October 2006 Archives by author

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Starting: Sun Oct 1 08:08:50 MST 2006
Ending: Tue Oct 31 23:41:54 MST 2006
Messages: 862

• [asterisk-dev] Dialplan -- Which version of Asterisk is running me?   Matt Riddell (IT)
• [asterisk-dev] Jitter Buffer   Martin Vít
• [asterisk-dev] Configurable T1 default   Martin Vít
• [asterisk-dev] Jitter Buffer and PLC   Martin Vít
• [asterisk-dev] Jitter Buffer   Martin Vít
• [asterisk-dev] Jitter Buffer and PLC   Martin Vít
• [asterisk-dev] Jitter Buffer   Martin Vít
• [asterisk-dev] Re: [svn-commits] oej: trunk r46392 - /trunk/channels/chan_sip.c   Martin Vít
• [asterisk-dev] 1.4 Beta and oracle   René Enskat [Teamware GmbH]
• [asterisk-dev] Patch to enable specifying SDP RTP IP address   Jon Schøpzinsky
• SV: [asterisk-dev] Patch to enable specifying SDP RTP IP address   Jon Schøpzinsky
• SV: SV: [asterisk-dev] Patch to enable specifying SDP RTP IP address   Jon Schøpzinsky
• [asterisk-dev] SIP MESSAGE methode   René Oertel
• [asterisk-dev] Re: Connection Pooling   Tomislav Parčina
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Rich Adamson
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Rich Adamson
• [asterisk-dev] [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Rich Adamson
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Rich Adamson
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Rich Adamson
• [asterisk-dev] call flow call transfer   Kamran Ahmad
• [asterisk-dev] Re: How to busy out PRI channels?   Imran Ahmed
• [asterisk-dev] Re: How to busy out PRI channels?   Imran Ahmed
• [asterisk-dev] Strange functionality of hardware SIP phones on Asterisk 1.4-branch   Volkov Alexei
• [asterisk-dev] Builtin transfer   Volkov Alexei
• [asterisk-dev] Builtin transfer   Volkov Alexei
• [asterisk-dev] Asterisk 1.4-beta3 builtin transfer   Volkov Alexei
• [asterisk-dev] SIP Address port remove suggestion   Alexandre Almeida
• [asterisk-dev] Re: autoconf issues for FreeBSD   Benny Amorsen
• [asterisk-dev] Re: A legal question   Benny Amorsen
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Benny Amorsen
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Benny Amorsen
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Benny Amorsen
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Benny Amorsen
• [asterisk-dev] IAX2 deadlock in trunk - out of free iax2 threads   Anton
• [asterisk-dev] IAX2 deadlock in trunk - out of free iax2 threads   Anton
• [asterisk-dev] IAX2 deadlock in trunk - out of free iax2 threads   Anton
• [asterisk-dev] SVN is old...   Anton
• [asterisk-dev] IAX2 deadlock in trunk - out of free iax2 threads   Anton
• [asterisk-dev] show vs. list   Anton
• [asterisk-dev] show vs. list   Anton
• [asterisk-dev] show vs. list   Anton
• [asterisk-dev] crash - Attempted to delete nonexistent schedule entry   Anton
• [asterisk-dev] Floating audio disappear issue   Anton
• [asterisk-dev] How to busy out PRI channels?   Anton
• [asterisk-dev] How to busy out PRI channels?   Anton
• [asterisk-dev] PRI HANGUPCAUSE   Anton
• [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 andZaptel-1.2.10 compile problems   Anton
• [asterisk-dev] SIP Channel does not release.   Anton
• [asterisk-dev] SIP Channel does not release.   Anton
• [asterisk-dev] SIP Channel does not release.   Anton
• [asterisk-dev] IAX2 still broken   Anton
• [asterisk-dev] IAX2 still broken   Anton
• [asterisk-dev] Problem with Asterisk realtime ?   Oded Arbel
• [asterisk-dev] Problem with Asterisk realtime ?   Oded Arbel
• [asterisk-dev] Problem with Asterisk realtime ?   Oded Arbel
• [asterisk-dev] Problem with Asterisk realtime ?   Oded Arbel
• [asterisk-dev] Problem with Asterisk realtime ?   Oded Arbel
• [asterisk-dev] Build instructions out of date?   Jay R. Ashworth
• [asterisk-dev] Build instructions out of date?   Jay R. Ashworth
• [asterisk-dev] What's the best source for architectural understanding?   Jay R. Ashworth
• [asterisk-dev] What's the best source for architectural understanding?   Jay R. Ashworth
• [asterisk-dev] What's the best source for architectural understanding?   Jay R. Ashworth
• [asterisk-dev] What's the best source for architectural understanding?   Jay R. Ashworth
• [asterisk-dev] Asterisk 2.0 -- any roadmap?   Jay R. Ashworth
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   Jay R. Ashworth
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Jay R. Ashworth
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Jay R. Ashworth
• [asterisk-dev] OT: Where Mailing List Replies Should Go (delete if you're not interested)   Jay R. Ashworth
• [asterisk-dev] OT: Where Mailing List Replies Should Go   Jay R. Ashworth
• [asterisk-dev] OT: Where Mailing List Replies Should Go   Jay R. Ashworth
• [asterisk-dev] OT: Where Mailing List Replies Should Go   Jay R. Ashworth
• [asterisk-dev] Daylight Savings Time Change 2007   Jay R. Ashworth
• [asterisk-dev] Daylight Savings Time Change 2007   Jay R. Ashworth
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Jay R. Ashworth
• [asterisk-dev] Speech Recognition   Jay R. Ashworth
• [asterisk-dev] OT: mailing-list-replies [was CLI: list vs show ?]   Jay R. Ashworth
• [asterisk-dev] Regarding SIP performance   Bob Atkins
• [asterisk-dev] Mantis acting up?   Dan Austin
• [asterisk-dev] gsm.h moved but codec_gsm.c not updated   Dan Austin
• [asterisk-dev] gsm.h moved but codec_gsm.c not updated   Dan Austin
• [asterisk-dev] 1.4.0-beta3 installs files in /   Dan Austin
• [asterisk-dev] Proposing meetme patch   Dan Austin
• [asterisk-dev] Trying to vfree() nonexistent vm area   Chris B.
• [asterisk-dev] Developer Summit Topics   Michiel van Baak
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Michiel van Baak
• [asterisk-dev] Development on Asterisk integration to MSC   Wasim Baig
• [asterisk-dev] autodial   Wasim Baig
• [asterisk-dev] chan_bluetooth with nokia 6230i   Boris Bakchiev
• [asterisk-dev] Fwd: [Iaxclient-devel] Video codec negotiation in IAX   Mihai Balea
• *****SPAM***** [asterisk-dev] asterisk 1.4 problem with call queues   Dean Bath
• *****SPAM***** [asterisk-dev] Segmentation Fault on 1.4 - Bug ref:8228   Dean Bath
• *****SPAM***** [asterisk-dev] Segmentation Fault - Logged on Bugs 8228   Dean Bath
• [asterisk-dev] Bugtracker suggestion   Peter Beckman
• [asterisk-dev] problem dialing to Local channel   Maxi Belino
• [asterisk-dev] Jitter Buffer   Nic Bellamy
• [asterisk-dev] Jitter Buffer   Nic Bellamy
• [asterisk-dev] RFC - "Improved" SIP MWI   Steve Blair
• [asterisk-dev] Developer Summit Topics   Marc Blanchet
• [asterisk-dev] Developer Summit Topics   Marc Blanchet
• [asterisk-dev] Developer Summit Topics   Marc Blanchet
• [asterisk-dev] Speed Dails   Al Bochter
• [asterisk-dev] Best practice requested   Greg Boehnlein
• [asterisk-dev] Asterisk Open File Limit   Greg Boehnlein
• [asterisk-dev] Developer Summit Topics   Greg Boehnlein
• [asterisk-dev] Connection Pooling   Greg Boehnlein
• [asterisk-dev] Dialplan code to simulate calls   Greg Boehnlein
• [asterisk-dev] Re: How to busy out PRI channels?   Greg Boehnlein
• [asterisk-dev] IAX2 still broken   Greg Boehnlein
• [Asterisk-Dev] multiple registrations of same credentials   Bradley
• [asterisk-dev] Astricon: Serious Asterisk Testing   Bradley
• [asterisk-dev] autoconf issues for FreeBSD   Russell Bryant
• [asterisk-dev] autoconf issues for FreeBSD   Russell Bryant
• [asterisk-dev] GPG signatures   Russell Bryant
• [asterisk-dev] Jitter Buffer   Russell Bryant
• [asterisk-dev] Re: deadlock in ast_custom_function_register?   Russell Bryant
• [asterisk-dev] Re: Pthread wrapper updates   Russell Bryant
• [asterisk-dev] Pthread wrapper updates   Russell Bryant
• [asterisk-dev] where do we put GUIs ?   Russell Bryant
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r46409 - /trunk/main/rtp.c   Russell Bryant
• [asterisk-dev] manager events to show file and line info ?   Russell Bryant
• [asterisk-dev] ASTERISK_FILE_VERSION   Russell Bryant
• [asterisk-dev] why 'o' (preserve original callerid) is not default in app_dial.c ?   Russell Bryant
• [asterisk-dev] Loading module cdr_mysql crashes 1.4b3   Russell Bryant
• [asterisk-dev] IAX2 still broken   Russell Bryant
• [asterisk-dev] NCS PacketCable Patch   Jason Burton
• [asterisk-dev] Daylight Savings Time Change 2007   Butler, Larry
• [asterisk-dev] Asterisk Call Time limit   Dome C.
• [asterisk-dev] How to contribute ?   Dome C.
• [asterisk-dev] How to reject calls   Paul Cadach
• [asterisk-dev] DTMF Gereration duplicated. Why?   Paul Cadach
• [asterisk-dev] DTMF Gereration duplicated. Why?   Paul Cadach
• [asterisk-dev] Dynamic extension registration problem   Paul Cadach
• [asterisk-dev] Build instructions out of date?   Brian Candler
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   Brian Candler
• [asterisk-dev] A legal question   Brian Candler
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Brian Candler
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Brian Candler
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Brian Candler
• [asterisk-dev] Regarding SIP performance   Brian Candler
• [asterisk-dev] Regarding SIP performance   Brian Candler
• [asterisk-dev] Regarding SIP performance   Brian Candler
• [asterisk-dev] Regarding SIP performance   Brian Candler
• [asterisk-dev] Regarding SIP performance   Brian Candler
• [asterisk-dev] Regarding SIP performance   Brian Candler
• [asterisk-dev] Regarding SIP performance   Brian Candler
• [asterisk-dev] Regarding SIP performance   Brian Candler
• [asterisk-dev] Regarding SIP direct media (new topic)   Brian Candler
• [asterisk-dev] Regarding SIP direct media (new topic)   Brian Candler
• [asterisk-dev] SIP Address port remove suggestion   Brian Candler
• [asterisk-dev] Asterisk Open File Limit   Brian Candler
• [asterisk-dev] Regarding SIP direct media (new topic)   Brian Candler
• [asterisk-dev] Regarding SIP direct media (new topic)   Brian Candler
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Brian Candler
• [asterisk-dev] chan_bluetooth patch for SVN trunk   Brian Candler
• [asterisk-dev] Re: chan_bluetooth patch for SVN trunk   Brian Candler
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Brian Candler
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Brian Candler
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Brian Candler
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Brian Candler
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Brian Candler
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Brian Candler
• [asterisk-dev] configure behavior wrt Postgres broken?   Brian Capouch
• [asterisk-dev] VoicemailMain: s preceding mailbox causes big weirdness   Brian Capouch
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Brian Capouch
• [asterisk-dev] res_config_pgsql in SVN-trunk?   Brian Capouch
• [asterisk-dev] job offers for asterisk gurus   michel Carette
• [asterisk-dev] SUBSCRIPTION for MWI support for multiple boxes   Chris Carey
• [asterisk-dev] SUBSCRIPTION for MWI support for multiple boxes   Chris Carey
• [asterisk-dev] SUBSCRIPTION for MWI support for multiple boxes   Chris Carey
• [asterisk-dev] SUBSCRIPTION for MWI support for multiple boxes   Chris Carey
• [asterisk-dev] Another bounty - app_reload   Juan Carlos Castro y Castro
• [asterisk-dev] About ast_waitfor_nandfs() in channel.c   Juan Carlos Castro y Castro
• [asterisk-dev] How to reject calls   Christian
• [asterisk-dev] How to reject calls   Christian
• [asterisk-dev] Re: [svn-commits] mogorman: branch 1.4 r1490 - in/branches/1.   Christian
• [asterisk-dev] mISDN_dsp and the poll parameter   Christian
• [asterisk-dev] mISDN_dsp and the poll parameter   Christian
• [asterisk-dev] Developer Summit Topics   Christian
• [asterisk-dev] svn syncing backed up again?   James Cloos
• [asterisk-dev] Re: [svn-commits] mogorman: branch 1.4 r1490 - in /branches/1.4: wctdm.c wctdm24xxp.c   Tzafrir Cohen
• [asterisk-dev] zaptel build and wct4xxp   Tzafrir Cohen
• [asterisk-dev] zaptel build and wct4xxp   Tzafrir Cohen
• [asterisk-dev] autoconf issues for FreeBSD   Tzafrir Cohen
• [asterisk-dev] autoconf issues for FreeBSD   Tzafrir Cohen
• [asterisk-dev] autoconf issues for FreeBSD   Tzafrir Cohen
• [asterisk-dev] A legal question   Tzafrir Cohen
• [asterisk-dev] autoconf issues for FreeBSD   Tzafrir Cohen
• [asterisk-dev] Daylight Savings Time Change 2007   Tzafrir Cohen
• [asterisk-dev] show vs. list   Tzafrir Cohen
• [asterisk-dev] show vs. list   Tzafrir Cohen
• [asterisk-dev] show vs. list   Tzafrir Cohen
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Tzafrir Cohen
• [asterisk-dev] Strange problem with new asterisk.   Tzafrir Cohen
• [asterisk-dev] Skype integration   Tzafrir Cohen
• [asterisk-dev] Skype integration   Tzafrir Cohen
• [asterisk-dev] How to busy out PRI channels?   Tzafrir Cohen
• [asterisk-dev] zaptel helper script   Tzafrir Cohen
• [asterisk-dev] zaptel helper script   Tzafrir Cohen
• [asterisk-dev] zaptel helper script   Tzafrir Cohen
• [asterisk-dev] more compact HTML documentation   Tzafrir Cohen
• [asterisk-dev] Strange code in chan_zap.c ?   Tzafrir Cohen
• [asterisk-dev] Developer Summit Topics   Dean Collins
• [asterisk-dev] Developer Summit Topics   Dean Collins
• [asterisk-dev] canreinvite=no behaviour changed between 1.2.x and1.4 ?   Kevin Collins
• [asterisk-dev] canreinvite=no behaviour changed between 1.2.xand1.4 ?   Kevin Collins
• [asterisk-dev] canreinvite=no behaviour changed between1.2.xand1.4 ?   Kevin Collins
• [asterisk-dev] IAX2 deadlock in trunk - out of free iax2 threads   Joshua Colp
• [asterisk-dev] Problem with Asterisk realtime ?   Joshua Colp
• [asterisk-dev] Mantis acting up?   Joshua Colp
• [asterisk-dev] Speech Recognition   Joshua Colp
• [asterisk-dev] Speech Recognition   Joshua Colp
• [asterisk-dev] 1.4.0-beta3 installs files in /   Joshua Colp
• [asterisk-dev] Developer Summit Topics   Joshua Colp
• [asterisk-dev] Asterisk 1.4 beta 3 on MAC OS X   Joshua Colp
• [asterisk-dev] IAX2 still broken   Joshua Colp
• [asterisk-dev] ast_mutex_lock   Juan Jose Comellas
• [asterisk-dev] Bug 0008069: PRI Channels become unavailable if too many call files are queued   Andre Courchesne
• [asterisk-dev] Developer Summit Topics   Andre Courchesne
• [asterisk-dev] Dialplan code to simulate calls   Andre Courchesne
• [asterisk-dev] Re: How to busy out PRI channels?   Alistair Cunningham
• [asterisk-dev] chan_bluetooth with nokia 6230i   Roel Cuppen
• [asterisk-dev] Realtime Application   Aaron Daniel
• [asterisk-dev] Realtime Application   Aaron Daniel
• [asterisk-dev] SIP compliance question   Klaus Darilion
• [asterisk-dev] SIP compliance question   Klaus Darilion
• [asterisk-dev] SIP Address port remove suggestion   Klaus Darilion
• [asterisk-dev] SIP compliance question   Klaus Darilion
• [asterisk-dev] Need Sip channel notify other channel to answer call   Derek
• [asterisk-dev] Need Sip channel notify other channel to answer call   Derek
• [asterisk-dev] Build instructions out of date?   Rod Dorman
• [asterisk-dev] Unable to compile chan_capi with Asterisk 1.4   Gregory Duchatelet
• [asterisk-dev] REGISTER uses 401 not 407?   Johansson Olle E
• [asterisk-dev] Need Sip channel notify other channel to answer call   Johansson Olle E
• [asterisk-dev] [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Johansson Olle E
• [asterisk-dev] canreinvite=no behaviour changed between 1.2.x and 1.4 ?   Johansson Olle E
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Johansson Olle E
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Johansson Olle E
• [asterisk-dev] SUBSCRIPTION for MWI support for multiple boxes   Johansson Olle E
• [asterisk-dev] canreinvite=no behaviour changed between 1.2.xand1.4 ?   Johansson Olle E
• [asterisk-dev] Regarding SIP performance   Johansson Olle E
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Johansson Olle E
• [asterisk-dev] SIP MESSAGE methode   Johansson Olle E
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Johansson Olle E
• [asterisk-dev] Regarding SIP performance   Johansson Olle E
• [asterisk-dev] Regarding SIP performance   Johansson Olle E
• [asterisk-dev] Regarding SIP performance   Johansson Olle E
• [asterisk-dev] Regarding SIP performance   Johansson Olle E
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Johansson Olle E
• [asterisk-dev] Regarding SIP performance   Johansson Olle E
• [asterisk-dev] Regarding SIP performance   Johansson Olle E
• [asterisk-dev] [patch] manager.c and http interface   Johansson Olle E
• [asterisk-dev] Regarding SIP direct media (new topic)   Johansson Olle E
• [asterisk-dev] Regarding SIP performance   Johansson Olle E
• [asterisk-dev] Regarding SIP direct media (new topic)   Johansson Olle E
• [asterisk-dev] Patch to enable specifying SDP RTP IP address   Johansson Olle E
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Johansson Olle E
• [asterisk-dev] Changes to SIP authentication   Johansson Olle E
• SV: [asterisk-dev] Patch to enable specifying SDP RTP IP address   Johansson Olle E
• [asterisk-dev] SIP Address port remove suggestion   Johansson Olle E
• [asterisk-dev] SIP compliance question   Johansson Olle E
• [asterisk-dev] SIP Address port remove suggestion   Johansson Olle E
• [asterisk-dev] SIP compliance question   Johansson Olle E
• [asterisk-dev] SIP compliance question   Johansson Olle E
• [asterisk-dev] Dialplan -- Which version of Asterisk is running me?   Johansson Olle E
• [asterisk-dev] Re: [asterisk-commits] oej: branch 1.4 r46252 - /branches/1.4/channels/chan_sip.c   Johansson Olle E
• [asterisk-dev] users and peers (and phones and trunks) ?   Johansson Olle E
• [asterisk-dev] where do we put GUIs ?   Johansson Olle E
• [asterisk-dev] users and peers (and phones and trunks) ?   Johansson Olle E
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Johansson Olle E
• [asterisk-dev] manager events to show file and line info ?   Johansson Olle E
• [asterisk-dev] Dynamically removing a provider registration entry   Johansson Olle E
• [asterisk-dev] [patch] adding username from registration to peer state ?   Johansson Olle E
• [asterisk-dev] Need explaination about ARA and Asterisk 1.4-beta3   Johansson Olle E
• [asterisk-dev] bug: autocreate peer + sippeers table entry => auth required   Johansson Olle E
• [asterisk-dev] Filed a chan_sip bug (1.4/trunk) ? Please re-test! *** IMPORTANT ***   Johansson Olle E
• [asterisk-dev] Configurable T1 default   Johansson Olle E
• [asterisk-dev] more compact HTML documentation   Johansson Olle E
• [asterisk-dev] [patch] adding username from registration to peer state ?   Johansson Olle E
• [asterisk-dev] [patch] adding username from registration to peer state ?   Johansson Olle E
• [asterisk-dev] [patch] adding username from registration to peer state ?   Johansson Olle E
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r46409 - /trunk/main/rtp.c   Johansson Olle E
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r46513 - in /trunk: funcs/ include/asterisk/ main/   Johansson Olle E
• [asterisk-dev] Bug resolution efficieny   Johansson Olle E
• [asterisk-dev] Multi-homed host routing problems with IAX   Stephan A. Edelman
• [asterisk-dev] Speech Recognition   Stephan A. Edelman
• [asterisk-dev] Speech Recognition   Stephan A. Edelman
• [asterisk-dev] Speech Recognition   Stephan A. Edelman
• [asterisk-dev] Speech Recognition   Stephan A. Edelman
• [asterisk-dev] Dynamically removing a provider registration entry   Craig Edwards
• [asterisk-dev] Dynamically removing a provider registration entry   Craig Edwards
• [asterisk-dev] Re: app_reload?   Steve Edwards
• [asterisk-dev] Bugtracker suggestion   Steve Edwards
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Steve Edwards
• [asterisk-dev] How to busy out PRI channels?   Steve Edwards
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Steve Edwards
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Steve Edwards
• [asterisk-dev] mutex type selection   SF Markus Elfring
• [asterisk-dev] build options for Pthread usage   SF Markus Elfring
• [asterisk-dev] static code analysis   SF Markus Elfring
• [asterisk-dev] handling of strncpy calls   SF Markus Elfring
• [asterisk-dev] Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] Re: Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] Re: Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] Pthread wrapper updates   SF Markus Elfring
• [asterisk-dev] Call drop and strange CDR records   CAHEN Fabrice
• RES: [asterisk-dev] Astricon: Serious Asterisk Testing   Walter Rodrigues Filho
• [asterisk-dev] Asterisk + TCP   Kevin P. Fleming
• [asterisk-dev] svn syncing backed up again?   Kevin P. Fleming
• [asterisk-dev] Best practice requested   Kevin P. Fleming
• [asterisk-dev] Build instructions out of date?   Kevin P. Fleming
• [asterisk-dev] Build instructions out of date?   Kevin P. Fleming
• [asterisk-dev] zaptel build and wct4xxp   Kevin P. Fleming
• [asterisk-dev] autoconf issues for FreeBSD   Kevin P. Fleming
• [asterisk-dev] zaptel build and wct4xxp   Kevin P. Fleming
• [asterisk-dev] Asterisk and "305 Use Proxy"   Kevin P. Fleming
• [asterisk-dev] Another bounty - app_reload   Kevin P. Fleming
• [asterisk-dev] proposed change to main/http.c   Kevin P. Fleming
• [asterisk-dev] autoconf issues for FreeBSD   Kevin P. Fleming
• [asterisk-dev] Transfer app and DTMF via SIP info   Kevin P. Fleming
• [asterisk-dev] GPG signatures   Kevin P. Fleming
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Kevin P. Fleming
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Kevin P. Fleming
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Kevin P. Fleming
• [asterisk-dev] proposed change to main/http.c   Kevin P. Fleming
• [asterisk-dev] autoconf issues for FreeBSD   Kevin P. Fleming
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Kevin P. Fleming
• [asterisk-dev] VoicemailMain: s preceding mailbox causes big weirdness   Kevin P. Fleming
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Kevin P. Fleming
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] oej: branch oej/iaxtrunkfix-1.2 r44779 - in /team/oej/iaxtrunkfix-1.2: ./ cha...   Kevin P. Fleming
• [asterisk-dev] Dynamic extension registration problem   Kevin P. Fleming
• [asterisk-dev] canreinvite=no behaviour changed between 1.2.x and 1.4 ?   Kevin P. Fleming
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Kevin P. Fleming
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Kevin P. Fleming
• [asterisk-dev] zero is a legitimate value for SIP CSeq numbers, right ?   Kevin P. Fleming
• [asterisk-dev] zero is a legitimate value for SIP CSeq numbers, right ?   Kevin P. Fleming
• [asterisk-dev] Re: [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2 r44779   Kevin P. Fleming
• [asterisk-dev] zero is a legitimate value for SIP CSeq numbers, right ?   Kevin P. Fleming
• [asterisk-dev] Changes to Manager API in 1.4 ?   Kevin P. Fleming
• [asterisk-dev] Regarding SIP performance   Kevin P. Fleming
• [asterisk-dev] Regarding SIP performance   Kevin P. Fleming
• [asterisk-dev] Multi-homed host routing problems with IAX   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r45219 - /trunk/main/manager.c   Kevin P. Fleming
• [asterisk-dev] Speech Recognition   Kevin P. Fleming
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Kevin P. Fleming
• [asterisk-dev] Bugtracker suggestion   Kevin P. Fleming
• [asterisk-dev] Patch to enable specifying SDP RTP IP address   Kevin P. Fleming
• [asterisk-dev] [patch] manager.c and http interface   Kevin P. Fleming
• [asterisk-dev] Regarding SIP performance   Kevin P. Fleming
• [asterisk-dev] Regarding SIP performance   Kevin P. Fleming
• [asterisk-dev] Regarding SIP performance   Kevin P. Fleming
• [asterisk-dev] Regarding SIP direct media (new topic)   Kevin P. Fleming
• [asterisk-dev] Speech Recognition   Kevin P. Fleming
• [asterisk-dev] Bugtracker suggestion   Kevin P. Fleming
• [asterisk-dev] Asterisk Open File Limit   Kevin P. Fleming
• [asterisk-dev] Speech Recognition   Kevin P. Fleming
• [asterisk-dev] 1.4 Beta and oracle   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r45551 - /trunk/main/manager.c   Kevin P. Fleming
• [asterisk-dev] License for New Sounds   Kevin P. Fleming
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Kevin P. Fleming
• [asterisk-dev] asterisk.h missing in /usr/include?   Kevin P. Fleming
• [asterisk-dev] Asterisk Open File Limit   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] mogorman: trunk r45571 - /trunk/main/manager.c   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r45551 - /trunk/main/manager.c   Kevin P. Fleming
• [asterisk-dev] asterisk.h missing in /usr/include?   Kevin P. Fleming
• [asterisk-dev] gsm.h moved but codec_gsm.c not updated   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] file: branch 1.4 r45817 - /branches/1.4/main/loader.c   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r45836 - /trunk/main/http.c   Kevin P. Fleming
• [asterisk-dev] Re: https support now in trunk (please read)   Kevin P. Fleming
• [asterisk-dev] Re: chan_bluetooth patch for SVN trunk   Kevin P. Fleming
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Kevin P. Fleming
• [asterisk-dev] Asterisk + MRCP for Speech Resources (TTS / ASR) ?   Kevin P. Fleming
• [asterisk-dev] Asterisk + MRCP for Speech Resources (TTS / ASR) ?   Kevin P. Fleming
• [asterisk-dev] Re: https support now in trunk (please read)   Kevin P. Fleming
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   Kevin P. Fleming
• [asterisk-dev] Jitter Buffer   Kevin P. Fleming
• [asterisk-dev] Jitter Buffer   Kevin P. Fleming
• [asterisk-dev] Jitter Buffer   Kevin P. Fleming
• [asterisk-dev] Fwd: [svn-commits] kpfleming: branch 1.4 r46153 - in /branches/1.4: channels/ main/   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] kpfleming: branch 1.4 r46200 - in /branches/1.4: apps/ cdr/ channels/ main/ p...   Kevin P. Fleming
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Kevin P. Fleming
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Kevin P. Fleming
• [asterisk-dev] Pthread wrapper updates   Kevin P. Fleming
• [asterisk-dev] TDM400P on hppa-linux   Kevin P. Fleming
• [asterisk-dev] Re: Pthread wrapper updates   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r46409 - /trunk/main/rtp.c   Kevin P. Fleming
• [asterisk-dev] manager events to show file and line info ?   Kevin P. Fleming
• [asterisk-dev] where do we put GUIs ?   Kevin P. Fleming
• [asterisk-dev] Asterisk on Embedded Platform   Kevin P. Fleming
• [asterisk-dev] res_config_pgsql in SVN-trunk?   Kevin P. Fleming
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r46513 - in /trunk: funcs/ include/asterisk/ main/   Kevin P. Fleming
• [asterisk-dev] usecount implementation in the new loader ?   Kevin P. Fleming
• [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 andZaptel-1.2.10 compile problems   Kevin P. Fleming
• [asterisk-dev] Subversion server maintenance   Kevin P. Fleming
• [asterisk-dev] [ast-dev] bridging active channels together [fwd]   Matt Florell
• [asterisk-dev] [ast-dev] bridging active channels together [fwd]   Matt Florell
• [asterisk-dev] [ast-dev] bridging active channels together [fwd]   Matt Florell
• [asterisk-dev] Asterisk Open File Limit   Matt Florell
• [asterisk-dev] Asterisk Open File Limit   Matt Florell
• [asterisk-dev] Asterisk Open File Limit   Matt Florell
• [asterisk-dev] Developer Summit Topics   Matt Florell
• [asterisk-dev] H.324M<->SIP/H.323 gateways   Guillaume Fraysse
• [asterisk-dev] Zaptel/Asterisk - Q.SIG status   Matthew Fredrickson
• [asterisk-dev] Re: How to busy out PRI channels?   Matthew Fredrickson
• [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 and Zaptel-1.2.10 compile problems   Michael J. Tubby G8TIC
• [asterisk-dev] Fedora Core 6 (FC6) andAsterisk-1.2.13 andZaptel-1.2.10 compile problems   Michael J. Tubby G8TIC
• [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 andZaptel-1.2.10 compile problems   Michael J. Tubby B.Sc. G8TIC
• [asterisk-dev] DTMF Gereration duplicated. Why?   Paulo Garcia
• [asterisk-dev] DTMF Gereration duplicated. Why?   Paulo Garcia
• [asterisk-dev] DTMF Gereration duplicated. Why?   Paulo Garcia
• [asterisk-dev] send_digit mechanism in ast_channel_tech   Paulo Garcia
• [asterisk-dev] ast_log and deadlock - they are related?   Paulo Garcia
• [asterisk-dev] Re: send_digit mechanism in ast_channel_tech   Paulo Garcia
• [asterisk-dev] Re: ast_log and deadlock - they are related?   Paulo Garcia
• [asterisk-dev] Testing Libpri without Asterisk   Paulo Garcia
• [asterisk-dev] Re: 1.4 Beta and oracle   René Enskat [Teamware GmbH]
• [asterisk-dev] 1.4 Beta and oracle   René Enskat [Teamware GmbH]
• [asterisk-dev] call dropped on transfer (sip) bugid 8064   Adam Goryachev
• [asterisk-dev] REGISTER uses 401 not 407?   Ed Greenberg
• [asterisk-dev] REGISTER uses 401 not 407?   Ed Greenberg
• [asterisk-dev] autoconf issues for FreeBSD   Edwin Groothuis
• [asterisk-dev] Re: autoconf issues for FreeBSD   Edwin Groothuis
• [asterisk-dev] autoconf issues for FreeBSD   Edwin Groothuis
• [asterisk-dev] Fixes / enhancements to the ENUM code   Edwin Groothuis
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Edwin Groothuis
• [asterisk-dev] Daylight Savings Time Change 2007   Edwin Groothuis
• [asterisk-dev] Bug resolution efficieny   Anu Gupta
• [asterisk-dev] Process flow internal Asterisk   Thong Lam Hai
• [asterisk-dev] SIP to IAX   Thong Lam Hai
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   Thong Lam Hai
• [asterisk-dev] Problem implementing call redirection on Zap channels   Johann Hanne
• [asterisk-dev] Problem implementing call redirection on Zap channels   Johann Hanne
• [asterisk-dev] Re: How to busy out PRI channels?   Freddi Hansen
• [asterisk-dev] Re: How to busy out PRI channels?   Freddi Hansen
• [asterisk-dev] Regarding SIP performance   Vlasis Hatzistavrou
• [asterisk-dev] PRI HANGUPCAUSE   Vlasis Hatzistavrou
• [asterisk-dev] Sporadic crash in mp3 code   Dave Hawkes
• [asterisk-dev] Builtin transfer   Paul Hewlett
• [asterisk-dev] Dial without phone   Ejay Hire
• 回覆: [asterisk-dev] sip authentication again   King Ho
• [asterisk-dev] TDM400P on hppa-linux   Kai Holthaus
• [asterisk-dev] Realtime Config Drivers   Marcus Hunger
• [asterisk-dev] Regarding SIP performance   Morten Isaksen
• [asterisk-dev] How to busy out PRI channels?   Morten Isaksen
• [asterisk-dev] Re: How to busy out PRI channels?   Morten Isaksen
• [asterisk-dev] Re: How to busy out PRI channels?   Morten Isaksen
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   raphael Jacquot
• [asterisk-dev] chan_skinny crashes asterisk (1.4)   Pavel Jezek
• [asterisk-dev] Jitter Buffer   Pavel Jezek
• [asterisk-dev] Jitter Buffer   Pavel Jezek
• [asterisk-dev] Jitter Buffer   Pavel Jezek
• [asterisk-dev] Jitter Buffer   Pavel Jezek
• [asterisk-dev] Jitter Buffer   Pavel Jezek
• [asterisk-dev] Jitter Buffer   Pavel Jezek
• [asterisk-dev] Jitter Buffer   Pavel Jezek
• [asterisk-dev] Jitter Buffer   Pavel Jezek
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Pavel Jezek
• [asterisk-dev] Zaptel/Asterisk - Q.SIG status   Pavel Jezek
• [asterisk-dev] REGISTER uses 401 not 407?   Olle E Johansson
• [asterisk-dev] AstriConVideo ! Paris Nov 20-22! Book your calendar!   Olle E Johansson
• [asterisk-dev] Changes to SIP authentication   Olle E Johansson
• [asterisk-dev] Realtime caching or something else?   Olle E Johansson
• [asterisk-dev] 1.4 Beta and oracle   Olle E Johansson
• [asterisk-dev] Fwd: [svn-commits] rizzo: trunk r45325 - /trunk/main/manager.c   Olle E Johansson
• [asterisk-dev] Fwd: [svn-commits] kpfleming: branch 1.4 r46153 - in /branches/1.4: channels/ main/   Olle E Johansson
• [asterisk-dev] SVN out of synch...   Olle E Johansson
• [asterisk-dev] November: The Asterisk month :-)   Olle E Johansson
• [asterisk-dev] Realtime caching or something else? (SIP retransmit #1)   Olle E Johansson
• [asterisk-dev] Asterisk on Embedded Platform   Prasad Kandikonda
• [asterisk-dev] Jitter Buffer   Steve Kann
• [asterisk-dev] Need An Astcc Mod!   Nate Kapi
• [asterisk-dev] REGISTER uses 401 not 407?   Roy Sigurd Karlsbakk
• [asterisk-dev] REGISTER uses 401 not 407?   Roy Sigurd Karlsbakk
• [asterisk-dev] REGISTER uses 401 not 407?   Roy Sigurd Karlsbakk
• [asterisk-dev] SIP compliance question   Roy Sigurd Karlsbakk
• [asterisk-dev] SIP compliance question   Roy Sigurd Karlsbakk
• [asterisk-dev] SIP compliance question   Roy Sigurd Karlsbakk
• [asterisk-dev] asterisk crashing at ast_reset_cdr()   Roy Sigurd Karlsbakk
• [asterisk-dev] SIP compliance question   Roy Sigurd Karlsbakk
• [asterisk-dev] SIP Address port remove suggestion   Roy Sigurd Karlsbakk
• [asterisk-dev] SIP compliance question   Roy Sigurd Karlsbakk
• [asterisk-dev] SIP compliance question   Roy Sigurd Karlsbakk
• [asterisk-dev] Asterisk Open File Limit   Roy Sigurd Karlsbakk
• [asterisk-dev] Memory leak issue   Roy Sigurd Karlsbakk
• [asterisk-dev] Asterisk Open File Limit   Roy Sigurd Karlsbakk
• [asterisk-dev] Audio Data Packets   Harish Kasiviswanathan
• [asterisk-dev] IMAP email storage proposal and question   Dax Kelson
• [asterisk-dev] autoconf issues for FreeBSD   Steve Kennedy
• [asterisk-dev] RFC - "Improved" SIP MWI   Kristian Kielhofner
• [asterisk-dev] Another bounty - app_reload   Kristian Kielhofner
• [asterisk-dev] Another bounty - app_reload   Kristian Kielhofner
• [asterisk-dev] Another bounty - app_reload   Kristian Kielhofner
• [asterisk-dev] RFC - "Improved" SIP MWI   Kristian Kielhofner
• [asterisk-dev] Another bounty - app_reload   Kristian Kielhofner
• [asterisk-dev] Re: app_reload?   Kristian Kielhofner
• [asterisk-dev] SUBSCRIPTION for MWI support for multiple boxes   Kristian Kielhofner
• [asterisk-dev] A legal question   Andrew Kirch
• [asterisk-dev] SR Level Developer with skills in C, C#, VOIP(Asterisk) and Linux   Andrew Kirch
• [asterisk-dev] asterisk.h missing in /usr/include?   Slav Klenov
• [asterisk-dev] asterisk.h missing in /usr/include?   Slav Klenov
• [asterisk-dev] Jitter Buffer   Slav Klenov
• [asterisk-dev] Jitter Buffer   Slav Klenov
• [asterisk-dev] Another bounty - app_reload   Andrew Kohlsmith
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Andrew Kohlsmith
• [asterisk-dev] IAX2 deadlock in trunk - out of free iax2 threads   Andrew Kohlsmith
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Andrew Kohlsmith
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Andrew Kohlsmith
• [asterisk-dev] Jitter Buffer   Andrew Kohlsmith
• [asterisk-dev] Jitter Buffer   Andrew Kohlsmith
• [asterisk-dev] Jitter Buffer   Andrew Kohlsmith
• [asterisk-dev] Re: How to busy out PRI channels?   Andrew Kohlsmith
• [asterisk-dev] Re: Pthread wrapper updates   Andrew Kohlsmith
• [asterisk-dev] Transfer app and DTMF via SIP info   Michael Konietzny
• [asterisk-dev] Transfer app and DTMF via SIP info   Michael Konietzny
• [asterisk-dev] Proposing meetme patch   Koopmann, Jan-Peter
• [asterisk-dev] Proposing meetme patch   Koopmann, Jan-Peter
• [asterisk-dev] Skype integration   Koopmann, Jan-Peter
• [asterisk-dev] How to reject calls   Kaloyan Kovachev
• [asterisk-dev] Jitter Buffer   Kaloyan Kovachev
• [asterisk-dev] Realtime caching or something else? (SIPretransmit#2)   Bill Kunyiha
• [asterisk-dev] Realtime caching or something else? (SIPretransmit #1)   Bill Kunyiha
• [asterisk-dev] Please help me!!   Anthony LaMantia
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   John Lange
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   John Lange
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   John Lange
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   John Lange
• [asterisk-dev] Changes to Manager API in 1.4 ?   John Lange
• [asterisk-dev] Speech Recognition   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] Jitter Buffer (patch included)   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] Jitter Buffer   John Lange
• [asterisk-dev] zaptel helper script   John Lange
• [asterisk-dev] zaptel helper script   John Lange
• [asterisk-dev] zaptel helper script   John Lange
• [asterisk-dev] Loading module cdr_mysql crashes 1.4b3   John Lange
• [asterisk-dev] Regarding SIP performance   Steve Langstaff
• [asterisk-dev] Regarding SIP performance   Steve Langstaff
• [asterisk-dev] Regarding SIP performance   Steve Langstaff
• [asterisk-dev] Regarding SIP performance   Steve Langstaff
• [asterisk-dev] Question about using Time-Warner Versapak Power T-12 with Asterisk   Michael Lavelle
• [asterisk-dev] show vs. list   Chris Lee
• [asterisk-dev] zaptel helper script   Chris Lee
• [asterisk-dev] Fixes / enhancements to the ENUM code   Otmar Lendl
• [asterisk-dev] Asterisk + TCP   Tilghman Lesher
• [asterisk-dev] [ast-dev] bridging active channels together [fwd]   Tilghman Lesher
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   Tilghman Lesher
• [asterisk-dev] VoicemailMain: s preceding mailbox causes big weirdness   Tilghman Lesher
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Tilghman Lesher
• [asterisk-dev] Getting started with Asterisk development   Tilghman Lesher
• [asterisk-dev] Asterisk Open File Limit   Tilghman Lesher
• [asterisk-dev] Asterisk Open File Limit   Tilghman Lesher
• [asterisk-dev] IMAP email storage proposal and question   Tilghman Lesher
• [asterisk-dev] ast_mutex_lock   Tilghman Lesher
• [asterisk-dev] Connection Pooling   Tilghman Lesher
• [asterisk-dev] Connection Pooling   Tilghman Lesher
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Tilghman Lesher
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Tilghman Lesher
• [asterisk-dev] where do we put GUIs ?   Tilghman Lesher
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Tilghman Lesher
• [asterisk-dev] OT: mailing-list-replies [was CLI: list vs show ?]   Tilghman Lesher
• [asterisk-dev] Asterisk Open File Limit   Richard Lyman
• [asterisk-dev] Asterisk Open File Limit   Richard Lyman
• [asterisk-dev] manager.c changes breaks my app :)   Richard Lyman
• [asterisk-dev] manager.c changes breaks my app :)   Julian Lyndon-Smith
• [asterisk-dev] manager.c changes breaks my app :)   Julian Lyndon-Smith
• [asterisk-dev] manager.c changes breaks my app :)   Julian Lyndon-Smith
• [asterisk-dev] chan_skinny crashes asterisk (1.4)   Leif Madsen
• [asterisk-dev] ODBCquery from dial plan   Leif Madsen
• [asterisk-dev] Another bounty - app_reload   Leif Madsen
• [asterisk-dev] job offers for asterisk gurus   Leif Madsen
• [asterisk-dev] REGISTER uses 401 not 407?   Mikael Magnusson
• [asterisk-dev] ACK during reinvite not sent through SIP Proxy   Mikael Magnusson
• [asterisk-dev] REGISTER uses 401 not 407?   Mikael Magnusson
• [asterisk-dev] Developer Summit Topics   Mikael Magnusson
• [asterisk-dev] Skype integration   Paulo Mannheimer
• [asterisk-dev] Skype integration   Paulo Mannheimer
• [asterisk-dev] Channel file descriptors   John Martin
• [asterisk-dev] [ANNOUNCE] chan_celliax, for managing cellphones via Asterisk, first release   Giovanni Maruzzelli
• [asterisk-dev] SR Level Developer with skills in C, C#, VOIP(Asterisk) and Linux   Roland Matte
• [asterisk-dev] Asterisk + MRCP for Speech Resources (TTS / ASR) ?   Josh McAllister
• [asterisk-dev] Asterisk + MRCP for Speech Resources (TTS / ASR) ?   Josh McAllister
• [asterisk-dev] Re: How to busy out PRI channels?   Josh McAllister
• [asterisk-dev] Best practice requested   Jeremy McNamara
• [asterisk-dev] Best practice requested   Jeremy McNamara
• [asterisk-dev] Another bounty - app_reload   Jeremy McNamara
• [asterisk-dev] Another bounty - app_reload   Jeremy McNamara
• [asterisk-dev] Another bounty - app_reload   Jeremy McNamara
• [asterisk-dev] Re: app_reload?   Jeremy McNamara
• [asterisk-dev] Re: app_reload?   Jeremy McNamara
• [asterisk-dev] SIP to IAX   Jeremy McNamara
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   Jeremy McNamara
• [asterisk-dev] Another bounty - app_reload   Jeremy McNamara
• [asterisk-dev] Astricon: Serious Asterisk Testing   Jeremy McNamara
• Summit Conference Call (was Re: [asterisk-dev] Developer Summit Topics)   Jeremy McNamara
• [asterisk-dev] Developers Summit Conference Call   Jeremy McNamara
• [asterisk-dev] Realtime caching or something else? (SIP retransmit #1)   Jeremy McNamara
• [asterisk-dev] How to contribute ?   Jeremy McNamara
• [asterisk-dev] Transfer app and DTMF via SIP info   Roystan Mendez
• [asterisk-dev] GPG signatures   Bill Merriam
• [asterisk-dev] GPG signatures   Bill Merriam
• [asterisk-dev] crash - Attempted to delete nonexistent schedule entry   Dawid Mielnik
• [asterisk-dev] Invitation to OpenSER Summit @ VoN Berlin   Daniel-Constantin Mierla
• [asterisk-dev] Dial without phone   Mir
• [asterisk-dev] Realtime caching or something else?   Dmytro Mishchenko
• [asterisk-dev] Asterisk and "305 Use Proxy"   Ariel Monaco
• [asterisk-dev] Dial without phone   Ariel Monaco
• [asterisk-dev] Re: Best practice requested   Tony Mountifield
• [asterisk-dev] Re: 1.4 Beta and oracle   Tony Mountifield
• [asterisk-dev] Re: https support now in trunk (please read)   Tony Mountifield
• [asterisk-dev] Re: deadlock in ast_custom_function_register?   Tony Mountifield
• [asterisk-dev] How to busy out PRI channels?   Tony Mountifield
• [asterisk-dev] Re: deadlock in ast_custom_function_register?   Tony Mountifield
• [asterisk-dev] Re: How to busy out PRI channels?   Tony Mountifield
• [asterisk-dev] Re: How to busy out PRI channels?   Tony Mountifield
• [asterisk-dev] Strange code in chan_zap.c ?   Tony Mountifield
• [asterisk-dev] Dialplan -- Which version of Asterisk is running me?   Steve Murphy
• [asterisk-dev] autodial   Santhosh N
• [asterisk-dev] autodial   Santhosh N
• [asterisk-dev] Best practice requested   Dinesh Nair
• [asterisk-dev] autoconf issues for FreeBSD   Dinesh Nair
• [asterisk-dev] Re: autoconf issues for FreeBSD   Dinesh Nair
• [asterisk-dev] Re: [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows   Dinesh Nair
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Dinesh Nair
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Dinesh Nair
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Dinesh Nair
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Dinesh Nair
• [asterisk-dev] Patch to enable specifying SDP RTP IP address   Dinesh Nair
• [asterisk-dev] Changes to SIP authentication   Dinesh Nair
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Dinesh Nair
• [asterisk-dev] Changes to SIP authentication   Dinesh Nair
• [asterisk-dev] why 'o' (preserve original callerid) is not default in app_dial.c ?   Dinesh Nair
• [Asterisk-Dev] multiple registrations of same credentials   Igor Neves
• [asterisk-dev] asterisk imap voicemail storage   Matt O'Gorman
• [asterisk-dev] VoiceMail modularization   Carlton O'Riley
• [asterisk-dev] VoiceMail Modularization   Carlton O'Riley
• [asterisk-dev] Development on Asterisk integration to MSC   Alfred M. Ocon
• [asterisk-dev] Is it a right workaround?   Sergey Okhapkin
• [asterisk-dev] Realtime caching or something else? (SIP retransmit #1)   Sergey Okhapkin
• [asterisk-dev] ODBCquery from dial plan   Alexandr Olekhnovich
• [asterisk-dev] Asterisk Call Time limit   Alexandr Olekhnovich
• [asterisk-dev] ODBCquery trouble   Alexandr Olekhnovich
• [asterisk-dev] Connection Pooling   Alexandr Olekhnovich
• [asterisk-dev] ast_mutex_lock   Alexandr Olekhnovich
• [asterisk-dev] threading   Alexandr Olekhnovich
• [asterisk-dev] Definitions   Alexandr Olekhnovich
• [asterisk-dev] ASTERISK_FILE_VERSION   Alexandr Olekhnovich
• [asterisk-dev] License for New Sounds   Jeffrey C. Ollie
• [asterisk-dev] 1.4.0-beta3 installs files in /   Jeffrey C. Ollie
• [asterisk-dev] Re: https support now in trunk (please read)   Jeffrey C. Ollie
• [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 andZaptel-1.2.10 compile problems   Jeffrey C. Ollie
• [asterisk-dev] Number of Asterisk responses returned   J. Oquendo
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   J. Oquendo
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   J. Oquendo
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   J. Oquendo
• [asterisk-dev] My misfire ... Asterisk DoS   J. Oquendo
• [asterisk-dev] OT: Where Mailing List Replies Should Go   J. Oquendo
• [asterisk-dev] OT: Where Mailing List Replies Should Go   J. Oquendo
• [asterisk-dev] Penalty approach for DoS   J. Oquendo
• [asterisk-dev] job offers for asterisk gurus   Ozed Oseghale
• [asterisk-dev] How to stream audio to external app forspeech recognition and recognize dtmf in parallel ?   Tim Panton
• [asterisk-dev] Developer Summit Topics   Tim Panton
• [asterisk-dev] Developer Summit Topics   Tim Panton
• [asterisk-dev] Developers Summit Conference Call   Tim Panton
• [asterisk-dev] Re: Pthread wrapper updates   Tim Panton
• [asterisk-dev] Asterisk 1.4 beta 3 on MAC OS X   David Parcerisa
• [asterisk-dev] Realtime Application   Jason Parker
• [asterisk-dev] Re: 1.4 Beta and oracle   Jason Parker
• [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 and Zaptel-1.2.10 compile problems   Jason Parker
• [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 andZaptel-1.2.10 compile problems   Patrick
• [asterisk-dev] Bugtracker suggestion   Oron Peled
• [asterisk-dev] Asterisk Open File Limit   Oron Peled
• [asterisk-dev] zaptel helper script   Oron Peled
• [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 andZaptel-1.2.10 compile problems   Oron Peled
• [asterisk-dev] Realtime Config Drivers   Richi Plana
• [asterisk-dev] Strange problem with new asterisk.   Jonson Player
• [asterisk-dev] Strange problem with new asterisk.   Jonson Player
• [asterisk-dev] Push to Talk options in Asterisk.   Jonson Player
• [asterisk-dev] Simple example for call transfer.   Jonson Player
• [asterisk-dev] zaptel build and wct4xxp   Daniel Pocock
• [asterisk-dev] zaptel build and wct4xxp   Daniel Pocock
• [asterisk-dev] zaptel build and wct4xxp   Daniel Pocock
• [asterisk-dev] Daylight Savings Time Change 2007   Daniel Pocock
• [asterisk-dev] Need explaination about ARA and Asterisk 1.4-beta3   Raffaele Porzio
• [asterisk-dev] bug: autocreate peer + sippeers table entry => auth required   Mark Price
• [asterisk-dev] SIP compliance question   Michael Procter
• [asterisk-dev] deadlock in ast_custom_function_register?   Yuan Qin
• [asterisk-dev] ACK during reinvite not sent through SIP Proxy   H Quintana
• [asterisk-dev] Re: ACK during reinvite not sent through SIP Proxy   H Quintana
• [asterisk-dev] Media transcoding   Ramachandran
• [asterisk-dev] asterisk crashing at ast_reset_cdr()   Atif Rasheed
• [asterisk-dev] astricon GUI demo   Bruce Reeves
• [asterisk-dev] Strange problem with new asterisk.   Mathieu Rene
• [asterisk-dev] canreinvite=no behaviour changed between 1.2.xand1.4 ?   Mathieu Rene
• [asterisk-dev] Bugtracker suggestion   Stefan Reuter
• [asterisk-dev] autoconf issues for FreeBSD   Luigi Rizzo
• [asterisk-dev] autoconf issues for FreeBSD   Luigi Rizzo
• [asterisk-dev] autoconf issues for FreeBSD   Luigi Rizzo
• [asterisk-dev] autoconf issues for FreeBSD   Luigi Rizzo
• [asterisk-dev] proposed change to main/http.c   Luigi Rizzo
• [asterisk-dev] autoconf issues for FreeBSD   Luigi Rizzo
• [asterisk-dev] merging manager.conf and http.conf ?   Luigi Rizzo
• [asterisk-dev] autoconf issues for FreeBSD   Luigi Rizzo
• [asterisk-dev] proposed change to main/http.c   Luigi Rizzo
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Luigi Rizzo
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Luigi Rizzo
• [asterisk-dev] canreinvite=no behaviour changed between 1.2.x and 1.4 ?   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] russell: trunk r44876 - /trunk/channels/chan_sip.c   Luigi Rizzo
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Luigi Rizzo
• [asterisk-dev] canreinvite=no behaviour changed between 1.2.xand1.4 ?   Luigi Rizzo
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Luigi Rizzo
• [asterisk-dev] zero is a legitimate value for SIP CSeq numbers, right ?   Luigi Rizzo
• [asterisk-dev] zero is a legitimate value for SIP CSeq numbers, right ?   Luigi Rizzo
• [asterisk-dev] zero is a legitimate value for SIP CSeq numbers, right ?   Luigi Rizzo
• [asterisk-dev] show vs. list   Luigi Rizzo
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Luigi Rizzo
• [asterisk-dev] manager.c   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] oej: trunk r45209 - /trunk/channels/chan_sip.c   Luigi Rizzo
• [asterisk-dev] [patch] manager.c and http interface   Luigi Rizzo
• [asterisk-dev] [patch] manager.c and http interface   Luigi Rizzo
• [asterisk-dev] Regarding SIP performance   Luigi Rizzo
• [asterisk-dev] Regarding SIP performance   Luigi Rizzo
• [asterisk-dev] Fwd: [svn-commits] rizzo: trunk r45325 - /trunk/main/manager.c   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] mogorman: trunk r45571 - /trunk/main/manager.c   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r45551 - /trunk/main/manager.c   Luigi Rizzo
• [asterisk-dev] sip authentication again   Luigi Rizzo
• ¦^ÂÐ: [asterisk-dev] sip authentication again   Luigi Rizzo
• [asterisk-dev] https support now in trunk (please read)   Luigi Rizzo
• [asterisk-dev] Developer Summit Topics   Luigi Rizzo
• [asterisk-dev] Re: https support now in trunk (please read)   Luigi Rizzo
• [asterisk-dev] Developer Summit Topics   Luigi Rizzo
• [asterisk-dev] Re: https support now in trunk (please read)   Luigi Rizzo
• [asterisk-dev] Re: https support now in trunk (please read)   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] kpfleming: branch 1.4 r46200 - in /branches/1.4: apps/ cdr/ channels/ main/ p...   Luigi Rizzo
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Luigi Rizzo
• [asterisk-dev] manager.c changes breaks my app :)   Luigi Rizzo
• [asterisk-dev] CLI: list vs show ? (Re: [asterisk-commits] oej: branch 1.4 r46216 - /branches/1.4/channels/chan_sip.c)   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] oej: branch 1.4 r46252 - /branches/1.4/channels/chan_sip.c   Luigi Rizzo
• [asterisk-dev] Re: [asterisk-commits] oej: branch 1.4 r46252 - /branches/1.4/channels/chan_sip.c   Luigi Rizzo
• [asterisk-dev] manager.c changes breaks my app :)   Luigi Rizzo
• [asterisk-dev] users and peers (and phones and trunks) ?   Luigi Rizzo
• [asterisk-dev] where do we put GUIs ?   Luigi Rizzo
• [asterisk-dev] where do we put GUIs ?   Luigi Rizzo
• [asterisk-dev] users and peers (and phones and trunks) ?   Luigi Rizzo
• [asterisk-dev] manager events to show file and line info ?   Luigi Rizzo
• [asterisk-dev] [patch] adding username from registration to peer state ?   Luigi Rizzo
• [asterisk-dev] [patch] adding username from registration to peer state ?   Luigi Rizzo
• [asterisk-dev] [patch] adding username from registration to peer state ?   Luigi Rizzo
• [asterisk-dev] [patch] adding username from registration to peer state ?   Luigi Rizzo
• [asterisk-dev] [patch] adding username from registration to peer state ?   Luigi Rizzo
• [asterisk-dev] Definitions   Luigi Rizzo
• [asterisk-dev] usecount implementation in the new loader ?   Luigi Rizzo
• [asterisk-dev] SIP Channel does not release.   Luigi Rizzo
• [asterisk-dev] why 'o' (preserve original callerid) is not default in app_dial.c ?   Luigi Rizzo
• [asterisk-dev] Speech Recognition   Adam Robins
• [asterisk-dev] Asterisk Open File Limit   Matthew J. Roth
• [asterisk-dev] How to busy out PRI channels?   Brian Roy
• [asterisk-dev] Re: How to busy out PRI channels?   Brian Roy
• [asterisk-dev] How to stream audio to external app forspeech recognition and recognize dtmf in parallel ?   Robert Rozman
• [asterisk-dev] Re: [svn-commits] mogorman: branch 1.4 r1490 - in/branches/1.   Michael Rozov
• [asterisk-dev] Another bounty - app_reload   Matthew Rubenstein
• Generic Manager API Client Apps (was: Re: [asterisk-dev] Another bounty - app_reload)   Matthew Rubenstein
• [asterisk-dev] Another bounty - app_reload   Matthew Rubenstein
• [asterisk-dev] merging manager.conf and http.conf ?   Matthew Rubenstein
• [asterisk-dev] merging manager.conf and http.conf ?   Matthew Rubenstein
• [asterisk-dev] Skype integration   Alberto Sagredo
• [asterisk-dev] Unable to compile chan_capi with Asterisk 1.4   Armin Schindler
• [asterisk-dev] Re: How to busy out PRI channels?   Christian Schlatter
• [asterisk-dev] Getting started with Asterisk development   Austin Seipp
• [asterisk-dev] How to reject calls   Hans Petter Selasky
• [asterisk-dev] How to reject calls   Hans Petter Selasky
• [asterisk-dev] How to reject calls   Hans Petter Selasky
• [asterisk-dev] Best practice requested   Mitch Sharp
• [asterisk-dev] Best practice requested   Mitch Sharp
• [asterisk-dev] [ast-dev] bridging active channels together [fwd]   Moises Silva
• [asterisk-dev] [ast-dev] bridging active channels together [fwd]   Moises Silva
• [asterisk-dev] What's the best source for architectural understanding?   Moises Silva
• [asterisk-dev] Another bounty - app_reload   Moises Silva
• [asterisk-dev] What's the best source for architectural understanding?   Moises Silva
• [asterisk-dev] Another bounty - app_reload   Moises Silva
• [asterisk-dev] Dial without phone   Moises Silva
• [asterisk-dev] [ast-dev] bridging active channels together [fwd]   Moises Silva
• [asterisk-dev] send_digit mechanism in ast_channel_tech   Moises Silva
• [asterisk-dev] threading   Moises Silva
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   Jared Smith
• [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)   Jared Smith
• [asterisk-dev] Developer Summit at AstriCon Dallas   Steven Sokol
• [asterisk-dev] AstriCon hotel full - check these alternates   Steven Sokol
• [asterisk-dev] [ast-dev] bridging active channels together [fwd]   Roberto Sottile
• [asterisk-dev] [patch] [issue 7837] MySQL error log   Andrea Spadaccini
• [asterisk-dev] zaptel build and wct4xxp   Steven
• [asterisk-dev] Rate limiting traffic to address potential DoS issues?   Steven
• [asterisk-dev] Audio Data Packets   Steven
• [asterisk-dev] Asterisk Open File Limit   Steven
• [asterisk-dev] zaptel beta2 tar ball corrupt?   Steven
• [asterisk-dev] Simple example for call transfer.   Steven
• [asterisk-dev] autoconf issues for FreeBSD   Rosli Sukri
• [asterisk-dev] Asterisk on Embedded Platform   Mike Taht
• [asterisk-dev] Asterisk 1.0.12 released - Security Vulnerability Fix   Asterisk Development Team
• [asterisk-dev] Asterisk 1.2.13 released - Security Vulnerability Fix   Asterisk Development Team
• [asterisk-dev] Asterisk 1.4.0-beta3 released!   Asterisk Development Team
• [asterisk-dev] Asterisk + TCP   George Thanos
• [asterisk-dev] Re: How to busy out PRI channels?   John Todd
• [asterisk-dev] Re: How to busy out PRI channels?   John Todd
• [asterisk-dev] VoiceMail Modularization   John Todd
• [asterisk-dev] Re: How to busy out PRI channels?   John Todd
• [asterisk-dev] Asterisk on Embedded Platform   Wojciech Tryc
• [asterisk-dev] Develop an Asterisk module   Justin Tunney
• [asterisk-dev] Regarding SIP performance   Chirag Vaishnav
• [asterisk-dev] Regarding SIP performance   Chirag Vaishnav
• [asterisk-dev] Regarding SIP performance   Chirag Vaishnav
• [asterisk-dev] Regarding SIP performance   Chirag Vaishnav
• [asterisk-dev] Memory leak issue   Chirag Vaishnav
• [asterisk-dev] Memory leak issue   Chirag Vaishnav
• [asterisk-dev] Regarding SIP performance   Chirag Vaishnav
• [asterisk-dev] Regarding SIP performance   Chirag Vaishnav
• [asterisk-dev] astricon GUI demo   Isack Waserman
• [asterisk-dev] VoiceMail Modularization   Watkins, Bradley
• [asterisk-dev] Realtime caching or something else? (SIP retransmit#1)   Watkins, Bradley
• [asterisk-dev] Realtime caching or something else? (SIP retransmit#1)   Watkins, Bradley
• [asterisk-dev] Potential change to outgoing codec offers (new topic)   BJ Weschke
• [asterisk-dev] manager events to show file and line info ?   BJ Weschke
• [asterisk-dev] How to reject calls   Eric "ManxPower" Wieling
• [asterisk-dev] What's the best source for architectural understanding?   Eric "ManxPower" Wieling
• [asterisk-dev] Dial without phone   Eric "ManxPower" Wieling
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Eric "ManxPower" Wieling
• [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?   Eric "ManxPower" Wieling
• [asterisk-dev] About ast_waitfor_nandfs() in channel.c   Eric "ManxPower" Wieling
• [asterisk-dev] Re: How to busy out PRI channels?   Eric "ManxPower" Wieling
• [asterisk-dev] Re: chan_bluetooth patch for SVN trunk   David Woodhouse
• [asterisk-dev] Re: chan_bluetooth patch for SVN trunk   David Woodhouse
• [asterisk-dev] Re: chan_bluetooth patch for SVN trunk   David Woodhouse
• [asterisk-dev] Asterisk + TCP   William Emmanuel S. Yu
• [asterisk-dev] Re: [asterisk-commits] oej: branch oej/iaxtrunkfix-1.2 r44779 - in /team/oej/iaxtrunkfix-1.2: ./ cha...   Zoa
• [asterisk-dev] Jitter Buffer   Zoa
• [asterisk-dev] mISDN_dsp and the poll parameter   a.spadaccini at mediatechnologies.it
• [asterisk-dev] Re: [svn-commits] mogorman: branch 1.4 r1490 - in/branches/1.   alex at pilosoft.com
• [asterisk-dev] I: [asterisk-users] Sip answer one side , ring other side   antonio
• [asterisk-dev] Music on hold and multi-language   edantie at canarias.org
• [asterisk-dev] Segmentation fault issue   flavio
• [asterisk-dev] Please help me!!   flavio
• [asterisk-dev] Develop an Asterisk module   Elkeir ismail
• [asterisk-dev] A legal question   mbodbg at gmx.net
• [asterisk-dev] Build instructions out of date?   mogorman
• [asterisk-dev] Skype integration   subrato roy
• [asterisk-dev] show vs. list   sam at bingner.com
• [asterisk-dev] Dropping extra frame of G.729 since we already have a VAD frame at the end   laurent schweizer
• [asterisk-dev] zaptel beta2 tar ball corrupt?   sean
• [asterisk-dev] Scansoft (Nuance) Vocon 3200 Support??   toto at ewoes.com
• [asterisk-dev] RE: Call Hold even   Xiaoming wang
• [asterisk-dev] Jitter Buffer   zoachien at securax.org
• [asterisk-dev] Jitter Buffer   zoachien at securax.org
• [asterisk-dev] Jitter Buffer   zoachien at securax.org
• [asterisk-dev] Jitter Buffer   zoachien at securax.org

Last message date: Tue Oct 31 23:41:54 MST 2006
Archived on: Tue Oct 31 23:41:55 MST 2006

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