| Re: [asterisk-dev] dahdi static device files - Tzafrir Cohen |
| [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX - Joseph Benden |
| Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX - Joseph Benden |
| Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX - Felipe Bergo |
| Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX - Stelios Koroneos |
| Re: [asterisk-dev] GPU Audio Codec Transcoding within Asterisk PBX - Steve Underwood |
| Re: [asterisk-dev] RTP interop with Sonus: hack - John Todd |
| Re: [asterisk-dev] RTP interop with Sonus: hack - Kristian Kielhofner |
| Re: [asterisk-dev] RTP interop with Sonus: hack - Kristian Kielhofner |
| Re: [asterisk-dev] [design] Realtime changes - Atis Lezdins |
| [asterisk-dev] Interesting SVN graphical representations of Asterisk project - John Todd |
| Re: [asterisk-dev] [design] Realtime changes - Atis Lezdins |
| Re: [asterisk-dev] Interesting SVN graphical representations of Asterisk project - Tzafrir Cohen |
| Re: [asterisk-dev] Interesting SVN graphical representations of Asterisk project - Jeff Peeler |
| Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed. - Klaus Darilion |
| Re: [asterisk-dev] [design] Realtime changes - Tilghman Lesher |
| Re: [asterisk-dev] [Code Review] Queue SIP requests/responses that cannot be immediately processed. - Dmitry Andrianov |
| Re: [asterisk-dev] RTP interop with Sonus: hack - Kristian Kielhofner |
| [asterisk-dev] OT: Paying people in faraway (Western) places. - Alex Balashov |
| [asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk - Kevin Fleming |
| [asterisk-dev] No emails from Mantis - Benny Amorsen |
| Re: [asterisk-dev] 2008 Post Count - Steven S. Critchfield |
| Re: [asterisk-dev] RTP interop with Sonus: hack - Joshua Colp |
| Re: [asterisk-dev] i extension does not match on initial context - bug or not? - Leif Madsen |
| Re: [asterisk-dev] [Code Review] MFC/R2 support for chan_dahdi - Moises Silva |
| [asterisk-dev] Looking for commit that fixed bug #14139, similar to #14173 - Alex Villacís Lasso |
| Re: [asterisk-dev] apps/app_page.c: fix buffer overflow and invalid memory access - John Todd |
| [asterisk-dev] origsvn.digium.com maintenance completed - Kevin P. Fleming |
| Re: [asterisk-dev] i extension does not match on initial context - bug or not? - Benny Amorsen |
| Re: [asterisk-dev] OT: Paying people in faraway (Western) places. - Nir Simionovich |
| Re: [asterisk-dev] Conversion to C++ (groups/asterisk-cpp)? - David A. Burgess |
| [asterisk-dev] G.729.1 - any interest? - John Todd |
| Re: [asterisk-dev] G.729.1 - any interest? - Kristian Kielhofner |
| [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree - Philip Prindeville |
| Re: [asterisk-dev] G.729.1 - any interest? - Kevin P. Fleming |
| Re: [asterisk-dev] calls problem - Russell Bryant |
| Re: [asterisk-dev] calls problem - Russell Bryant |
| Re: [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree - Philip Prindeville |
| Re: [asterisk-dev] reset/remove SipAddHeader() headers - Klaus Darilion |
| Re: [asterisk-dev] reset/remove SipAddHeader() headers - Klaus Darilion |
| [asterisk-dev] Asterisk 1.4.23-rc4 Now Available - Asterisk Development Team |
| Re: [asterisk-dev] voicemail: storing vmsecret in /var/spool/asterisk/voicemail - Klaus Darilion |
| Re: [asterisk-dev] AEL macro backward compatibility for 1.6 - Dmitry Andrianov |
| Re: [asterisk-dev] [Code Review] Convert character pointers in sip_request structure to be offsets from the beginning of the buffer - Mark Michelson |
| Re: [asterisk-dev] SIPAdd/RemoveHeader apps - Tilghman Lesher |
| Re: [asterisk-dev] SIPAdd/RemoveHeader apps - Chris Tooley |
| Re: [asterisk-dev] SIPAdd/RemoveHeader apps - Klaus Darilion |
| [asterisk-dev] [0014275] Group does not count all channels - Marcin J. Kowalczyk |
| Re: [asterisk-dev] dahdi sysfs branch - Tzafrir Cohen |
| Re: [asterisk-dev] How ti let execution continue to the next priorité after callee or caller hangup - Dmitry Andrianov |
| [asterisk-dev] app_queue, ringinuse, joinempty combination problems - Wolfgang Pichler |
| [asterisk-dev] SIP channel/owner question - Klaus Darilion |
| Re: [asterisk-dev] Access to encryption functions from dial plan - Russell Bryant |
| Re: [asterisk-dev] app_queue, ringinuse, joinempty combination problems - Wolfgang Pichler |
| Re: [asterisk-dev] Access to encryption functions from dial plan - Benny Amorsen |
| [asterisk-dev] ChannelRedirect and channels on Dial - Gabriel Ortiz Lour |
| Re: [asterisk-dev] Welcome to David Vossel - Leif Madsen |
| [asterisk-dev] X-Asterisk-HangupCause header only added in auto fallthrough mode - Klaus Darilion |
| Re: [asterisk-dev] X-Asterisk-HangupCause header only added in auto fallthrough mode - Johansson Olle E |
| Re: [asterisk-dev] SIP channel/owner question - Klaus Darilion |
| Re: [asterisk-dev] SIP channel/owner question - Klaus Darilion |
| [asterisk-dev] Newbie Developer Resource? - David Phillips |
| Re: [asterisk-dev] [Code Review] This patch implements CEL in trunk. - Russell Bryant |
| Re: [asterisk-dev] New AGI manager command: PlaySound - Nir Simionovich |
| Re: [asterisk-dev] Newbie Developer Resource? - Grey Man |
| Re: [asterisk-dev] Newbie Developer Resource? - Grey Man |
| Re: [asterisk-dev] Using SIP REFER with a non bridged channel - is there a workaround? - Russell Bryant |
| [asterisk-dev] Using SIP REFER with a non bridged channel - is there a workaround? - Dan Julius |
| Re: [asterisk-dev] Newbie Developer Resource? - Nicolas Chapleau |
| Re: [asterisk-dev] Building asterisk apps, features, etc. outside of Asterisk tree - Philip A. Prindeville |
| Re: [asterisk-dev] Adding Bluetooth Pairing Support to chan_mobile - Steve Murphy |
| Re: [asterisk-dev] DTMF queuing - Russell Bryant |
| Re: [asterisk-dev] [Code Review] Add common implementation for a scheduler context with a dedicated thread - Russell Bryant |
| Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes - Russell Bryant |
| Re: [asterisk-dev] [Code Review] properly report ast_func_read errors in getvar AMI action - Russell Bryant |
| Re: [asterisk-dev] chan_sip SIP Authentication - Johansson Olle E |
| Re: [asterisk-dev] chan_sip SIP Authentication - Philipp Kempgen |
| Re: [asterisk-dev] chan_sip SIP Authentication - Benny Amorsen |
| Re: [asterisk-dev] chan_sip SIP Authentication - Johansson Olle E |
| Re: [asterisk-dev] chan_sip SIP Authentication - Klaus Darilion |
| Re: [asterisk-dev] chan_sip SIP Authentication - Johansson Olle E |
| Re: [asterisk-dev] chan_sip SIP Authentication - Jesus Rodriguez |
| Re: [asterisk-dev] chan_sip SIP Authentication - Klaus Darilion |
| Re: [asterisk-dev] chan_sip SIP Authentication - Klaus Darilion |
| Re: [asterisk-dev] chan_sip SIP Authentication - Johansson Olle E |
| Re: [asterisk-dev] Multiple Server - Jeff Peeler |
| Re: [asterisk-dev] Multiple Server - Chris Tooley |
| Re: [asterisk-dev] [Code Review] Fix feature inheritance when using builtin features - Terry Wilson |
| Re: [asterisk-dev] chan_sip SIP Authentication - Philipp Kempgen |
| Re: [asterisk-dev] chan_sip SIP Authentication - Johansson Olle E |
| Re: [asterisk-dev] Asterisk RFC2833 SDP fmtp 0-16? - Johansson Olle E |
| Re: [asterisk-dev] DTMF queuing - James Lamanna |
| Re: [asterisk-dev] chan_sip.c - no reply to critical packet - Gregory Boehnlein |
| Re: [asterisk-dev] chan_sip.c - no reply to critical packet - James Golovich |
| Re: [asterisk-dev] Asterisk RFC2833 SDP fmtp 0-16? - Leif Madsen |
| Re: [asterisk-dev] [Code Review] Bridging API for Conference Bridge purposes - Joshua Colp |
| Re: [asterisk-dev] chan_sip.c - no reply to critical packet - Benny Amorsen |
| Re: [asterisk-dev] Need to start in development - Michiel van Baak |
| Re: [asterisk-dev] ast_careful_fwrite: fflush() returned error: Broken pipe - Chris Maciejewski |
| [asterisk-dev] Quoting in AGI data - Alistair Cunningham |
