Mail Index

• Thread Index

• Re: [asterisk-dev] Apple TV and Asterisk - who was it?
  • From: Johansson Olle E
• [asterisk-dev] dahdi static device files
  • From: Tzafrir Cohen
• Re: [asterisk-dev] Apple TV and Asterisk - who was it?
  • From: John Todd
• Re: [asterisk-dev] dahdi static device files
  • From: Philipp Kempgen
• Re: [asterisk-dev] dahdi static device files
  • From: Tzafrir Cohen
• [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX
  • From: Joseph Benden
• Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX
  • From: Felipe Bergo
• Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX
  • From: Joseph Benden
• Re: [asterisk-dev] AES encryption in IAX2
  • From: Tim Panton
• Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX
  • From: Steve Underwood
• Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX
  • From: Stelios Koroneos
• Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX
  • From: Steve Underwood
• Re: [asterisk-dev] [Code Review] Add support in AEL for macro return values and direct assignment of them to variables and functions.
  • From: Watkins, Bradley
• Re: [asterisk-dev] [Code Review] Add support in AEL for macro return values and direct assignment of them to variables and functions.
  • From: Steve Murphy
• Re: [asterisk-dev] [Code Review] New queue CLI and Manager commands to facilitate fine-grained reloading
  • From: Mark Michelson
• Re: [asterisk-dev] [design] Realtime changes
  • From: Tilghman Lesher
• Re: [asterisk-dev] [design] Realtime changes
  • From: Atis Lezdins
• [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] [design] Realtime changes
  • From: Dmitry Andrianov
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: John Todd
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Joshua Colp
• Re: [asterisk-dev] [design] Realtime changes
  • From: Tilghman Lesher
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Gregory Boehnlein
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Joshua Colp
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4
  • From: Mark Michelson
• [asterisk-dev] Asterisk CLI got freezed!!
  • From: Max Alex
• Re: [asterisk-dev] Asterisk CLI got freezed!!
  • From: Steve Howes
• [asterisk-dev] External MWI & IMAP
  • From: Jay R. Worthington
• Re: [asterisk-dev] [design] Realtime changes
  • From: Matthew Nicholson
• Re: [asterisk-dev] [design] Realtime changes
  • From: Atis Lezdins
• [asterisk-dev] Interesting SVN graphical representations of Asterisk project
  • From: John Todd
• [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Kevin Fleming
• Re: [asterisk-dev] Interesting SVN graphical representations of Asterisk project
  • From: Tzafrir Cohen
• [asterisk-dev] Dean Collins added you as a business connection on Plaxo
  • From: Dean Collins
• Re: [asterisk-dev] Dean Collins added you as a business connection on Plaxo
  • From: Steve Totaro
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Johansson Olle E
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Klaus Darilion
• Re: [asterisk-dev] [design] Realtime changes
  • From: Kevin P. Fleming
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Kevin P. Fleming
• Re: [asterisk-dev] Dean Collins added you as a business connection on Plaxo
  • From: Dean Collins
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Klaus Darilion
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Dmitry Andrianov
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Dmitry Andrianov
• Re: [asterisk-dev] Interesting SVN graphical representations of Asterisk project
  • From: Jeff Peeler
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Fernando Urzedo
• [asterisk-dev] Events generated by followme
  • From: Mirko Tochtermann
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Kevin P. Fleming
• Re: [asterisk-dev] Events generated by followme
  • From: BJ Weschke
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Kevin Fleming
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Kevin Fleming
• [asterisk-dev] RFC: Capacity Study
  • From: Kristian Kielhofner
• Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed.
  • From: Joshua Colp
• Re: [asterisk-dev] wctdm 4/8/24 VMWI generation capabilities
  • From: Alec Davis
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] [design] Realtime changes
  • From: Tilghman Lesher
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] RFC: Capacity Study
  • From: Alex Balashov
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Joshua Colp
• [asterisk-dev] OT: Paying people in faraway (Western) places.
  • From: Alex Balashov
• Re: [asterisk-dev] OT: Paying people in faraway (Western) places.
  • From: Alex Balashov
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] RFC: Capacity Study
  • From: Tilghman Lesher
• [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk
  • From: Kevin Fleming
• Re: [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk
  • From: Russell Bryant
• Re: [asterisk-dev] OT: Paying people in faraway (Western) places.
  • From: Chris Mason (Lists)
• [asterisk-dev] No emails from Mantis
  • From: Benny Amorsen
• Re: [asterisk-dev] No emails from Mantis
  • From: Leif Madsen
• [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Klaus Darilion
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Leif Madsen
• Re: [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk
  • From: Kevin Fleming
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Nir Simionovich
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Klaus Darilion
• Re: [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk
  • From: Russell Bryant
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Nir Simionovich
• [asterisk-dev] 2008 Post Count
  • From: Tilghman Lesher
• [asterisk-dev] Asterisk 1.2.31, 1.4.22.1, and 1.6.0.3 released
  • From: Asterisk Development Team
• [asterisk-announce] AST-2009-001: Information leak in IAX2 authentication
  • From: Asterisk Security Team
• [asterisk-dev] AST-2009-001: Information leak in IAX2 authentication
  • From: Asterisk Security Team
• Re: [asterisk-dev] 2008 Post Count
  • From: Steven S. Critchfield
• Re: [asterisk-dev] 2008 Post Count
  • From: Russell Bryant
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Kristian Kielhofner
• Re: [asterisk-dev] RTP interop with Sonus: hack
  • From: Joshua Colp
• [asterisk-dev] SRTP and Dialplan control
  • From: Terry Wilson
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Leif Madsen
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Dmitry Andrianov
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Benny Amorsen
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Klaus Darilion
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Philipp Kempgen
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Nir Simionovich
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Leif Madsen
• [asterisk-dev] Request for Testing: Issue 12312, DNS SRV lookups causing re-registrations problems
  • From: Leif Madsen
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Dmitry Andrianov
• [asterisk-dev] [Code Review] [bug] Asterisk does not choose same SRV record when attempting to re-register
  • From: Mark Michelson
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Dmitry Andrianov
• Re: [asterisk-dev] [Code Review] [bug] Asterisk does not choose same SRV record when attempting to re-register
  • From: Mark Michelson
• [asterisk-dev] make config update-rc.d on Debian
  • From: Philipp Kempgen
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Tzafrir Cohen
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Dmitry Andrianov
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Steve Murphy
• Re: [asterisk-dev] [Code Review] MFC/R2 support for chan_dahdi
  • From: Moises Silva
• [asterisk-dev] origsvn.digium.com maintenance
  • From: Kevin P. Fleming
• [asterisk-dev] Looking for commit that fixed bug #14139, similar to #14173
  • From: Alex Villací­s Lasso
• [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access
  • From: Alex Villací­s Lasso
• Re: [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access
  • From: John Todd
• [asterisk-dev] origsvn.digium.com maintenance completed
  • From: Kevin P. Fleming
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Benny Amorsen
• Re: [asterisk-dev] i extension does not match on initial context - bug or not?
  • From: Benny Amorsen
• [asterisk-dev] 1.6.0.3 chan_dahdi - can not dial FXS until call is first recieved - including fix
  • From: Karl Fife
• Re: [asterisk-dev] OT: Paying people in faraway (Western) places.
  • From: Nir Simionovich
• Re: [asterisk-dev] 1.6.0.3 chan_dahdi - can not dial FXS until call is first recieved - including fix
  • From: Tzafrir Cohen
• Re: [asterisk-dev] Looking for commit that fixed bug #14139, similar to #14173
  • From: Rajkumar S
• Re: [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access
  • From: Leif Madsen
• Re: [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access
  • From: Alex Villací­s Lasso
• Re: [asterisk-dev] Looking for commit that fixed bug #14139, similar to #14173
  • From: Alex Villací­s Lasso
• [asterisk-dev] Patch: wcfxo; Failed to initailize DAA, giving up...
  • From: Timothy Allen
• [asterisk-dev] Asterisk burns 100% cpu when pressing "#" in Voicemail
  • From: John Lange
• [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?
  • From: Tony Mountifield
• Re: [asterisk-dev] Asterisk burns 100% cpu when pressing "#" in Voicemail
  • From: Matthew Nicholson
• Re: [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?
  • From: Russell Bryant
• [asterisk-dev] Bug 14153 was closed by accident [Fwd: Re: [asterisk-users] bug 14153 and svn checkout.]
  • From: Philipp Kempgen
• Re: [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?
  • From: David A. Burgess
• Re: [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?
  • From: Tony Mountifield
• Re: [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)?
  • From: Tony Mountifield
• Re: [asterisk-dev] Bug 14153 was closed by accident [Fwd: Re: [asterisk-users] bug 14153 and svn checkout.]
  • From: Mark Michelson
• Re: [asterisk-dev] UDPTL crash anyone?
  • From: Atis Lezdins
• Re: [asterisk-dev] UDPTL crash anyone?
  • From: Steve Underwood
• Re: [asterisk-dev] UDPTL crash anyone?
  • From: Atis Lezdins
• Re: [asterisk-dev] Asterisk burns 100% cpu when pressing "#" in Voicemail
  • From: Steve Mathers
• Re: [asterisk-dev] seanbright: tools/trunk r5656 - /tools/trunk/dahdi_monitor.c
  • From: Russell Bryant
• Re: [asterisk-dev] seanbright: tools/trunk r5656 - /tools/trunk/dahdi_monitor.c
  • From: Sean Bright
• Re: [asterisk-dev] seanbright: tools/trunk r5656 - /tools/trunk/dahdi_monitor.c
  • From: Kevin P. Fleming
• [asterisk-dev] reset/remove SipAddHeader() headers
  • From: Klaus Darilion
• Re: [asterisk-dev] seanbright: tools/trunk r5656 - /tools/trunk/dahdi_monitor.c
  • From: Russell Bryant
• [asterisk-dev] strange spiral handling in chan_sip
  • From: Klaus Darilion
• Re: [asterisk-dev] reset/remove SipAddHeader() headers
  • From: John Todd
• Re: [asterisk-dev] strange spiral handling in chan_sip
  • From: Mark Michelson
• Re: [asterisk-dev] G.729.1 - any interest?
  • From: Kevin P. Fleming
• [asterisk-dev] G.729.1 - any interest?
  • From: John Todd
• Re: [asterisk-dev] G.729.1 - any interest?
  • From: Kristian Kielhofner
• Re: [asterisk-dev] G.729.1 - any interest?
  • From: Stéphane Van Geystelen
• [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree
  • From: Philip Prindeville
• Re: [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree
  • From: Tilghman Lesher
• Re: [asterisk-dev] G.729.1 - any interest?
  • From: Stéphane Van Geystelen
• Re: [asterisk-dev] G.729.1 - any interest?
  • From: Dmitry Andrianov
• Re: [asterisk-dev] reset/remove SipAddHeader() headers
  • From: Klaus Darilion
• Re: [asterisk-dev] G.729.1 - any interest?
  • From: Kevin P. Fleming
• [asterisk-dev] calls problem
  • From: Ali Haydar GÜMÜŞ
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps (was: oej: trunk r168638 - /trunk/channels/chan_sip.c)
  • From: Russell Bryant
• Re: [asterisk-dev] calls problem
  • From: Russell Bryant
• Re: [asterisk-dev] calls problem
  • From: Ali Haydar GÜMÜŞ
• Re: [asterisk-dev] calls problem
  • From: Russell Bryant
• Re: [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree
  • From: Philip Prindeville
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Klaus Darilion
• Re: [asterisk-dev] reset/remove SipAddHeader() headers
  • From: Klaus Darilion
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Johansson Olle E
• Re: [asterisk-dev] reset/remove SipAddHeader() headers
  • From: Klaus Darilion
• Re: [asterisk-dev] reset/remove SipAddHeader() headers
  • From: Leif Madsen
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Tilghman Lesher
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Johansson Olle E
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Russell Bryant
• [asterisk-dev] codec preference
  • From: David Hansen
• Re: [asterisk-dev] codec preference
  • From: Adam Moffett
• Re: [asterisk-dev] codec preference
  • From: David Hansen
• [asterisk-dev] Asterisk 1.4.23-rc4 Now Available
  • From: Asterisk Development Team
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Benny Amorsen
• [asterisk-dev] Trying to do a transfer
  • From: David Jones
• [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer
  • From: Mark Michelson
• [asterisk-dev] Reminder: Emerging Communications Conference Early-Bird
  • From: Lee S Dryburgh
• [asterisk-dev] development possibilities with existing SW
  • From: Steven Hoffmann
• Re: [asterisk-dev] development possibilities with existing SW
  • From: Moises Silva
• Re: [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail
  • From: Klaus Darilion
• Re: [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail
  • From: Tzafrir Cohen
• [asterisk-dev] AST_LIST_LOCK/UNLOCK
  • From: Chandrakant Solanki
• [asterisk-dev] AEL macro backward compatibility for 1.6
  • From: Dmitry Andrianov
• [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail
  • From: Klaus Darilion
• Re: [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail
  • From: Klaus Darilion
• Re: [asterisk-dev] AEL macro backward compatibility for 1.6
  • From: Steve Murphy
• Re: [asterisk-dev] AEL macro backward compatibility for 1.6
  • From: Dmitry Andrianov
• Re: [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer
  • From: Kevin Fleming
• Re: [asterisk-dev] AST_LIST_LOCK/UNLOCK
  • From: Tilghman Lesher
• Re: [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer
  • From: Mark Michelson
• Re: [asterisk-dev] reset/remove SipAddHeader() headers
  • From: Klaus Darilion
• Re: [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer
  • From: Kevin Fleming
• Re: [asterisk-dev] dahdi sysfs branch
  • From: Tzafrir Cohen
• Re: [asterisk-dev] dahdi sysfs branch
  • From: Shaun Ruffell
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Tilghman Lesher
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Chris Tooley
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Klaus Darilion
• [asterisk-dev] [0014275] Group does not count all channels
  • From: Marcin J. Kowalczyk
• Re: [asterisk-dev] dahdi sysfs branch
  • From: Tzafrir Cohen
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Tilghman Lesher
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Chris Tooley
• Re: [asterisk-dev] [0014275] Group does not count all channels
  • From: John Todd
• [asterisk-dev] Minimum packages for building and running asterisk
  • From: Julian Lyndon-Smith
• Re: [asterisk-dev] Minimum packages for building and running asterisk
  • From: Russell Bryant
• [asterisk-dev] How ti let execution continue to the next priorité after callee or caller hangup
  • From: Paul Isidore
• Re: [asterisk-dev] How ti let execution continue to the next priorité after callee or caller hangup
  • From: Dmitry Andrianov
• [asterisk-dev] app_queue, ringinuse, joinempty combination problems
  • From: Wolfgang Pichler
• Re: [asterisk-dev] SIPAdd/RemoveHeader apps
  • From: Benny Amorsen
• [asterisk-dev] SIP channel/owner question
  • From: Klaus Darilion
• Re: [asterisk-dev] SIP channel/owner question
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] properly report ast_func_read errors in getvar AMI action
  • From: Russell Bryant
• Re: [asterisk-dev] [svn-commits] tilghman: trunk r97651 - in /trunk: ./ apps/ channels/ configs/ include/aster...
  • From: Marcus Hunger
• Re: [asterisk-dev] app_queue, ringinuse, joinempty combination problems
  • From: Mark Michelson
• [asterisk-dev] Access to encryption functions from dial plan
  • From: Benny Amorsen
• Re: [asterisk-dev] app_queue, ringinuse, joinempty combination problems
  • From: Wolfgang Pichler
• Re: [asterisk-dev] Access to encryption functions from dial plan
  • From: Russell Bryant
• [asterisk-dev] Asterisk 1.4.23 Now Available!
  • From: Asterisk Development Team
• Re: [asterisk-dev] Access to encryption functions from dial plan
  • From: Benny Amorsen
• Re: [asterisk-dev] Access to encryption functions from dial plan
  • From: Leif Madsen
• Re: [asterisk-dev] Access to encryption functions from dial plan
  • From: Benny Amorsen
• Re: [asterisk-dev] Asterisk 1.4.23 Now Available!
  • From: John Lange
• Re: [asterisk-dev] Asterisk 1.4.23 Now Available!
  • From: Sean Bright
• [asterisk-dev] ChannelRedirect and channels on Dial
  • From: Gabriel Ortiz Lour
• [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile
  • From: Matthew Nicholson
• [asterisk-dev] Asterisk 1.6.0.4 Release Candidate 1 Now Available
  • From: Asterisk Development Team
• Re: [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions
  • From: Russell Bryant
• [asterisk-dev] Welcome to David Vossel
  • From: Russell Bryant
• Re: [asterisk-dev] BugID 014021, Zap/DAHDI timers, internal timing and packetization
  • From: Dan Austin
• Re: [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile
  • From: Vlasis Hatzistavrou (KTI)
• Re: [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile
  • From: Matthew Nicholson
• [asterisk-dev] X-Asterisk-HangupCause header only added in auto fallthrough mode
  • From: Klaus Darilion
• Re: [asterisk-dev] X-Asterisk-HangupCause header only added in auto fallthrough mode
  • From: Johansson Olle E
• Re: [asterisk-dev] Welcome to David Vossel
  • From: Leif Madsen
• Re: [asterisk-dev] SIP channel/owner question
  • From: Klaus Darilion
• Re: [asterisk-dev] SIP channel/owner question
  • From: Miguel Molina
• Re: [asterisk-dev] SIP channel/owner question
  • From: Johansson Olle E
• Re: [asterisk-dev] SIP channel/owner question
  • From: Klaus Darilion
• Re: [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile
  • From: Steve Murphy
• Re: [asterisk-dev] Welcome to David Vossel
  • From: Steven S. Critchfield
• [asterisk-dev] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
  • From: Asterisk Development Team
• Re: [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile
  • From: David Bowerman
• Re: [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions
  • From: Russell Bryant
• [asterisk-dev] New AGI manager command: PlaySound
  • From: nik600
• Re: [asterisk-dev] New AGI manager command: PlaySound
  • From: Russell Bryant
• Re: [asterisk-dev] New AGI manager command: PlaySound
  • From: nik600
• Re: [asterisk-dev] New AGI manager command: PlaySound
  • From: Moises Silva
• [asterisk-dev] Call Detection
  • From: David Phillips
• Re: [asterisk-dev] New AGI manager command: PlaySound
  • From: Nir Simionovich
• Re: [asterisk-dev] [Code Review] This patch implements CEL in trunk.
  • From: Russell Bryant
• [asterisk-dev] Newbie Developer Resource?
  • From: David Phillips
• Re: [asterisk-dev] Newbie Developer Resource?
  • From: Sergey Okhapkin
• [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread
  • From: Russell Bryant
• [asterisk-dev] p2p asterisk/gtalk
  • From: Mark Spowage
• Re: [asterisk-dev] Call Detection
  • From: Nir Simionovich
• Re: [asterisk-dev] Newbie Developer Resource?
  • From: David Phillips
• Re: [asterisk-dev] Newbie Developer Resource?
  • From: Grey Man
• Re: [asterisk-dev] Newbie Developer Resource?
  • From: David Phillips
• Re: [asterisk-dev] Newbie Developer Resource?
  • From: Grey Man
• Re: [asterisk-dev] Newbie Developer Resource?
  • From: John Todd
• [asterisk-dev] Using SIP REFER with a non bridged channel - is there a workaround?
  • From: Dan Julius
• Re: [asterisk-dev] Newbie Developer Resource?
  • From: Nicolas Chapleau
• [asterisk-dev] [Code Review] unload/load/reload support for chan_skinny
  • From: Michiel van Baak
• Re: [asterisk-dev] Using SIP REFER with a non bridged channel - is there a workaround?
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] unload/load/reload support for chan_skinny
  • From: Russell Bryant
• Re: [asterisk-dev] New AGI manager command: PlaySound
  • From: Tilghman Lesher
• [asterisk-dev] Answering Machine Detection (was: Re: Call Detection)
  • From: Philipp Kempgen
• [asterisk-dev] [Code Review] Resolve some race conditions in chan_iax2 scheduler handling
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Calendaring API for Asterisk
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Calendaring API for Asterisk
  • From: Russell Bryant
• Re: [asterisk-dev] New AGI manager command: PlaySound
  • From: nik600
• Re: [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree
  • From: Philip A. Prindeville
• Re: [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Resolve some race conditions in chan_iax2 scheduler handling
  • From: Mark Michelson
• Re: [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile
  • From: Steve Murphy
• Re: [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread
  • From: Russell Bryant
• [asterisk-dev] [Code Review] Patch for issue 12312 (DNS SRV lookups causing re-registration problems)
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] Patch for issue 12312 (DNS SRV lookups causing re-registration problems)
  • From: Johansson Olle E
• [asterisk-dev] p2p asterisk action
  • From: Mark Spowage
• Re: [asterisk-dev] [Code Review] Patch for issue 12312 (DNS SRV lookups causing re-registration problems)
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread
  • From: Russell Bryant
• Re: [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile
  • From: Matthew Nicholson
• Re: [asterisk-dev] [Code Review] This patch implements CEL in trunk.
  • From: Steve Murphy
• Re: [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread
  • From: Russell Bryant
• Re: [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile
  • From: David Bowerman
• Re: [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread
  • From: Russell Bryant
• [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails
  • From: Jeff Peeler
• [asterisk-dev] DTMF queuing
  • From: James Lamanna
• Re: [asterisk-dev] DTMF queuing
  • From: Russell Bryant
• Re: [asterisk-dev] DTMF queuing
  • From: James Lamanna
• Re: [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] properly report ast_func_read errors in getvar AMI action
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] This patch implements CEL in trunk.
  • From: Steve Murphy
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] This patch implements CEL in trunk.
  • From: Russell Bryant
• Re: [asterisk-dev] DTMF queuing
  • From: John Todd
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Joshua Colp
• Re: [asterisk-dev] [Code Review] properly report ast_func_read errors in getvar AMI action
  • From: Russell Bryant
• [asterisk-dev] SMDI Function Call Redux
  • From: Bill Sofko
• Re: [asterisk-dev] SMDI Function Call Redux
  • From: Russell Bryant
• Re: [asterisk-dev] SMDI Function Call Redux
  • From: Bill Sofko
• Re: [asterisk-dev] SMDI Function Call Redux
  • From: Russell Bryant
• [asterisk-dev] chan_sip SIP Authentication
  • From: asterisk
• Re: [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] added AES_ENCRYPT and AES_DECRYPT dialplan functions
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails
  • From: Jeff Peeler
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Philipp Kempgen
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Philipp Kempgen
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Benny Amorsen
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Jesus Rodriguez
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: asterisk
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• [asterisk-dev] Multiple Server
  • From: Chandrakant Solanki
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Joshua Colp
• [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features
  • From: Terry Wilson
• Re: [asterisk-dev] Multiple Server
  • From: Jeff Peeler
• Re: [asterisk-dev] Multiple Server
  • From: Chris Tooley
• Re: [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] Backport of state_interface for app_queue in Asterisk 1.4
  • From: Atis Lezdins
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Jesus Rodriguez
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Jesus Rodriguez
• Re: [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features
  • From: Terry Wilson
• Re: [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features
  • From: Terry Wilson
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Terry Wilson
• Re: [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features
  • From: Mark Michelson
• [asterisk-dev] Asterisk 1.6.1 Release Candidate 1 Now Available
  • From: Asterisk Development Team
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Mark Michelson
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Philipp Kempgen
• Re: [asterisk-dev] [Code Review] unload/load/reload support for chan_skinny
  • From: Dan Austin
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Benny Amorsen
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Joshua Colp
• Re: [asterisk-dev] [Code Review] unload/load/reload support for chan_skinny
  • From: Russell Bryant
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Mark Michelson
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Benny Amorsen
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• [asterisk-dev] Asterisk RFC2833 SDP fmtp 0-16?
  • From: Kristian Kielhofner
• Re: [asterisk-dev] Asterisk RFC2833 SDP fmtp 0-16?
  • From: Johansson Olle E
• [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: Gregory Boehnlein
• Re: [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: Kevin P. Fleming
• Re: [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: Gregory Boehnlein
• Re: [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: Gregory Boehnlein
• Re: [asterisk-dev] DTMF queuing
  • From: James Lamanna
• Re: [asterisk-dev] jpeeler: branch 1.0 r465 - in /branches/1.0: menuselect.c menuselect.h
  • From: Russell Bryant
• Re: [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: Johansson Olle E
• [asterisk-dev] ast_careful_fwrite: fflush() returned error: Broken pipe
  • From: Chris Maciejewski
• Re: [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: Benny Amorsen
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Joshua Colp
• Re: [asterisk-dev] Asterisk RFC2833 SDP fmtp 0-16?
  • From: Leif Madsen
• Re: [asterisk-dev] ast_careful_fwrite: fflush() returned error: Broken pipe
  • From: Russell Bryant
• Re: [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: Gregory Boehnlein
• Re: [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: James Golovich
• Re: [asterisk-dev] chan_sip.c - no reply to critical packet
  • From: Gregory Boehnlein
• [asterisk-dev] Need to start in development
  • From: bilal ghayyad
• Re: [asterisk-dev] Need to start in development
  • From: Michiel van Baak
• Re: [asterisk-dev] ast_careful_fwrite: fflush() returned error: Broken pipe
  • From: Chris Maciejewski
• [asterisk-dev] Quoting in AGI data
  • From: Alistair Cunningham
• Re: [asterisk-dev] Quoting in AGI data
  • From: Johansson Olle E
• Re: [asterisk-dev] Quoting in AGI data
  • From: Alistair Cunningham
• Re: [asterisk-dev] Quoting in AGI data
  • From: Philipp Kempgen
• [asterisk-dev] IT Expo, Miami FL
  • From: Nir Simionovich
• Re: [asterisk-dev] Quoting in AGI data
  • From: Mark Michelson
• [asterisk-dev] T38 and digium 420
  • From: Clara Chan
• Re: [asterisk-dev] Quoting in AGI data
  • From: Alistair Cunningham
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Vadim Lebedev
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Joshua Colp
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Vadim Lebedev
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Joshua Colp
• Re: [asterisk-dev] T38 and digium 420
  • From: Matt Riddell
• Re: [asterisk-dev] T38 and digium 420
  • From: Clara Chan
• Re: [asterisk-dev] [svn-commits] seanbright: trunk r172778 - /trunk/channels/chan_sip.c
  • From: Johansson Olle E
• Re: [asterisk-dev] [svn-commits] r172778 - svn:log
  • From: Johansson Olle E
• [asterisk-dev] menuselect.h change to check asterisk.makeopts
  • From: Johansson Olle E
• [asterisk-dev] Implementing priodic announces in Dial
  • From: Saúl Ibarra
• Re: [asterisk-dev] Implementing priodic announces in Dial
  • From: Grey Man
• Re: [asterisk-dev] Implementing priodic announces in Dial
  • From: Saúl Ibarra
• Re: [asterisk-dev] Implementing priodic announces in Dial
  • From: Nicolas
• Re: [asterisk-dev] Implementing priodic announces in Dial
  • From: Saúl Ibarra
• [asterisk-dev] Request for Testing: Auto-register realtime contexts
  • From: Leif Madsen
• Re: [asterisk-dev] strange spiral handling in chan_sip
  • From: NeoTel Lists
• Re: [asterisk-dev] [svn-commits] oej: trunk r172818 - /trunk/channels/chan_sip.c
  • From: Mark Michelson
• Re: [asterisk-dev] strange spiral handling in chan_sip
  • From: Mark Michelson
• Re: [asterisk-dev] IT Expo, Miami FL
  • From: Jeff Gehlbach
• [asterisk-dev] Wrong REFER handling
  • From: Martin Drasar
• Re: [asterisk-dev] [svn-commits] oej: trunk r172818 - /trunk/channels/chan_sip.c
  • From: Kevin P. Fleming
• Re: [asterisk-dev] menuselect.h change to check asterisk.makeopts
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Russell Bryant
• [asterisk-dev] Problems with Firewall
  • From: Karsten Asche
• Re: [asterisk-dev] Problems with Firewall
  • From: Steve Totaro
• [asterisk-dev] audio in console using /dev/dsp via OSS
  • From: Robert Prince
• Re: [asterisk-dev] Problems with Firewall
  • From: Steve Totaro
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Vadim Lebedev
• Re: [asterisk-dev] Problems with Firewall
  • From: Karsten Asche
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Russell Bryant
• Re: [asterisk-dev] Problems with Firewall
  • From: Karsten Asche
• Re: [asterisk-dev] jpeeler: branch 1.0 r465 - in /branches/1.0: menuselect.c menuselect.h
  • From: Jeff Peeler
• Re: [asterisk-dev] Implementing priodic announces in Dial
  • From: Leif Madsen
• Re: [asterisk-dev] Wrong REFER handling
  • From: Leif Madsen
• Re: [asterisk-dev] Implementing priodic announces in Dial
  • From: Mark Michelson
• Re: [asterisk-dev] [svn-commits] oej: trunk r172818 - /trunk/channels/chan_sip.c
  • From: Johansson Olle E
• Re: [asterisk-dev] Implementing priodic announces in Dial
  • From: Russell Bryant
• Re: [asterisk-dev] murf: trunk r172890 - in /trunk: ./ apps/ include/asterisk/ main/
  • From: Russell Bryant
• Re: [asterisk-dev] [svn-commits] rmudgett: branch 1.4 r172962 - in /branches/1.4: channels/ configs/
  • From: Johansson Olle E
• Re: [asterisk-dev] [svn-commits] rmudgett: branch 1.4 r172962 - in /branches/1.4: channels/ configs/
  • From: Michiel van Baak
• [asterisk-dev] [Code Review] set ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path
  • From: Michiel van Baak
• Re: [asterisk-dev] mmichelson: trunk r173028 - in /trunk: CHANGES main/manager.c
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Joshua Colp
• [asterisk-dev] How To - Close IAX2 channel when there has been a silence in sending and receiving for x number of minutes
  • From: Chandrakant Solanki
• Re: [asterisk-dev] [svn-commits] rmudgett: branch 1.4 r172962 - in /branches/1.4: channels/ configs/
  • From: Johansson Olle E
• Re: [asterisk-dev] [svn-commits] mmichelson: manager logout cli
  • From: Johansson Olle E
• Re: [asterisk-dev] Implementing priodic announces in Dial
  • From: Saúl Ibarra
• Re: [asterisk-dev] strange spiral handling in chan_sip
  • From: Klaus Darilion
• Re: [asterisk-dev] IT Expo, Miami FL
  • From: Nir Simionovich
• Re: [asterisk-dev] [asterisk-commits] tilghman: trunk r173130 - in /trunk: ./ apps/ include/asterisk/ main/
  • From: Kevin P. Fleming
• Re: [asterisk-dev] [svn-commits] mmichelson: manager logout cli
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] inotify support for checking timezone file modification
  • From: Russell Bryant
• [asterisk-dev] dahdi-linux 2.1.0.4 released
  • From: Asterisk Team
• Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] IAX2 Transfer Fix
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] IAX2 Transfer Fix
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] Keep bridge up if parking attempt fails
  • From: Russell Bryant
• Re: [asterisk-dev] [Code Review] IAX2 Transfer Fix
  • From: David Vossel
• [asterisk-dev] Update: Asterisk DTMF issues with Sonus
  • From: Kristian Kielhofner
• Re: [asterisk-dev] Update: Asterisk DTMF issues with Sonus
  • From: Joshua Colp
• Re: [asterisk-dev] [Code Review] IAX2 Transfer Fix
  • From: Russell Bryant
• [asterisk-dev] Close IAX2 channel when there has been a silence
  • From: Chandrakant Solanki
• Re: [asterisk-dev] Close IAX2 channel when there has been a silence
  • From: Steven S. Critchfield
• [asterisk-dev] Looking for an Asterisk Developer
  • From: John Marosi - Director Technology Operations
• Re: [asterisk-dev] Update: Asterisk DTMF issues with Sonus
  • From: Kristian Kielhofner
• [asterisk-dev] eComm developer "scholarships" available
  • From: Lee S Dryburgh
• [asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus
  • From: Gregory Boehnlein
• Re: [asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus
  • From: Kristian Kielhofner
• Re: [asterisk-dev] Close IAX2 channel when there has been a silence
  • From: Chandrakant Solanki
• Re: [asterisk-dev] Update: Asterisk DTMF issues with Sonus
  • From: Maxim Sobolev
• Re: [asterisk-dev] Update: Asterisk DTMF issues with Sonus
  • From: Maxim Sobolev
• Re: [asterisk-dev] Close IAX2 channel when there has been a silence
  • From: Chandrakant Solanki
• [asterisk-dev] AEL and switch breaks ${EXTEN}, bug in AEL compiler?
  • From: Klaus Darilion
• [asterisk-dev] 1.4 and CDRs -- The Breaking Point
  • From: Steve Murphy
• [asterisk-dev] [Code Review] Limit addition of Contact header in SIP responses
  • From: Matthew Nicholson
• [asterisk-dev] Asterisk B2BUA Media Offer Handling
  • From: Matthew Nicholson
• Re: [asterisk-dev] [Code Review] Limit addition of Contact header in SIP responses
  • From: Mark Michelson
• Re: [asterisk-dev] [Code Review] Patch for issue 12312 (DNS SRV lookups causing re-registration problems)
  • From: Mark Michelson
• [asterisk-dev] [Code Review] Adding immediate yes/no option to iax.conf
  • From: David Vossel
• Re: [asterisk-dev] [Code Review] Limit addition of Contact header in SIP responses
  • From: Klaus Darilion
• [asterisk-dev] Destroy Mini Frames (IAX2)
  • From: Chandrakant Solanki
• Re: [asterisk-dev] 1.4 and CDRs -- The Breaking Point
  • From: Johansson Olle E
• Re: [asterisk-dev] [Code Review] Limit addition of Contact header in SIP responses
  • From: Olle E Johansson
• [asterisk-dev] RxFax doesn't receive anything
  • From: tylmann
• Re: [asterisk-dev] RxFax doesn't receive anything
  • From: Hoggins!
• [asterisk-dev] SRTP support on 1.4.23.1 ?
  • From: Serge Berney
• Re: [asterisk-dev] SRTP support on 1.4.23.1 ?
  • From: Johansson Olle E
• Re: [asterisk-dev] [Code Review] Adding immediate yes/no option to iax.conf
  • From: Russell Bryant
• Re: [asterisk-dev] SRTP support on 1.4.23.1 ?
  • From: Serge Berney
• [asterisk-dev] Hague declaration
  • From: Vadim Lebedev
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Johansson Olle E
• Re: [asterisk-dev] SRTP support on 1.4.23.1 ?
  • From: Mark Michelson
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Klaus Darilion
• Re: [asterisk-dev] RxFax doesn't receive anything
  • From: tylmann
• Re: [asterisk-dev] 1.4 and CDRs -- The Breaking Point
  • From: John Lange
• Re: [asterisk-dev] RxFax doesn't receive anything
  • From: Jeff Gehlbach
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Philipp Kempgen
• Re: [asterisk-dev] chan_sip SIP Authentication
  • From: Michael Neuhauser
• Re: [asterisk-dev] [Code Review] set ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path
  • From: Matthew Nicholson
• [asterisk-dev] [policy] Language specific prompts
  • From: Tilghman Lesher
• Re: [asterisk-dev] [Code Review] Adding immediate yes/no option to iax.conf
  • From: David Vossel
• [asterisk-dev] [Code Review] Improve behavior of jitterbuffer when maxjitterbuffer is set
  • From: Matthew Nicholson
• Re: [asterisk-dev] [Code Review] Improve behavior of jitterbuffer when maxjitterbuffer is set
  • From: Matthew Nicholson
• Re: [asterisk-dev] [policy] Language specific prompts
  • From: Tzafrir Cohen
• Re: [asterisk-dev] [policy] Language specific prompts
  • From: Philipp Kempgen