November 2011 Archives by thread
Starting: Tue Nov 1 01:18:10 CDT 2011
Ending: Wed Nov 30 19:33:34 CST 2011
Messages: 497
- [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned
wdoekes
- [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups
irroot
- [asterisk-dev] How to get ast_debug() logged
Mark Michelson
- [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned
Terry Wilson
- [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned
Terry Wilson
- [asterisk-dev] externaddr in chan_sip.c - locking required?
Ian Pilcher
- [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned
Tilghman Lesher
- [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned
David Vossel
- [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned
David Vossel
- [asterisk-dev] [Code Review]: Astobj2 locking enhancement
rmudgett
- [asterisk-dev] [Code Review] Fix more res_jabber resource leaks
opticron
- [asterisk-dev] [Code Review] Mothership cmenuselect easteregg addition
jrose
- [asterisk-dev] [Code Review] Adds barriers to the menuselect easter egg
jrose
- [asterisk-dev] [Code Review] NetBorder Call Analyzer (Answering Machine Detection) support
Konstantin M.
- [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip.
rmudgett
- [asterisk-dev] [Code Review] ast_indicate(chan, -1) don't stop playing tones
may213
- [asterisk-dev] Scheduler and event callbacks - what thread?
Ian Pilcher
- [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned
wdoekes
- [asterisk-dev] Memory failure - channels in meetme
Kevin P. Fleming
- [asterisk-dev] New Asterisk community developers
Jason Parker
- [asterisk-dev] [Code Review] ensure that ast_string_field_pool base + used is always aligned
Terry Wilson
- [asterisk-dev] [Code Review] Add hold status with CEL and setting a channel var HOLDING via ast_moh_stop/start
Terry Wilson
- [asterisk-dev] [Code Review] fix excess warnings after r1395 (and disallow empty names always)
wdoekes
- [asterisk-dev] [Code Review] fix sqlite crash on unset config_table / unbreak sqlite realtime_multi_handler
wdoekes
- [asterisk-dev] [Code Review] Just remove the silly SIP registertrying option
Terry Wilson
- [asterisk-dev] [Code Review] chan_sip registertrying lacks a global option
Terry Wilson
- [asterisk-dev] [asterisk-commits] wdoekes: trunk r343163 - in /trunk: ./ include/asterisk/ main/
Kevin P. Fleming
- [asterisk-dev] [Code Review] Add #includeif statement
Terry Wilson
- [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge
Terry Wilson
- [asterisk-dev] [Code Review] Add #includeif statement
wdoekes
- [asterisk-dev] Optional api and weak symbol problem
Yaroslav Panych
- [asterisk-dev] [Code Review] BLF Subscriptions Causes SIP Deadlock
opticron
- [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge
David Vossel
- [asterisk-dev] [Code Review] use autoconf to detect whether alignment should be enforced
Kevin Fleming
- [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge
mjordan
- [asterisk-dev] Setvar= for manager accounts
Kevin P. Fleming
- [asterisk-dev] [Code Review] Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.
jrose
- [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)
rmudgett
- [asterisk-dev] DAHDI-Linux 2.6.0-rc1 and DAHDI-Tools 2.6.0-rc1 Released
Asterisk Development Team
- [asterisk-dev] changing sizeof to __alignof__
Walter Doekes
- [asterisk-dev] [Code Review]: [patch] Improve debug of ast_hangup
wdoekes
- [asterisk-dev] [Code Review] SIP: peer matching by callbackextension
wdoekes
- [asterisk-dev] [Code Review] fix default udptl port range which differs in source / udptl.conf.sample
wdoekes
- [asterisk-dev] [Code Review]: sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'
wdoekes
- [asterisk-dev] [Code Review] Fix deadlock between subscription event RWLOCK and dialogs container lock in chan_sip. (simplified)
David Vossel
- [asterisk-dev] [Code Review] PreDial - Ability to run dialplan on callee channel and caller channel right before actual Dial
kobaz
- [asterisk-dev] [Code Review] Added VM_INFO dialplan function to return voicemail user information
shawkris
- [asterisk-dev] [Code Review] Doubly linked lists unit test and update to implementation.
rmudgett
- [asterisk-dev] [Code Review] sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'
Tzafrir Cohen
- [asterisk-dev] [Code Review]: sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'
Alec Davis
- [asterisk-dev] [Code Review] app_voicemail [general] variables zonetag and locale are not set on mailbox until after reload
wdoekes
- [asterisk-dev] [Code Review] Make TestSuite Asterisk class create and install config files after instantiation
mjordan
- [asterisk-dev] [Code Review] Fix several bugs with SDP parsing and well-formedness of responses
Matthew Nicholson
- [asterisk-dev] [Code Review] regression test for r1570 (app_voicemail, bad load_config order)
wdoekes
- [asterisk-dev] [Code Review] use __alignof__ instead of sizeof for alignment of ast_string_field_allocation
wdoekes
- [asterisk-dev] SIP, NAT, security concerns, oh my!
Kevin P. Fleming
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Kevin P. Fleming
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Bruce B
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Terry Wilson
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Walter Doekes
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Terry Wilson
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Bruce B
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Olle E. Johansson
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Bruce B
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Matt Riddell
- [asterisk-dev] Summary: SIP, NAT, security concerns, oh my!
Terry Wilson
- [asterisk-dev] [Code Review] Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain
Terry Wilson
- [asterisk-dev] [Code Review] Add a unit test for ast_sockaddr_split_hostport
Terry Wilson
- [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes
Terry Wilson
- [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes
Terry Wilson
- [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused
rmudgett
- [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes
Terry Wilson
- [asterisk-dev] DAHDI analog: Monitoring call setup by backward audio connection
Pavel Troller
- [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes
Olle E Johansson
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Andrew Olmsted
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Andrew Olmsted
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Olle E Johansson
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Olle E Johansson
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Andrew Olmsted
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Olle E Johansson
- [asterisk-dev] [Code Review] chan_sip: removing some non-compliant code in v10 and minor fixes
Terry Wilson
- [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes
Terry Wilson
- [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes
Olle E Johansson
- [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes
Terry Wilson
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Andrew Olmsted
- [asterisk-dev] Asterisk 10.0.0-rc1 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.8.8.0-rc3 Now Available
Asterisk Development Team
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Olle E Johansson
- [asterisk-dev] [Code Review] mxml puts quotes inside multiline opaque data
wdoekes
- [asterisk-dev] $500 bounty for #18039: Realtime music restarts from beginning each time
Alistair Cunningham
- [asterisk-dev] [Code Review] fix using external MP3 player and res_timing_dahdi
elguero
- [asterisk-dev] [Code Review] Don't forget to rescan MOH files for cached realtime MOH classes
Terry Wilson
- [asterisk-dev] [Code Review] Eliminate redundant and possibly dangerous close/fclose pairs in chan_sip and tcptls.
jrose
- [asterisk-dev] [Code Review] Video format treated as audio when removed from the file playback scheduler
mjordan
- [asterisk-dev] Google voice patch
sean darcy
- [asterisk-dev] [Code Review] Move password entry for new Comedian Mail users to the end of setup.
jrose
- [asterisk-dev] [Code Review]: Remove need for registration strings in sip.conf
wdoekes
- [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.
rmudgett
- [asterisk-dev] [Code Review] Don't read past then end of our int when calling write() in __ast_queue_frame
Terry Wilson
- [asterisk-dev] [Code Review] disallow tcp/tls transport in peers when tcpenable=no
wdoekes
- [asterisk-dev] [Code Review] fix odd logic in get_msg_text() and a couple of related quirks
wdoekes
- [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER
wdoekes
- [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER
wdoekes
- [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused
garlew
- [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Andrew Olmsted
- [asterisk-dev] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters
Kevin P. Fleming
- [asterisk-dev] [Code Review] unbreaking <sip:username> from-uri support for REGISTER
wdoekes
- [asterisk-dev] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters
Kevin P. Fleming
- [asterisk-dev] [Code Review] Update basic call parking test.
rmudgett
- [asterisk-dev] [Code Review] Restore SIP DTMF overlap dialing method.
mjordan
- [asterisk-dev] [Code Review] Multiple recipients and urgency.
jrose
- [asterisk-dev] Asterisk World 2012 Call for Papers
Bryan Johns
- [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - Trunk - Ubuntu Lucid (10.04) 1227
Matthew Jordan
- [asterisk-dev] Asterisk 1.8.8.0-rc4 Now Available
Asterisk Development Team
- [asterisk-dev] [Code Review] Add a test for verifying when username information can be leaked via differing nat settings
Terry Wilson
- [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 10 - Ubuntu Lucid (10.04) - amd64 357
Matthew Jordan
- [asterisk-dev] [Code Review] Change the default nat setting to "yes"
Terry Wilson
- [asterisk-dev] [svn-commits] mjordan: testsuite/asterisk/trunk r2810 - /asterisk/trunk/tests/channels/SIP/...
Paul Belanger
- [asterisk-dev] [Code Review] Permit/deny negation, plus multiple-specification with commas
Tilghman Lesher
- [asterisk-dev] [Code Review] Update astdb documentation to how it actually works
Tilghman Lesher
- [asterisk-dev] asterisk dial plan questions
Jamuel Starkey
- [asterisk-dev] Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: un defined symbol
shalu dhamija
- [asterisk-dev] [Code Review] Add #includeif statement
Paul Belanger
- [asterisk-dev] [Code Review] Change the default nat setting to "yes"
mjordan
- [asterisk-dev] [Code Review] Change the default nat setting to "yes"
wdoekes
- [asterisk-dev] (no subject)
Charles Wang
- [asterisk-dev] [Code Review] fix ast_str_truncate signedness warning and documentation
wdoekes
- [asterisk-dev] [Code Review] Don't keep the STUN socket open between STUN monitor checks.
rmudgett
- [asterisk-dev] [Code Review] Don't play a new song for every new hold within a call with caching realtime MOH
Terry Wilson
- [asterisk-dev] [Code Review] SendMessage strips extension from To: Header in SIP MESSAGE
mjordan
- [asterisk-dev] [Code Review] chan_dahdi: create and destroy channels at run-time
Tzafrir Cohen
- [asterisk-dev] How to save CLI log to file
SIP IMS
- [asterisk-dev] [Code Review] Separate verbose level for logging
Tilghman Lesher
- [asterisk-dev] [Code Review] Separate verbose level for logging
Tilghman Lesher
- [asterisk-dev] [Code Review] rtp / rtcp set debug ip doesnt work correctly in 1.8
schmidts
- [asterisk-dev] Intended sip.conf matching behaviour?
Alistair Cunningham
- [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - Trunk - Ubuntu Lucid (10.04) - amd64 1250
Matthew Jordan
- [asterisk-dev] [Code Review] safe_asterisk: Allow overriding TTY and CONSOLE in startup.d
Tzafrir Cohen
- [asterisk-dev] asterisk-core-sounds-ru
Tzafrir Cohen
- [asterisk-dev] [Code Review] make testsuite runtests a bit less warny
wdoekes
- [asterisk-dev] [Code Review] update samples to use [public] context
Paul Belanger
- [asterisk-dev] Adding languages to say.conf
Nir Simionovich
- [asterisk-dev] IAX Hardphone (again)
Bill Shaw
- [asterisk-dev] [Code Review] update samples to use [public] context
Tilghman Lesher
Last message date:
Wed Nov 30 19:33:34 CST 2011
Archived on: Thu Dec 1 08:26:57 CST 2011
This archive was generated by
Pipermail 0.09 (Mailman edition).