October 2011 Archives by author

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Starting: Sat Oct 1 02:07:13 CDT 2011
Ending: Mon Oct 31 22:18:51 CDT 2011
Messages: 463

• [asterisk-dev] simple register doesn't work using 1.8.7.1 on armv5tel   S Adrian
• [asterisk-dev] simple register doesn't work using 1.8.7.1 on armv5tel   S Adrian
• [asterisk-dev] simple register doesn't work using 1.8.7.1 on armv5tel   S Adrian
• [asterisk-dev] Questions about dialplan function MessageSend in Asterisk 10   Kamal Ahmed
• [asterisk-dev] AppKonference 2.0   Paul Albrecht
• [asterisk-dev] Reg : Asterisk Support of SIPConnect 1.1   Neeharika Allanki
• [asterisk-dev] [Code Review] Asterisk Support of SIP Connect 1.1   Neeharika Allanki
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Benny Amorsen
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Örn Arnarson
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Örn Arnarson
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Örn Arnarson
• [asterisk-dev] New codec support in Asterisk 10   Jay Ashworth
• [asterisk-dev] New codec support in Asterisk 10   Jay Ashworth
• [asterisk-dev] [Code Review] Add #includeif statement   Paul Belanger
• [asterisk-dev] [Code Review] Add #includeif statement   Paul Belanger
• [asterisk-dev] [Code Review] Add #includeif statement   Paul Belanger
• [asterisk-dev] [Code Review] Add #includeif statement   Paul Belanger
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first. (Another version)   Paul Belanger
• [asterisk-dev] [svn-commits] qwell: trunk r940 - /trunk/menuselect.c   Paul Belanger
• [asterisk-dev] [svn-commits] qwell: trunk r940 - /trunk/menuselect.c   Paul Belanger
• [asterisk-dev] SIP now target of botnets?   Paul Belanger
• [asterisk-dev] SIP now target of botnets?   Paul Belanger
• [asterisk-dev] SHA1 and MD5 code?   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) 843   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 10 - Ubuntu Lucid (10.04) 236   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) - amd64 840   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) - amd64 840   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) 847   Paul Belanger
• [asterisk-dev] [Code Review] Clean module info documentation tags for the purpose of menuselect category arrangement.   Paul Belanger
• [asterisk-dev] [Code Review] menuselect by support level categories   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - Trunk - FreeBSD 8.1 375   Paul Belanger
• [asterisk-dev] Failure to write to tcp/tls socket.   Paul Belanger
• [asterisk-dev] Failure to write to tcp/tls socket.   Paul Belanger
• [asterisk-dev] Many sip dialog/ opened channels.   Paul Belanger
• [asterisk-dev] SHA1 and MD5 code?   Paul Belanger
• [asterisk-dev] SHA1 and MD5 code?   Paul Belanger
• [asterisk-dev] [Code Review] Allow linking/building against an external editline library.   Paul Belanger
• [asterisk-dev] Audio is not always working.   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - Trunk - Ubuntu Lucid (10.04) 1128   Paul Belanger
• [asterisk-dev] Google voice patch   Paul Belanger
• [asterisk-dev] [Bamboo] Paul Belanger commented on Asterisk - Trunk - Ubuntu Lucid (10.04) 1130   Paul Belanger
• [asterisk-dev] [Code Review] chan_sip: removing some non-compliant code in v10 and minor fixes   Paul Belanger
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Paul Belanger
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   Paul Belanger
• [asterisk-dev] Astricon: GPG Key signing event   Paul Belanger
• [asterisk-dev] [Code Review] "sipusers" only exists in documentation, and "sipfriends" is still used in documentation   Paul Belanger
• [asterisk-dev] SHA1 and MD5 code?   Marc Blanchet
• [asterisk-dev] building asterisk-scf   Marc Blanchet
• [asterisk-dev] Things Google   David Bowerman
• [asterisk-dev] Bounty   Russell Bryant
• [asterisk-dev] [svn-commits] qwell: trunk r940 - /trunk/menuselect.c   Russell Bryant
• [asterisk-dev] SHA1 and MD5 code?   Russell Bryant
• [asterisk-dev] SHA1 and MD5 code?   Russell Bryant
• [asterisk-dev] SHA1 and MD5 code?   Russell Bryant
• [asterisk-dev] SHA1 and MD5 code?   Russell Bryant
• [asterisk-dev] [Code Review]: Preserve DTMF length in main/features.c   Russell Bryant
• [asterisk-dev] Volume Control Methods in app_voicemail   Byron Clark
• [asterisk-dev] [Code Review] update SHA1 code to RFC6234   Tzafrir Cohen
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first.   Tzafrir Cohen
• [asterisk-dev] SHA1 and MD5 code?   Tzafrir Cohen
• [asterisk-dev] SHA1 and MD5 code?   Tzafrir Cohen
• [asterisk-dev] [Code Review] Allow linking/building against an external editline library.   Tzafrir Cohen
• [asterisk-dev] [Code Review] svnmerge: avoid a separator when merging a single commit   Tzafrir Cohen
• [asterisk-dev] [Code Review]: svnmerge: avoid a separator when merging a single commit   Tzafrir Cohen
• [asterisk-dev] [Code Review]: svnmerge: avoid a separator when merging a single commit   Tzafrir Cohen
• [asterisk-dev] [Code Review] svnmerge: avoid a separator when merging a single commit   Tzafrir Cohen
• [asterisk-dev] [Code Review]: svnmerge: avoid a separator when merging a single commit   Tzafrir Cohen
• [asterisk-dev] [Code Review] ensure that ast_string_field_pool base + used is always aligned   Tzafrir Cohen
• [asterisk-dev] Things Google   Saúl Ibarra Corretgé
• [asterisk-dev] Memory failure - channels in meetme   Saúl Ibarra Corretgé
• [asterisk-dev] [Code Review] sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'   Alec Davis
• [asterisk-dev] [Code Review]: sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'   Alec Davis
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   Alec Davis
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   Alec Davis
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   Alec Davis
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   Alec Davis
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   Alec Davis
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   Alec Davis
• [asterisk-dev] [Code Review] Don't try to lookup purely numeric hostnames that don't match peers or try to send requests to failed lookups   Walter Doekes
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Walter Doekes
• [asterisk-dev] simple register doesn't work using 1.8.7.1 on armv5tel   Walter Doekes
• [asterisk-dev] simple register doesn't work using 1.8.7.1 on armv5tel   Walter Doekes
• [asterisk-dev] Enabling strictrtp by default in 10   Walter Doekes
• [asterisk-dev] Enabling strictrtp by default in 10   Walter Doekes
• [asterisk-dev] [Code Review] Allow dialplan control of faxdetection via faxdetect framehook / options in func CHANNEL   Kevin Fleming
• [asterisk-dev] [Code Review] Reject v17 skinny devices in Asterisk10   Kevin Fleming
• [asterisk-dev] [Code Review] update SHA1 code to RFC6234   Kevin Fleming
• [asterisk-dev] [Code Review] Store route-set from provisional SIP responses so early-dialog requests can be routed properly   Kevin Fleming
• [asterisk-dev] [Code Review]: Store route-set from provisional SIP responses so early-dialog requests can be routed properly   Kevin Fleming
• [asterisk-dev] [Code Review] Store route-set from provisional SIP responses so early-dialog requests can be routed properly   Kevin Fleming
• [asterisk-dev] [Code Review] Don't send in-dialog requests when Asterisk has not yet received a Contact URI from a UAS   Kevin Fleming
• [asterisk-dev] [Code Review] Don't send in-dialog requests when Asterisk has not yet received a Contact URI from a UAS   Kevin Fleming
• [asterisk-dev] [Code Review]: svnmerge: avoid a separator when merging a single commit   Kevin Fleming
• [asterisk-dev] [Code Review]: Add option to check state when state is unknown and optomise the checking to reduce existing load.   Kevin Fleming
• [asterisk-dev] Bounty   Kevin P. Fleming
• [asterisk-dev] Completely Off Topic - Getting ready for Astricon   Kevin P. Fleming
• [asterisk-dev] Seeking built-in channel waiter function   Kevin P. Fleming
• [asterisk-dev] Seeking built-in channel waiter function   Kevin P. Fleming
• [asterisk-dev] chan_sip source: Flags definition: Is this a bug, or am I stupid ?   Kevin P. Fleming
• [asterisk-dev] SIP now target of botnets?   Kevin P. Fleming
• [asterisk-dev] Local channel origination issue.   Kevin P. Fleming
• [asterisk-dev] Things Google   Kevin P. Fleming
• [asterisk-dev] SIP now target of botnets?   Kevin P. Fleming
• [asterisk-dev] MRCP support for asterisk   Kevin P. Fleming
• [asterisk-dev] SHA1 and MD5 code?   Kevin P. Fleming
• [asterisk-dev] Replace voicemail PIN with voice biometric authentication   Kevin P. Fleming
• [asterisk-dev] SHA1 and MD5 code?   Kevin P. Fleming
• [asterisk-dev] SHA1 and MD5 code?   Kevin P. Fleming
• [asterisk-dev] SHA1 and MD5 code?   Kevin P. Fleming
• [asterisk-dev] New codec support in Asterisk 10   Kevin P. Fleming
• [asterisk-dev] New codec support in Asterisk 10   Kevin P. Fleming
• [asterisk-dev] Best keepalive for peers   Kevin P. Fleming
• [asterisk-dev] building asterisk-scf   Kevin P. Fleming
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Kevin P. Fleming
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Kevin P. Fleming
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Kevin P. Fleming
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Kevin P. Fleming
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Kevin P. Fleming
• [asterisk-dev] [asterisk-commits] pabelanger: branch group/astricon2011 r342540 - /team/group/astricon2011/   Kevin P. Fleming
• [asterisk-dev] paging and intecom   SIP IMS
• [asterisk-dev] [Code Review]: Check for ANY to place calls on hold for SIP   Olle E Johansson
• [asterisk-dev] [Code Review] Saving route header information even on a non-final response for proper UPDATE messages   Olle E Johansson
• [asterisk-dev] [Code Review]: Implement Externaddr on a sip device basis   Olle E Johansson
• [asterisk-dev] [Code Review]: Implement Externaddr on a sip device basis   Olle E Johansson
• [asterisk-dev] [Code Review] "sipusers" only exists in documentation, and "sipfriends" is still used in documentation   Olle E Johansson
• [asterisk-dev] [Code Review] Enable support for early media in AMI originate action   Olle E Johansson
• [asterisk-dev] [Code Review] Dialplan function for manager account checks - AMI_CLIENT()   Olle E Johansson
• [asterisk-dev] [Code Review]: Implement Externaddr on a sip device basis   Olle E Johansson
• [asterisk-dev] Bounty   Olle E. Johansson
• [asterisk-dev] SHA1 and MD5 code?   Olle E. Johansson
• [asterisk-dev] Dahdi file descriptors and meetme   Olle E. Johansson
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Olle E. Johansson
• [asterisk-dev] [Code Review]: Preserve DTMF length in main/features.c   Olle E. Johansson
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Olle E. Johansson
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Olle E. Johansson
• [asterisk-dev] Memory failure - channels in meetme   Olle E. Johansson
• [asterisk-dev] Setvar= for manager accounts   Olle E. Johansson
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   Matthew Jordan
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 10 - Ubuntu Lucid (10.04) 236   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 10 - Ubuntu Lucid (10.04) 236   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 10 - Ubuntu Lucid (10.04) 236   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) 840   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) 843   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) - amd64 840   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) 847   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) - amd64 840   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - Trunk - FreeBSD 8.1 375   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) - amd64 860   Matthew Jordan
• [asterisk-dev] [Bamboo] Matthew Jordan commented on Asterisk - Trunk - Ubuntu Lucid (10.04) - amd64 1122   Matthew Jordan
• [asterisk-dev] building asterisk-scf   Vik Killa
• [asterisk-dev] SIP now target of botnets?   Jeremy Kister
• [asterisk-dev] STUN support idea   Jeremy Kister
• [asterisk-dev] AMI Events for Skinny   Jacob.E.Miles at L-3Com.com
• [asterisk-dev] Audio is not always working.   Jacob.E.Miles at L-3Com.com
• [asterisk-dev] Audio is not always working.   Jacob.E.Miles at L-3Com.com
• [asterisk-dev] Audio is not always working.   Jacob.E.Miles at L-3Com.com
• [asterisk-dev] building asterisk-scf   Andrew Latham
• [asterisk-dev] [Code Review]: Modify merge message so commit log message comes first.   Tilghman Lesher
• [asterisk-dev] [Code Review] rework cdr_csv parsing (2nd attempt)   Tilghman Lesher
• [asterisk-dev] [Code Review] Add #includeif statement   Tilghman Lesher
• [asterisk-dev] [Code Review] Add #includeif statement   Tilghman Lesher
• [asterisk-dev] SHA1 and MD5 code?   Tilghman Lesher
• [asterisk-dev] SHA1 and MD5 code?   Tilghman Lesher
• [asterisk-dev] SHA1 and MD5 code?   Tilghman Lesher
• [asterisk-dev] SHA1 and MD5 code?   Tilghman Lesher
• [asterisk-dev] SHA1 and MD5 code?   Tilghman Lesher
• [asterisk-dev] SHA1 and MD5 code?   Tilghman Lesher
• [asterisk-dev] SHA1 and MD5 code?   Tilghman Lesher
• [asterisk-dev] New codec support in Asterisk 10   Tilghman Lesher
• [asterisk-dev] New codec support in Asterisk 10   Tilghman Lesher
• [asterisk-dev] New codec support in Asterisk 10   Tilghman Lesher
• [asterisk-dev] New codec support in Asterisk 10   Tilghman Lesher
• [asterisk-dev] New codec support in Asterisk 10   Tilghman Lesher
• [asterisk-dev] Best keepalive for peers   Tilghman Lesher
• [asterisk-dev] [Code Review] svnmerge: avoid a separator when merging a single commit   Tilghman Lesher
• [asterisk-dev] Potential 1.8 behavior change in realtime (especially for the CURL backend)   Tilghman Lesher
• [asterisk-dev] How to change length of variable   Tilghman Lesher
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Tilghman Lesher
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Tilghman Lesher
• [asterisk-dev] [Code Review]: fopen(REF_FILE) fails to check for success   Tilghman Lesher
• [asterisk-dev] simple register doesn't work using 1.8.7.1 on armv5tel   Tilghman Lesher
• [asterisk-dev] [Code Review]: Add PAUSED tag to queue log when adding queue member as paused   Leif Madsen
• [asterisk-dev] [Code Review] Update prep_tarball script to pull down pre-exported documentation when creating an Asterisk tarball   Leif Madsen
• [asterisk-dev] [Code Review] Update prep_tarball script to pull down pre-exported documentation when creating an Asterisk tarball   Leif Madsen
• [asterisk-dev] [Code Review] Update prep_tarball script to pull down pre-exported documentation when creating an Asterisk tarball   Leif Madsen
• [asterisk-dev] Video patch for Cisco 9971 phones   Artem Makhutov
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Artem Makhutov
• [asterisk-dev] AMI Events for Skinny   Stoyan Marinov
• [asterisk-dev] Audio is not always working.   Stoyan Marinov
• [asterisk-dev] [asterisk-users] Queue calls to agent end prematurely with diastatus cancel   Chris Miller
• [asterisk-dev] Dahdi file descriptors and meetme   Tony Mountifield
• [asterisk-dev] Video patch for Cisco 9971 phones   Richard Mudgett
• [asterisk-dev] Local channel origination issue.   Richard Mudgett
• [asterisk-dev] Commit message format updated. Please update and reinstall your repotools.   Richard Mudgett
• [asterisk-dev] [svn-commits] schmidts: trunk r340720 - /trunk/channels/chan_sip.c   Richard Mudgett
• [asterisk-dev] [svn-commits] schmidts: trunk r340720 - /trunk/channels/chan_sip.c   Richard Mudgett
• [asterisk-dev] [svn-commits] irroot: trunk r340771 - /trunk/   Richard Mudgett
• [asterisk-dev] ASTERISK-14769 r342112 OBJ_KEY   Richard Mudgett
• [asterisk-dev] [Code Review]: Allow dialplan control of faxdetection via faxdetect framehook / options in func CHANNEL   Matthew Nicholson
• [asterisk-dev] [Code Review] Allow dialplan control of faxdetection via faxdetect framehook / options in func CHANNEL   Matthew Nicholson
• [asterisk-dev] [Code Review] Load the proper XML documentation when multiple modules document the same application   Matthew Nicholson
• [asterisk-dev] [Code Review]: Load the proper XML documentation when multiple modules document the same application   Matthew Nicholson
• [asterisk-dev] [Code Review] Load the proper XML documentation when multiple modules document the same application   Matthew Nicholson
• [asterisk-dev] [Code Review] Check for ANY to place calls on hold for SIP   Matthew Nicholson
• [asterisk-dev] [Code Review] Add test execution modes to the Asterisk Testsuite   Matthew Nicholson
• [asterisk-dev] [Code Review] Allow fine grained control over test version asterisk compatibility specification   Matthew Nicholson
• [asterisk-dev] [Code Review]: Allow fine grained control over test version asterisk compatibility specification   Matthew Nicholson
• [asterisk-dev] [Code Review] Add option to check state when state is unknown and optomise the checking to reduce existing load.   Matthew Nicholson
• [asterisk-dev] [Bamboo] Matthew Nicholson commented on Asterisk - 1.8 - Ubuntu Lucid (10.04) - amd64 873   Matthew Nicholson
• [asterisk-dev] [Bamboo] Matthew Nicholson commented on Asterisk - 10 - Ubuntu Lucid (10.04) 272   Matthew Nicholson
• [asterisk-dev] [Bamboo] Matthew Nicholson commented on Asterisk - Trunk - Ubuntu Lucid (10.04) 1140   Matthew Nicholson
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   Matthew Nicholson
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   Matthew Nicholson
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] SHA1 and MD5 code?   Jeffrey Ollie
• [asterisk-dev] AST-2011-012: Remote crash vulnerability in SIP channel driver   Jeffrey Ollie
• [asterisk-dev] Seeking built-in channel waiter function   Yaroslav Panych
• [asterisk-dev] Seeking built-in channel waiter function   Yaroslav Panych
• [asterisk-dev] Seeking built-in channel waiter function   Yaroslav Panych
• [asterisk-dev] Local channel origination issue.   Yaroslav Panych
• [asterisk-dev] Local channel origination issue.   Yaroslav Panych
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Yaroslav Panych
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Yaroslav Panych
• [asterisk-dev] [Code Review] Add #includeif statement   Jason Parker
• [asterisk-dev] New codec support in Asterisk 10   Jason Parker
• [asterisk-dev] AST-2011-012: Remote crash vulnerability in SIP channel driver   Jason Parker
• [asterisk-dev] Asterisk 1.8.8.0-rc2 Now Available   Jason Parker
• [asterisk-dev] pinned-spans DAHDI: plug&play start working...   Oron Peled
• [asterisk-dev] SHA1 and MD5 code?   Simon Perreault
• [asterisk-dev] SHA1 and MD5 code?   Simon Perreault
• [asterisk-dev] New codec support in Asterisk 10   Simon Perreault
• [asterisk-dev] [Code Review] Don't try to lookup purely numeric hostnames that don't match peers or try to send requests to failed lookups   Simon Perreault
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Simon Perreault
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Simon Perreault
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Simon Perreault
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Simon Perreault
• [asterisk-dev] STUN support idea   Ian Pilcher
• [asterisk-dev] How to get ast_debug() logged   Ian Pilcher
• [asterisk-dev] How to change length of variable   Kirill Popov
• [asterisk-dev] How to change length of variable   Kirill Popov
• [asterisk-dev] SIP now target of botnets?   Philip Prindeville
• [asterisk-dev] SIP now target of botnets?   Philip Prindeville
• [asterisk-dev] [Code Review] Add #includeif statement   Grzegorz Pycia
• [asterisk-dev] [Code Review] Add #includeif statement   Matt Riddell
• [asterisk-dev] [Code Review]: Preserve DTMF length in main/features.c   Jonathan Rose
• [asterisk-dev] Failure to write to tcp/tls socket.   Catalin S.
• [asterisk-dev] Many sip dialog/ opened channels.   Catalin S.
• [asterisk-dev] Many sip dialog/ opened channels.   Catalin S.
• [asterisk-dev] Many sip dialog/ opened channels.   Catalin S.
• [asterisk-dev] Best keepalive for peers   Catalin S.
• [asterisk-dev] Best keepalive for peers   Catalin S.
• [asterisk-dev] Failure to write to tcp/tls socket.   Catalin S.
• [asterisk-dev] [svn-commits] schmidts: trunk r340720 - /trunk/channels/chan_sip.c   Stefan Schmidt
• [asterisk-dev] Many sip dialog/ opened channels.   Stefan Schmidt
• [asterisk-dev] Completely Off Topic - Getting ready for Astricon   Nir Simionovich
• [asterisk-dev] Completely Off Topic - Getting ready for Astricon   Nir Simionovich
• [asterisk-dev] Completely Off Topic - Getting ready for Astricon   Nir Simionovich
• [asterisk-dev] Add hold status with CEL and setting a channel var HOLDING via ast_moh_stop/start and potential res_musiconhold fixup.   Shane R. Spencer
• [asterisk-dev] Add hold status with CEL and setting a channel var HOLDING via ast_moh_stop/start and potential res_musiconhold fixup.   Shane R. Spencer
• [asterisk-dev] Asterisk 1.8.8.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Scheduled Maintenance for Asterisk Project community services   Asterisk Development Team
• [asterisk-dev] Scheduled Maintenance for Asterisk Project community services   Asterisk Development Team
• [asterisk-dev] Correction: Scheduled Maintenance for Asterisk Project community services   Asterisk Development Team
• [asterisk-dev] Asterisk 1.8.7.1 Now Available (Security Release)   Asterisk Development Team
• [asterisk-dev] Scheduled Maintenance for Asterisk Project community services   Asterisk Development Team
• [asterisk-dev] DAHDI-Linux 2.5.0.2 and DAHDI-Tools 2.5.0.2 Released   Asterisk Development Team
• [asterisk-dev] AST-2011-012: Remote crash vulnerability in SIP channel driver   Asterisk Security Team
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Steve Totaro
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Steve Totaro
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Steve Totaro
• [asterisk-dev] [Code Review] SIP 484 handling fix and new Incomplete app control frame   Pavel Troller
• [asterisk-dev] chan_sip source: Flags definition: Is this a bug, or am I stupid ?   Pavel Troller
• [asterisk-dev] [Code Review] Overlap dialling improvement for chan_sip.   Pavel Troller
• [asterisk-dev] Bounty   Venefax
• [asterisk-dev] [Code Review] Adjust formats of chan_local when channel we proxying changes   David Vossel
• [asterisk-dev] Questions about dialplan function MessageSend in Asterisk 10   David Vossel
• [asterisk-dev] [Code Review] Clean module info documentation tags for the purpose of menuselect category arrangement.   David Vossel
• [asterisk-dev] [Code Review] menuselect by support level categories   David Vossel
• [asterisk-dev] [Code Review] chan_sip: removing some non-compliant code in v10 and minor fixes   David Vossel
• [asterisk-dev] [Code Review] chan_sip: removing some non-compliant code in v10 and minor fixes   David Vossel
• [asterisk-dev] STUN support idea   David Vossel
• [asterisk-dev] [Code Review] Implement Externaddr on a sip device basis   David Vossel
• [asterisk-dev] [Code Review]: ensure that ast_string_field_pool base + used is always aligned   David Vossel
• [asterisk-dev] Memory failure - channels in meetme   David Vossel
• [asterisk-dev] [Code Review] Astobj2 locking enhancement   David Vossel
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Ryan Wagoner
• [asterisk-dev] Audio is not always working.   Damien Wedhorn
• [asterisk-dev] Audio is not always working.   Damien Wedhorn
• [asterisk-dev] How to get ast_debug() logged   Damien Wedhorn
• [asterisk-dev] Many sip dialog/ opened channels.   Eric Wieling
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   Terry Wilson
• [asterisk-dev] [Code Review]: rewrite of matching code in chan_sip.c realtime_peer()   Terry Wilson
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   Terry Wilson
• [asterisk-dev] [Code Review] Add some realtime SIP tests via res_config_sqlite   Terry Wilson
• [asterisk-dev] [Code Review] Add a utility to convert Asterisk 10s SQLite3 astdb back to Berkeley DB...just in case.   Terry Wilson
• [asterisk-dev] [Code Review] Add a utility to convert Asterisk 10s SQLite3 astdb back to Berkeley DB...just in case.   Terry Wilson
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   Terry Wilson
• [asterisk-dev] [Code Review] Force consistency for ast_realtime_load_multientry when no rows are returned   Terry Wilson
• [asterisk-dev] [Code Review] Force consistency for ast_realtime_load_multientry when no rows are returned   Terry Wilson
• [asterisk-dev] Add hold status with CEL and setting a channel var HOLDING via ast_moh_stop/start and potential res_musiconhold fixup.   Terry Wilson
• [asterisk-dev] SHA1 and MD5 code?   Terry Wilson
• [asterisk-dev] SHA1 and MD5 code?   Terry Wilson
• [asterisk-dev] SHA1 and MD5 code?   Terry Wilson
• [asterisk-dev] [Code Review] Don't try to lookup purely numeric hostnames that don't match peers or try to send requests to failed lookups   Terry Wilson
• [asterisk-dev] [Code Review] Add getnameinfo() to ast_sockaddr_resolve()   Terry Wilson
• [asterisk-dev] [Code Review] Asterisk Support of SIP Connect 1.1   Terry Wilson
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Terry Wilson
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   Terry Wilson
• [asterisk-dev] Potential 1.8 behavior change in realtime (especially for the CURL backend)   Terry Wilson
• [asterisk-dev] How to change length of variable   Terry Wilson
• [asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1   Terry Wilson
• [asterisk-dev] Google voice patch   Terry Wilson
• [asterisk-dev] [Code Review]: Add some realtime SIP tests via res_config_sqlite   Terry Wilson
• [asterisk-dev] SIP, NAT, security concerns, oh my!   Terry Wilson
• [asterisk-dev] [Code Review]: Preserve DTMF length in main/features.c   Terry Wilson
• [asterisk-dev] [Code Review] Make sure the member count is correct after a reload   Terry Wilson
• [asterisk-dev] [Code Review] Make sure the member count is correct after a reload   Terry Wilson
• [asterisk-dev] [Code Review]: Make sure the member count is correct after a reload   Terry Wilson
• [asterisk-dev] [Code Review] Make sure the member count is correct after a reload   Terry Wilson
• [asterisk-dev] [Code Review] Make sure the member count is correct after a reload   Terry Wilson
• [asterisk-dev] Replace voicemail PIN with voice biometric authentication   Nick Wise
• [asterisk-dev] Replace voicemail PIN with voice biometric authentication   Nick Wise
• [asterisk-dev] SHA1 and MD5 code?   Hans Witvliet
• [asterisk-dev] Google voice patch   sean darcy
• [asterisk-dev] [Code Review] Add PAUSED tag to queue log when adding queue member as paused   garlew
• [asterisk-dev] MRCP support for asterisk   ik
• [asterisk-dev] [Code Review] Allow dialplan control of faxdetection via faxdetect framehook / options in func CHANNEL   irroot
• [asterisk-dev] [Code Review] Allow dialplan control of faxdetection via faxdetect framehook / options in func CHANNEL   irroot
• [asterisk-dev] [Code Review]: Allow dialplan control of faxdetection via faxdetect framehook / options in func CHANNEL   irroot
• [asterisk-dev] [Code Review] Allow dialplan control of faxdetection via faxdetect framehook / options in func CHANNEL   irroot
• [asterisk-dev] [Code Review] Fix several bugs with SDP parsing and well-formedness of responses   irroot
• [asterisk-dev] [Code Review] Add option to check state when state is unknown and optomise the checking to reduce existing load.   irroot
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review]: app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review] Implement Externaddr on a sip device basis   irroot
• [asterisk-dev] [Code Review] "sipusers" only exists in documentation, and "sipfriends" is still used in documentation   irroot
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   irroot
• [asterisk-dev] [Code Review] Allow linking/building against an external editline library.   jcollie
• [asterisk-dev] [Code Review] Allow linking/building against an external editline library.   jcollie
• [asterisk-dev] [Code Review] Allow linking/building against an external editline library.   jcollie
• [asterisk-dev] [Code Review] Clean up reference leaks in res_jabber   jrose
• [asterisk-dev] [Code Review] Clean up reference leaks in res_jabber   jrose
• [asterisk-dev] [Code Review] menuselect by support level categories   jrose
• [asterisk-dev] [Code Review] menuselect by support level categories   jrose
• [asterisk-dev] [Code Review] Clean module info documentation tags for the purpose of menuselect category arrangement.   jrose
• [asterisk-dev] [Code Review] Preserve DTMF length in main/features.c   jrose
• [asterisk-dev] [Code Review]: Preserve DTMF length in main/features.c   jrose
• [asterisk-dev] [Code Review]: Preserve DTMF length in main/features.c   jrose
• [asterisk-dev] [Code Review] app_queue change to have one devstate per device not per member / Janitorial cleanups   jrose
• [asterisk-dev] [Code Review] Clean up reference leaks in res_jabber   jrose
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   jrose
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   jrose
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   jrose
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   jrose
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   jrose
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   jrose
• [asterisk-dev] [Code Review] SIP TLS Registration Test   jrose
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   jrose
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   jrose
• [asterisk-dev] [Code Review] Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.   jrose
• [asterisk-dev] [Code Review] Get CDR Congestion Logging Test (and improving whole concept behind CDRTestCase) working.   jrose
• [asterisk-dev] [Code Review] a driver for single-port FXO cards based on Si3052 chip + Si3011/17/18 DAA (Motorola 52-6116-2A, PM560MS etc.)   mhej
• [asterisk-dev] [Code Review] Check for ANY to place calls on hold for SIP   mjordan
• [asterisk-dev] [Code Review] Allow fine grained control over test version asterisk compatibility specification   mjordan
• [asterisk-dev] [Code Review] Add test execution modes to the Asterisk Testsuite   mjordan
• [asterisk-dev] [Code Review] Add test execution modes to the Asterisk Testsuite   mjordan
• [asterisk-dev] [Code Review] Add mute all participants; play participant count to ConfBridge   mjordan
• [asterisk-dev] [Code Review] TestSuite: Add tests for codec negotiation   mjordan
• [asterisk-dev] [Code Review] Fix for invalid memory access in adding extension to pattern match tree   mjordan
• [asterisk-dev] [Code Review] Fix for invalid memory access in adding extension to pattern match tree   mjordan
• [asterisk-dev] [Code Review] Clean up http.conf parsing   mjordan
• [asterisk-dev] [Code Review] Clean up cel.conf parsing   mjordan
• [asterisk-dev] [Code Review] Clean up logger.conf parsing   mjordan
• [asterisk-dev] [Code Review] Clean up udptl.conf parsing   mjordan
• [asterisk-dev] [Code Review]: Clean up http.conf parsing   mjordan
• [asterisk-dev] [Code Review] TestSuite: Add tests for codec negotiation   mjordan
• [asterisk-dev] [Code Review] Fix several bugs with SDP parsing and well-formedness of responses   mjordan
• [asterisk-dev] [Code Review]: TestSuite: Add tests for codec negotiation   mjordan
• [asterisk-dev] [Code Review] Add some realtime SIP tests via res_config_sqlite   mjordan
• [asterisk-dev] [Code Review]: Add some realtime SIP tests via res_config_sqlite   mjordan
• [asterisk-dev] [Code Review] TestSuite: Add tagging functionality to test framework   mjordan
• [asterisk-dev] [Code Review] ensure that ast_string_field_pool base + used is always aligned   mjordan
• [asterisk-dev] [Code Review] SIP TLS Registration Test   mjordan
• [asterisk-dev] [Code Review] Fix several bugs with SDP parsing and well-formedness of responses   opticron
• [asterisk-dev] [Code Review] Fix several bugs with SDP parsing and well-formedness of responses   opticron
• [asterisk-dev] [Code Review]: Fix several bugs with SDP parsing and well-formedness of responses   opticron
• [asterisk-dev] [Code Review] TestSuite: Add tests for codec negotiation   opticron
• [asterisk-dev] [Code Review]: TestSuite: Add tests for codec negotiation   opticron
• [asterisk-dev] [Code Review] TestSuite: Add tagging functionality to test framework   opticron
• [asterisk-dev] [Code Review] Fix several bugs with SDP parsing and well-formedness of responses   opticron
• [asterisk-dev] [Code Review]: Fix several bugs with SDP parsing and well-formedness of responses   opticron
• [asterisk-dev] [Code Review] TestSuite: Add tests for codec negotiation   opticron
• [asterisk-dev] [Code Review] ConfBridge User Announce Feature Set   opticron
• [asterisk-dev] [Code Review] Clean up reference leaks in res_jabber   opticron
• [asterisk-dev] [Code Review] Clean up reference leaks in res_jabber   opticron
• [asterisk-dev] [Code Review] Fix for invalid memory access in adding extension to pattern match tree   opticron
• [asterisk-dev] [Code Review] Modify merge message so commit log message comes first. (Another version)   rmudgett
• [asterisk-dev] [Code Review] svnmerge: avoid a separator when merging a single commit   rmudgett
• [asterisk-dev] [Code Review]: svnmerge: avoid a separator when merging a single commit   rmudgett
• [asterisk-dev] [Code Review] Clean up http.conf parsing   rmudgett
• [asterisk-dev] [Code Review]: Clean up http.conf parsing   rmudgett
• [asterisk-dev] [Code Review]: Clean up cel.conf parsing   rmudgett
• [asterisk-dev] [Code Review] fopen(REF_FILE) fails to check for success   rmudgett
• [asterisk-dev] [Code Review] Make sure the member count is correct after a reload   rmudgett
• [asterisk-dev] [Code Review] Make sure the member count is correct after a reload   rmudgett
• [asterisk-dev] [Code Review] fopen(REF_FILE) fails to check for success   rmudgett
• [asterisk-dev] [Code Review]: fopen(REF_FILE) fails to check for success   rmudgett
• [asterisk-dev] [Code Review]: fopen(REF_FILE) fails to check for success   rmudgett
• [asterisk-dev] [Code Review] Make sure the member count is correct after a reload   rmudgett
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   rmudgett
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   rmudgett
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   rmudgett
• [asterisk-dev] [Code Review] Astobj2 locking enhancement   rmudgett
• [asterisk-dev] [Code Review] Saving route header information even on a non-final response for proper UPDATE messages   schmidts
• [asterisk-dev] [Code Review] Saving route header information even on a non-final response for proper UPDATE messages   schmidts
• [asterisk-dev] [Code Review] Store route-set from provisional SIP responses so early-dialog requests can be routed properly   schmidts
• [asterisk-dev] [Code Review] Store route-set from provisional SIP responses so early-dialog requests can be routed properly   schmidts
• [asterisk-dev] [Code Review] didnt sent out an update request in an early dialog state if we dont have a contact header   schmidts
• [asterisk-dev] Asterisk does not hangup a channel after endpoint hangs up   tsengeljab tsogoo
• [asterisk-dev] [Code Review]: rewrite of matching code in chan_sip.c realtime_peer()   wdoekes
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   wdoekes
• [asterisk-dev] [Code Review]: rewrite of matching code in chan_sip.c realtime_peer()   wdoekes
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   wdoekes
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   wdoekes
• [asterisk-dev] [Code Review] Load the proper XML documentation when multiple modules document the same application   wdoekes
• [asterisk-dev] [Code Review]: Load the proper XML documentation when multiple modules document the same application   wdoekes
• [asterisk-dev] [Code Review] Load the proper XML documentation when multiple modules document the same application   wdoekes
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   wdoekes
• [asterisk-dev] [Code Review] Force consistency for ast_realtime_load_multientry when no rows are returned   wdoekes
• [asterisk-dev] [Code Review] "sipusers" only exists in documentation, and "sipfriends" is still used in documentation   wdoekes
• [asterisk-dev] [Code Review]: rewrite of matching code in chan_sip.c realtime_peer()   wdoekes
• [asterisk-dev] [Code Review] rewrite of matching code in chan_sip.c realtime_peer()   wdoekes
• [asterisk-dev] [Code Review] chan_sip: removing some non-compliant code in v10 and minor fixes   wdoekes
• [asterisk-dev] [Code Review] sparc64 compile error, undefined reference to `__sync_fetch_and_add_4'   wdoekes
• [asterisk-dev] [Code Review]: chan_sip: removing some non-compliant code in v10 and minor fixes   wdoekes
• [asterisk-dev] [Code Review] Allow linking/building against an external editline library.   wdoekes
• [asterisk-dev] [Code Review] fopen(REF_FILE) fails to check for success   wdoekes
• [asterisk-dev] [Code Review] chan_sip registertrying lacks a global option   wdoekes
• [asterisk-dev] [Code Review] fopen(REF_FILE) fails to check for success   wdoekes
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   wdoekes
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   wdoekes
• [asterisk-dev] [Code Review]: Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   wdoekes
• [asterisk-dev] [Code Review] Implement Externaddr on a sip device basis   wdoekes
• [asterisk-dev] [Code Review] Resolve Sequence Number Rollover causing codec increase in res_rtp_multicast   wdoekes
• [asterisk-dev] [Code Review] ensure that ast_string_field_pool base + used is always aligned   wdoekes
• [asterisk-dev] [Code Review] use autoconf to detect whether alignment should be enforced   wdoekes
• [asterisk-dev] [Code Review] ensure that ast_string_field_pool base + used is always aligned   wdoekes
• [asterisk-dev] [Code Review] "sipusers" only exists in documentation, and "sipfriends" is still used in documentation   wdoekes
• [asterisk-dev] [Code Review] Reject v17 skinny devices in Asterisk10   wedhorn
• [asterisk-dev] [Code Review] Add hold status with CEL and setting a channel var HOLDING via ast_moh_stop/start   whardier

Last message date: Mon Oct 31 22:18:51 CDT 2011
Archived on: Tue Nov 1 08:18:38 CDT 2011

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