August 2009 Archives by thread
Starting: Sat Aug 1 00:20:27 CDT 2009
Ending: Mon Aug 31 19:16:07 CDT 2009
Messages: 295
- [asterisk-dev] Fwd: Inquiry:Asterisk supporting hash (#) key
hadi motamedi
- [asterisk-dev] Fwd: Inquiry : Asterisk hash key
hadi motamedi
- [asterisk-dev] [Asterisk-Dev] HylaFAX and spandsp
Thomas Kenyon
- [asterisk-dev] HELP....reading asterisk code
Ruddy Gbaguidi
- [asterisk-dev] gentoo linux problems with dahdi-tools-2.2.0
Randall Degges
- [asterisk-dev] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement
Asterisk Team
- [asterisk-dev] h extension not considered within a macro forAsterisk 1.6.0.6 and later
Serge Berney
- [asterisk-dev] Integration with legacy systems ...
Mauro Sergio Ferreira Brasil
- [asterisk-dev] chan_mobile - chan_usbb3g sinalization problems
Rafael Seste
- [asterisk-dev] Why do a DNS search on sip_devicestate
Gabriel Ortiz Lour
- [asterisk-dev] [Code Review] data api gsoc2009
Eliel Sardañons
- [asterisk-dev] trunk asterisk/main/file.c connected line updates intended?
D Tucny
- [asterisk-dev] T38 gateway - consultation
Daniel Ferenci
- [asterisk-dev] Hangup channel
Rafael Seste
- [asterisk-dev] [Code Review] CLI filtering [GSoC 2009]
Russell Bryant
- [asterisk-dev] [Code Review] CLI filtering [GSoC 2009]
Eliel Sardañons
- [asterisk-dev] [Code Review] CLI filtering [GSoC 2009]
Mark Michelson
- [asterisk-dev] [Code Review] Add COLP support to libpri for ETSI PTP, ETSI PTMP, and Q.SIG.
rmudgett at digium.com
- [asterisk-dev] [Code Review] Add COLP support to chan_dahdi/sig_pri.
rmudgett at digium.com
- [asterisk-dev] Hold Option
Venkateshwarlu Kakkireni
- [asterisk-dev] Compiling app_cbmysql for asterisk 1.6.2
Alex Villacís Lasso
- [asterisk-dev] SIP users authentication ...
Mauro Sergio Ferreira Brasil
- [asterisk-dev] DeadAgi application issue
Max Alex
- [asterisk-dev] ChanSpy Trouble
Matt D.
- [asterisk-dev] about driver wct4xxp
leexx
- [asterisk-dev] ast_channel_free
Johann Steinwendtner
- [asterisk-dev] rtp data in an app question
Gallmeier, Jonathan
- [asterisk-dev] Analog Lines Answer, busy, etc... supervision.
Jose Hector Galimberti
- [asterisk-dev] About chan_mobile
Carlos Ruiz Diaz
- [asterisk-dev] Asterisk 1.2.34, 1.4.26.1, 1.6.0.12, and 1.6.1.4 release announcement
Asterisk Team
- [asterisk-dev] 1.6.0.13: sounds 1.4.14, but rc2 was 1.4.15
sean darcy
- [asterisk-dev] [Code Review] CLI filtering [GSoC 2009]
Mark Michelson
- [asterisk-dev] Flag to disable Packet2packet bridging
Richard Brady
- [asterisk-dev] Retrieve current channel from inside ARA driver method ...
Mauro Sergio Ferreira Brasil
- [asterisk-dev] qualifyfreq backport to 1.4
Albert Siersema
- [asterisk-dev] New group CCSS branch
Mark Michelson
- [asterisk-dev] dahdi_genconf and CAS configuration
Tzafrir Cohen
- [asterisk-dev] Transfer after pickup
Benny Amorsen
- [asterisk-dev] Realtime behavior questions ...
Mauro Sergio Ferreira Brasil
- [asterisk-dev] Customizable Voicemail Menu
Giuseppe Sucameli
- [asterisk-dev] strange outbound-registration loop
Klaus Darilion
- [asterisk-dev] Platform decision ...
Mauro Sergio Ferreira Brasil
- [asterisk-dev] Why does the outbound SIP channel use jointcapability instead of capability?
Alex Hermann
- [asterisk-dev] Asterisk project changes Music-On-Hold provider
Asterisk Development Team
- [asterisk-dev] Call routing between two Asterisk boxes using SIP not working ...
Mauro Sergio Ferreira Brasil
- [asterisk-dev] Asterisk 1.6.0.14-rc1 and 1.6.1.5-rc1 Now Available
Asterisk Development Team
- [asterisk-dev] 1.6.2.0-beta4 - SIP TCP or TLS - Ringing/OK ignored
Stefan Tichy
- [asterisk-dev] loading modules help
Boehm, Matthew
- [asterisk-dev] [Code Review] SIP: Re-send non-100 provisional responses every 60 seconds until a final response is sent
Terry Wilson
- [asterisk-dev] [Code Review] SIP: incorrect usage of 503 message
David Vossel
- [asterisk-dev] Action: Originate + CDR
Chandrakant Solanki
- [asterisk-dev] bug 14538 reappeared
Kaloyan Kovachev
- [asterisk-dev] app_fax.c :: Channel Hangup => Transmission Error but fax transmitted successfully !
Serge Berney
- [asterisk-dev] AMI - Originate Action with "NO ANSWER"
Chandrakant Solanki
- [asterisk-dev] Header from REFFER
Gomtesh Jain
- [asterisk-dev] Asterisk 1.6.1 branch and deprecated "call-limit" setting in sip.conf / device state / hints
Jared Smith
- [asterisk-dev] No unique identifier for CDR
Nick Lewis
- [asterisk-dev] [Code Review] SIP uri parsing cleanup
David Vossel
- [asterisk-dev] No unique identifier for CDR
Nick Lewis
- [asterisk-dev] [Code Review] SIP uri parsing cleanup
Nick Lewis
- [asterisk-dev] No unique identifier for CDR
Nick Lewis
- [asterisk-dev] [Code Review] SIP uri parsing cleanup
Nick Lewis
- [asterisk-dev] No unique identifier for CDR
Nick Lewis
- [asterisk-dev] No unique identifier for CDR
Nick Lewis
- [asterisk-dev] [Code Review] SIP: peer matching by callbackextension
David Vossel
- [asterisk-dev] No unique identifier for CDR
Nick Lewis
- [asterisk-dev] [Code Review] SIP: peer matching bycallbackextension
Nick Lewis
- [asterisk-dev] Queue Auto-Answer
João Castilho
- [asterisk-dev] QoS channel stats - definition and export
Klaus Darilion
- [asterisk-dev] AGI Server Stop Down issue
Sandip
- [asterisk-dev] [Code Review] SIP: peer matching by callbackextension
Olivier Krief
- [asterisk-dev] Asterisk 1.6.0.14 and 1.6.1.5 Now Available
Asterisk Development Team
- [asterisk-dev] Asterisk MWI issue
ilker Aktuna
- [asterisk-dev] Upgrading Asterisk in Trixbox installation
ilker Aktuna
- [asterisk-dev] ISDN Calling Sub Address and Called Sub Address for the branches
Alec Davis
- [asterisk-dev] Validate Phone Number with Asterisk
Chandrakant Solanki
- [asterisk-dev] Asterisk 1.6.2.0-rc1 Now Available
Asterisk Development Team
- [asterisk-dev] [Code Review] Add mutestream manager action and MUTESTREAM() dialplan function
Olle E Johansson
Last message date:
Mon Aug 31 19:16:07 CDT 2009
Archived on: Mon Aug 31 19:16:11 CDT 2009
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