| [Asterisk-Dev] Re: New Channel for alsa - Jaime Lopez |
| [Asterisk-Dev] Re: New Channel for alsa - Steve McMahon |
| [Asterisk-Dev] Refined Voice CallerID Announce - Shawn L. Djernes |
| [Asterisk-Dev] Refined Voice CallerID Announce & Partial Apologies to Steven Critchfield ... Still an ass - Steve McMahon |
| [Asterisk-Dev] calling one application from other application - Wolfgang Pichler |
| [Asterisk-Dev] Refined Voice CallerID Announce - Adam Goryachev |
| [Asterisk-Dev] Refined Voice CallerID Announce - Steven Critchfield |
| [Asterisk-Dev] Refined Voice CallerID Announce - Andreas Sikkema |
| [Asterisk-Dev] Refined Voice CallerID Announce & Partial Apologies to Steven Critchfield ... Still an ass - Greg Boehnlein |
| [Asterisk-Dev] ClueCon in Chicago June 8th to the 10th. - Brian West |
| [Asterisk-Dev] variable sample period? - Roy Sigurd Karlsbakk |
| [Asterisk-Dev] Re: calling one application from other application - Jeremy McNamara |
| [Asterisk-Dev] variable sample period? - Peter Svensson |
| [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58 - Constantine Filin |
| [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58 - Sergey Kuznetsov |
| [Asterisk-Dev] Asterisk Compilation using ARM GCC - Geetha |
| [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58 - Kevin P. Fleming |
| [Asterisk-Dev] Re: variable sample period? - Tony Mountifield |
| [Asterisk-Dev] Re: variable sample period? - Steve Kann |
| [Asterisk-Dev] Refined Voice CallerID Announce - Richard Lyman |
| [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58 - Sergey Kuznetsov |
| [Asterisk-Dev] Receive calls without be registered - Sadie Louise |
| [Asterisk-Dev] ClueCon in Chicago June 8th to the 10th. - Tilghman Lesher |
| [Asterisk-Dev] Receive calls without be registered - Steven Critchfield |
| [Asterisk-Dev] new jitterbuffer in 1.2? - rsen...@harrislogic.com |
| [Asterisk-Dev] new jitterbuffer in 1.2? - Mike Taht |
| [Asterisk-Dev] dev conf topic: better CDRs - Race Vanderdecken |
| [Asterisk-Dev] RE: Asterisk-Dev Digest, Vol 7, Issue 58 - Race Vanderdecken |
| [Asterisk-Dev] new jitterbuffer in 1.2? - Steve Kann |
| [Asterisk-Dev] new jitterbuffer in 1.2? - Andrew Kohlsmith |
| [Asterisk-Dev] dev conf topic: better CDRs - C. Maj |
| [Asterisk-Dev] dev conf topic: better CDRs - Derrick D. Daugherty |
| [Asterisk-Dev] dev conf topic: better CDRs - Matthew Boehm |
| [Asterisk-Dev] dev conf topic: better CDRs - Matthew Boehm |
| [Asterisk-Dev] dev conf topic: better CDRs - Clint Guillot |
| [Asterisk-Dev] new jitterbuffer in 1.2? - Mike Taht |
| [Asterisk-Dev] dev conf topic: better CDRs - Chris Wade |
| [Asterisk-Dev] dev conf topic: better CDRs - Steve Kann |
| [Asterisk-Dev] dev conf topic: better CDRs - Chris Wade |
| [Asterisk-Dev] variable sample period? - Jared Smith |
| [Asterisk-Dev] Refined Voice CallerID Announce - Tom Dickenson |
| [Asterisk-Dev] dev conf topic: better CDRs - Tilghman Lesher |
| [Asterisk-Dev] Error in Res make file - Claus Futtrup |
| [Asterisk-Dev] Receive calls without be registered - Sadie Louise |
| [Asterisk-Dev] Asterisk Compilation using ARM GCC - Vikramsinh Katkar |
| [Asterisk-Dev] Re: calling one application from other application - Tom Dickenson |
| [Asterisk-Dev] Re: Error in Res make file - Claus Futtrup |
| [Asterisk-Dev] Asterisk Compilation using ARM GCC - Geetha |
| [Asterisk-Dev] dev conf topic: better CDRs - Kevin P. Fleming |
| [Asterisk-Dev] dev conf topic: better CDRs - Kevin P. Fleming |
| [Asterisk-Dev] Request for feedback: Overriding codec in dialplan - nie...@wxn.nl |
| [Asterisk-Dev] dev conf topic: better CDRs - Kevin P. Fleming |
| [Asterisk-Dev] H323 Trunking - Huddleston, Robert |
| [Asterisk-Dev] dev conf topic: better CDRs - Tilghman Lesher |
| [Asterisk-Dev] How to monitor Agen Voice channal? - Preston Garrison |
| [Asterisk-Dev] dev conf topic: better CDRs - C. Maj |
| [Asterisk-Dev] changing codec during call - Jesse Kaijen |
| [Asterisk-Dev] changing codec during call - Race Vanderdecken |
| [Asterisk-Dev] changing codec during call - Michael Giagnocavo |
| [Asterisk-Dev] Speaking in Chicago - Brian West |
| [Asterisk-Dev] changing codec during call - Steve Kann |
| [Asterisk-Dev] changing codec during call - Race Vanderdecken |
| [Asterisk-Dev] changing codec during call - Steve Kann |
| [Asterisk-Dev] changing codec during call - Steve Kann |
| [Asterisk-Dev] changing codec during call - Steve Underwood |
| [Asterisk-Dev] changing codec during call - Jesse Kaijen |
| [Asterisk-Dev] changing codec during call - Steve Kann |
| [Asterisk-Dev] changing codec during call - Steve Kann |
| [Asterisk-Dev] changing codec during call - Daniel Bichara |
| [Asterisk-Dev] dev conf topic: better CDRs - Tom Dickenson |
| [Asterisk-Dev] Helps needed for Tethereal and sip - ph...@bevertec.com |
| [Asterisk-Dev] Helps needed for Tethereal and sip - Race Vanderdecken |
| [Asterisk-Dev] Helps needed for Tethereal and sip - Andrew Pyles |
| [Asterisk-Dev] "click to dial extension number" functionality ? - Terje Myhre |
| [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch) - Jared Smith |
| [Asterisk-Dev] Re: rfc2833 DTMFs sent with bad timestamps (patch) - Paul Cadach |
| [Asterisk-Dev] Re: rfc2833 DTMFs sent with bad timestamps (patch) - Frank van Dijk |
| [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch) - Brian West |
| [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch) - Brian West |
| [Asterisk-Dev] RTP not sending UDP checksums? - Andrew Lindh |
| [Asterisk-Dev] file descriptors per call? - Roy Sigurd Karlsbakk |
| [Asterisk-Dev] Keeping queue status between reloads? - Kevin P. Fleming |
| [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch) - Frank van Dijk |
| [Asterisk-Dev] file descriptors per call? - Steven Critchfield |
| [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch) - Brian West |
| [Asterisk-Dev] setting up fromuser - Kevin P. Fleming |
| [Asterisk-Dev] setting up fromuser - Olle E. Johansson |
| [Asterisk-Dev] changing codec during call - Jesse Kaijen |
| [Asterisk-Dev] h323-channel dynamic endpoint registration - Michael Platov |
| [Asterisk-Dev] Bristuff and Realtime - Alessio Focardi |
| [Asterisk-Dev] -user questions in -dev - Jerris, Michael MI |
| [Asterisk-Dev] -user questions in -dev - Andrew Kohlsmith |
| [Asterisk-Dev] Re: how to use ast_channel_setwhentohangup to allocat maximum time for call - Kamran Ahmad |
| [Asterisk-Dev] Dev Conference - Chris Wade |
| [Asterisk-Dev] Silence suppression in asterisk/chan_sip? - Olle E. Johansson |
| [Asterisk-Dev] Silence suppression in asterisk/chan_sip? - Kevin P. Fleming |
| [Asterisk-Dev] Silence suppression in asterisk/chan_sip? - Roy Sigurd Karlsbakk |
| [Asterisk-Dev] how to design digim clone card? - dev2...@mail.ustc.edu.cn |
| [Asterisk-Dev] how to design digim clone card? - Andrew Kohlsmith |
| [Asterisk-Dev] AMP set httpd to asterisk user - ePonkFiria |
