September 2005 Archives by thread
Starting: Thu Sep 1 01:41:47 MST 2005
Ending: Fri Sep 30 21:02:18 MST 2005
Messages: 822
- reason values (was: [Asterisk-Dev] Re: Originate Call and Unique ID)
Joerg Lauer
- [Asterisk-Dev] smsq
Drew Wells
- [Asterisk-Dev] asterisk doesn't send '200 result..." after a
VERBOSE command
Arnaud Ligot
- [Asterisk-Dev] question about ast_async_goto
Sergio Chersovani
- [Asterisk-Dev] ZyXel 2000W
John Matthews
- [Asterisk-Dev] undefined symbol: ast_config_load error with latest
cvs install
ddiffa
- [Asterisk-Dev] ASTCC-adding more than one trunk to one route
Faramarz Amidafshar
- [Asterisk-Dev] undefined symbol: ast_config_load error with
latest cvs install
ddiffa
- [Asterisk-Dev] Account Code?
Matt
- [Asterisk-Dev] Doxygen and comments
José Pablo Ezequiel Fernández
- [Asterisk-Dev] Realtime agents.
José Pablo Ezequiel Fernández
- [Asterisk-Dev] advance debugging?
david.j as
- [Asterisk-Dev] Compiling Asterisk for dual architectue
Aarno Syvänen
- [Asterisk-Dev] Device State
Mark Edwards
- [Asterisk-Dev] spelling error in iax.conf.sample
Michiel van Baak
- [Asterisk-Dev] Zaptel compile error after kernel switch to SMP
David Cook
- [Asterisk-Dev] cable modems?
julian howard
- [Asterisk-Dev] cable modems?
Colin Anderson
- [Asterisk-Dev] Fwd: CallerID and CDR
Sherwood McGowan
- [Asterisk-Dev] Re: How to measure delay in meetme?
Clive Nicolson
- [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP
Matt
- [Asterisk-Dev] Error while starting Asterisk
jibumathewemail-ast at yahoo.com
- [Asterisk-Dev] Is CVS-HEAD the next 1.2 Beta?
Brian Capouch
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Daniel Pocock
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Paradise Dove
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Tzafrir Cohen
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Daniel Pocock
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
David Woodhouse
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Jeremy McNamara
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Adam Goryachev
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Andrew Kohlsmith
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Daniel Pocock
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Andrew Kohlsmith
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Andrew Kohlsmith
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Tzafrir Cohen
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Paradise Dove
- [Asterisk-Dev] Open G.729 / G.723.1 update, fixed memory leak
Edwin Groothuis
- [Asterisk-Dev] Input and testing.
Brian West
- [Asterisk-Dev] How to write an asterisk application
Thang Nguyen
- [Asterisk-Dev] [patch] IM Manager Event
Andreas Anderson
- [Asterisk-Dev] Call and video bandwidths
John Martin
- [Asterisk-Dev] I need help to make a simple iax switch/proxy
Goldenear
- [Asterisk-Dev] Call and video bandwidths
John Martin
- [Asterisk-Dev] Ring requested on channel already in use
alan
- [Asterisk-Dev] Authorisation field for voicemail.conf
Mark Elkins
- [Asterisk-Dev] Strange code in chan_zap.c
steve at daviesfam.org
- [Asterisk-Dev] undefined symbol: ast_config_load error with
latest cvs install
ddiffa
- [Asterisk-Dev] Ring requested on channel already in use
alan
- [Asterisk-Dev] * Crashed
Chee Foong
- [Asterisk-Dev] play tone for auto monitoring
Keiron Liddle
- [Asterisk-Dev] Asterisk Compiled on Arm with static libraries
Jawhny Cooke
- [Asterisk-Dev] IAX PBX responds to IAX registration with expires
time=0
Maciek
- [Asterisk-Dev] PrivacyManager on a per user basis?
Sherwood McGowan
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 14, Issue 19
alan
- [Asterisk-Dev] RSA auth broken in IAX2?
Olle E. Johansson
- [Asterisk-Dev] RSA auth broken in IAX2?
Jerris, Michael MI
- [Asterisk-Dev] callerid spill
Gurminder Arora
- [Asterisk-Dev] Answering Machine detection
Nitin Joshi
- [Asterisk-Dev] Ring requested on channel already in use
alan
- [Asterisk-Dev] Asterisk Bounty VoiceMail-n-Email Synchronization
=$1125
Matthew Butt
- [Asterisk-Dev] Current status on _outgoing_ Swedish/Dutch DTMFCLIP
for TDM400 FXS interfaces?
Werner Johansson
- [Asterisk-Dev] Agents in realtime
José Pablo Ezequiel Fernández
- [Asterisk-Dev] need help to compile asterisk under ubuntu
julienasterisk t
- [Asterisk-Dev] rfc3389 status?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] app_dial.c ANSWEREDTIME value is not return when
caller cancels the call
Raymond Chen
- [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriate when
ALAW is in use
steve at daviesfam.org
- [Asterisk-Dev] Subscribe/Notify support
David (Dalei) Liu
- [Asterisk-Dev] rfc3389 status?
Jerris, Michael MI
- [Asterisk-Dev] chan_zap bug - pri spans end up stuck DOWN
steve at daviesfam.org
- [Asterisk-Dev] Question about RTP port usage
Bryan Field-Elliot
- [Asterisk-Dev] Maximum wait for I/O in bridged channels
Juan Jose Comellas
- [Asterisk-Dev] spandsp crashing asterisk
Atif Rasheed
- [Asterisk-Dev] Question about RTP port usage
Jerris, Michael MI
- [Asterisk-Dev] res_monitor improvement
Tamas
- [Asterisk-Dev] getting started
Jonathan k. Creasy
- [Asterisk-Dev] getting started
Jonathan k. Creasy
- [Asterisk-Dev] getting started
Jonathan k. Creasy
- [Asterisk-Dev] getting started
Jonathan k. Creasy
- [Asterisk-Dev] zaptel "rest value" of 0xff is not appropriatewhen
ALAW is in use
Michael Procter
- [Asterisk-Dev] Re: Answering Machine detection (steve@daviesfam.org)
manish kumar
- [Asterisk-Dev] startup
himanshoo kumar saxena
- [Asterisk-Dev] Asterisk and AMD64 - Crashes
Joseph
- [Asterisk-Dev] Asterisk and AMD64 - Crashes
Brian C. Fertig
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 14, Issue 31
Adam Gundy
- [Asterisk-Dev] Bug in SIP message Handling
Gene Willingham
- [Asterisk-Dev] Asterisk and AMD64 - Crashes
Brian C. Fertig
- <SPAM>RE: [Asterisk-Dev] Asterisk and AMD64 - Crashes
Jerris, Michael MI
- [Asterisk-Dev] Asterisk and AMD64 - Crashes
Woody Anderson
- [Asterisk-Dev] AgentCallbackLogin, 1.2-beta1 issues
alan
- [Asterisk-Dev] IAX busy indication but not invalidating the call
Lee Howard
- [Asterisk-Dev] IAX timing of VOICE frames
Lee Howard
- [Asterisk-Dev] pass through g.723/g.729 as a b2b ua
Ma Zhiyong
- [Asterisk-Dev] Re: Answering Machine detection
Nitin Joshi
- [Asterisk-Dev] zaptel "rest value" of 0xff is not
appropriatewhenALAW is in use
Michael Procter
- [Asterisk-Dev] codec issues
Dov Bigio
- [Asterisk-Dev] Asterisk and AMD64 - Crashes
Jerris, Michael MI
- [Asterisk-Dev] Re: AgentCallbackLogin, 1.2-beta1 issues
alan
- [Asterisk-Dev] Is the ChanIsAvail command thread safe?
hugolivude
- [Asterisk-Dev] Real-time Linux claims single-digit microsecond
responsiveness
Carlos Antunes
- [Asterisk-Dev] Asterisk and AMD64 - Crashes
Woody Anderson
- [Asterisk-Dev] Asterisk and AMD64 - Crashes
Woody Anderson
- [Asterisk-Dev] Re: Answering Machine detection
Gilmore, Gerry
- [Asterisk-Dev] PLEASE HELP!! CALLERID FAILS!!
John Hill
- [Asterisk-Dev] Re: Answering Machine detection
Gilmore, Gerry
- [Asterisk-Dev] asterisk 'stable'?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] asterisk 'stable'?
Jerris, Michael MI
- [Asterisk-Dev] Asterisk run as non root
Joseph
- [Asterisk-Dev] Digium Dual T1 Spans 2xxp
Ricky Keele
- [Asterisk-Dev] Impossible for an extension to send Flash to Bell CO?
hugolivude
- [Asterisk-Dev] Dual Span T1
Ricky Keele
- [Asterisk-Dev] Not for 1.2,
was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev
Roy Sigurd Karlsbakk
- [Asterisk-Dev] Asterisk 1.0.9 long term stability
Sig Lange
- [Asterisk-Dev] R1.502 of chan_zap.c kills callerid on a x101p
John Hill
- [Asterisk-Dev] asterisk 'stable'?
Woody Anderson
- [Asterisk-Dev] SIP INVITE vs TO URI
William Lloyd
- [Asterisk-Dev] asterisk 'stable'?
Nathan C. Smith
- [Asterisk-Dev] asterisk 'stable'?
Jonathan k. Creasy
- [Asterisk-Dev] config mgmt plan?
Woody Anderson
- [Asterisk-Dev] asterisk 'stable'?
Nathan C. Smith
- [Asterisk-Dev] NoOp not returning anything?
Sherwood McGowan
- [Asterisk-Dev] SIP INVITE vs TO URI
Jerris, Michael MI
- [Asterisk-Dev] H323/SIP<--->PSTN with asterisk. Is it Possible?
Reynold Blur
- [Asterisk-Dev] Newbie needs advice
jose luis campos
- [Asterisk-Dev] zaptel API description?
Bob Smith
- [Asterisk-Dev] Re: [Asterisk-cvs] asterisk logger.c,1.81,1.82
Brian K. West
- [Asterisk-Dev] Can't Add SDP Madness
Matthew Boehm
- [Asterisk-Dev] Re: AgentCallbackLogin, 1.2-beta1 issues
alan
- [Asterisk-Dev] Properly Destroying Variables
Matthew Boehm
- [Asterisk-Dev] chan_h323 1.128 compile fix
Brad Borgald
- [Asterisk-Dev] Bug# 4824: CLI prompt always shown underneath cli
command output
Corey Frang
- [Asterisk-Dev] INVITE URI from contact?
William Lloyd
- [Asterisk-Dev] CVS Servers
Alexander Lopez
- [Asterisk-Dev] Re: [Asterisk-Users] MusicOnHold not working
Gurminder Arora
- [Asterisk-Dev] [Fwd: 802.1p support via SO_Priority& 8021q module
in RTP.c]
Alex Y Fadeyev
- [Asterisk-Dev] CVS Servers
Alexander Lopez
- [Asterisk-Dev] FW: [Asterisk-Users] RxFax/TxFax - Compile Problem
Alexander Lopez
- [Asterisk-Dev] SIPit17 in Stockholm :: Asterisk under pressure!!
Olle E. Johansson
- [Asterisk-Dev] FW: [Asterisk-Users] RxFax/TxFax - Compile Problem
Steve Hanselman
- [Asterisk-Dev] Caller Name: Asterisk reading too fast
J Thomas
- [Asterisk-Dev] Caller Name: Asterisk reading too fast
Alexander Lopez
- [Asterisk-Dev] SIP Jitter Buffer Testing
George Pajari
- [Asterisk-Dev] Call dropped 100% of time when incoming IAX routed
to outgoing CAPI
Christopher Mylonas
- [Asterisk-Dev] Call dropped 100% of time when incoming IAX
routedto outgoing CAPI
Jerris, Michael MI
- Re [Asterisk-Dev] SIP Jitter Buffer Testing
George Pajari
- [Asterisk-Dev] fxotune.c broken by design?
Rob Thomas
- [Asterisk-Dev] hangupcause - tracethrough from PRI
Mark Edwards
- [Asterisk-Dev] App directed pickup (bug #4865)
asterisk at ntplx.net
- [Asterisk-Dev] odbc problems (newbie odbc developer)
Julian Lyndon-Smith
- [Asterisk-Dev] need help
julien bossart
- [Asterisk-Dev] need help
Brian C. Fertig
- [Asterisk-Dev] what do I need to implement odbc
Julian Lyndon-Smith
- [Asterisk-Dev] H.263 format Video
Areski K
- [Asterisk-Dev] Bug with toll free between 2 asterisk servers
(special toll free signaling?)
John Lange
- [Asterisk-Dev] hints not working on CVS HEAD
Paradise Dove
- [Asterisk-Dev] hints not working on CVS HEAD
Rob Thomas
- [Asterisk-Dev] asterisk-oh323: New versions 0.6.7 and 0.7.3
Michael Manousos
- [Asterisk-Dev] High Availability again
Sergio Serrano
- [Asterisk-Dev] Request Netiquite
Alexander Lopez
- [Asterisk-Dev] Request Netiquite
Alexander Lopez
- [Asterisk-Dev] Bug 2905 - patch for macro arguments in M() modifier
Ed Greenberg
- [Asterisk-Dev] Suspected memory leak in 1.0.9
Craig Guy
- [Asterisk-Dev] How to submit code - res_config_pgsql ?
Daniel Swarbrick
- [Asterisk-Dev] #3986 bounty
Roy Sigurd Karlsbakk
- [Asterisk-Dev] A couple of bugs?
Sherwood McGowan
- [Asterisk-Dev] maximum concurrent ZAP channels .... max conf ports
...
Vamsi Pottangi
- [Asterisk-Dev] Questions
Brian K. West
- [Asterisk-Dev] need help
Jonathan k. Creasy
- [Asterisk-Dev] Problem with Parking announcement and RedirectAction
Josip Gracin
- [Asterisk-Dev] Input and testing.
Dan Austin
- [Asterisk-Dev] ast realtime and ast_mutex_lock
Daniel Swarbrick
- [Asterisk-Dev] BOUNTY: 100$ Name before Voicemail Playback
Matthew Gibson
- [Asterisk-Dev] problem receiveing fax whit capi
roberto.ziano
- [Asterisk-Dev] How not to record music on hold?
Josip Gracin
- [Asterisk-Dev] Asterisk Dialogic channel
Matt Florell
- [Asterisk-Dev] Re: #3986 bounty
Andreas Sikkema
- [Asterisk-Dev] Re: Ring requested on channel already in use
alan
- [Asterisk-Dev] Re: Asterisk Dialogic channel
Gilmore, Gerry
- [Asterisk-Dev] Stable change log suggestion?
Rich Adamson
- [Asterisk-Dev] Open source time card application for Asterisk
Chuck Bunn
- [Asterisk-Dev] Open source time card application for Asterisk
Gilmore, Gerry
- [Asterisk-Dev] removing depreciated code?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] Re: Asterisk Time Card
Chuck Bunn
- [Asterisk-Dev] Open source time card application for Asterisk
Gilmore, Gerry
- [Asterisk-Dev] goiax expanded with free us domestic calling
Matthew Simpson
- [Asterisk-Dev] how to fix this bug?E400P port2 and port 4 can not
work
oncemore
- [Asterisk-Dev] VoicemailDetect
steve at daviesfam.org
- [Asterisk-Dev] Open Source time card application for Asterisk
Rob McKrill
- [Asterisk-Dev] removing depreciated code?
Jerris, Michael MI
- [Asterisk-Dev] Asterisk AGI php,
and Asterisk PHP are written as functions instead
of classes and objects.
Chuck Bunn
- [Asterisk-Dev] Problem with 'cvs diff', getting empty diffs
Josip Gracin
- [Asterisk-Dev] MATH() vs expressions?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] delayed sip answer
Mark Willis
- [Asterisk-Dev] Echo cancellation on GPU
steve at daviesfam.org
- [Asterisk-Dev] Echo cancellation on GPU
Richard Scobie
- [Asterisk-Dev] Licensing of Application over Asterisk
Amit Dang
- [Asterisk-Dev] Redirection reason
Kristian Nielsen
- [Asterisk-Dev] Open source time card application for Asterisk
Chuck Bunn
- [Asterisk-Dev] Database backed Asterisk
Serge Sozonoff
- [Asterisk-Dev] Interesting Core Dump
Brian C. Fertig
- [Asterisk-Dev] Open source time card application for Asterisk &
then hotel software
Christopher Mylonas
- [Asterisk-Dev] Bug in the blindxfer and atxfer commands?
hugolivude
- [Asterisk-Dev] TDMoE and tap device
Dome Charoenyost
- [Asterisk-Dev] ast_channel_tech
flavio
- [Asterisk-Dev] TDMoE and tap device
Colin Anderson
- [Asterisk-Dev] TDMoE and tap device
Colin Anderson
- [Asterisk-Dev] GUI's
Serge Sozonoff
- [Asterisk-Dev] Voicemail password change + Ast Realtime
Daniel Swarbrick
- [Asterisk-Dev] How complete is Q.SIG support?
Tony Mountifield
- [Asterisk-Dev] ACD: Limiting time spent with Agent
Begumisa Gerald M
- [Asterisk-Dev] ISDN SETUP on incoming SMS?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] ISDN SETUP on incoming SMS?
Jerris, Michael MI
- [Asterisk-Dev] CSTA interface comments
John Todd
- [Asterisk-Dev] Mac Mini on Debian - Asterisk
Tom Lo
- [Asterisk-Dev] SIP failover using Asterisk and openais
Steven Dake
- [Asterisk-Dev] "register" statement: why can't it reference an
account?
Enzo Michelangeli
- [Asterisk-Dev] Channels capactiy of asterisk
Dario Flores
- [Asterisk-Dev] ISDN SETUP on incoming SMS?
Michael Procter
- [Asterisk-Dev] Problem redirecting to voicemail through a SIPproxy
(Looks like a bug)
Chris St Denis
- [Asterisk-Dev] Hardware HDLC in zaptel
Matthew Fredrickson
- [Asterisk-Dev] [OT] [Very OT] socket connection from perl to visual
basic
Chris Wade
- [Asterisk-Dev] Interesting AEL / Variable issue
Corey Frang
- [Asterisk-Dev] Interesting AEL / Variable issue
Jerris, Michael MI
- [Asterisk-dev] sip calleridnum
Michal Olejnik
- [Asterisk-Dev] Fraud Possibility in CDR
Sherwood McGowan
- [Asterisk-Dev] (no subject)
flavio
- [Asterisk-Dev] ast_channel_tech
flavio
- [Asterisk-Dev] - Call progress event?
Daniel Montejo Biosca (hotmail)
- [Asterisk-Dev] ast_channel_tech
Jerris, Michael MI
- [Asterisk-Dev] Want to add a VM delete event - where in
app_voicemail.c?
Jordan Bean
- [Asterisk-Dev] Want to add a VM delete event - where
inapp_voicemail.c?
Jordan Bean
- [Asterisk-Dev] oh323 inband-info AST_CONTROL_PROGRESS
Igor - BZ
- [Asterisk-Dev] ast_xxx function..?
Matt Hess
- [Asterisk-Dev] ast_xxx function..?
Jerris, Michael MI
- [Asterisk-Dev] The CallerIDPres not working?
Sherwood McGowan
- [Asterisk-Dev] * debug problem
Matt
- [Asterisk-Dev] Want to add a VM delete event -
wherein app_voicemail.c?
Jordan Bean
Last message date:
Fri Sep 30 21:02:18 MST 2005
Archived on: Tue Sep 5 14:27:38 MST 2006
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