mantis ID 13495 in trunk.
KNK
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
Kevin Fleming
- [asterisk-dev] [Code Review] No valid transports available, falling back to 'udp'
wdoekes
- [asterisk-dev] [Code Review] Add ability to clone ao2 containers.
rmudgett
- [asterisk-dev] [Code Review] sync chan_dahdi->p->outgoing with sig_analog->p-outgoing
Alec Davis
- [asterisk-dev] [Code Review] Fixing the regression from wrong set route information on provisional sip responses
schmidts
- [asterisk-dev] [Code Review] regression test for ASTERISK-19358 part 2 (send ACK to 200-Contact, not 1xx-Contact)
wdoekes
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
rmudgett
- [asterisk-dev] [Code Review] Fix memory leak, possible crash in sip show peers
rmudgett
- [asterisk-dev] [Code Review] Make ast_unload_resource actually remove the module from the module list when it is unloaded
Terry Wilson
- [asterisk-dev] [Code Review] ast_channel opaquification: most pointers, integer types
Terry Wilson
- [asterisk-dev] [Code Review]: PreDial - Ability to run dialplan on callee channel and caller channel right before actual Dial
schmidts
- [asterisk-dev] [Code Review] Convert app_page to use app_confbridge internally
Joshua Colp
- [asterisk-dev] possible Bug in Confbridge?
Gunnar Schaller
- [asterisk-dev] [Code Review] Lightweight NAT Support
Joshua Colp
- [asterisk-dev] [Code Review] Lightweight NAT Support Test
Joshua Colp
- [asterisk-dev] [Code Review] Enable gosub use in connected line, redirecting, and ccss
opticron
- [asterisk-dev] [Code Review] Add tests for connected line, redirecting, and csss to test new gosub calls
opticron
- [asterisk-dev] [Code Review] fix compile warnings with app_rpt
Paul Belanger
- [asterisk-dev] [Code Review] Remove chan_usbradio and app_rpt.
Russell Bryant
- [asterisk-dev] (ASTERISK-19336) h exten is not run in the context that calls a AEL macro
Johan Wilfer
- [asterisk-dev] [Code Review] testing CDR(accountcode) being carried over into local channels
wdoekes
- [asterisk-dev] [Code Review]: PreDial - Ability to run dialplan on callee channel and caller channel right before actual Dial
kobaz
- [asterisk-dev] [Code Review] Fix test_utils.c and test_substitution.c for latest opaquification commit
Terry Wilson
- [asterisk-dev] [Code Review] fix compiling warnings for chan_ooh323
Paul Belanger
- [asterisk-dev] RTP/SDP for using unknown codecs
Beñat Urteaga
- [asterisk-dev] [Code Review] Add option to prevent SIP diversion headers from being sent
jrose
- [asterisk-dev] [Code Review] Opaquify ast_format structs in the ast_channel
Terry Wilson
- [asterisk-dev] [Code Review] fix that ./runtests.py -t something doesn't match something_else
wdoekes
- [asterisk-dev] ast_sockaddr_stringify
Johann Steinwendtner
- [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - Compile - pbuilder-maverick-amd64 42
Paul Belanger
- [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - Compile 42
Matthew Jordan
- [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - Compile 42
Matthew Jordan
- [asterisk-dev] Asterisk 1.8 and SIP Diversion: header
Pavel Troller
- [asterisk-dev] [Code Review] Audit and reformat IAX2 definitions to match the IANA-registered values.
Sean Bright
- [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - Compile 42
Paul Belanger
- [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - Compile 42
Paul Belanger
- [asterisk-dev] [Code Review] Second attempt at choosing optimal translation paths, specifically when resampling signed linear formats.
Mark Michelson
- [asterisk-dev] [Code Review] Astobj2 locking enhancement
Kevin Fleming
- [asterisk-dev] [Code Review] Astobj2 locking enhancement
rmudgett
- [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk Testing - Asterisk 1.8 Branch 5
Matthew Jordan
- [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk Testing - Asterisk 1.8 Branch - Asterisk 1.8 CentOS 6 64-Bit 5
Matthew Jordan
- [asterisk-dev] [Code Review] Opaquify ast_channel lists and structs
Terry Wilson
- [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk Testing - Asterisk 10 Branch - Asterisk CentOS 6 32-Bit 2
Matthew Jordan
- [asterisk-dev] [svn-commits] seanbright: trunk r357005 - in /trunk: channels/ include/asterisk/ main/
Mark Michelson
- [asterisk-dev] [Code Review] FastAGI: Add IPv6 support
Sean Bright
- [asterisk-dev] [Code Review] copy accountcode to accountcode, not peeraccount in app_dial
wdoekes
- [asterisk-dev] [Code Review] Create STACK_PEEK to view calling context, extension, and priority
Tilghman Lesher
- [asterisk-dev] [Code Review] Add IPv6 Address Support To Security Events Framework
elguero
- [asterisk-dev] [Code Review] Triggers dialplan actions when specific CONTROL_FRAMES are detected on a channel
Marco Signorini
- [asterisk-dev] [Code Review] Fix case-sensitivity for device-specific event subscriptions and CCSS
opticron
- [asterisk-dev] [Code Review] Opaquify some ast_channel typedefs, file descriptor arrays, and the _softhangup flag
Terry Wilson
- [asterisk-dev] [Code Review] Prevent race condition that can cause ast_read to "miss" important frames
Mark Michelson
- [asterisk-dev] [Code Review] Add F option from dial to app_queue
jrose
Last message date:
Wed Feb 29 20:46:30 CST 2012
Archived on: Thu Mar 1 08:10:58 CST 2012
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