November 2005 Archives by thread
      
      Starting: Tue Nov  1 04:12:38 MST 2005
         Ending: Wed Nov 30 23:49:00 MST 2005
         Messages: 736
     
- [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you can
	never use pthread_mutex_t?
 
steve at daviesfam.org
- [Asterisk-Dev] Zap gain changing patch
 
Denis Smirnov
- [Asterisk-Dev] chan_misdn in asterisk beta 2 ?
 
Zoa
- [Asterisk-Dev] chan_misdn in asterisk beta 2 ? - my mistake 
 
Zoa
- [Asterisk-Dev] Asterisk make clean and make looping??
 
Robert Webb
- [Asterisk-Dev] reload causing SIP phone to not be registered
 
James Armstrong
- [Asterisk-Dev] reload causing SIP phone to not be registered
 
Brian C. Fertig
- [Asterisk-Dev] New ChangeLog in CVS HEAD Asterisk tree
 
Kevin P. Fleming
- [Asterisk-Dev] Re: multiple registrations of same credentials
 
Jonathan k. Creasy
- [Asterisk-Dev] ztcfg and wctdm/wcfxo
 
Tzafrir Cohen
- [Asterisk-Dev] Re: multiple registrations of same credentials
 
Jonathan k. Creasy
- [Asterisk-Dev] Missing audio from Zaptel channels
 
Rod Bacon
- [Asterisk-Dev] Billing and consultative transfer
 
Sebastian Zaprzalski
- [Asterisk-Dev] reload causing SIP phone to not be registered
 
Asterisk User
- [Asterisk-Dev] Re: multiple registrations of same credentials
 
Michael Procter
- [Asterisk-Dev] Send tone to caller on answer
 
Ed Greenberg
- [Asterisk-Dev] Re: multiple registrations of same credentials
 
Michael Procter
- [Asterisk-Dev] New app: app_page comments
 
Edwin Horton
- [Asterisk-Dev] since 2005-10-28,
 pthread_mutex_t can never be used in apps that
 include asterisk's lock.h ??
 
steve at daviesfam.org
- [Asterisk-Dev] Local channel optimisation
 
Tony Mountifield
- [Asterisk-Dev] lock.h change on 10-28 (rev 1.39) means you
	cannever use pthread_mutex_t?
 
Jerris, Michael MI
- [Asterisk-Dev] Re: [Asterisk-Users] Warning -- chan_iax2.c:
	ast_sched_runq tasks
 
Rich Adamson
- [Asterisk-Dev] echo, delay and dropped packets in channel driver
 
Ben Kramer
- [Asterisk-Dev] echo, delay and dropped packets in channel driver
 
Kris Boutilier
- [Asterisk-Dev] echo, delay and dropped packets in channel driver
 
Boris Bakchiev
- [Asterisk-Dev] Re: Fw: Portace JUDR. Jan Grosam
 
Martin Vit
- [Asterisk-Dev] Send tone to caller on answer
 
Andreas Sikkema
- [Asterisk-Dev] [repost] cli output issue
 
Christopher L. Wade
- [Asterisk-Dev] [repost] cli output issue
 
Jerris, Michael MI
- [Asterisk-Dev] ODBC Voicemail Storage
 
Edwin Horton
- [Asterisk-Dev] Additions to function DUNDiLookup
 
Watkins, Bradley
- [Asterisk-Dev] Changes to the way DISA and CDR work together
 
Ed Greenberg
- [Asterisk-Dev] echo, delay and dropped packets in channel driver
 
Kris Boutilier
- [Asterisk-Dev] echo, delay and dropped packets in channel driver
 
Boris Bakchiev
- [Asterisk-Dev] ZT_DIAL Operation
 
Sergio Serrano
- [Asterisk-Dev] Asterisk Locking - queue application
 
Dov Bigio
- [Asterisk-Dev] always_inline change...
 
Luigi Rizzo
- [Asterisk-Dev] Meetme: Sending DTMF to other users in a conference
 
Vamsi Pottangi
- [Asterisk-Dev] other -Werror issues - time_t printf format...
 
Luigi Rizzo
- [Asterisk-Dev] RFC2833 Event Duration
 
Ryan Courtnage
- [Asterisk-Dev] Asterisk or Polycom Bug?
 
Will McCown
- [Asterisk-Dev] chan_modem_i4l (only noise is received,
	reason for the problem)
 
David Arendt
- [Asterisk-Dev] musiconhold -vs- musicclass problems setting the
	differnt class of music 
 
Ronald Hartmann
- [Asterisk-Dev] jb-rtp-sip.scx.new.patch compiled error in Fedora 3
 
Raymond Chen
- [Asterisk-Dev] AST_FRAME_DATA
 
Christian Richter
- [Asterisk-Dev] Security issue mumblings
 
John Todd
- [Asterisk-Dev] 
	extensions.conf limitations, and how to overcome them
 
Quinn Weaver
- [Asterisk-Dev] has somebody ever tested faxdetection ?
 
Thomas Häger
- [Asterisk-Dev] Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.
 
Harald Baron
- [Asterisk-Dev] E1 connection problem
 
Hugh Jackman
- [Asterisk-Dev] Re: TDMoE + kernel badness
 
Fabio Ferrari
- [Asterisk-Dev] Current HEAD: res_config_odbc.c compilation failure
 
Patrick
- [Asterisk-Dev] How to start?
 
Isack Waserman
- [Asterisk-Dev] [OTAnn] Feedback
 
shenanigans
- [Asterisk-Dev] Patch to allow Dial() to ignore call forward
 
John Lange
- [Asterisk-Dev] say.c cleanup
 
Luigi Rizzo
- [Asterisk-Dev] chan_bluetooth
 
Sven Boeckelmann
- [Asterisk-Dev] asterisk.spec and the redhat dir
 
Jason Pyeron
- [Asterisk-Dev] trivial patch requesting comments before filing
 
Jason Pyeron
- [Asterisk-Dev] [OTAnn] Feedback
 
Jonathan k. Creasy
- [Asterisk-Dev] trying to generate bt,
	but info threads not working...
 
Ron Arts
- [Asterisk-Dev] New manager API command
 
Saul Diaz
- [Asterisk-Dev] New manager API command
 
Jonathan k. Creasy
- [Asterisk-Dev] Asterisk 1.2.0-rc1 Released!
 
Kevin P. Fleming
- [Asterisk-Dev] CVS HEAD tree frozen until release
 
Kevin P. Fleming
- [Asterisk-Dev] sip show peers doesn't work?
 
Chih-Wei Huang
- [Asterisk-Dev] Asterisk Fax support using T.38
 
Lilantha Karunaratne
- [Asterisk-Dev] bug 4252 - increasing delay over time channels in
	MeetMe
 
Chih-Wei Huang
- [Asterisk-Dev] Bug 5590 and Release Candidates
 
Paul Davidson
- [Asterisk-Dev] Can anoncvs make 'cvs commit' in CVS asterisk ?
 
Juan Manuel Guerrero de los Ríos
- [Asterisk-Dev] Inconsistencies between .call files and Manager API
 
Tony Mountifield
- [Asterisk-Dev] Develop SoftPhone with IAXclient!!
 
Alberto Lavariega Arista
- [Asterisk-Dev] E1 channel allocation
 
Denis Galvão
- [Asterisk-Dev] Collect Calls
 
IT
- [Asterisk-Dev] Voicemail password changes not sticking
 
James Armstrong
- [Asterisk-Dev] Monitor Function broken since Beta-2 ?
 
Max Bressel
- [Asterisk-Dev] Asterisk 1.2.0-beta2 ParkAndAnnounce question
 
Steve Blair
- [Asterisk-Dev] UA
 
Isack Waserman
- [Asterisk-Dev] Recommendations for SIP PBX/SP interoperability draft
 
John Todd
- [Asterisk-Dev] Asterisk on Cygiwn
 
Jerris, Michael MI
- [Asterisk-Dev] Get number function
 
Tristram Graham
- [Asterisk-Dev] say.c cleanup -feedback wanted
 
Luigi Rizzo
- [Asterisk-Dev] Problems with zap lines in 1.2 Beta 2 or 1.2 RC1,
	but ok in Beta1
 
Will McCown
- [Asterisk-Dev] Calling apps and parsing return values
 
Ed Greenberg
- [Asterisk-Dev] More work on DISA
 
Ed Greenberg
- [Asterisk-Dev] Asterisk 1.2 will have support for SIMPLE
	notifications of device presence
 
harry gaillac
- [Asterisk-Dev] Another weird dialplan question
 
asterisk at funkspiel.org
- [Asterisk-Dev] Another weird dialplan question
 
Jerris, Michael MI
- [Asterisk-Dev] 1.2.0-beta1 MeetMe bug?
 
Dan Austin
- [Asterisk-Dev] $10 Bounty for Bug #5407
 
Eric "ManxPower" Wieling
- [Asterisk-Dev] $10 Bounty for Bug #5407
 
Alexander Lopez
- [Asterisk-Dev] Asterisk 1.2.0-rc2 Released!
 
Kevin P. Fleming
- [Asterisk-Dev] Re: 1.2.0-beta1 MeetMe bug?
 
Dan Austin
- [Asterisk-Dev] chan_exosip2 for asterisk
 
harry gaillac
- [Asterisk-Dev] Asterisk on Cygiwn
 
don vanfossen
- [Asterisk-Dev] chan_exosip2 for asterisk
 
Jerris, Michael MI
- [Asterisk-Dev] Asterisk on Cygiwn
 
Jerris, Michael MI
- [Asterisk-Dev] How to get Referred-By header
 
Steve Blair
- [Asterisk-Dev] chan_exosip2
 
harry gaillac
- [Asterisk-Dev] Hardware Manual
 
Jerris, Michael MI
- [Asterisk-Dev] Interrupted playback of messages (ast_waitstream)
 
Lars Sundqvist
- [Asterisk-Dev] MeetMe participant time tracking
 
Dan Austin
- [Asterisk-Dev] MeetMe participant time tracking
 
Dan Austin
- [Asterisk-Dev] MeetMe participant time tracking
 
Dan Austin
- [Asterisk-Dev] another little try
 
Roy Sigurd Karlsbakk
- [Asterisk-Dev] banned from the *-users list?
 
Roy Sigurd Karlsbakk
- [Asterisk-Dev] Realtime fails using unixODBC and FreeTDS against
	MSSQL
 
Trælnes AS
- [Asterisk-Dev] Asterisk on Cygiwn
 
Jonathan k. Creasy
- [Asterisk-Dev] app_datetime unused ? and sayunixtime cleanup...
 
Luigi Rizzo
- [Asterisk-Dev] Small question about CDR's
 
Jan Saell
- [Asterisk-Dev] Changing Port 5060 with 81
 
Abdul Lateef Khan
- [Asterisk-Dev] ooh323 crash
 
Ma Zhiyong
- [Asterisk-Dev] Re: mtp-2 (janvb@caselaboratories.com)
 
Hadi Jadallah
- [Asterisk-Dev] ooh323 cause asterisk reload broken
 
Ma Zhiyong
- [Asterisk-Dev] Getting caller id
 
ast guy
- [Asterisk-Dev] Delphi ActiveX component
 
contato at tomasfrota.eti.br
- [Asterisk-Dev] Delphi ActiveX component
 
Jerris, Michael MI
- [Asterisk-Dev] LDAP RealTime driver
 
Manuel Guesdon
- [Asterisk-Dev] 1.2.0 bug or mis-config? chan_sip.c
 
Bryan Heitman
- [Asterisk-Dev] Asterisk 1.2 Released!
 
Asterisk Development Team
- [Asterisk-Dev] Handling AST_FORMAT_SLINEAR at 48000Hz
 
voipwala hindustani
- [Asterisk-Dev] A question about G option of Dial
 
Chih-Wei Huang
- [Asterisk-Dev] RE: Changing Port 5060 with 81
 
Abdul Lateef Khan
- [Asterisk-Dev] New asterisk management tool
 
snacktime
- [Asterisk-Dev] Install destination on Solaris
 
Simon Lockhart
- [Asterisk-Dev] gmake error for asterisk-addons-1.2.0
 
Vahan Yerkanian
- [Asterisk-Dev] call forwarding
 
Abdul Lateef Khan
- [Asterisk-Dev] Module that works in Asterisk 1.0.x and 1.2.x
 
Juan Jose Comellas
- [Asterisk-Dev] INSTALL_PREFIX
 
Hans Fugal
- [Asterisk-Dev] Version 1.2.0 chan_h323 (ooh323c)
 
Dan Austin
- [Asterisk-Dev] Version 1.2.0 chan_h323 (ooh323c)
 
Dan Austin
- [Asterisk-Dev] Version 1.2.0 chan_h323 (ooh323c)
 
Dan Austin
- [Asterisk-Dev] app_page
 
Michael Anderson
- [Asterisk-Dev] ast_load_realtime
 
José Pablo Ezequiel Fernández
- [Asterisk-Dev] Asterisk app_ices
 
Daniel Mikusa
- [Asterisk-Dev] cvs revision tags
 
Rod Dorman
- [Asterisk-Dev] Need help for developing application like
	"musiconhold" . . .
 
Pratik Thakkar
- [Asterisk-Dev] reload and hints problem
 
Sergio Chersovani
- [Asterisk-Dev] Sox Mixing Command With Monitor?
 
Nate Kapi
- [Asterisk-Dev] CVS server problem?
 
Brian Capouch
- [Asterisk-Dev] generation of config files from source code ?
 
Luigi Rizzo
- [Asterisk-Dev] CAC in Asterisk
 
Mohamed A. Gombolaty
- [Asterisk-Dev] Compiler debug switch in the main Asterisk Makefile
 
Steve Rodgers
- [Asterisk-Dev] ChanIsAvail() works for SIP?
 
Guo-Wei Chiuan
- [Asterisk-Dev] Problem in call conference
 
Vinay
- [Asterisk-Dev] AGIphp Installation
 
Abdul Lateef Khan
- [Asterisk-Dev] AGIphp Installation
 
Abdul Lateef Khan
- [Asterisk-Dev] AGIphp Installation
 
Abdul Lateef Khan
- [Asterisk-Dev] Re: [Asterisk-Users] Asterisk crash: "using
 deprecated BYE/Also	transfer method"
 
Olle E. Johansson
- [Asterisk-Dev] Silence Detection in G729 payload
 
suresh suresh
- [Asterisk-Dev] error while building zaptal
 
Isack Waserman
- [Asterisk-Dev] Queue - Transfer & server stability
 
Dov Bigio
- [Asterisk-Dev] Compiler debug switch in the main Asterisk
	Makefile
 
hwstar at rodgers.sdcoxmail.com
- [Asterisk-Dev] Call-Limit
 
Rushowr
- [Asterisk-Dev] Compiler debug switch in the main
	Asterisk	Makefile
 
hwstar at rodgers.sdcoxmail.com
- [Asterisk-Dev] FYI: Internet-Draft for IAX: Inter-Asterisk eXchange
	Version 2
 
Rod Dorman
- [Asterisk-Dev] Source Code for PalmTool 1.47 available anywhere?
 
Ralf P. Loserth
- [Asterisk-Dev] 482 Loop Detected problem
 
Charles Huang
- [Asterisk-Dev] Re: Asterisk make clean and make looping?? (Possible
	solution)
 
David Taylor
- [Asterisk-Dev] Action item :: Check if your patch is up to date!
 
Olle E. Johansson
- [Asterisk-Dev] Echo cancel on PRI
 
imran ahmed
- [Asterisk-Dev] regular expression behavior
 
snacktime
- [Asterisk-Dev] voicemail
 
Isack Waserman
- [Asterisk-Dev] Re: Queue - Transfer & server stability
 
alan
- [Asterisk-Dev] get register info from peer section in sip.conf ?
 
Luigi Rizzo
- [Asterisk-Dev] Sox Mixing Command With Monitor?
 
Nate Kapi
- [Asterisk-Dev] chan_bluetooth and asterisk-1.2
 
Dan
- [Asterisk-Dev] Problem with hanging up a SIP channel
 
Marc Haisenko
- [Asterisk-Dev] Chan_sip version 1, 2 and NG: 3
 
Olle E. Johansson
- [Asterisk-Dev] Tr: Re: [Asterisk-Users] open letter
 
harry gaillac
- [Asterisk-Dev] Nat
 
harry gaillac
- [Asterisk-Dev] nat
 
harry gaillac
- [Asterisk-Dev] hello
 
harry gaillac
- [Asterisk-Dev] 
	Chan_sip: video capabilities, call bandwidth and RTCP
 
John Martin
- [Asterisk-Dev] festival voice
 
Abdul Lateef Khan
- [Asterisk-Dev] 	Chan_sip: video capabilities,
	call bandwidthand RTCP
 
John Martin
- [Asterisk-Dev] G.729 SDP bug in sip_chan.c  (asterisk v1.2.0)
 
hcb+asterisk-dev at unco.de
- [Asterisk-Dev] HELP! on disconnecting stale calls.
 
Paradise Dove
- [Asterisk-Dev] Sip Invite in application
 
Markus Monka
- [Asterisk-Dev] asterisk 1.2  g729  compile errors
 
Diyanat Ali
- [Asterisk-Dev] no DNID copied when bridging channels ?
 
Luigi Rizzo
- [Asterisk-Dev] bug or expected behaviour : vars in template
	contexts (1.2.0/HEAD)
 
Bruno Voigt
- [Asterisk-Dev] MeetMe assumes 20ms...
 
Dan Austin
- [Asterisk-Dev] MeetMe assumes 20ms...
 
Dan Austin
- [Asterisk-Dev] Require Help in Detection of Human Voice or
	Answering machine on Called telephone number
 
Muhammad Asim Sajjad
- [Asterisk-Dev] possible bug in app_dial.c ?
 
Luigi Rizzo
- [Asterisk-Dev] Tr: RE : [Asterisk-Users] Asterisk doesn't start
 
harry gaillac
- [Asterisk-Dev] Help Connecting Cisco Router to Asterisk
 
ddiffa
- [Asterisk-Dev] dialplan hint case sensitive
 
Sergio Chersovani
- [Asterisk-Dev] Conditional Goto and GoSub use 'n'?
 
Rushowr
- [Asterisk-Dev] sip gateway provider recommendations?
 
John Brookes
- [Asterisk-Dev] Line status detection for Zap ifaces ( fxo )
 
Gabriel Rojas
- [Asterisk-Dev] Help X101p kewlstart
 
MZ
- [Asterisk-Dev] Require Help in Detection of Human Voice or
	Answering machine on Called telephone number
 
Justin Newman
- [Asterisk-Dev] app_festival normal behaviour??
 
Moises Silva
- [Asterisk-Dev] res_config_mysql.c connection problems => bug
 
Loic DIDELOT
- [Asterisk-Dev] Line status detection for Zap ifaces ( fxo ) (
	cont'd )
 
Gabriel Rojas
- [Asterisk-Dev] Re: Require Help in Detection of Human Voice or
	Answering machine on Called telephone number
 
Claude Klimos
- [Asterisk-Dev] DID number not saved when incoming fax detected on
	Zap
 
James Armstrong
- [Asterisk-Dev] Asterisk project converts to Subversion version
	control system
 
Asterisk Development Team
- [Asterisk-Dev] gmake issues
 
Luigi Rizzo
- [Asterisk-Dev] SIP media negotiation problem
 
anthony thomas
- [Asterisk-Dev] Subversion file permissions
 
brett at websmyths.com
- [Asterisk-Dev] chan_sip/sip.conf
 
bosinclar at laposte.net
- [Asterisk-Dev] zaptel-1.2.0 compilation fails on 2.5.15-rc2-git6
 
Patrick
- [Asterisk-Dev] Subversion checkouts
 
Richard Scobie
- [Asterisk-Dev] New SVN email subject
 
Mark Hulber
- [Asterisk-Dev] How are SIP calls connected/bridged ?
 
 Arnaud 
- [Asterisk-Dev] How are SIP calls connected/bridged ?
 
Alexander Lopez
- [Asterisk-Dev] sco connection error: connection reset by peer
 
Khaled
- [Asterisk-Dev] Missing RTP header in iLBC
 
Jan Saell
- [Asterisk-Dev] 1.2.0 Manager Action: Agents bug?
 
alan
- [Asterisk-Dev] asterisk 1.2  g729  compile errors
 
Mik Cheez
- [Asterisk-Dev] AST_FLAG_DEFER_DTMF (would like dialogic-r4 like
	semantics)
 
Greg Lim
- [Asterisk-Dev] 1.2.0 Manager Action: Agents bug?
 
Alexander Lopez
- [Asterisk-Dev] An Ast-user question for Ast-dev experts
 
Hugh Jackman
- [Asterisk-Dev] Disposition failed in Asterisk-1.2.0-stable
 
Aaron Daniel
- [Asterisk-Dev] New Jersey ATT Vocie T1 Asterisk Toll free number
	not working
 
Charles Huang
- [Asterisk-Dev] app_conference errors
 
Diyanat Ali
    
      Last message date: 
       Wed Nov 30 23:49:00 MST 2005
    Archived on: Tue Sep  5 14:27:44 MST 2006
    
   
     
     
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