September 2006 Archives by thread
Starting: Fri Sep 1 00:13:56 MST 2006
Ending: Sat Sep 30 21:54:29 MST 2006
Messages: 845
- [asterisk-dev] #7848 was closed - why ?
Julian Lyndon-Smith
- [asterisk-dev] HANGUP_LOCALUSERS undeclared (asterisk-addons)
Jason Parker
- [asterisk-dev] HANGUP_LOCALUSERS undeclared (asterisk-addons)
Kaloyan Kovachev
- [asterisk-dev] Measuring PDD
Jean-Michel Hiver
- [asterisk-dev] Outboundproxy Settings
Aaron Clauson
- [asterisk-dev] recommended version for testing voicemail+imap?
Arnd Vehling
- [asterisk-dev] Anyone got "SIP/RTP" Working reliably using svn
trunk?
Joshua Colp
- [asterisk-dev] Question from rtp.c
Hunt, Bill
- [asterisk-dev] Re: file: trunk r41694 - /trunk/main/file.c
Tony Mountifield
- [asterisk-dev] RTP Packetization (bugid 5162)
Dan Austin
- [asterisk-dev] VLDTMF eats DTMF,
news at 11 (or look at bugid 7868 if you love DTMF)
Dan Austin
- [asterisk-dev] VLDTMF eats DTMF,
news at 11 (or look at bugid7868 if you love DTMF)
Dan Austin
- [asterisk-dev] Building an embedded Asterisk PBX
David Rowe
- [asterisk-dev] VLDTMF eats DTMF,
news at 11 (or look at bugid7868if you love DTMF)
Dan Austin
- [asterisk-dev] What's the difference ?
Julian Lyndon-Smith
- [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaper than
existing one.
Kannaiyan Natesan
- [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaper
than existing one.
Kannaiyan Natesan
- [asterisk-dev] Re: [asterisk-biz] G729 Replacement Codec - FREE or
may ne cheaper than existing one.
Kannaiyan Natesan
- [asterisk-dev] Re: [asterisk-biz] G729 Replacement Codec - FREE or
may ne cheaper than existing one.
Kannaiyan Natesan
- [asterisk-dev] Outboundproxy Settings
Aaron Clauson
- [asterisk-dev] digium closing the source?
Roy Sigurd Karlsbakk
- [asterisk-dev] crash when entering WaitExten with moh class
specified
Pavel Jezek
- [asterisk-dev] OT: RSS Feed for bugs page (Mantis)
Matt Riddell (IT)
- [asterisk-dev] Need some help
sudhir kumar
- [asterisk-dev] wideband support?
Roy Sigurd Karlsbakk
- [asterisk-dev] Various things
Jean-Michel Hiver
- [asterisk-dev] Zaptel-1.2.8 compile problem
Vidura Senadeera
- [asterisk-dev] Data Modem calls passthrough
Anton
- [asterisk-dev] Re: [asterisk-users] Digum g729 and g723
Kannaiyan Natesan
- [asterisk-dev] Re: [asterisk-biz] Re: G729 Replacement Codec - FREE
or may ne cheaper than existing one.
Kannaiyan Natesan
- [asterisk-dev] UDP packets are transmiting in small package or big
package?
Ma Zhiyong
- [asterisk-dev] Re: [asterisk-users] Digum g729 and g723
Jeremy McNamara
- [asterisk-dev] asterisk-addons/ooh323c
harrygaillac-sip at yahoo.fr
- [asterisk-dev] Realtime crash
Peter Spikings
- [asterisk-dev] Call priority and Dial()
Andrew Kohlsmith
- [asterisk-dev] iax2 connection drops after some minutes
Pavel Jezek
- [asterisk-dev] Query of IAX Registration
sudhir kumar
- [asterisk-dev] files version..
Dov Bigio
- [asterisk-dev] Packetization update (bugid 5162)
Dan Austin
- [asterisk-dev] Wishlist
Jean-Michel Hiver
- [asterisk-dev] Packetization update (bugid 5162)
Dan Austin
- [asterisk-dev] Packetization update (bugid 5162)
Dan Austin
- [asterisk-dev] UUI in calls
Daniel Montejo
- [asterisk-dev] Crash when using custom application when Slav's
Abstract Jitterbuffer is enabled
Mikael Bjerkeland
- [asterisk-dev] Digium G.729 codec binaries updated
Kevin P. Fleming
- [asterisk-dev] Digium's response to posting of G.729 and G.723
source code
Kevin P. Fleming
- [asterisk-dev] Digium's response to posting of G.729 and
G.723source code
Dan Austin
- [asterisk-dev] Digium's response to posting of G.729
andG.723source code
Dan Austin
- [asterisk-dev] Digium G.729 codec binaries updated (or DEA is a
bonehead)
Dan Austin
- RV: [asterisk-dev] UUI in calls
Daniel Montejo
- [asterisk-dev] Crash when using custom application when Slav's
Abstract Jitterbuffer is enabled
Mikael Bjerkeland
- [asterisk-dev] Change to PRI called number presentation length
between 1.2.10 and 1.2.11?
Steve Hanselman
- [asterisk-dev] codec amr
gennaro
- [asterisk-dev] Transfer
Kai Ober
- [asterisk-dev] SIP Session establish thru IAX servers
sudhir kumar
- [asterisk-dev] Digium G.729 codec binaries updated (or DEA is
abonehead)
Dan Austin
- [asterisk-dev] asterisk trunk registration problem URGENT
harrygaillac-sip at yahoo.fr
- [asterisk-dev] registration failed
harrygaillac-sip at yahoo.fr
- [asterisk-dev] registration failed
harrygaillac-sip at yahoo.fr
- [asterisk-dev] Open source G.729 and G.723.1 release for 1.2 and 1.4
Daniel Pocock
- [asterisk-dev] jabber static realtime
Theo Belder
- [asterisk-dev] SVN-trunk-r41990 IMAP storage
harrygaillac-sip at yahoo.fr
- [asterisk-dev] Iaxclient-devel
Giovanni Miano
- [asterisk-dev] app queue crash
harrygaillac-sip at yahoo.fr
- [asterisk-dev] Disabling Call Parking
Wildheart
- [asterisk-dev] Asterisk Makefile
Maxi Belino
- [asterisk-dev] Developer's meeting @von
Olle E Johansson
- [asterisk-dev] E&M Flash Hook Support
Rob Wilson
- [asterisk-dev] G729 codec for trunk updated again... now it works
Kevin P. Fleming
- [asterisk-dev] G729 codec for trunk updated again... now it works
Dan Austin
- [asterisk-dev] G729 codec for trunk updated again... now it works
Dan Austin
- [asterisk-dev] Re: app_queue member naming and AMI events
Kevin P. Fleming
- [asterisk-dev] app_queue member naming and AMI events
Kevin P. Fleming
- [asterisk-dev] G729 codec for trunk updated again... now it works
Dan Austin
- [asterisk-dev] Asterisk 1.2.12 and Zaptel 1.2.9 released!
The Asterisk Development Team
- [asterisk-dev] G729 codec for trunk updated again... now it works
Dan Austin
- [asterisk-dev] Re: kpfleming: tag 1.2.12 r42471 - /tags/1.2.12/
Tony Mountifield
- [asterisk-dev] dialplan apps
Robert Bielik
- [asterisk-dev] internal_timing on 1.2? clock drift?
Daniel Pocock
- [asterisk-dev] Re: file: trunk r42569 - /trunk/main/rtp.c
Tony Mountifield
- [asterisk-dev] chan_skype?
Matthew Rubenstein
- [asterisk-dev] chan_skype?
Matthew Rubenstein
- [asterisk-dev] bug #7911
Raphael Jacquot
- [asterisk-dev] Try chan_skype
Paulo Mannheimer
- [asterisk-dev] rewriting standard stuff?
Roy Sigurd Karlsbakk
- [asterisk-dev] PRI: help to understand why Asterisk drops calls
(Got restart ack)
Giorgio Incantalupo
- [asterisk-dev] Temporary Voicemail prompt suggestion
John Lange
- [asterisk-dev] Re: asterisk-dev Digest, Vol 26, Issue 40
Matthew Rubenstein
- [asterisk-dev] Solution to the Codec Negotiation Problem
Chan Kwang Mien
- [asterisk-dev] Warning and/or question (Zaptel MMX)
Steve Davies
- [asterisk-dev] asterisk-1.2.12 iLBC decode cause core dump
Ma Zhiyong
- [asterisk-dev] ast_channel_masquerade in a queue consultation
Daniel Montejo
- [asterisk-dev] Forwarding sip requests from none local domains
harrygaillac-sip at yahoo.fr
- [asterisk-dev] mute calls through a sip proxy
Antonio Ceccatelli
- [asterisk-dev] Voicemail + IMAP Problems again (svn trunk)
Arnd Vehling
- [asterisk-dev] test
harrygaillac-sip at yahoo.fr
- [asterisk-dev] Re: Power suply for TDM2400p for FXs
sword power
- [asterisk-dev] zap channel allocation
gaddam purna
- [asterisk-dev] Asterisk SIP authentication
Andrea Spadaccini
- [asterisk-dev] Outgoing callerid in AMI
Mir
- [asterisk-dev] Calls on hold
Mir
- [asterisk-dev] Adding own info in AMI
Mir
- [asterisk-dev] Hardcoded res_features.c attended transfer timeout
Vahan Yerkanian
- [asterisk-dev] Developer's meeting @von
Olle E Johansson
- [asterisk-dev] zap call works
gaddam purna
- [asterisk-dev] mute calls through a sip proxy
Antonio Ceccatelli
- [asterisk-dev] Asterisk&Aculab compatibility issue with Brooktrout.
Brian Quinn
- [asterisk-dev] Setting up imap based voicemail / invalid remote
specification
Arnd Vehling
- [asterisk-dev] Asterisk Appliance?
John Lange
- [asterisk-dev] chan_h323 warnings
Jean-Michel Hiver
- [asterisk-dev] Possible app_transcribe?
Slim Shady
- [asterisk-dev] Backing up Asterisk configuration during testing
Frank Church
- [asterisk-dev] Copyright issues with libcurl and OpenSSL
James Jones
- [asterisk-dev] Re: [asterisk-users] Copyright issues with libcurl
and OpenSSL
James Jones
- [asterisk-dev] chan_jingle
Theo Belder
- [asterisk-dev] Transforming a 2 party MeetMe conference into a
simple call
ISAC Flavius
- [asterisk-dev] Transfer capabilities inherited after transfer
Kai Ober
- [asterisk-dev] Note about mpg123
Brian Candler
- [asterisk-dev] Monitoring of Call Hold
Mir
- [asterisk-dev] [asterisk-users] DIAL and automatic/manual co line
acces
Kai Ober
- [asterisk-dev] Compilation problem with menuselect.makeopts
Frank Church
- [asterisk-dev] Asterisk and "305 Use Proxy"
Wolfgang Hottgenroth
- [asterisk-dev] chan_jingle failure
Theo Belder
- [asterisk-dev] Note about mpg123
Raphaël Jacquot
- [asterisk-dev] WAIT FOR DIGIT not working
Joel Lansden
- [asterisk-dev] Is DTMF monitory part of trunk?
Frank Church
- [asterisk-dev] RE: [asterisk-biz] looking for Wifi Ip Phones
Matthew Rubenstein
- [asterisk-dev] Capture Text in Dialplan
Roland Ndaka Fru
- [asterisk-dev] Problems compiling team/jcollie/bug6082 branch -
revision 4294
Frank Church
- [asterisk-dev] 491 request pending
harrygaillac-sip at yahoo.fr
- [asterisk-users] [asterisk-dev] 491 request pending
Patrick
- [asterisk-dev] IMAP email storage proposal and question
Dax Kelson
- [asterisk-dev] Asterisk Appliance?
Robin Getz
- [asterisk-dev] manager api no cdr event
Raymond Chen
- [asterisk-dev] Asterisk and Openais
raman kumar
- [asterisk-dev] about asterisk, aculab & brooktrout
Brian Quinn
- [asterisk-dev] Asterisk&Aculab compatibility issue with Brooktrout.
Brian Quinn
- [asterisk-dev] func_math.c
Kai Ober
- [asterisk-dev] NAT traversal always disabled when Asterisk INVITEs
peer
Yoann Aubineau
- [asterisk-dev] RTCP port always bound to 0.0.0.0?
Tony Mountifield
- [asterisk-dev] open letter
harrygaillac-sip at yahoo.fr
- [asterisk-dev] Change of Codec For Blind Transfer
Chan Kwang Mien
- [asterisk-dev] Re: Note about mpg123
Matthew Rubenstein
- [asterisk-dev] Asterisk 1.2.12.1 and Zaptel 1.2.9.1 Released
The Asterisk Development Team
- [asterisk-dev] NFAS Primary-Secondary D Issue
Butler, Larry
- [asterisk-dev] Zaptel issue in FC6
Aryanto Rachmad
- [asterisk-dev] Need help for 4E1 config
Dome C.
- [asterisk-dev] IPv6
Michiel van Baak
- [asterisk-dev] error in build on latest trunk
Julian Lyndon-Smith
- [asterisk-dev] Re: POSIX timers
Tony Mountifield
- [asterisk-dev] last stable Asterisk version
techno e
- [asterisk-dev] ooh323 & radius authentification
rio at traveltele.com
- [asterisk-dev] using Hardware TDM switch
M Desai
- [asterisk-dev] Bug 7966 RPID - works in SVN Trunk - What changed?
Ed Greenberg
- [asterisk-dev] chan_skinny - call to turned off phone causes
deadlock (r43208)
Pavel Jezek
- [asterisk-dev] Clarification on packetization feature
Dan Austin
- [asterisk-dev] SIP satck of asterisk
sudhir kumar
- [asterisk-dev] Underwood's spandsp
isamar at isamarmaia.org
- [asterisk-dev] RHEL "2.6.9" kernel detection
Tzafrir Cohen
- [asterisk-dev] Re: (IPv6) (punit)
rajesh singh
- [asterisk-dev] streaming Asterisk audio online
Christian Croft
- [asterisk-dev] last stable Asterisk version
techno e
- [asterisk-dev] Are there any voice prompt changes in Asterisk 1.4?
Stuart
- [asterisk-dev] Trunk 43282 does not build
Dan Austin
- [asterisk-dev] Trunk 43282 does not build
Dan Austin
- [asterisk-dev] iax2 trunking - little packets out of trunk?
Ma Zhiyong
- [asterisk-dev] Call forward with CFU?
Roy Sigurd Karlsbakk
- [asterisk-dev] Asterisk logging system.
Dmytro Mishchenko
- [asterisk-dev] Are there any voice prompt changes in Asterisk 1.4?
Stuart
- [asterisk-dev] Asterisk Voicemail with Sonus?
Chris Carey
- [asterisk-dev] Codec Framing/Packetization limits
Dan Austin
- [asterisk-dev] Codec Framing/Packetization limits
Dan Austin
- [asterisk-dev] CallerID retain on internal transfer
Olivier Krief
- [asterisk-dev] RTT in rtcp debug
André Abrantes
- [asterisk-dev] The queue call distribute
li yuqian
- [asterisk-dev] automated response
tomc
- [asterisk-dev] SIP to SIP unhold and no voice
Aragon Gouveia
- [asterisk-dev] Curl not avalable
Elpidio Ramos
- [asterisk-dev] Cannot build latest trunk
Julian Lyndon-Smith
- [asterisk-dev] To bweschke regarding app FollowMe
Denis Galvão
- [asterisk-dev] Asterisk, Asterisk-Addons,
Zaptel and Libpri 1.4 betas released!
Asterisk Development Team
- [asterisk-dev] SIP Interoperability in Asterisk?
John Lange
- [asterisk-dev] Management of branches, bug fixes, etc.
Kevin P. Fleming
- [asterisk-dev] A rewrite of Asterisk::AGI
Tzafrir Cohen
- [asterisk-dev] Re: [asterisk-users] 488 Not acceptable here sent by
Asterisk - SIPdebug follows
Dinesh Nair
- [asterisk-dev] Hello
Alexandr Olekhnovich
- [asterisk-dev] Asterisk Presence
Alexandr Olekhnovich
- [asterisk-dev] Advice of charge
Tomislav Parčina
- [asterisk-dev] (no subject)
Alexandr Olekhnovich
- [asterisk-dev] core dump asterisk 1.2.12.1 - ChanSpy ???
Dov Bigio
- [asterisk-dev] (no subject)
Alexandr Olekhnovich
- [asterisk-dev] AstriConVideo ! Paris Nov 20-22! Book your calendar!
Olle E Johansson
- [asterisk-dev] 1.4.0-beta2 and g729
Dan Austin
- [asterisk-dev] core dump asterisk 1.2.12.1 - ChanSpy ???
Sebastian Auriol
- [asterisk-dev] Digium G.729 codec binaries updated for Asterisk 1.4
beta
Jason Parker
- [asterisk-dev] Solution allowing Asterisk send and receiving serial
data from/to pbx
Paulo Garcia
- [asterisk-dev] Developer Summit at Astricon USA 2006
Kevin P. Fleming
- [asterisk-dev] 1.4.0-beta2 and g729
Dan Austin
- [asterisk-dev] OT But So Ungodly Important
Rushowr
- [asterisk-dev] Some 1.4.0-beta2 and Solaris 10 issues...
Lee Essen
- [asterisk-dev] Problems compiling chan_h323 of 1.4beta2 version
Vlasis Hatzistavrou Mailing Lists Account
- [asterisk-dev] Asterisk 1.4 and spandsp3/txfax/rxfax
asterisk at ntplx.net
- [asterisk-dev] Question about hup_handler & deadlocks
Jay Hoover
- [asterisk-dev] Voicemail imap storage.
Sergey Okhapkin
- [asterisk-dev] fixes to zaptel makefile
Tzafrir Cohen
- [asterisk-dev] Dual core
Tomislav Parčina
- [asterisk-dev] what am i doing wrong/ DIAL (local)/ always busy
Kai Ober
- [asterisk-dev] LIMIT_TIMEOUT_FILE leaves called party connected
Alistair Cunningham
- [asterisk-dev] (no subject)
Alexandr Olekhnovich
- [asterisk-dev] Codec not changed when making an Attended xfer
(REFER)
Chan Kwang Mien
- [asterisk-dev] Problem with 1.4b2 and native sounds
Dan Austin
- [asterisk-dev] ZAPTEL in trunk - bug/hardcoded d-chan in zaptel?
E1-T1 issue.
Anton
- [asterisk-dev] CDR-csv records
Bro
- [asterisk-dev] Fastagi question
Yelson Vivas
- [asterisk-dev] Problem with 1.4b2 and native sounds
Dan Austin
- [asterisk-dev] Re: Advice of charge
Tomislav Parčina
- [asterisk-dev] Re: Advice of charge
Tomislav Parčina
- [asterisk-dev] Re: Dual core
Tomislav Parčina
- [asterisk-dev] AMI Documentation
rizwan hisham
- [asterisk-dev] ast_frfree
Steve Underwood
- [asterisk-dev] Re: CDR-csv records
Bro
- [asterisk-dev] Problem with 1.4b2 and native sounds
Dan Austin
- [asterisk-dev] New codec support in chan_skinny
Dan Austin
- [asterisk-dev] Rate limiting traffic to address potential DoS
issues?
Kevin P. Fleming
- [asterisk-dev] New codec support in chan_skinny
Dan Austin
- [asterisk-dev] Policy change for those with commit access
Kevin P. Fleming
- [asterisk-dev] displaysystemname in manager.conf
Stefan Reuter
- [asterisk-dev] Core Dump from res_musiconhold.c Crash
Steve Totaro
- [asterisk-dev] Re: Rate limiting and firewall failures on DoS
J. Oquendo
- [asterisk-dev] Re: [asterisk-commits] qwell: trunk r43666 - in
/trunk: configs/logger.conf.sample main/logger.c
James Ainslie
- [asterisk-dev] Extension exact match or alphabetical still?
Anton
- [asterisk-dev] AGI SayNumber gender option
asterisk-dev at agk.nnov.ru
- [asterisk-dev] How to send SIP replies to another Asterisk?
Hauke Joachim Zuehl
- [asterisk-dev] chan_sccp rtp patch for 1.4?
Andreas Anderson
- [asterisk-dev] Passing DTMF through MeetMe
Tony Mountifield
- [asterisk-dev] Re: Advice of charge
Tomislav Parčina
- [asterisk-dev] G.729 codec problem
Rosa De Santis
- [asterisk-dev] Bug 5811
DANIEL, AARON MATTHEW
- [asterisk-dev] unable to call AT&T audio conference bridge
asterisk-user
- [asterisk-dev] List of Zaptel/Libpri 1.2->1.4 changes
Matthew Fredrickson
- [asterisk-dev] Crash with latest 1.2 svn r43186
Martin Vít
- [asterisk-dev] Fwd: Configuring Asterisk 1.4-beta2 to work with
jingle
Raffaele Porzio
- [asterisk-dev] shutting down zaptel spans
Tzafrir Cohen
- [asterisk-dev] Fwd: How can I unistall Asterisk?
Raffaele Porzio
- [asterisk-dev] how to compile and install chan_jingle.so
Raffaele Porzio
- [asterisk-dev] Checking h323.h presence... no???
Vlasis Hatzistavrou
- [asterisk-dev] Asterisk and Nextone
anthony guan
- [asterisk-dev] MOH not working on custom channel driver
Earle Clubb
- [asterisk-dev] Zaptel 1.4beta won't compile on SLES 10
John Lange
- [asterisk-dev] Digium G.729 codec binaries updated for Asterisk 1.4
on Solaris
Jason Parker
- [asterisk-dev] Re: kpfleming: branch 1.2 r43895 -
/branches/1.2/cli.c
Tony Mountifield
- [asterisk-dev] Re: file: branch 1.4 r43913 -
/branches/1.4/main/cli.c
Tony Mountifield
- [asterisk-dev] chan_skinny crashes asterisk (1.4)
Pavel Jezek
- [asterisk-dev] Audio processing
Harish Kasiviswanathan
- [asterisk-dev] several compile problems after SVN 44056 changes
Luigi Rizzo
- [asterisk-dev] developers help
Michael Rozov
- [asterisk-dev] Re: [asterisk-commits] pcadach: branch 1.4 r44090 -
/branches/1.4/main/rtp.c
Kevin P. Fleming
Last message date:
Sat Sep 30 21:54:29 MST 2006
Archived on: Sat Sep 30 21:54:31 MST 2006
This archive was generated by
Pipermail 0.09 (Mailman edition).