October 2006 Archives by thread
Starting: Sun Oct 1 08:08:50 MST 2006
Ending: Tue Oct 31 23:41:54 MST 2006
Messages: 862
- [asterisk-dev] call dropped on transfer (sip) bugid 8064
Adam Goryachev
- [asterisk-dev] Music on hold and multi-language
edantie at canarias.org
- [asterisk-dev] I: [asterisk-users] Sip answer one side ,
ring other side
antonio
- [asterisk-dev] Asterisk + TCP
George Thanos
- [asterisk-dev] Problem implementing call redirection on Zap channels
Johann Hanne
- [asterisk-dev] How to stream audio to external app forspeech
recognition and recognize dtmf in parallel ?
Robert Rozman
- [asterisk-dev] svn syncing backed up again?
James Cloos
- [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] REGISTER uses 401 not 407?
Ed Greenberg
- [asterisk-dev] chan_skinny crashes asterisk (1.4)
Pavel Jezek
- [asterisk-dev] Process flow internal Asterisk
Thong Lam Hai
- [asterisk-dev] Re: [svn-commits] mogorman: branch 1.4 r1490 - in
/branches/1.4: wctdm.c wctdm24xxp.c
Tzafrir Cohen
- [asterisk-dev] Build instructions out of date?
Brian Candler
- [asterisk-dev] Re: [svn-commits] mogorman: branch 1.4 r1490 -
in/branches/1.
Michael Rozov
- [asterisk-dev] [ast-dev] bridging active channels together [fwd]
Roberto Sottile
- [asterisk-dev] DTMF Gereration duplicated. Why?
Paulo Garcia
- [asterisk-dev] Realtime Config Drivers
Richi Plana
- [asterisk-dev] call flow call transfer
Kamran Ahmad
- [asterisk-dev] Development on Asterisk integration to MSC
Alfred M. Ocon
- [asterisk-dev] Sporadic crash in mp3 code
Dave Hawkes
- [asterisk-dev] zaptel build and wct4xxp
Daniel Pocock
- [asterisk-dev] autoconf issues for FreeBSD
Luigi Rizzo
- [asterisk-dev] RFC - "Improved" SIP MWI
Kristian Kielhofner
- [asterisk-dev] Another bounty - app_reload
Kristian Kielhofner
- [asterisk-dev] Asterisk and "305 Use Proxy"
Ariel Monaco
- [asterisk-dev] Re: app_reload?
Jeremy McNamara
- [asterisk-dev] Another bounty - app_reload
Matthew Rubenstein
- [asterisk-dev] SIP to IAX
Thong Lam Hai
- [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)
Thong Lam Hai
- [asterisk-dev] autodial
Santhosh N
- [asterisk-dev] autodial
Santhosh N
- [asterisk-dev] Fixes / enhancements to the ENUM code
Otmar Lendl
- [asterisk-dev] Asterisk 2.0 -- any roadmap?
Jay R. Ashworth
- [asterisk-dev] Need Sip channel notify other channel to answer call
Derek
- [asterisk-dev] Dial without phone
Mir
- [asterisk-dev] proposed change to main/http.c
Luigi Rizzo
- [asterisk-dev] Another bounty - app_reload
Matthew Rubenstein
- [asterisk-dev] GPG signatures
Bill Merriam
- [asterisk-dev] Need An Astcc Mod!
Nate Kapi
- [asterisk-dev] merging manager.conf and http.conf ?
Luigi Rizzo
- [asterisk-dev] A legal question
mbodbg at gmx.net
- [Asterisk-Dev] multiple registrations of same credentials
Bradley
- [asterisk-dev] Transfer app and DTMF via SIP info
Michael Konietzny
- [asterisk-dev] A legal question
Andrew Kirch
- [asterisk-dev] merging manager.conf and http.conf ?
Matthew Rubenstein
- [asterisk-dev] ODBCquery from dial plan
Alexandr Olekhnovich
- [asterisk-dev] ACK during reinvite not sent through SIP Proxy
H Quintana
- [asterisk-dev] Bug 0008069: PRI Channels become unavailable if too
many call files are queued
Andre Courchesne
- [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] Is it a right workaround?
Sergey Okhapkin
- [asterisk-dev] Asterisk Call Time limit
Alexandr Olekhnovich
- [asterisk-dev] Realtime Application
Aaron Daniel
- [asterisk-dev] configure behavior wrt Postgres broken?
Brian Capouch
- [asterisk-dev] VoicemailMain: s preceding mailbox causes big
weirdness
Brian Capouch
- [asterisk-dev] IAX2 deadlock in trunk - out of free iax2 threads
Anton
- [asterisk-dev] My misfire ... Asterisk DoS
J. Oquendo
- [asterisk-dev] ODBCquery trouble
Alexandr Olekhnovich
- [asterisk-dev] SIP compliance question
Roy Sigurd Karlsbakk
- [asterisk-dev] Re: [asterisk-users] 488 Not acceptable here sent
by Asterisk - SIPdebug follows
Dinesh Nair
- [asterisk-dev] Developer Summit at AstriCon Dallas
Steven Sokol
- [asterisk-dev] Daylight Savings Time Change 2007
Butler, Larry
- [asterisk-dev] Getting started with Asterisk development
Austin Seipp
- [asterisk-dev] Invitation to OpenSER Summit @ VoN Berlin
Daniel-Constantin Mierla
- [asterisk-dev] Re: [asterisk-commits] oej: branch
oej/iaxtrunkfix-1.2 r44779 - in /team/oej/iaxtrunkfix-1.2: ./ cha...
Kevin P. Fleming
- [asterisk-dev] SVN is old...
Anton
- [asterisk-dev] [Fwd: [svn-commits] oej: branch oej/iaxtrunkfix-1.2
r44779
Rich Adamson
- [asterisk-dev] Re: ACK during reinvite not sent through SIP Proxy
H Quintana
- [asterisk-dev] Dynamic extension registration problem
Paul Cadach
- [asterisk-dev] SR Level Developer with skills in C, C#,
VOIP(Asterisk) and Linux
Roland Matte
- [asterisk-dev] SR Level Developer with skills in C, C#,
VOIP(Asterisk) and Linux
Andrew Kirch
- [asterisk-dev] canreinvite=no behaviour changed between 1.2.x and
1.4 ?
Luigi Rizzo
- [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: [asterisk-commits] russell: trunk r44876 -
/trunk/channels/chan_sip.c
Luigi Rizzo
- [asterisk-dev] Audio Data Packets
Harish Kasiviswanathan
- [asterisk-dev] SUBSCRIPTION for MWI support for multiple boxes
Chris Carey
- [asterisk-dev] RE: Call Hold even
Xiaoming wang
- [asterisk-dev] Call drop and strange CDR records
CAHEN Fabrice
- [asterisk-dev] zero is a legitimate value for SIP CSeq numbers,
right ?
Luigi Rizzo
- [asterisk-dev] Problem with Asterisk realtime ?
Oded Arbel
- [asterisk-dev] AstriCon hotel full - check these alternates
Steven Sokol
- [asterisk-dev] Mantis acting up?
Dan Austin
- [asterisk-dev] Changes to Manager API in 1.4 ?
John Lange
- [asterisk-dev] show vs. list
sam at bingner.com
- [asterisk-dev] Regarding SIP performance
Chirag Vaishnav
- [asterisk-dev] Multi-homed host routing problems with IAX
Stephan A. Edelman
- [asterisk-dev] Speech Recognition
Stephan A. Edelman
- [asterisk-dev] job offers for asterisk gurus
michel Carette
- [asterisk-dev] SIP MESSAGE methode
René Oertel
- [asterisk-dev] Segmentation fault issue
flavio
- [asterisk-dev] Trying to vfree() nonexistent vm area
Chris B.
- [asterisk-dev] crash - Attempted to delete nonexistent schedule
entry
Dawid Mielnik
- [asterisk-dev] manager.c
Luigi Rizzo
- [asterisk-dev] Re: [asterisk-commits] oej: trunk r45209 -
/trunk/channels/chan_sip.c
Luigi Rizzo
- [asterisk-dev] Speech Recognition
Stephan A. Edelman
- [asterisk-dev] AstriConVideo ! Paris Nov 20-22! Book your calendar!
Olle E Johansson
- [asterisk-dev] [patch] manager.c and http interface
Luigi Rizzo
- [asterisk-dev] Astricon: Serious Asterisk Testing
Jeremy McNamara
- [asterisk-dev] Bugtracker suggestion
Steve Edwards
- [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r45219 -
/trunk/main/manager.c
Kevin P. Fleming
- [asterisk-dev] Patch to enable specifying SDP RTP IP address
Jon Schøpzinsky
- [asterisk-dev] Changes to SIP authentication
Olle E Johansson
- [asterisk-dev] Speech Recognition
Stephan A. Edelman
- [asterisk-dev] Regarding SIP performance
Chirag Vaishnav
- [asterisk-dev] Scansoft (Nuance) Vocon 3200 Support??
toto at ewoes.com
- SV: [asterisk-dev] Patch to enable specifying SDP RTP IP address
Jon Schøpzinsky
- [asterisk-dev] 1.4 Beta and oracle
René Enskat [Teamware GmbH]
- SV: SV: [asterisk-dev] Patch to enable specifying SDP RTP IP address
Jon Schøpzinsky
- [asterisk-dev] SIP Address port remove suggestion
Alexandre Almeida
- [asterisk-dev] Realtime caching or something else?
Olle E Johansson
- [asterisk-dev] Strange functionality of hardware SIP phones on
Asterisk 1.4-branch
Volkov Alexei
- [asterisk-dev] Please help me!!
flavio
- [asterisk-dev] Asterisk Open File Limit
Matt Florell
- [asterisk-dev] Fwd: [svn-commits] rizzo: trunk r45325 -
/trunk/main/manager.c
Olle E Johansson
- [asterisk-dev] About ast_waitfor_nandfs() in channel.c
Juan Carlos Castro y Castro
- [asterisk-dev] License for New Sounds
Jeffrey C. Ollie
- [asterisk-dev] Media transcoding
Ramachandran
- [asterisk-dev] Memory leak issue
Chirag Vaishnav
- [asterisk-dev] Testing Libpri without Asterisk
Paulo Garcia
- [asterisk-dev] asterisk.h missing in /usr/include?
Slav Klenov
- [asterisk-dev] Re: [asterisk-commits] rizzo: trunk r45551 -
/trunk/main/manager.c
Kevin P. Fleming
- [asterisk-dev] Re: [asterisk-commits] mogorman: trunk r45571 -
/trunk/main/manager.c
Luigi Rizzo
- [asterisk-dev] Jitter Buffer
John Lange
- [asterisk-dev] Jitter Buffer and PLC
Martin Vít
- [asterisk-dev] Speed Dails
Al Bochter
- [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] Connection Pooling
Alexandr Olekhnovich
- [asterisk-dev] Memory leak issue
Chirag Vaishnav
- [asterisk-dev] Re: Connection Pooling
Tomislav Parčina
- [asterisk-dev] Builtin transfer
Volkov Alexei
- [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] Asterisk 1.4-beta3 builtin transfer
Volkov Alexei
- [asterisk-dev] Dialplan -- Which version of Asterisk is running me?
Steve Murphy
- [asterisk-dev] mISDN_dsp and the poll parameter
a.spadaccini at mediatechnologies.it
- [asterisk-dev] Jitter Buffer
Pavel Jezek
- [asterisk-dev] Strange problem with new asterisk.
Jonson Player
- [asterisk-dev] NCS PacketCable Patch
Jason Burton
- [asterisk-dev] 1.4.0-beta3 installs files in /
Dan Austin
- [asterisk-dev] IMAP email storage proposal and question
Dax Kelson
- [asterisk-dev] chan_bluetooth with nokia 6230i
Roel Cuppen
- [asterisk-dev] zaptel beta2 tar ball corrupt?
sean
- [asterisk-dev] Developer Summit Topics
Joshua Colp
- [asterisk-dev] Develop an Asterisk module
Elkeir ismail
- [asterisk-dev] https support now in trunk (please read)
Luigi Rizzo
- [asterisk-dev] Push to Talk options in Asterisk.
Jonson Player
- [asterisk-dev] problem dialing to Local channel
Maxi Belino
- [asterisk-dev] ast_mutex_lock
Alexandr Olekhnovich
- [asterisk-dev] chan_bluetooth patch for SVN trunk
Brian Candler
- [asterisk-dev] threading
Alexandr Olekhnovich
- [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] Asterisk + MRCP for Speech Resources (TTS / ASR) ?
Josh McAllister
- [asterisk-dev] [ANNOUNCE] chan_celliax,
for managing cellphones via Asterisk, first release
Giovanni Maruzzelli
- [asterisk-dev] Asterisk + MRCP for Speech Resources (TTS / ASR) ?
Josh McAllister
- [asterisk-dev] chan_bluetooth with nokia 6230i
Boris Bakchiev
- [asterisk-dev] deadlock in ast_custom_function_register?
Yuan Qin
- *****SPAM***** [asterisk-dev] asterisk 1.4 problem with call queues
Dean Bath
- [asterisk-dev] Developers Summit Conference Call
Jeremy McNamara
- [asterisk-dev] How to busy out PRI channels?
Tony Mountifield
- [asterisk-dev] Simple example for call transfer.
Jonson Player
- [asterisk-dev] mutex type selection
SF Markus Elfring
- [asterisk-dev] build options for Pthread usage
SF Markus Elfring
- [asterisk-dev] Fwd: [svn-commits] kpfleming: branch 1.4 r46153 - in
/branches/1.4: channels/ main/
Olle E Johansson
- [asterisk-dev] static code analysis
SF Markus Elfring
- [asterisk-dev] handling of strncpy calls
SF Markus Elfring
- [asterisk-dev] Proposing meetme patch
Koopmann, Jan-Peter
- [asterisk-dev] Re: [asterisk-commits] kpfleming: branch 1.4 r46200
- in /branches/1.4: apps/ cdr/ channels/ main/ p...
Luigi Rizzo
- [asterisk-dev] Proposing meetme patch
Dan Austin
- [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 :)
Julian Lyndon-Smith
- [asterisk-dev] Fwd: [Iaxclient-devel] Video codec negotiation in IAX
Mihai Balea
- [asterisk-dev] Pthread wrapper updates
SF Markus Elfring
- [asterisk-dev] TDM400P on hppa-linux
Kai Holthaus
- [asterisk-dev] Floating audio disappear issue
Anton
- [asterisk-dev] Regarding SIP performance
Chirag Vaishnav
- [asterisk-dev] Re: [asterisk-commits] oej: branch 1.4 r46252 -
/branches/1.4/channels/chan_sip.c
Luigi Rizzo
- [asterisk-dev] Skype integration
subrato roy
- [asterisk-dev] Re: Pthread wrapper updates
SF Markus Elfring
- *****SPAM***** [asterisk-dev] Segmentation Fault on 1.4 - Bug ref:8228
Dean Bath
- [asterisk-dev] users and peers (and phones and trunks) ?
Luigi Rizzo
- [asterisk-dev] where do we put GUIs ?
Luigi Rizzo
- [asterisk-dev] Pthread wrapper updates
SF Markus Elfring
- [asterisk-dev] Skype integration
Paulo Mannheimer
- [asterisk-dev] manager events to show file and line info ?
Luigi Rizzo
- [asterisk-dev] Asterisk 1.4 beta 3 on MAC OS X
David Parcerisa
- [asterisk-dev] Dynamically removing a provider registration entry
Craig Edwards
- [asterisk-dev] [patch] adding username from registration to peer
state ?
Luigi Rizzo
- [asterisk-dev] Re: How to busy out PRI channels?
Freddi Hansen
- [asterisk-dev] Re: How to busy out PRI channels?
Freddi Hansen
- [asterisk-dev] zaptel helper script
Tzafrir Cohen
- [asterisk-dev] Need explaination about ARA and Asterisk 1.4-beta3
Raffaele Porzio
- [asterisk-dev] [patch] [issue 7837] MySQL error log
Andrea Spadaccini
- [asterisk-dev] Question about using Time-Warner Versapak Power T-12
with Asterisk
Michael Lavelle
- [asterisk-dev] Unable to compile chan_capi with Asterisk 1.4
Gregory Duchatelet
- [asterisk-dev] bug: autocreate peer + sippeers table entry => auth
required
Mark Price
- [asterisk-dev] VoiceMail Modularization
Carlton O'Riley
- [asterisk-dev] res_config_pgsql in SVN-trunk?
Brian Capouch
- [asterisk-dev] Channel file descriptors
John Martin
- [asterisk-dev] Re: [svn-commits] oej: trunk r46392 -
/trunk/channels/chan_sip.c
Martin Vít
- [asterisk-dev] more compact HTML documentation
Tzafrir Cohen
- [asterisk-dev] Filed a chan_sip bug (1.4/trunk) ? Please re-test!
*** IMPORTANT ***
Johansson Olle E
- [asterisk-dev] Regarding SIP performance
Chirag Vaishnav
- [asterisk-dev] SVN out of synch...
Olle E Johansson
- [asterisk-dev] November: The Asterisk month :-)
Olle E Johansson
- [asterisk-dev] astricon GUI demo
Isack Waserman
- [asterisk-dev] VoiceMail Modularization
Watkins, Bradley
- [asterisk-dev] H.324M<->SIP/H.323 gateways
Guillaume Fraysse
- [asterisk-dev] Re: [asterisk-commits] oej: trunk r46409 -
/trunk/main/rtp.c
Kevin P. Fleming
- *****SPAM***** [asterisk-dev] Segmentation Fault - Logged on Bugs 8228
Dean Bath
- [asterisk-dev] Re: [asterisk-commits] oej: trunk r46513 - in
/trunk: funcs/ include/asterisk/ main/
Kevin P. Fleming
- [asterisk-dev] Re: How to busy out PRI channels?
Josh McAllister
- [asterisk-dev] Definitions
Alexandr Olekhnovich
- [asterisk-dev] Fedora Core 6 (FC6) and Asterisk-1.2.13 and
Zaptel-1.2.10 compile problems
Michael J. Tubby G8TIC
- [asterisk-dev] Bug resolution efficieny
Anu Gupta
- [asterisk-dev] PRI HANGUPCAUSE
Anton
- [asterisk-dev] ASTERISK_FILE_VERSION
Alexandr Olekhnovich
- [asterisk-dev] Dropping extra frame of G.729 since we already have
a VAD frame at the end
laurent schweizer
- [asterisk-dev] Zaptel/Asterisk - Q.SIG status
Pavel Jezek
- [asterisk-dev] Realtime caching or something else? (SIP retransmit
#1)
Olle E Johansson
- [asterisk-dev] Realtime caching or something else? (SIP
retransmit#1)
Watkins, Bradley
- [asterisk-dev] usecount implementation in the new loader ?
Luigi Rizzo
- [asterisk-dev] Strange code in chan_zap.c ?
Tony Mountifield
- [asterisk-dev] Realtime caching or something else? (SIP
retransmit#1)
Watkins, Bradley
- [asterisk-dev] How to contribute ?
Dome C.
- [asterisk-dev] Dialplan code to simulate calls
Andre Courchesne
- [asterisk-dev] Re: How to busy out PRI channels?
Matthew Fredrickson
- [asterisk-dev] SIP Channel does not release.
Anton
- [asterisk-dev] asterisk imap voicemail storage
Matt O'Gorman
- [asterisk-dev] Loading module cdr_mysql crashes 1.4b3
John Lange
- [asterisk-dev] Subversion server maintenance
Kevin P. Fleming
- [asterisk-dev] why 'o' (preserve original callerid) is not default
in app_dial.c ?
Luigi Rizzo
- [asterisk-dev] IAX2 still broken
Anton
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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