499 messages

com.digium.lists.asterisk-dev [All Lists]

2008 January [All Months]

Page 1 (Messages 1 to 100): 1 2 3 4 5

[asterisk-dev] How to develop a new channel driver for asterisk - Kevin P. Fleming
[asterisk-dev] Call parking bug - Antonio Gallo
[asterisk-dev] SIP call-limit and Realtime - Johansson Olle E
[asterisk-dev] A question about the nonce generation and checking - Klaus Darilion
[asterisk-dev] my Asterisk is missing time... - David Boyd
[asterisk-dev] my Asterisk is missing time... - Atis Lezdins
[asterisk-dev] A question about the nonce generation and checking - Isaac Lee
[asterisk-dev] my Asterisk is missing time... - Atis Lezdins
[asterisk-dev] Trunk configure bombs, but the files are there and 1.4.x builds fine - Tilghman Lesher
[asterisk-dev] Rhino incorporating into zaptel tree - Bob
[asterisk-dev] Rhino incorporating into zaptel tree - Tilghman Lesher
[asterisk-dev] Rhino incorporating into zaptel tree - Oron Peled
[asterisk-dev] Rhino incorporating into zaptel tree - Jared Smith
[asterisk-dev] Rhino incorporating into zaptel tree - Jared Smith
[asterisk-dev] Rhino incorporating into zaptel tree - Brian Capouch
[asterisk-dev] Rhino incorporating into zaptel tree - James Finstrom
[asterisk-dev] Rhino incorporating into zaptel tree - Russell Bryant
[asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call - MENEAULT Maxime
[asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon - Dan Austin
[asterisk-dev] (((Getting debug level))) - Ed Greenberg
[asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon - Russell Bryant
[asterisk-dev] ChannelRedirect improvments - Russell Bryant
[asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon - Russell Bryant
[asterisk-dev] (((Getting debug level))) - Jim Capp
[asterisk-dev] Challenging a send-only INVITE - SCG
[asterisk-dev] Challenging a sendonly INVITE - SCG2
[asterisk-dev] measure call quality for performance test - Di-Shi Sun
[asterisk-dev] measure call quality for performance test - Alex Balashov
[asterisk-dev] When can I AIG? - Steven S. Critchfield
[asterisk-dev] measure call quality for performance test - Jared Smith
[asterisk-dev] When can I AIG? - Evan Ruff
[asterisk-dev] New Team Member: Jeff Peeler - Russell Bryant
[asterisk-dev] ChannelRedirect improvments (was: Asterisk 1.4.18 and 1.6.0-beta2 coming soon) - Johan Wilfer
[asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available - The Asterisk Development Team
[asterisk-dev] New Team Member: Jeff Peeler - jca...@anteil.com
[asterisk-dev] (((Getting debug level))) - Dmitry Andrianov
[asterisk-dev] measure call quality for performance test - Benny Amorsen
[asterisk-dev] Challenging a sendonly INVITE - Maxim Sobolev
[asterisk-dev] [svn-commits] oej: branch 1.4 r100740 - /branches/1.4/channels/chan_sip.c - Russell Bryant
[asterisk-dev] Challenging a sendonly INVITE - Johansson Olle E
[asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon - Johansson Olle E
[asterisk-dev] Asterisk 1.4.18 and 1.6.0-beta2 coming soon - Benny Amorsen
[asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available - Jim Capp
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - The Asterisk Development Team
[asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available - Russell Bryant
[asterisk-dev] Patch: SIP: choosing common codec and select "free" codec for transcoding - Jared Smith
[asterisk-dev] Patch: SIP: choosing common codec and select "free"codec for transcoding - asterisk
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - bkruse
[asterisk-dev] SOA & Web Services for Asterisk - Kevin Bouchard
[asterisk-dev] Dependencies on zaptel - Andre Courchesne - Prival
[asterisk-dev] [asterisk-commits] murf: branch group/CDRfix5 r101034 - in /team/group/CDRfix5: ./ agi/ apps/ bu... - Sean Bright
[asterisk-dev] Patch: SIP: choosing common codec and select "free" codec for transcoding - Andrey Sofronov
[asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call - syd wonder
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Atis Lezdins
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Atis Lezdins
[asterisk-dev] Dependencies on zaptel - Tzafrir Cohen
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Brandon Kruse
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Brandon Kruse
[asterisk-dev] Zaptel-1.4.8; Zap trunk reverts to dialtone instead of processing call - MENEAULT Maxime
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Russell Bryant
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Russell Bryant
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Joshua Colp
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Namal Gunawardene
[asterisk-dev] SOA & Web Services for Asterisk - Stefan Reuter
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Joel Vandal
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Matt Florell
[asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available - James Finstrom
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Atis Lezdins
[asterisk-dev] Asterisk 1.4.18-rc3 Now Available - The Asterisk Development Team
[asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available - Jim Capp
[asterisk-dev] minor (one line) patch for app_voicemail - Kurt Lidl
[asterisk-dev] Request for Comment Analog states. - James Finstrom
[asterisk-dev] Request for Comment Analog states. - James Finstrom
[asterisk-dev] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available - Russell Bryant
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Atis Lezdins
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Russell Bryant
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Atis Lezdins
[asterisk-dev] Kill the user - a murder that needs testing - Johansson Olle E
[asterisk-dev] ztscan checks __ZT_SIG_DACS for a digital span - Tzafrir Cohen
[asterisk-dev] Patch: SIP: choosing common codec and select "free"codec for transcoding - Andrey Sofronov
[asterisk-dev] ztscan checks __ZT_SIG_DACS for a digital span - Kevin P. Fleming
[asterisk-dev] Attn Olle: closed bug #9239 - "xfersound" - Johansson Olle E
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Matt Riddell
[asterisk-dev] Asterisk 1.4.18-rc2 Now Available - Matt Riddell
[asterisk-dev] Request for Comment Analog states. - Matt Riddell
[asterisk-dev] Request for Comment Analog states. - Matt Riddell
[asterisk-dev] Attn Olle: closed bug #9239 - "xfersound" - Steve Davies
[asterisk-dev] Request for Comment Analog states. - MENEAULT Maxime
[asterisk-dev] Asterisk mishandling user busy isdn releases - Ken Leland III
[asterisk-dev] problem using Cagi - Yelson Vivas
[asterisk-dev] Attn Olle: closed bug #9239 - "xfersound" - Johansson Olle E
[asterisk-dev] Asterisk 1.4.18-rc4 Now Available - The Asterisk Development Team
[asterisk-dev] DEBUG_THREADS and unreleased locks - Russell Bryant
[asterisk-dev] DEBUG_THREADS and unreleased locks - Norman Franke
[asterisk-dev] [svn-commits] mvanbaak: branch mvanbaak/cli-command-audit r101579 - in /team/mvanbaak/cli-c... - Russell Bryant
[asterisk-dev] [svn-commits] tilghman: branch tilghman/res_odbc_astobj2 r101392 - /team/tilghman/res_odbc_... - Russell Bryant
[asterisk-dev] [svn-commits] tilghman: branch tilghman/res_odbc_astobj2 r101392 - /team/tilghman/res_odbc_... - Tilghman Lesher
[asterisk-dev] RFC 3264 - RTP media stream should be set to recvonly when recording messages via a SIP channel - Iain McBride
[asterisk-dev] Another dial-option, catching hangup of caller party - Johan Wilfer
[asterisk-dev] [svn-commits] mvanbaak: branch mvanbaak/cli-command-audit r101579 - in /team/mvanbaak/cli-c... - Michiel van Baak

Page 1 (Messages 1 to 100): 1 2 3 4 5