September 2009 Archives by thread
Starting: Tue Sep 1 03:52:37 CDT 2009
Ending: Wed Sep 30 19:12:22 CDT 2009
Messages: 485
- [asterisk-dev] [Code Review] SIP: peer matchingbycallbackextension
Nick Lewis
- [asterisk-dev] Peer matching in trunk - matching on contact?
Nick Lewis
- [asterisk-dev] Segmentation Fault on 1.4.24.1
Dario Busso
- [asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy
Olle E Johansson
- [asterisk-dev] [Code Review] Add mutestream manager action and MUTESTREAM() dialplan function
Russell Bryant
- [asterisk-dev] Pinetree :: For Asterisk SIP trunks behind a SIPproxy
Nick Lewis
- [asterisk-dev] - [Iax2 Implementation] Reagrding Attendted Transfer
Kumar Subramanian
- [asterisk-dev] [Code Review] Add mutestream manager action and MUTESTREAM() dialplan function
Olle E Johansson
- [asterisk-dev] DNS A queries for unregister extension
thiago.fernandes
- [asterisk-dev] [Code Review] Add service message support for the 'national' switchtype
Kevin Fleming
- [asterisk-dev] [Code Review] Add service message support for the 'national' switchtype
rmudgett at digium.com
- [asterisk-dev] [asterisk-commits] oej: trunk r215110 - /trunk/channels/chan_sip.c
David Vossel
- [asterisk-dev] [Code Review] SIP uri parsing cleanup
David Vossel
- [asterisk-dev] [Code Review] SIP: Re-send non-100 provisional responses every 60 seconds until a final response is sent
Olle E Johansson
- [asterisk-dev] Pinetree :: For Asterisk SIP trunks behind aSIPproxy
Nick Lewis
- [asterisk-dev] [Code Review] SIP uri parsing cleanup
Nick Lewis
- [asterisk-dev] Pinetree :: For Asterisk SIP trunks behindaSIPproxy
Nick Lewis
- [asterisk-dev] Pinetree :: For Asterisk SIP trunksbehindaSIPproxy
Nick Lewis
- [asterisk-dev] Pinetree :: For Asterisk SIPtrunksbehindaSIPproxy
Nick Lewis
- [asterisk-dev] how to increase dial timeout
Giorgio Incantalupo
- [asterisk-dev] 1.6.2.0-beta4 - SIP TCP or TLS - Ringing/OK ignored
Stefan Tichy
- [asterisk-dev] [Code Review] Dynamic parking lots
Michiel van Baak
- [asterisk-dev] [Code Review] PubSub-based distributed events
marquis42 at gmail.com
- [asterisk-dev] definition of RTP jitter - potential bug in Asterisk
Klaus Darilion
- [asterisk-dev] [Code Review] Add service message support for the 'national' switchtype
rmudgett at digium.com
- [asterisk-dev] Asterisk 1.2.35, 1.4.26.2, 1.6.0.15, and 1.6.1.6 Now Available
Asterisk Development Team
- [asterisk-dev] definition of RTP jitter - potential bug inAsterisk
Nick Lewis
- [asterisk-dev] please please please open ViewVC for svn.digium.com again
Klaus Darilion
- [asterisk-dev] Muting a noisy warning in a specific case
Pavel Troller
- [asterisk-dev] please please please open ViewVC for svn.digium.comagain
Nick Lewis
- [asterisk-dev] definition of RTP jitter - potential bugin Asterisk
Nick Lewis
- [asterisk-dev] [Code Review] Optionally build apps in utils/ directory
Sean Bright
- [asterisk-dev] [Code Review] String field test module
Sean Bright
- [asterisk-dev] [Code Review] SIP: peer matching by address with TCP/TLS
David Vossel
- [asterisk-dev] [Code Review] SIP: peer matching TCP/TLS 1.6.0
David Vossel
- [asterisk-dev] Distributed events via XMPP
kael
- [asterisk-dev] add filter to verbose (CLI) for a context and src or dst match.
Germán Aracil Boned
- [asterisk-dev] [svn-commits] dvossel: trunk r216594 -/trunk/channels/chan_sip.c
Nick Lewis
- [asterisk-dev] Linksys SPA962 losing registration
Jeff LaCoursiere
- [asterisk-dev] Asterisk CLI commands not running !!!!!
abdelkader
- [asterisk-dev] Patch: Manager api, Posibility to send two channels in different direcitons
Håkon Nessjøen
- [asterisk-dev] How to find crashing problems (cause maybe double free) in my own module?
Håkon Nessjøen
- [asterisk-dev] [svn-commits] dvossel: trunk r216594 -/trunk/channels/chan_sip.c
David Vossel
- [asterisk-dev] [svn-commits] dvossel: trunk r216594 -/trunk/channels/chan_sip.c
Nick Lewis
- [asterisk-dev] [svn-commits] dvossel: trunk r216594 -/trunk/channels/chan_sip.c
David Vossel
- [asterisk-dev] [svn-commits] dvossel: trunk r216594 -/trunk/channels/chan_sip.c
David Vossel
- [asterisk-dev] Seeking Dallas based Developer
Michial Thompson
- [asterisk-dev] MeetMe and kernel module dependency
Hans Petter Selasky
- [asterisk-dev] [Code Review] SIP: port configuration
David Vossel
- [asterisk-dev] Dahdi compilation error on ARM
Rafael Seste
- [asterisk-dev] SIP: handling multiple m=video or m= audio lines
David Vossel
- [asterisk-dev] SIP: handling multiple m=video or m= audio lines
David Vossel
- [asterisk-dev] [Code Review] SIP: peer matching by callbackextension
David Vossel
- [asterisk-dev] developing pecl ectension for my asterisk system
Tamer Higazi
- [asterisk-dev] How to start
ABBAS SHAKEEL
- [asterisk-dev] Newbie question about ast_pthread_create_background
Alex Massover
- [asterisk-dev] oej: trunk r216805 - /trunk/channels/chan_sip.c
Russell Bryant
- [asterisk-dev] channel driver question
Gallmeier, Jonathan
- [asterisk-dev] Welcome to our new interns!
Russell Bryant
- [asterisk-dev] [Code Review] IAX2: encryption regression
David Vossel
- [asterisk-dev] Strange behaviour of Incomplete() application
Pavel Troller
- [asterisk-dev] [asterisk-commits] tilghman: trunk r217916 - in /trunk: channels/ contrib/scripts/
Kevin P. Fleming
- [asterisk-dev] [Asterisk 0014810]: [patch] channel-specific hangupcauses
Marcus Hunger
- [asterisk-dev] Problem with chan->_bridge Pointer in answered Macro
Stefan Schmidt
- [asterisk-dev] Problem with chan->_bridge Pointer in answered Macro
Stefan Schmidt
- [asterisk-dev] "hangupcause" field from "sip_pvt" structure shouldn't have the channel especific hangup cause ?
Mauro Sergio Ferreira Brasil
- [asterisk-dev] [Code Review] Make Asterisk able to connect to XMPP chatrooms
Philippe Sultan
- [asterisk-dev] [Code Review] New application JabberReceive, implement SendText in chan_gtalk and chan_jingle
Philippe Sultan
- [asterisk-dev] Opening temp file in action_command
Milad Rastian
- [asterisk-dev] Cisco IAD2421 Configuration
Michial Thompson
- [asterisk-dev] DAHDI_CHECK_HOOKSTATE and setting rxisoffhook
Tzafrir Cohen
- [asterisk-dev] astridevcon 2009
Clod Patry
- [asterisk-dev] Queue Hold Time Not Updated on Timeout or Break Out
Paul C Diem
- [asterisk-dev] Bug in SendFax/ReceiveFax ?
Håkon Nessjøen
- [asterisk-dev] New committer: Russ Meyerriecks
Russell Bryant
- [asterisk-dev] Need help about transmit_bearer_capability
Roger Schreiter
- [asterisk-dev] [Code Review] Fix race condition that may lead to a crash in ast_hangup() procedure.
Matthew Nicholson
- [asterisk-dev] [Code Review] Format change when queue still has frames of the old format
Tilghman Lesher
- [asterisk-dev] A Curious Question Cisco IAD
Michial Thompson
- [asterisk-dev] [dahdi-commits] tzafrir: tools/trunk r7134 - /tools/trunk/
Kevin P. Fleming
- [asterisk-dev] CNG fax detection on RTP
Gregory Boehnlein
- [asterisk-dev] [Code Review] AST-33: Create a list of channel variables to be posted within AMI call events
Tilghman Lesher
- [asterisk-dev] Help me test a new feature for AMI Redirect() command
Håkon Nessjøen
- [asterisk-dev] [Code Review] Document Asterisk open source issue tracker workflow
Russell Bryant
- [asterisk-dev] [Code Review] Fix RURI generation in chan_sip so that URIs are correct when SRV records are involved or when port 5060 is specified
Matthew Nicholson
- [asterisk-dev] [Code Review] Change internal structures in chan_sip - sip_request and sip_pkt merge and more stuff
Olle E Johansson
- [asterisk-dev] Asterisk 1.4.27-rc1, 1.6.0.16-rc1, 1.6.1.7-rc1, and 1.6.2.0-rc2 Now Available!
Asterisk Development Team
- [asterisk-dev] [Code Review] SIP: INVITE w/Replaces deadlock fix
David Vossel
- [asterisk-dev] calls drop during attended transfer with PRI line
Giorgio Incantalupo
- [asterisk-dev] Help me test a new feature for AMI Redirect() command
Håkon Nessjøen
- [asterisk-dev] [Code Review] Document Asterisk open source issue tracker workflow
Michiel van Baak
- [asterisk-dev] [Code Review] Force stream to really stop when ast_stopstream() is called
Tilghman Lesher
- [asterisk-dev] how to remove "remote hold" in notify message from pri protocol
Giorgio Incantalupo
- [asterisk-dev] qeustion about app_dial.c do_forward function getting the forwarding peer
Stefan Schmidt
- [asterisk-dev] Dahdi-linux embedded
Raphael Amorim
- [asterisk-dev] Manager Action sendText
ruben.buron at gmail.com
- [asterisk-dev] Manager Action sendText
ruben.buron at gmail.com
- [asterisk-dev] ISDN Calling/Caller/Redirecting Subaddress support work.
Alec Davis
- [asterisk-dev] [Code Review] tcptls_session memory leak
David Vossel
- [asterisk-dev] Automatic sending of progress indication - a possible problem
Pavel Troller
- [asterisk-dev] make -j
Klaus Darilion
- [asterisk-dev] [Code Review] Change SSRC when a peer sends a re-invite
Terry Wilson
- [asterisk-dev] multiple call parking
Ullas Sutaria
- [asterisk-dev] [svn-commits] mvanbaak: branch 1.4 r220027 - /branches/1.4/build_tools/mkpkgconfig
Sean Bright
- [asterisk-dev] ISDN Calling/Caller/Redirecting Subaddress support work.
Richard Mudgett
- [asterisk-dev] Asterisk 1.6.2.0-rc2 - srtp_init() problem
Stefan Tichy
- [asterisk-dev] ISDN Calling/Caller/Redirecting Subaddress support work.
Richard Mudgett
- [asterisk-dev] [Asterisk 0015465]: crash in bridging api
Marcus Hunger
- [asterisk-dev] [Code Review] Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.
Kevin Fleming
- [asterisk-dev] Dahdi-tools cross-compiling problem
Raphael Amorim
- [asterisk-dev] Inquiry:How to get remote access via Asterisk?
hadi motamedi
- [asterisk-dev] [Code Review] SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.
David Vossel
- [asterisk-dev] AST-XXX issues ?
Daniel Ferrer
- [asterisk-dev] SIP TLS handshake needs a timeout
David Vossel
- [asterisk-dev] Asterisk 1.6.2-rc2 suddenly restart
Eric Osvaldo R
- [asterisk-dev] valgrind errors
Benny Amorsen
- [asterisk-dev] Review request(s)
Brent Thomson
- [asterisk-dev] [Code Review] Added mohsuggest info to output for CLI: sip show peer {name}
Sean Bright
- [asterisk-dev] [Code Review] Added ability to specify different SIP and media addresses
Jared Smith
- [asterisk-dev] [Code Review] Added ability to perform SRV lookups for AGI URIs
Jared Smith
- [asterisk-dev] [Code Review] Added ability to perform SRV lookups for AGI URIs
Olle E Johansson
- [asterisk-dev] [Code Review] Added mohsuggest info to output for CLI: sip show peer {name}
Olle E Johansson
- [asterisk-dev] [Code Review] Added ability to specify different SIP and media addresses
Olle E Johansson
- [asterisk-dev] [Code Review] SIP: deadlock in local_attended_transfer()
David Vossel
- [asterisk-dev] Destinatio operator recognition by LNP Distinctive tone during "DIAL"
Isamar Maia
- [asterisk-dev] [Code Review] Fix ao2_iterator API to hold references to containers being iterated.
Kevin Fleming
- [asterisk-dev] Run extension script on SIP peer registration
Kirill 'Big K' Katsnelson
- [asterisk-dev] [Code Review] Added mohsuggest info to output for CLI: sip show peer {name}
Sean Bright
- [asterisk-dev] reasoning behind char[0]
Boehm, Matthew
- [asterisk-dev] segfault in asterisk 1.6.1.6 chan_iax2 after approx. 75.000 calls.
David Vossel
- [asterisk-dev] [Code Review] Don't return an embedded frame from a filestream
Russell Bryant
- [asterisk-dev] [Code Review] Don't return an embedded frame from a filestream
Kevin Fleming
- [asterisk-dev] [Code Review] Deadlock in channel masquerade handling
David Vossel
Last message date:
Wed Sep 30 19:12:22 CDT 2009
Archived on: Wed Sep 30 19:12:37 CDT 2009
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