| [Asterisk-Dev] HEAD - Advanced voicemail behaviour change - ste...@daviesfam.org |
| [Asterisk-Dev] HEAD - Advanced voicemail behaviour change - denon |
| [Asterisk-Dev] MGCP Clients - J C |
| [Asterisk-Dev] Some (lack of) answers regarding the wakeup ca ll application... - Luckcuck Nick-LCKN001 |
| [Asterisk-Dev] Re: [Core Asterisk 0001760]: Macros do not clear argument variables when calling other macros - Eric Wieling |
| [Asterisk-Dev] Re: [Applications 0001758]: VoicemailMain2 does not respond in anticipated manner - ste...@daviesfam.org |
| [Asterisk-Dev] ast_data code first public release ready for testing - Rob Gagnon |
| [Asterisk-Dev] Re: [Applications 0001758]: VoicemailMain2 doesnot respond in anticipated manner - Kevin Walsh |
| [Asterisk-Dev] ast_data code first public release ready fortesting - brian |
| [Asterisk-Dev] ast_data code first public release ready fortesting - Greg Varga |
| [Asterisk-Dev] Problem: Sip/Zap (ONE WAY AUDIO) CVS 02-06-2004 10:00AM PST - Steve McMahon |
| [Asterisk-Dev] VoIP x PSTN Gateway - Steven Sokol |
| [Asterisk-Dev] Variable manipulation in extensions.conf - oi geli |
| [Asterisk-Dev] XTunnels: freeware for SIP tunnelling - John Todd |
| [Asterisk-Dev] Call authentication in Asterisk - Aram Ter-Martirosyan |
| [Asterisk-Dev] HEAD - Advanced voicemail behaviour change - aste...@jdennis.net |
| [Asterisk-Dev] Re: Call authentication in Asterisk - Peter Nixon |
| [Asterisk-Dev] zaptel wcusb will not load - Wichert Akkerman |
| [Asterisk-Dev] New Channel - Serial + Sound Card -> GSM Mobile - Adam Goryachev |
| [Asterisk-Dev] PATCH - Adds a set'able text string for each channel - Adam Goryachev |
| [Asterisk-Dev] PATCH - Adds a set'able text string for each channel - James Golovich |
| [Asterisk-Dev] Hardware question (wrong list?) LG GDK Digital phones - Steve Hanselman |
| [Asterisk-Dev] chan_sip2 outboundproxy - David Beckemeyer |
| [Asterisk-Dev] Continuing dial plan after caller hangs up - Alric |
| [Asterisk-Dev] Re: Creating An Asterisk Data Model - Peter Nixon |
| [Asterisk-Dev] Re: Creating An Asterisk Data Model - Paul Mahler |
| [Asterisk-Dev] Adding a dial option to update the DB on bridge - Steve Rodgers |
| [Asterisk-Dev] Error on Chan_zap and prid_dchannel - george bush |
| [Asterisk-Dev] STUN support - Jeremy McNamara |
| [Asterisk-Dev] STUN support - Karl Brose |
| [Asterisk-Dev] Asterisk SIP Problem - Luis Mata |
| [Asterisk-Dev] Passing overlap digits from one pri-E1 to another - Peter Svensson |
| [Asterisk-Dev] Error installing Prepaid App - oi geli |
| [Asterisk-Dev] freebsd rtp.c resource not available errors - Olle E. Johansson |
| [Asterisk-Dev] Error installing Prepaid App - Soren Rathje |
| [Asterisk-Dev] Error installing Prepaid App - reseaux |
| [Asterisk-Dev] Problem with X100P & Local Telco - Steve McMahon |
| [Asterisk-Dev] Problem with X100P & Local Telco - Steven Critchfield |
| [Asterisk-Dev] Bluetooth handsfree channel - Andreas Bayer |
| [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel - Ryan Courtnage |
| [Asterisk-Dev] Problems with ring debounce on TDM400P FXO channel - Rich Adamson |
| [Asterisk-Dev] Centralized voicemail - Soren Rathje |
| [Asterisk-Dev] Overlap digits on pri-E1 etc in Sweden - Peter Svensson |
| [Asterisk-Dev] Overlap digits on pri-E1 etc in Sweden - brian k. west |
| IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel - Dan |
| IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel - jbarre |
| IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel - Rainer Jochem |
| IAX bluetooth [GSM mobile] gateway - was Re: [Asterisk-Dev] Bluetooth handsfree channel - Dan |
| [Asterisk-Dev] Centralized voicemail - Rob Gagnon |
| [Asterisk-Dev] Making libiax2 speak TCP (through udp tunnelin g) - Whisker, Peter |
| [Asterisk-Dev] GR-303 support? - Dave Weis |
| [Asterisk-Dev] Centralized voicemail - Soren Rathje |
| [Asterisk-Dev] DTMF digits relay with Quintum to Oh323 version 0.5.10 - Luis Mata |
| [Asterisk-Dev] Building Zaptel on MIPS - Christian Hecimovic |
| [Asterisk-Dev] IAX2 no compatible codecs - Jason Penton |
| [Asterisk-Dev] Eicon diva server 2M problem - Scannachiappolo |
| [Asterisk-Dev] cvs [server aborted]: could not find desired version ... - Dr. Rich Murphey |
| [Asterisk-Dev] Re: cvs [server aborted]: could not find desired version ... - Tony Mountifield |
| [Asterisk-Dev] Simple kde frontend for astersisk - Steven Critchfield |
| [Asterisk-Dev] Streaming File in Conversation - aste...@aconectarse.com |
| [Asterisk-Dev] Simple kde frontend for astersisk - Andreas Bayer |
| [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE - Dr. Rich Murphey |
| [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE - Alfred R. Nurnberger |
| [Asterisk-Dev] Status of autoconf integration? - Robert Bedell |
| [Asterisk-Dev] PTHREAD_MUTEX_RECURSIVE - ste...@daviesfam.org |
| [Asterisk-Dev] SIP Peer handling for outbound calls - Kristopher Lalletti |
| [Asterisk-Dev] Status of autoconf integration? - aste...@jdennis.net |
| [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/apps app_dial.c,1.72,1.73 - Tais M. Hansen |
| [Asterisk-Dev] IAX2 timestamping issues identified from trace analysis - ste...@daviesfam.org |
| [Asterisk-Dev] chan_h323 dtmf - Kelvin Chua |
| [Asterisk-Dev] Problems with inuse counter. - Claus Futtrup |
| [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_modem.c,1.22,1.23 chan_modem_i4l.c,1.15,1.16 - Trevor Peirce |
| [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_modem.c,1.22,1.23 chan_modem_i4l.c,1.15,1.16 - Niklas Ögren |
| [Asterisk-Dev] Codecs G729 and G723.1 - Sudhir Kumar |
| [Asterisk-Dev] Re: [Asterisk-Users] using 2 single pri cards on 1 server - Mike Sturdee |
| [Asterisk-Dev] Re: Codecs G729 and G723.1 - Sudhir Kumar |
| [Asterisk-Dev] Channel Status - Ryan Courtnage |
| [Asterisk-Dev] Re: Codecs G729 and G723.1 - Kevin P. Fleming |
| [Asterisk-Dev] current error with today cvs - Rob Gagnon |
| [Asterisk-Dev] Getting URL to IAX Client Agent - Steve Kann |
| [Asterisk-Dev] Stable branch usable? Development branch better? - Brian |
| [Asterisk-Dev] HylaFAX and spandsp - Kevin P. Fleming |
| [Asterisk-Dev] HylaFAX and spandsp - Steve Underwood |
| [Asterisk-Dev] Small Zaptel patch - Kevin Walsh |
| [Asterisk-Dev] Advanced ADSI scripts - TC |
| [Asterisk-Dev] Advanced ADSI scripts - Andrew Kohlsmith |
| [Asterisk-Dev] Manager Command Reference - Rooster |
| [Asterisk-Dev] Manager Command Reference - Jason Penton |
| [Asterisk-Dev] Test POP's - Kurtz |
| [Asterisk-Dev] OS X 10.3 Patch - Jeremy McNamara |
| [Asterisk-Dev] PRI U2U display messages - Alfred R. Nurnberger |
| [Asterisk-Dev] enumLookup - Duane |
| [Asterisk-Dev] HylaFAX and spandsp - Terry Wilson |
| [Asterisk-Dev] Bugfix for CVS-HEAD-06/26/04-21:56:45 - programmer_ted |
| [Asterisk-Dev] How to get the SIP Response in AGI Application. - santosh bettad |
| [Asterisk-Dev] Problems in chan_zap.c with libr2 support. - CW_ASN |
| [Asterisk-Dev] Calling Reload Remotely - brian |
| [Asterisk-Dev] Calling Reload Remotely - Jeremy McNamara |
| [Asterisk-Dev] sip call routing - mark spowage |
| [Asterisk-Dev] Calling Reload Remotely - Michael Sandee |
