December 2009 Archives by thread
Starting: Tue Dec 1 00:32:05 CST 2009
Ending: Thu Dec 31 23:28:34 CST 2009
Messages: 366
- [asterisk-dev] Insulting source code comments in main/channel.c
Kirill 'Big K' Katsnelson
- [asterisk-dev] New committer - odicha
Odicha
- [asterisk-dev] SIP URI checking in chan_sip.c
Nick Lewis
- [asterisk-dev] SIP URI checking in chan_sip.c
Nick Lewis
- [asterisk-dev] Can asterisk PRI/BRI support redirect calls
Alec Davis
- [asterisk-dev] [Code Review] fix for double close of file descriptors raised in issue 16305
thedavidfactor at gmail.com
- [asterisk-dev] team creation (novice question)
Odicha
- [asterisk-dev] Working in useful examples... and freenum/e.164dialing in extensions.conf.example
Nick Lewis
- [asterisk-dev] [asterisk-commits] tilghman: trunk r232164 - in /trunk: ./ include/asterisk/ main/
Kevin P. Fleming
- [asterisk-dev] Asterisk-Addons 1.4.10, 1.6.0.4, and 1.6.1.2 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.6.2.0-rc7 released, and Asterisk-Addons 1.6.2.0-rc2 re-released
Asterisk Development Team
- [asterisk-dev] Meetme AMI actions
Nick Lewis
- [asterisk-dev] chan_sip - sip_write error
Chandrakant Solanki
- [asterisk-dev] [Code Review] Fix minor bugs preventing EIVR socket client from working
thedavidfactor at gmail.com
- [asterisk-dev] State of FAX (primarily T.38) in Asterisk trunk (planning for 1.8 release)
Kevin P. Fleming
- [asterisk-dev] * 1.6.1.11 res_config_sqlite No application SQLITE()
Andrea Cristofanini
- [asterisk-dev] diruggles: branch 1.6.0 r232811 - /branches/1.6.0/apps/app_externalivr.c
Tony Mountifield
- [asterisk-dev] New feature in app_queue: Add support for ring indication when calling member
Håkon Nessjøen
- [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf
Ryan Finnie
- [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf
John Todd
- [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf
Michiel van Baak
- [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf
Ryan Finnie
- [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf
John Todd
- [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf
Olle E. Johansson
- [asterisk-dev] app_meetme call for testing: Roll call, eject all, mute all, record in-conf
Russell Bryant
- [asterisk-dev] tilghman: trunk r232660 - in /trunk: include/asterisk/ res/
Russell Bryant
- [asterisk-dev] bad file prompt play
Chandrakant Solanki
- [asterisk-dev] Asterisk and multiple SIP registrations to the same host
asterisk-dev at eurower.com
- [asterisk-dev] Makefile.rules MAKE_DEPS usage
Luigi Rizzo
- [asterisk-dev] Can Asterik act as a SIP Proxy
Gayathri G
- [asterisk-dev] Asterisk and multiple SIP registrations to the samehost
Nick Lewis
- [asterisk-dev] g722 question
Saúl Ibarra
- [asterisk-dev] [svn-commits] mnick: branch 1.6.0 r233236 - in /branches/1.6.0: ./ pbx/pbx_config.c
Richard Mudgett
- [asterisk-dev] Asterisk 1.6.2: CDR missing for .call files
Mario Moran
- [asterisk-dev] ChanSpy mod
Gabriel Ortiz Lour
- [asterisk-dev] [Code Review] [16388] MOH refcount fix
Tilghman Lesher
- [asterisk-dev] Forbidden - wrong password on authentication for REGISTER
Nick Lewis
- [asterisk-dev] why reset p->method in chan_sip?
David Vossel
- [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop?
JR Richardson
- [asterisk-dev] [Code Review] Resolve format module dependency issue for app_voicemail
Russell Bryant
- [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop?
JR Richardson
- [asterisk-dev] Add attributes to Asterisk T38 invite for MAX TNT Interop? Update
JR Richardson
- [asterisk-dev] [Code Review] Cancel retries for all dialogs when the remote host isn't answering
Tilghman Lesher
- [asterisk-dev] How to Add New Language Prompts to Asterisk (Officially to get along with distribution)
Kannaiyan Natesan
- [asterisk-dev] Asterisk Release Candidates Available (1.4.28-rc1, 1.6.0.20-rc1, 1.6.1.12-rc1, and 1.6.2.0-rc8)
Asterisk Development Team
- [asterisk-dev] ASTERISK INCOMMING CALLER ID
Dulip Ravindra
- [asterisk-dev] Asterisk and multiple SIP registrations to the same host
Nick Lewis
- [asterisk-dev] ASTERISK INCOMMING CALLER ID
Nick Lewis
- [asterisk-dev] TDMOE protocol
Antonio
- [asterisk-dev] fxo ring parameter adjust
晶讯田工
- [asterisk-dev] Music on hold after an attended transfer to a queue.
nico kooijman
- [asterisk-dev] Asterisk Caller ID Pop Using Dial Event in 5038 manager port.
Dulip Ravindra
- [asterisk-dev] Addons1.6.1.2 compile issue
Bryant Zimmerman
- [asterisk-dev] Addons1.6.1.2 compile issue
Bryant Zimmerman
- [asterisk-dev] Asterisk Caller ID Pop Using Dial Event in 5038manager port.
Nick Lewis
- [asterisk-dev] ISDN remote HOLD DAHDI
Andrea Cristofanini
- [asterisk-dev] Disabling frame cache to workaround #16374 - negative impact ?
Pavel Troller
- [asterisk-dev] [Code Review] parkinglots with different parking extensions
Russell Bryant
- [asterisk-dev] [Code Review] RTP monitoring branch: initial review
Russell Bryant
- [asterisk-dev] Fwd: Inquiry:Asterisk sip server?
hadi motamedi
- [asterisk-dev] AGI question
Ngo-Vi Hoai-Anh
- [asterisk-dev] Make menu select for asterisk
Julian Lyndon-Smith
- [asterisk-dev] [Code Review] Unit Test Framework
David Vossel
- [asterisk-dev] [Code Review] Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing
Alec Davis
- [asterisk-dev] Asterisk and multiple SIP registrations to thesame host
Nick Lewis
- [asterisk-dev] Asterisk and multiple SIP registrations tothesame host
Nick Lewis
- [asterisk-dev] Peer matching architecture
Nick Lewis
- [asterisk-dev] Asterisk and multiple SIP registrationstothesame host
Nick Lewis
- [asterisk-dev] SIP replies to via or to ip/port (according toRFC)
Nick Lewis
- [asterisk-dev] SIP replies to via or to ip/port (accordingtoRFC)
Nick Lewis
- [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?
JR Richardson
- [asterisk-dev] Asterisk Core Dump
postalforall
- [asterisk-dev] Peer matching architecture
Nick Lewis
- [asterisk-dev] dahdi-linux: don't mknot on install by default
Tzafrir Cohen
- [asterisk-dev] dahdi-dude
Tzafrir Cohen
- [asterisk-dev] DAHDI-Linux 2.2.1-rc2 and DAHDI-Tools 2.2.1-rc2 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.4.28 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.6.0.20 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.6.1.12 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk 1.6.2.0 Now Available!
Asterisk Development Team
- [asterisk-dev] Compiling asterisk as single binary / minimalist OS
Julian Lyndon-Smith
- [asterisk-dev] SIP replies to via or to ip/port (according to RFC)
Nick Lewis
- [asterisk-dev] Peer matching architecture
Nick Lewis
- [asterisk-dev] Peer matching architecture
Nick Lewis
- [asterisk-dev] [Code Review] Add notificiation of intrupted file and fix bug
thedavidfactor at gmail.com
- [asterisk-dev] [Code Review] SIP: decode uri after parsing
David Vossel
- [asterisk-dev] Expression logic problem
Mueller, Alexander
- [asterisk-dev] 900 second call drop - 420 Bad Extension error w/ SIP, IOS GW
Jared Mauch
- [asterisk-dev] [Code Review] SIP: decode uri after parsing
Nick Lewis
- [asterisk-dev] [Code Review] SIP: decode uri after parsing
Nick Lewis
- [asterisk-dev] Asterisk 1.6.2.0 Now Available!
Faidon Liambotis
- [asterisk-dev] 1.6.1.12 Asterisk Crashes after 100 Bridged Calls in SIP?
JR Richardson
- [asterisk-dev] Release Schedules and plans for Asterisk 1.8
Russell Bryant
- [asterisk-dev] Regarding predictive dialing
Shivamurthy
- [asterisk-dev] Unit Test Framework Now Available!
David Vossel
- [asterisk-dev] [Code Review] ast_uri_encode() behavior change
David Vossel
- [asterisk-dev] asterisk 1.6.2.0 realtime queue
hooble at gmx.net
- [asterisk-dev] Happy Holidays from OpSys Consulting Group
Alexander Lopez
- [asterisk-dev] Failed to record Radius CDR record!
Zhang Shukun
- [asterisk-dev] DNID and RDNIS are not carried across IAX2
Nir Simionovich
- [asterisk-dev] No CDR for non-bridged outgoing calls
Jim Gottlieb
- [asterisk-dev] CallerID issue using DTMF on 1.2
Stelios Koroneos
- [asterisk-dev] Possible Call Event Logging (CEL) Bugs
Nic Colledge
- [asterisk-dev] de-latinisation of the web - http://blog.collins.net.pr/2009/12/de-latinisation-of-web.html
Dean Collins
- [asterisk-dev] Skype-Asterisk DTMF issues with Originate command
srinivas antarvedi
- [asterisk-dev] Issue related to CDR for calls dialed using Asterisk Manager Originate Command
Amit Patkar | Avhan Technologies Pvt. Ltd.
- [asterisk-dev] Propagation of cidnum/cidname to the called party in Originate cmd (1.6.1.12)
Pavel Troller
- [asterisk-dev] [Code Review] [15609] Prevent a stream of warnings about the voice frame queue too long
Tilghman Lesher
- [asterisk-dev] [asterisk-commits] oej: branch oej/deluxepine-1.4 r236897 - in /team/oej/deluxepine-1.4: include...
Kevin P. Fleming
- [asterisk-dev] [Code Review] Added ability to perform SRV lookups for AGI URIs
Brent Thomson
- [asterisk-dev] Regarding the "progressinband" SIP option
Pavel Troller
Last message date:
Thu Dec 31 23:28:34 CST 2009
Archived on: Thu Dec 31 23:28:39 CST 2009
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