January 2017 Archives by subject

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Starting: Tue Jan 3 05:56:07 CST 2017
Ending: Tue Jan 31 17:14:05 CST 2017
Messages: 72

• [asterisk-dev] [BOUNTY] 26414 - ExternalIVR attended transfer - no audio   Chris Maciejewski
• [asterisk-dev] [BOUNTY] 26414 - ExternalIVR attended transfer - no audio   bala murugan
• [asterisk-dev] [DAHDI] How to set FXO channel busy   dpn at switchvoice.com
• [asterisk-dev] [DAHDI] How to set FXO channel busy   dpn at switchvoice.com
• [asterisk-dev] app_queue: RINGNOANSWER event   Tomec Martin
• [asterisk-dev] app_queue: RINGNOANSWER event   Troy Bowman
• [asterisk-dev] app_queue: RINGNOANSWER event   Kevin Harwell
• [asterisk-dev] app_queue: RINGNOANSWER event   Tomec Martin
• [asterisk-dev] ARI Bridge Behavior   Michael Petruzzello
• [asterisk-dev] ARI Bridge Behavior   Joshua Colp
• [asterisk-dev] ARI Bridge Behavior   Michael Petruzzello
• [asterisk-dev] ARI Bridge Behavior   bala murugan
• [asterisk-dev] ARI Bridge Behavior   Matt Fredrickson
• [asterisk-dev] ARI Bridge Behavior   Phil Mickelson
• [asterisk-dev] ARI Bridge Behavior   BJ Weschke
• [asterisk-dev] ARI Bridge Behavior   Matt Fredrickson
• [asterisk-dev] ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint   Ross Beer
• [asterisk-dev] ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint   Joshua Colp
• [asterisk-dev] ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint   Joshua Colp
• [asterisk-dev] ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint   Ross Beer
• [asterisk-dev] Asterisk 13 GIT Deadlock   Ross Beer
• [asterisk-dev] Asterisk 13 GIT Deadlock   Joshua Colp
• [asterisk-dev] Asterisk Load Performance   bala murugan
• [asterisk-dev] Asterisk Logger find-replace in log messages   info at asteriskite.com
• [asterisk-dev] Asterisk Logger find-replace in log messages   Matt Fredrickson
• [asterisk-dev] Asterisk Logger find-replace in log messages   Steve Edwards
• [asterisk-dev] Asterisk Logger find-replace in log messages   info at asteriskite.com
• [asterisk-dev] Asterisk Logger find-replace in log messages   info at asteriskite.com
• [asterisk-dev] Asterisk or Jansson issue?   Joshua Colp
• [asterisk-dev] Caller ID in FXO channel   dpn at switchvoice.com
• [asterisk-dev] detect called channel hang-up even with dial application g argument   Fred Muteesa
• [asterisk-dev] detect called channel hang-up even with dial application g argument   Matt Fredrickson
• [asterisk-dev] detect called channel hang-up even with dial application g argument   Yves
• [asterisk-dev] detect called channel hang-up even with dial application g argument   Fred Muteesa
• [asterisk-dev] detect called channel hang-up even with dial application g argument   Yves
• [asterisk-dev] Fw: Caller ID in FXO channel   dpn at switchvoice.com
• [asterisk-dev] G.729 licensing   Tzafrir Cohen
• [asterisk-dev] G.729 licensing   Matt Fredrickson
• [asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required   Dan Jenkins
• [asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required   Sebastian Gutierrez
• [asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required   Dan Jenkins
• [asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required   Dan Jenkins
• [asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required   Matt Fredrickson
• [asterisk-dev] libpri 1.6.0 Now Available   Asterisk Development Team
• [asterisk-dev] Local channel variable propagation: expired hack or a bug?   Kirill Katsnelson
• [asterisk-dev] Local channel variable propagation: expired hack or a bug?   Richard Mudgett
• [asterisk-dev] Local channel variable propagation: expired hack or a bug?   Kirill Katsnelson
• [asterisk-dev] Local channel variable propagation: expired hack or a bug?   Richard Mudgett
• [asterisk-dev] Local channel variable propagation: expired hack or a bug?   Kirill Katsnelson
• [asterisk-dev] Random crashes with Asterisk 13.12.2   Ludovic Gasc
• [asterisk-dev] Registration state for SIP over TCP or TLS   Steve Davies
• [asterisk-dev] Registration state for SIP over TCP or TLS   Joshua Colp
• [asterisk-dev] Registration state for SIP over TCP or TLS   Olle E. Johansson
• [asterisk-dev] Regression: dash a not a character any more?   Kirill Katsnelson
• [asterisk-dev] Regression: dash a not a character any more?   Richard Mudgett
• [asterisk-dev] Regression: dash a not a character any more?   Kirill Katsnelson
• [asterisk-dev] Regression: dash a not a character any more?   Richard Mudgett
• [asterisk-dev] SDP API Wiki Page   Mark Michelson
• [asterisk-dev] Stream Support Wiki Page   Joshua Colp
• [asterisk-dev] Thank you Asterisk community!   Digium's Asterisk Development Team
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Michael Maier
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Matt Fredrickson
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Matthew Jordan
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   George Joseph
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Matthew Jordan
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Michael Maier
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Michael Maier
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Joshua Colp
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Michael Maier
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Joshua Colp
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Michael Maier
• [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip   Joshua Colp

Last message date: Tue Jan 31 17:14:05 CST 2017
Archived on: Tue Jan 31 17:14:15 CST 2017

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