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- [asterisk-dev] Build instructions out of date?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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