August 2005 Archives by thread
Starting: Mon Aug 1 01:00:56 MST 2005
Ending: Wed Aug 31 20:35:53 MST 2005
Messages: 764
- [Asterisk-Dev] Re: ZTdummy/RTC and genrtc/rtc
Tony Mountifield
- [Asterisk-Dev] Inter-server queries
Thomas Andrews
- [Asterisk-Dev] Asterisk High availability
Sergio Serrano
- [Asterisk-Dev] IP-ID in RTP UDP packets
Aj
- [Asterisk-Dev] testing
Russell Bryant
- [Asterisk-Dev] I just recieved some -users mail, is -dev back?
Rob Thomas
- [Asterisk-Dev] IP-ID in RTP UDP packets
Aj
- [Asterisk-Dev] List messages: CVS replay
John Todd
- [Asterisk-Dev] Questions on Asterisk and CallerID
Ganbold Tsagaankhuu
- [Asterisk-Dev] List messages: CVS replay
Rob Thomas
- [Asterisk-Dev] Inter-server queries
Thomas Andrews
- [Asterisk-Dev] Asterisk High availability
Sergio Serrano
- [Asterisk-Dev] Bug 3203 "Get RT without runaway risk" revisited
Kristian Nielsen
- [Asterisk-Dev] Bug report 4783 and RFC 3326: The Reason Header
Field for SIP
Mikael Magnusson
- [Asterisk-Dev] Asterisk support for Solaris 10 X86
Simon Lockhart
- [Asterisk-Dev] Getting ISDN line restart problem with TE110P
Nahid Hossain
- [Asterisk-Dev] FXO PCI Master abort
Mark Burton
- [Asterisk-Dev] RTCP-support
Filip Olsson
- [Asterisk-Dev] meetme list cid num+name patch
Jared Mauch
- [Asterisk-Dev] global vars in shared objects
Wolfgang Pichler
- [Asterisk-Dev] storage binary data through RealTime interface
Timur V. Elzhov
- [Asterisk-Dev] storage binary data through RealTime interface
Brian Jones
- [Asterisk-Dev] storage binary data through RealTime interface
Nathan C. Smith
- [Asterisk-Dev] Route Calls Based on Caller ID
Doug Logan
- [Asterisk-Dev] Parsing caller ID info from serial port CallerID
device to Asterisk?
Carl Andersson
- [Asterisk-Dev] AGI deaf to DTMF after Zap dialout
Jim Gottlieb
- [Asterisk-Dev] ignore this
Russell Bryant
- [Asterisk-Dev] asterisk.org beta site up!
Matt Brooks
- [Asterisk-Dev] Few questions about Asterisk
Ganbold Tsagaankhuu
- [Asterisk-Dev] Route Calls Based on Caller ID [RE-SEND]
Bryce Chidester
- [Asterisk-Dev] Asterisk wants to use GMT and not local time
Frank Tarczynski
- [Asterisk-Dev] PostgreSQL support in Asterisk 1.2?
Daniel Swarbrick
- [Asterisk-Dev] Accessing astdb through perl
Umar Sear
- [Asterisk-Dev] Route Calls Based on Caller ID - Thanks
Doug Logan
- [Asterisk-Dev] fix request...
Dov Bigio
- [Asterisk-Dev] Today's Complie on Fresh System
Matthew Boehm
- [Asterisk-Dev] PostgreSQL support in Asterisk 1.2?
Daniel Swarbrick
- [Asterisk-Dev] PostgreSQL support in Asterisk 1.2?
Daniel Swarbrick
- [Asterisk-Dev] PostgreSQL support in Asterisk 1.2?
Daniel Swarbrick
- [Asterisk-Dev] Asterisk Network Troubleshooting Help Needed - Will
Pay $$$
Adam Robins
- [Asterisk-Dev] asterisk functions return codes
Timur V. Elzhov
- [Asterisk-Dev] setting up a good user database for asterisk?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] The killer app for Asterisk in corporate deployment
peter webier
- [Asterisk-Dev] h323 CALL PROBLEM TO / FROM AVAYA(UCENT) DEFİ
Nİ TY
Ersan ERSOY
- [Asterisk-Dev] h323 CALL PROBLEM TO / FROM AVAYA(UCENT) DEFİ
Nİ TY
Ersan ERSOY
- [Asterisk-Dev] h323 CALL PROBLEM TO / FROM AVAYA(UCENT) DEFİ
Nİ TY
Ersan ERSOY
- [Asterisk-Dev] Welcome to the "Asterisk-Dev" mailing list
Ersan ERSOY
- [Asterisk-Dev] Memory Leak in Stable?
Greg Boehnlein
- [Asterisk-Dev] STUN support
someshwarak
- [Asterisk-Dev] OPAL now supports IAX2
Brian West
- [Asterisk-Dev] Spelling type error
Nathan Alberti
- [Asterisk-Dev] Asterisk in ACD configuration
Mohsen Behnam
- [Asterisk-Dev] PostgreSQL support in Asterisk 1.2?
Rob Thomas
- [Asterisk-Dev] PostgreSQL support in Asterisk 1.2?
Andreas Sikkema
- [Asterisk-Dev] PostgreSQL support in Asterisk 1.2?
Rob Thomas
- [Asterisk-Dev] PostgreSQL support in Asterisk 1.2?
Andreas Sikkema
- [Asterisk-Dev] transfer an incoming ringing channel
Sergio Chersovani
- [Asterisk-Dev] sip call forwarding does not set accountcode
Deti Fliegl
- [Asterisk-Dev] Asterisk-to-IVR Problem
Rollin Weeks
- [Asterisk-Dev] Re: Asterisk-to-IVR Problem
Rollin Weeks
- [Asterisk-Dev] problem using ExtensionState
John Covici
- [Asterisk-Dev] t38 passthrough
Rpingar
- [Asterisk-Dev] Making uniqueID unique in a multiple system
environment.
Chris A. Icide
- [Asterisk-Dev] High-Bandwidth codecs (again) G.722.1
John Todd
- [Asterisk-Dev] wait_for_sysfs issue?
Rich Adamson
- [Asterisk-Dev] Update with new feature for app_mp3.c
Michel Koenen
- [Asterisk-Dev] 0002847: [patch] - Code to allow reverse polarity to
indicate a hang-up on the channel
JR Richardson
- [Asterisk-Dev] A few basic questions..
Geoff Robertson
- [Asterisk-Dev] UMA integration
Freddi Hansen
- [Asterisk-Dev] Newbie question about thread blocking/choppy sound
in meetme
Steve Edwards
- [Asterisk-Dev] AGI-PHP, variable values are not showing
Lambert
- [Asterisk-Dev] stress testing asterisk before putting it into
production
Roy Sigurd Karlsbakk
- [Asterisk-Dev] stress testing asterisk?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] Asterisk stress test?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] sorry - asterisk ml server fsckup?
Roy Sigurd Karlsbakk
- [Asterisk-Dev] tos values ignored in cvs-head current
Vahan Yerkanian
- [Asterisk-Dev] tos values ignored in cvs-head current
Jerris, Michael MI
- [Asterisk-Dev] AGI-PHP, variable values are not showing
Brian C. Fertig
- [Asterisk-Dev] Mac questions
julian howard
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 13, Issue 31
julian howard
- [Asterisk-Dev] Echo canceller testing
Iain Barker
- [Asterisk-Dev] mutagenix_asterisk livecd
Dan Barber
- [Asterisk-Dev] ENUM Lookup REGEXP Size
Andy Paumann
- [Asterisk-Dev] (no subject)
Jeremy Salmon
- [Asterisk-Dev] pseudo realtime and load issue
Steven Critchfield
- [Asterisk-Dev] adding SIPFROMUSER and SIPFROMDOMAIN variables /
pbx_builtin_getvar_helper problem
Günther Starnberger
- [Asterisk-Dev] Outbound proxy: Asterisk not using fromdomain when
REGISTERing
Michael Lunsford
- [Asterisk-Dev] Outbound proxy: Asterisk not using fromdomain when
REGISTERing
Michael Lunsford
- [Asterisk-Dev] [patch] 4854 - does it break deadagi functionality?
Sławomir Małota
- [Asterisk-Dev] Unable to open pseudo channel for timing
Timur V. Elzhov
- [Asterisk-Dev] Where to buy Sangoma cards?
Christian Victor
- [Asterisk-Dev] Intel media processing
John Todd
- [Asterisk-Dev] [patch] 4854 - does it break deadagi functionality?
Jerris, Michael MI
- [Asterisk-Dev] recent ENUM buffer mods
lconroy
- [Asterisk-Dev] Developing as application in Asterisk
Salim
- [Asterisk-Dev] TDMoE
luisprata
- [Asterisk-Dev] Asterisk Addons - H323 Module
Brian C. Fertig
- [Asterisk-Dev] Asterisk Addons - H323 Module
Brian C. Fertig
- [Asterisk-Dev] cvs down? fc3 udev question?
Rich Adamson
- [Asterisk-Dev] dtmfmode=inband broken since ver. 1.6
Joseph
- [Asterisk-Dev] Henning G. Schulzrinne quote on IAX2 from von
magazine
Mike Taht
- [Asterisk-Dev] asterisk with "-p" switch
Joseph
- [Asterisk-Dev] Bug in realtime SIP
Chris A. Icide
- [Asterisk-Dev] Asterisk Addons - H323 Module
Dan Austin
- [Asterisk-Dev] (no subject)
Steve Underwood
- [Asterisk-Dev] can ztdummy be used with a monolithic kernel? (2.6)
Rev. Jeffrey Paul
- [Asterisk-Dev] MS Live Communications Server
bubuk
- [Asterisk-Dev] tos values ignored in cvs-head current
Vahan Yerkanian
- [Asterisk-Dev] Crash while iax2 connection
Stefan Gofferje
- [Asterisk-Dev] Asterisk ISDN Channels Restarting
Richard Doiban
- [Asterisk-Dev] mgcp useragent in asterisk
jiangtao
- [Asterisk-Dev] regexten
Matt Riddell
- [Asterisk-Dev] Hmm... *confused*
Vedran Dakic
- [Asterisk-Dev] asterisk CV HEAD
harry gaillac
- [Asterisk-Dev] res_monitor improvement
Boris Bakchiev
- [Asterisk-Dev] res_monitor improvement
Jerris, Michael MI
- [Asterisk-Dev] res_monitor improvement
Boris Bakchiev
- [Asterisk-Dev] res_monitor improvement
Boris Bakchiev
- [Asterisk-Dev] res_monitor improvement
Jerris, Michael MI
- [Asterisk-Dev] asterisk CV HEAD
harry gaillac
- [Asterisk-Dev] asterisk CV HEAD
harry gaillac
- [Asterisk-Dev] Possible res_config Bounty / Contract
Nathan Alberti
- [Asterisk-Dev] Signal timing can bring down Asterisk
Hans Petter Selasky
- [Asterisk-Dev] X101P register map data please?
Mark Burton
- [Asterisk-Dev] SIP channels not cleared
Chee Foong Chiew
- [Asterisk-Dev] FC3 up2date and Zaptel
Steve Murphy
- [Asterisk-Dev] X101P register map data please?
Mark Burton
- [Asterisk-Dev] Wctdm.c and structure of indirects_reg[]
Jeremy Salmon
- [Asterisk-Dev] couldn't hear anything while playing file
Timur V. Elzhov
- [Asterisk-Dev] New Astmanproxy Mailing List, and New Version 1.11
David C. Troy
- [Asterisk-Dev] FXI PCI Master Abort (give up?)
Mark Burton
- [Asterisk-Dev] asterisk CVS HEAD + presence
harry gaillac
- [Asterisk-Dev] asterisk CVS HEAD + presence + IM
harry gaillac
- [Asterisk-Dev] ASTERISK + IM
harry gaillac
- [Asterisk-Dev] Asterisk IM + Presence
harry gaillac
- [Asterisk-Dev] Right place to plug in a CSTA(partial) implementation
dorn hetzel
- [Asterisk-Dev] Help IP phone project
Prakash N
- [Asterisk-Dev]
Help needed w app_meetme.c, willing to pay $$$$ (fwd)
Steve Edwards
- [Asterisk-Dev] IM patch
harry gaillac
- [Asterisk-Dev] libpri-s makefile
Tzafrir Cohen
- [Asterisk-Dev] REGEX Function
Alessio Focardi
- [Asterisk-Dev] Emprego
Roberto
- [Asterisk-Dev] Problem transferring queue agents
Tony Mountifield
- [Asterisk-Dev] Is this a bug in rtp.c?
Andrew Yager
- [Asterisk-Dev] Warning Unable to allocate socket
Kamran Ahmad
- [Asterisk-Dev] SIP codes/behaviors
Mark Willis
- [Asterisk-Dev] How to at 16 khz slinear codec to asterisk?
john smith
- [Asterisk-Dev] q931 dial errors
Matt
- [Asterisk-Dev] Re: Warning Unable to allocate socket
Kamran Ahmad
- [Asterisk-Dev] chan_bluetooth and Jabra BT250v headset
Whoopie
- [Asterisk-Dev] SIP changes
Olle E. Johansson
- [Asterisk-Dev] SIP freeze in HEAD using SIP Realtime
Chris A. Icide
- [Asterisk-Dev] Severe ISDN signal distortion in CVS-HEAD with
octoBRI
Elwin Andriol
- [Asterisk-Dev] Embedded HW: asterisk with USB ISDN TA on
NSLU2/Debian (fwd)
Ralf Ackermann
- [Asterisk-Dev] SIP codes/behaviors
www.IPKall.com
- [Asterisk-Dev] SIP codes/behaviors
www.IPKall.com
- [Asterisk-Dev] CVS-HEAD 08/13/2005 SIP DEADLOCKING
Sherwood McGowan
- [Asterisk-Dev] AEL parsing oddity with Curl
Beau Hargis
- [Asterisk-Dev] mec2 fix #6937 ?
Florian Overkamp
- [Asterisk-Dev] mec2 fix #6937 ?
Jerris, Michael MI
- [Asterisk-Dev] Cisco 7970 SCCP Configs.
Paul Duffy
- [Asterisk-Dev] Experienced Sysadmin/Programmer having major trouble
with British Telecom Caller ID & Distinctive Ring
Cats Muvva
- [Asterisk-Dev] installing pystre
chinmaya
- [Asterisk-Dev] Asterisk Manager API
Bonelli Sergio
- Antw: [Asterisk-Dev] Asterisk Manager API
Hoai-Anh Ngo-Vi
- [Asterisk-Dev] C/C++ socket to communicate to Asterisk ?
彭建翔
- FW: [Asterisk-Dev] Experienced Sysadmin/Programmer having
majortrouble with British Telecom Caller ID & Distinctive Ring
Cats Muvva
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 13, Issue 75
oliver at e-dxd.com
- [Asterisk-Dev] Accessing Call Id and IP address of the OUI from the
ast_channel structure
Dmitry Isakbayev
- [Asterisk-Dev] Job Opening - Release Engineer
Paul Mahler
- [Asterisk-Dev] ANI2 AKA Info Digits not supported?
Steve Edwards
- [Asterisk-Dev] app_rpt and iaxcomm ptt
Andreas Bayer
- [Asterisk-Dev] CVS HEAD behavior change: SIP Realtime caching
Kevin P. Fleming
- [Asterisk-Dev] CVS HEAD behavior change: SIP Realtime caching
Jerris, Michael MI
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 13, Issue 76
oliver at e-dxd.com
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Matt
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Olle E. Johansson
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Slav Klenov
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Matt Hess
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Matt Hess
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Slav Klenov
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Matt Hess
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Zoa
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk - now aimed for
1.3 dev
Olle E. Johansson
- [Asterisk-Dev] Not for 1.2,
was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev
Tilghman Lesher
- [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in
Asterisk - now aimed for 1.3 dev
Beau Hargis
- [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk
- now aimed for 1.3 dev
Kevin P. Fleming
- [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk
- now aimed for 1.3 dev
Eric Wieling aka ManxPower
- [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk
- now aimed for 1.3 dev
Arnaud
- [Asterisk-Dev] Not for 1.2,
was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev
Matt
- [Asterisk-Dev] Not for 1.2,
was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev
Matt
- [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk
- now aimed for 1.3 dev
Steve Kann
- [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk
- now aimed for 1.3 dev
Zoa
- [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev
Andrew Kohlsmith
- [Asterisk-Dev] Not for 1.2,
was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev
Matt
- [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev
Tilghman Lesher
- [Asterisk-Dev] SIP RTP JitterBuffer in Asterisk
Matt Hess
- [Asterisk-Dev] Minor makefile typo..
Rob Thomas
- [Asterisk-Dev] Minor makefile typo..
Rob Thomas
- [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT ***
Olle E. Johansson
- [Asterisk-Dev] app_rpt and iaxcomm ptt
hwstar at rodgers.sdcoxmail.com
- [Asterisk-Dev] Test
Malcolm Davenport
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 13, Issue 77
oliver at e-dxd.com
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 13, Issue 78
oliver at e-dxd.com
- [Asterisk-Dev] app_rpt and iaxcomm ptt
hwstar at rodgers.sdcoxmail.com
- [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP
Olle E. Johansson
- [Asterisk-Dev] Asterisk test-o-matic
Kristian Kielhofner
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 13, Issue 79
Huddleston, Robert
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 13, Issue 79
oliver at e-dxd.com
- [Asterisk-Dev] Asterisk test-o-matic
Jerris, Michael MI
- [Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 13, Issue 79
Jerris, Michael MI
- [Asterisk-Dev] Extension "Unavailable" Status
Darren Younger
- [Asterisk-Dev] fax codec problem
Daniel Grad
- [Asterisk-Dev] H263 file format
Christophe Guerin
- [Asterisk-Dev] Not for 1.2,
was SIP RTP JitterBuffer in Asterisk- now aimed for 1.3 dev
Jerris, Michael MI
- [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP
Jerris, Michael MI
- [Asterisk-Dev] Not for 1.2,
was SIP RTP JitterBuffer in Asterisk- now aimed for 1.3 dev
Jerris, Michael MI
- [Asterisk-Dev] Not for 1.2,
was SIP RTP JitterBuffer in Asterisk-now aimed for 1.3 dev
Jerris, Michael MI
- [Asterisk-Dev] SIP Benchmarking / Stress Testing
Sherwood McGowan
- [Asterisk-Dev] app_rpt and iaxcomm ptt
hwstar at rodgers.sdcoxmail.com
- [Asterisk-Dev] Asterisk 1.2.0-beta1 Released
Kevin P. Fleming
- [Asterisk-Dev] Re: [Asterisk-cvs] asterisk/codecs codec_speex.c,
1.17, 1.18 slin_speex_ex.h, 1.1, 1.2
SteveK
- [Asterisk-Dev] problems compiling 1.2 beta
Julian Lyndon-Smith
- [Asterisk-Dev] Tarball of Asterisk's CVSROOT available?
Stefan Reuter
- [Asterisk-Dev] Tarball of Asterisk's CVSROOT available?
Jerris, Michael MI
- [Asterisk-Dev] Tarball of Asterisk's CVSROOT available?
Jerris, Michael MI
- [Asterisk-Dev] problems compiling 1.2 beta
Damon Estep
- [Asterisk-Dev] [BUG?] chan_sip,
RFC 3261 and multiple UACs registering for one account
Hartwig Deneke
- [Asterisk-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups
James Jones
- [Asterisk-Dev] versions of zaptel, libpri for asterisk 1.2
Tzafrir Cohen
- [Asterisk-Dev] Compile problems with 1.2 beta1
Julian Lyndon-Smith
- [Asterisk-Dev] Asterisk + AstLinux testing images now available
Kristian Kielhofner
- [Asterisk-Dev] IM/presence support in asterisk
harry gaillac
- [Asterisk-Dev] DIALSTATUS on Originate
saket setu
- [Asterisk-Dev] DIALSTATUS for Originate
saket setu
- [Asterisk-Dev] How to configure Cisco AS5800 - Asterisk ??
kaws elchamal
- [Asterisk-Dev] Snom 360
Dovid B - Asterisk Dev.
- [Asterisk-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups
ewr at erols.com
- [Asterisk-Dev] where/when is app_flash loaded
Tim Allen
- [Asterisk-Dev] Segfault
Tamas Jalsovszky
- [Asterisk-Dev] How to measure delay in meetme?
Steve Edwards
- [Asterisk-Dev] IAX2 encryption
Derek Smithies
- [Asterisk-Dev] Images in IAX2
Derek Smithies
- [Asterisk-Dev] How to measure delay in meetme?
Jerris, Michael MI
- [Asterisk-Dev] SIP channels not cleared
Jerris, Michael MI
- [Asterisk-Dev] SIP channels not cleared
Jerris, Michael MI
- [Asterisk-Dev] Asterisk 1.2.0-beta1 tarball re-released
Kevin P. Fleming
- [Asterisk-Dev] RFC2833, Asterisk, and Cisco
Alistair Cunningham
- [Asterisk-Dev] hi
manish kumar
- [Asterisk-Dev] Video VoiceMail
Christophe Guerin
- [Asterisk-Dev] voicemessages table
harry gaillac
- [Asterisk-Dev] Moving to New Zealand
James Jones
- [Asterisk-Dev] confifiguration of Asterisk with Cisco hardware?
kaws elchamal
- [Asterisk-Dev] Is it possiple to run Asterisk with Cisco AS5800 ??
Brian C. Fertig
- [Asterisk-Dev] Two underscores
José Pablo Ezequiel Fernández
- [Asterisk-Dev] IAX2 exten@context dialing removed?
Chris A. Icide
- [Asterisk-Dev] SIP presence notification updated (#3644)
Olle E. Johansson
- [Asterisk-Dev] Makefile problem
Kristian Kielhofner
- [Asterisk-Dev] [PATCH] gcc 4.0.2 warning
Alfred E. Heggestad
- [Asterisk-Dev] voicemessages table
Jerris, Michael MI
- [Asterisk-Dev] SIP presence notification updated (#3644)
Adam Gundy
- [Asterisk-Dev] Question related to zaptel driver development
Henry Margies
- [Asterisk-Dev] Asterisk Bounty VoiceMail-n-Email Synchronization =
$1125
support at sjobeck.com
- [Asterisk-Dev] Jackd and Asterisk
Mike Taht
- [Asterisk-Dev] Jackd and Asterisk
Jerris, Michael MI
- [Asterisk-Dev] Manipulating CALLERIDNUM
Chad Brown
- [Asterisk-Dev] Asterisk Bounty VoiceMail-n-Email Synchronization
= $1125
support at sjobeck.com
- [Asterisk-Dev] RPID Support
Brian West
- [Asterisk-Dev] PCI Master Aborts effect multiple subsystems?
Mark Burton
- [Asterisk-Dev] Originate Call and Unique ID
Joerg Lauer
- [Asterisk-Dev] feature needed
mutta at tiscali.it
- [Asterisk-Dev] www.govarion.com E1 Card
Dome Charoenyost
- [Asterisk-Dev] chan_bluetooth not compiling on current 1.0 cvs
Leandro
- [Asterisk-Dev] Muting DTMF in app_meetme.c
Steve Edwards
- [Asterisk-Dev] locked sip channels
Dov Bigio
- [Asterisk-Dev] bug in chan_iax2.c to enable qualifysmoothing
Brad Borgald
Last message date:
Wed Aug 31 20:35:53 MST 2005
Archived on: Tue Sep 5 14:27:35 MST 2006
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