April 2012 Archives by author

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Starting: Sun Apr 1 00:06:26 CDT 2012
Ending: Mon Apr 30 13:26:02 CDT 2012
Messages: 571

• [asterisk-users] Account code script needed.   Raj Mathur ( राज माथुर )
• [asterisk-users] Music as ringtone   Ashish Agarwal
• [asterisk-users] Voicemail crashs asterisk   Carlos Alvarez
• [asterisk-users] Dial Plan - Routing via Caller ID   Carlos Alvarez
• [asterisk-users] Dial Plan - Routing via Caller ID   Carlos Alvarez
• [asterisk-users] Dial Plan - Routing via Caller ID   Carlos Alvarez
• [asterisk-users] Dial Plan - Routing via Caller ID   Carlos Alvarez
• [asterisk-users] Dial Plan - Routing via Caller ID   Carlos Alvarez
• [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp   Carlos Alvarez
• [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp   Carlos Alvarez
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Carlos Alvarez
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Carlos Alvarez
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Carlos Alvarez
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Carlos Alvarez
• [asterisk-users] Set SIP peer state busy   Benny Amorsen
• [asterisk-users] Disable services on startup   Satria Anamarta
• [asterisk-users] Disable services on startup   Satria Anamarta
• [asterisk-users] Caller ID problem   Satria Anamarta
• [asterisk-users] Caller ID problem   Satria Anamarta
• [asterisk-users] Caller ID problem   Satria Anamarta
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call   Satria Anamarta
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call   Satria Anamarta
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call   Satria Anamarta
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Satria Anamarta
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call   Satria Anamarta
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Satria Anamarta
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Satria Anamarta
• [asterisk-users] Simple Gateway to setup calls bewteen two or 3 asterisk systems .   Arstan
• [asterisk-users] Dahdi QSIG with Tadiran Coral - not working   Arstan
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Dan Austin
• [asterisk-users] meetme identify user number   Dan Austin
• [asterisk-users] Call Transfer not working   Chris Bagnall
• [asterisk-users] Limit Call ?   Bakko
• [asterisk-users] Asterisk 1.8 and DeadAGI   Alex Balashov
• [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp   Alex Balashov
• [asterisk-users] Mac OS X sip client with Video support   Alex Balashov
• [asterisk-users] Flashphoner   Alex Balashov
• [asterisk-users] Flashphoner   Alex Balashov
• [asterisk-users] Flashphoner   Alex Balashov
• [asterisk-users] Call status register   Daniel Bareiro
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Satish Barot
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Satish Barot
• [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)   Paul Belanger
• [asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??   Paul Belanger
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Niccolò Belli
• [asterisk-users] T38 gateway issue   Niccolò Belli
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Niccolò Belli
• [asterisk-users] Pickup calls coming from queues   Niccolò Belli
• [asterisk-users] SNOM phones? Please test this patch (broken hints with notifycid=yes)   Niccolò Belli
• [asterisk-users] Pickup calls coming from queues   Niccolò Belli
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Niccolò Belli
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Niccolò Belli
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Niccolò Belli
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Niccolò Belli
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Niccolò Belli
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Niccolò Belli
• [asterisk-users] Pickup calls coming from queues   Niccolò Belli
• [asterisk-users] Transcoding degradation G711<->iLBC   Gustavo Garcia Bernardo
• [asterisk-users] g729 freezes 1.8   Jeff Brower
• [asterisk-users] Limit Call ?   Olivier CALVANO
• [asterisk-users] Limit Call ?   Olivier CALVANO
• [asterisk-users] Change extension for international ?   Olivier CALVANO
• [asterisk-users] Set variables from one asterisk ta a second.   Olivier CALVANO
• [asterisk-users] Set variables from one asterisk ta a second.   Olivier CALVANO
• [asterisk-users] Set variables from one asterisk ta a second.   Olivier CALVANO
• [asterisk-users] Set variables from one asterisk ta a second.   Olivier CALVANO
• [asterisk-users] Set variables from one asterisk ta a second.   Olivier CALVANO
• [asterisk-users] Set variables from one asterisk ta a second.   Olivier CALVANO
• [asterisk-users] Delete "Session timer" ?   Olivier CALVANO
• [asterisk-users] Delete "Session timer" ?   Olivier CALVANO
• [asterisk-users] No extension found ?   Olivier CALVANO
• [asterisk-users] Asterisk don't use "context="   Olivier CALVANO
• [asterisk-users] No extension found ?   Olivier CALVANO
• [asterisk-users] Strange problem on ougoing call   Olivier CALVANO
• [asterisk-users] Strange problem on ougoing call   Olivier CALVANO
• [asterisk-users] Strange problem on ougoing call   Olivier CALVANO
• [asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??   Olivier CALVANO
• [asterisk-users] Strange problem on ougoing call   Olivier CALVANO
• [asterisk-users] Strange problem on ougoing call   Olivier CALVANO
• [asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??   Olivier CALVANO
• [asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??   Olivier CALVANO
• [asterisk-users] cdr documentation - new fields   Marek Cervenka
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Carlos Chavez
• [asterisk-users] Asterisk - Nortel transfer problem   Carlos Chavez
• [asterisk-users] Asterisk - Nortel transfer problem   Carlos Chavez
• [asterisk-users] DAHDI works, but returns CHANUNAVAIL ??   Tzafrir Cohen
• [asterisk-users] Unable to access the running directory (Permission denied).   Tzafrir Cohen
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   Tzafrir Cohen
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Tzafrir Cohen
• [asterisk-users] medooze MCU versus confbridge   Bart Coninckx
• [asterisk-users] Company info   Josué Conti
• [asterisk-users] Company info   Josué Conti
• [asterisk-users] Company info   Josué Conti
• [asterisk-users] GXP1400   Doug Crompton
• [asterisk-users] GXP1400   Doug Crompton
• [asterisk-users] GXP1400   Doug Crompton
• [asterisk-users] GXP1400   Doug Crompton
• [asterisk-users] Asterisk ACL   Leandro Dardini
• [asterisk-users] Combining multiple SIP providers   Leandro Dardini
• [asterisk-users] Advice on Asterisk Conference   Leandro Dardini
• [asterisk-users] Strange problem on ougoing call   Leandro Dardini
• [asterisk-users] Set SIP peer state busy   Leandro Dardini
• [asterisk-users] Asterisk 10 & app_swift problem   Brent Davidson
• [asterisk-users] Asterisk 10 & app_swift problem   Brent Davidson
• [asterisk-users] Asterisk ACL   Steve Davies
• [asterisk-users] CONNECTEDLINE() updated during SIP events?   Steve Davies
• [asterisk-users] CONNECTEDLINE() updated during SIP events?   Steve Davies
• [asterisk-users] CONNECTEDLINE() updated during SIP events?   Steve Davies
• [asterisk-users] keep dst cdr record if context change   Alec Davis
• [asterisk-users] issue with Digium TDM410P   Alec Davis
• [asterisk-users] issue with Digium TDM410P   Alec Davis
• [asterisk-users] Pickup calls coming from queues   Alec Davis
• [asterisk-users] FXO -> GSM Gateway Problem   Alec Davis
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Danny Dias
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Danny Dias
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Danny Dias
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Danny Dias
• [asterisk-users] POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)   Danny Dias
• [asterisk-users] POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)   Danny Dias
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   Danny Dias
• [asterisk-users] No UDPTL ports remaining   Mike Diehl
• [asterisk-users] Digium D40 Direction map 'X' key not functioning   Dennis Dryden
• [asterisk-users] Set variables from one asterisk ta a second.   Eduardo
• [asterisk-users] Asterisk 1.8 and DeadAGI   Steve Edwards
• [asterisk-users] Voicemail crashs asterisk   Steve Edwards
• [asterisk-users] Unable to access the running directory (Permission denied).   Steve Edwards
• [asterisk-users] another non-root problem: unable to set utime ??   Steve Edwards
• [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN   Steve Edwards
• [asterisk-users] Set variables from one asterisk ta a second.   Steve Edwards
• [asterisk-users] Process a variable in a string.   Steve Edwards
• [asterisk-users] Custom Application recording problem   Steve Edwards
• [asterisk-users] Advice on Asterisk Conference   Steve Edwards
• [asterisk-users] Looking for IAX trunk/DID to replace Junction Networks   Steve Edwards
• [asterisk-users] 404 Response to Invite - Should be 401   Stuart Elvish
• [asterisk-users] [SOLVED] 404 Response to Invite - Should be 401   Stuart Elvish
• [asterisk-users] Open source speech recognition engine?   Carl-Fredrik Enell
• [asterisk-users] Unable to access the running directory (Permission denied).   Noah Engelberth
• [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN   Noah Engelberth
• [asterisk-users] Asterisk ACL   Mark Farmer
• [asterisk-users] Asterisk ACL   Mark Farmer
• [asterisk-users] fax tone testing   Kevin P. Fleming
• [asterisk-users] T.30 Fax and Echo Cancelation   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Kevin P. Fleming
• [asterisk-users] g729 freezes 1.8   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Kevin P. Fleming
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   Kevin P. Fleming
• [asterisk-users] g729 freezes 1.8   Kevin P. Fleming
• [asterisk-users] g729 freezes 1.8   Kevin P. Fleming
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6 on Ubuntu   Kevin P. Fleming
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   Kevin P. Fleming
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   Kevin P. Fleming
• [asterisk-users] CONNECTEDLINE() updated during SIP events?   Kevin P. Fleming
• [asterisk-users] Hangup Cause and SIP Response Code   Kevin P. Fleming
• [asterisk-users] Hangup Cause and SIP Response Code   Kevin P. Fleming
• [asterisk-users] Playing Multiple File ( simultaneously ) on Channel   Kevin P. Fleming
• [asterisk-users] Playing Multiple File ( simultaneously ) on Channel   Kevin P. Fleming
• [asterisk-users] Hangup Cause and SIP Response Code   Kevin P. Fleming
• [asterisk-users] Master Registrations?   Kevin P. Fleming
• [asterisk-users] extending fallback numbers   Phil Frost
• [asterisk-users] Change extension for international ?   Phil Frost
• [asterisk-users] Recent FreePBX vulnerability attacks   Jared Geiger
• [asterisk-users] Question about asterisk to Cisco   Jerry Geis
• [asterisk-users] concurrent channels limit   Israel Gottlieb
• [asterisk-users] Asterisk - How to trggier some specical reject message from Asterisk server?   Gu, Cheng
• [asterisk-users] ChannelRedirect with callee channel   Gunnar
• [asterisk-users] fax tone testing   Anita Hall
• [asterisk-users] Combining multiple SIP providers   Anita Hall
• [asterisk-users] T.30 Fax and Echo Cancelation   Anita Hall
• [asterisk-users] meetme timeout if only one participant   Matt Hamilton
• [asterisk-users] device state of a realtime queue member   Matt Hamilton
• [asterisk-users] device state of a realtime queue member   Matt Hamilton
• [asterisk-users] device state of a realtime queue member   Matt Hamilton
• [asterisk-users] Problem with blank/empty voicemails   Freddi Hansen
• [asterisk-users] Call Transfer not working   Rizwan Hisham
• [asterisk-users] Call Transfer not working   Rizwan Hisham
• [asterisk-users] Voicemail crashs asterisk   Thomas Hoellriegel
• [asterisk-users] Looking for IAX trunk/DID to replace Junction Networks   Russell Horn
• [asterisk-users] Looking for IAX trunk/DID to replace Junction Networks   Russell Horn
• [asterisk-users] Far end nat traversal for media is not working always   Arif Hossain
• [asterisk-users] Far end nat traversal not working   Arif Hossain
• [asterisk-users] Voicemail crashs asterisk   Steven Howes
• [asterisk-users] Company info   Steven Howes
• [asterisk-users] Fwd: Flashphoner   Steven Howes
• [asterisk-users] FXO -> GSM Gateway Problem   DHAVAL INDRODIYA
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Alejandro Imass
• [asterisk-users] ISDN incoming call disconnected after picking up phone   Ivo
• [asterisk-users] Invite + decreasing sequence number => 500 Error?   Olle E. Johansson
• [asterisk-users] Advice on Asterisk Conference   Mitchell Johnson
• [asterisk-users] Voicemail crashs asterisk   Matthew Jordan
• [asterisk-users] process_sdp: Multiple audio streams are not supported   Matthew Jordan
• [asterisk-users] SNOM phones? Please test this patch (broken hints with notifycid=yes)   Matthew Jordan
• [asterisk-users] Invite + decreasing sequence number => 500 Error?   Matthew Jordan
• [asterisk-users] Incoming SIP call is rejected always.   Matthew Jordan
• [asterisk-users] Incoming SIP call is rejected always.   Matthew Jordan
• [asterisk-users] Question for a Jira bug marshal   Matthew Jordan
• [asterisk-users] Which file is loading these lines?   Joseph
• [asterisk-users] Which file is loading these lines?   Joseph
• [asterisk-users] Which file is loading these lines?   Joseph
• [asterisk-users] priorityjumping - asterisk 1.8   Joseph
• [asterisk-users] priorityjumping - asterisk 1.8   Joseph
• [asterisk-users] priorityjumping - asterisk 1.8   Joseph
• [asterisk-users] [SOLVED] priorityjumping - asterisk 1.8   Joseph
• [asterisk-users] BUSY vs. CONGESTION   Joseph
• [asterisk-users] deleting: res_ldap.conf   Joseph
• [asterisk-users] deleting: res_ldap.conf   Joseph
• [asterisk-users] deleting: res_ldap.conf   Joseph
• [asterisk-users] [SOLVED] deleting: res_ldap.conf   Joseph
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Joseph
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Joseph
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Joseph
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Joseph
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Joseph
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Joseph
• [asterisk-users] Looking for IAX trunk/DID to replace Junction Networks   Joseph
• [asterisk-users] Caller id issues   Arstan Jusupov
• [asterisk-users] Caller id issues   Arstan Jusupov
• [asterisk-users] Caller id issues   Arstan Jusupov
• [asterisk-users] Custom Application recording problem   Billy Kaye
• [asterisk-users] Custom Application recording problem   Billy Kaye
• [asterisk-users] Custom Application recording problem   Billy Kaye
• [asterisk-users] Custom Application recording problem   Billy Kaye
• [asterisk-users] dial rule problems( on e1 interface) after upgrading 1.8   Oguzhan Kayhan
• [asterisk-users] Set SIP peer state busy   Jonas Kellens
• [asterisk-users] Set SIP peer state busy   Jonas Kellens
• [asterisk-users] Set SIP peer state busy   Jonas Kellens
• [asterisk-users] Set SIP peer state busy   Jonas Kellens
• [asterisk-users] Flashphoner   Don Kelly
• [asterisk-users] Voicemail crashs asterisk   Vik Killa
• [asterisk-users] Voicemail crashs asterisk   Vik Killa
• [asterisk-users] Dial Plan - Routing via Caller ID   John Kiniston
• [asterisk-users] Release Announcement: Adhearsion 2.0 for Asterisk 1.8+   Ben Klang
• [asterisk-users] meetme identify user number   Daniel Knoll
• [asterisk-users] Experience with virtual servers?   Bruce Komito
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Anton Kvashenkin
• [asterisk-users] OpenVPN design w/ Yealink   Jeff LaCoursiere
• [asterisk-users] Calendar Integration Problem   Bharat Lalcheta
• [asterisk-users] Restart single dahdi span   James Lamanna
• [asterisk-users] Strange Asterisk port behavior   Mikhail Lischuk
• [asterisk-users] Max number of PCIe cards   Patrick Lists
• [asterisk-users] Dahdi-2.4.0+2.4.0 means ??   Patrick Lists
• [asterisk-users] Dahdi-2.4.0+2.4.0 means ??   Patrick Lists
• [asterisk-users] Asterisk 10 & app_swift problem   Patrick Lists
• [asterisk-users] Transcoding degradation G711<->iLBC   Patrick Lists
• [asterisk-users] OT - Which Sugarcm 6.2 plugin to add click-to-dial (with asterisk) ?   Patrick Lists
• [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN   Doug Lytle
• [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN   Doug Lytle
• [asterisk-users] Company info   Doug Lytle
• [asterisk-users] Playing Multiple File ( simultaneously ) on Channel   Leif Madsen
• [asterisk-users] device state of a realtime queue member   Ishfaq Malik
• [asterisk-users] Call recording and transfer issue (asterisk 1.8)   Ishfaq Malik
• [asterisk-users] ReceiveFax and multiple pages   Ishfaq Malik
• [asterisk-users] ReceiveFax and multiple pages   Ishfaq Malik
• [asterisk-users] Cannot send mail from System command   Ishfaq Malik
• [asterisk-users] Cannot send mail from System command [SOLVED]   Ishfaq Malik
• [asterisk-users] Call Transfer not working   Takehiro Matsushima
• [asterisk-users] Call Transfer not working   Takehiro Matsushima
• [asterisk-users] Pickup calls coming from queues   Mark Michelson
• [asterisk-users] syntax error from digium fax manual ??   Barry Miller
• [asterisk-users] Delete "Session timer" ?   Barry Miller
• [asterisk-users] Delete "Session timer" ?   Barry Miller
• [asterisk-users] Delete "Session timer" ?   Barry Miller
• [asterisk-users] POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)   John Millican
• [asterisk-users] Mac OS X sip client with Video support   Arjan Kroon | Mobillion
• [asterisk-users] 10.3 : sip loses registration ?   Larry Moore
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Larry Moore
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Larry Moore
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Larry Moore
• [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)   Larry Moore
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   Tony Mountifield
• [asterisk-users] Set SIP peer state busy   Michal Mruškovič
• [asterisk-users] Set SIP peer state busy   Michal Mruškovič
• [asterisk-users] Which file is loading these lines?   Richard Mudgett
• [asterisk-users] Call Deflection with DAHDISendCallreroutingFacility   Richard Mudgett
• [asterisk-users] DAHDI inter-digit timeout = 0   Richard Mudgett
• [asterisk-users] CONNECTEDLINE() updated during SIP events?   Richard Mudgett
• [asterisk-users] Cannot resume call on hold   Mark Murawski
• [asterisk-users] sip pregi net account registration   Gopalakrishnan N
• [asterisk-users] sip pregi net account registration   Gopalakrishnan N
• [asterisk-users] sip pregi net account registration   Gopalakrishnan N
• [asterisk-users] sip pregi net account registration   Gopalakrishnan N
• [asterisk-users] Nicaragua PSTN Frequency Parameters   Gopalakrishnan N
• [asterisk-users] DAHDI FXO Call Issues / Indication Types   Tim Nelson
• [asterisk-users] Limit Call ?   Danny Nicholas
• [asterisk-users] meetme timeout if only one participant   Danny Nicholas
• [asterisk-users] Change extension for international ?   Danny Nicholas
• [asterisk-users] Combining multiple SIP providers   Danny Nicholas
• [asterisk-users] Google TTS - Asterisk   Danny Nicholas
• [asterisk-users] MessageSend, SIP, and call files   Danny Nicholas
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Danny Nicholas
• [asterisk-users] MessageSend, SIP, and call files   Danny Nicholas
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Danny Nicholas
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Danny Nicholas
• [asterisk-users] Dial Plan - Routing via Caller ID   Danny Nicholas
• [asterisk-users] Asterisk 1.8.12.0-rc1   Danny Nicholas
• [asterisk-users] GXP1400   Danny Nicholas
• [asterisk-users] Question about asterisk to Cisco   Danny Nicholas
• [asterisk-users] priorityjumping - asterisk 1.8   Danny Nicholas
• [asterisk-users] Disable services on startup   Danny Nicholas
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Danny Nicholas
• [asterisk-users] Caller ID problem   Danny Nicholas
• [asterisk-users] Set variables from one asterisk ta a second.   Danny Nicholas
• [asterisk-users] Set variables from one asterisk ta a second.   Danny Nicholas
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Danny Nicholas
• [asterisk-users] Caller ID problem   Danny Nicholas
• [asterisk-users] ExtensionStatus event   Danny Nicholas
• [asterisk-users] Account code script needed.   Danny Nicholas
• [asterisk-users] Incoming SIP call is rejected always.   Danny Nicholas
• [asterisk-users] Delete "Session timer" ?   Danny Nicholas
• [asterisk-users] BUSY vs. CONGESTION   Danny Nicholas
• [asterisk-users] device state of a realtime queue member   Danny Nicholas
• [asterisk-users] Experience with virtual servers?   Danny Nicholas
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Danny Nicholas
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Danny Nicholas
• [asterisk-users] Looking for IAX trunk/DID to replace Junction Networks   Danny Nicholas
• [asterisk-users] Set SIP peer state busy   Danny Nicholas
• [asterisk-users] Custom Application recording problem   Dale Noll
• [asterisk-users] Custom Application recording problem   Dale Noll
• [asterisk-users] Custom Application recording problem   Dale Noll
• [asterisk-users] Looking for IAX trunk/DID to replace Junction Networks   John Novack
• [asterisk-users] Max number of PCIe cards   Olivier
• [asterisk-users] Does Cisco 79XX with SIP firmware support asterisk's BLF ?   Olivier
• [asterisk-users] Experiences with Polycom-Kirk 6000 and DECT/GAP handsets   Olivier
• [asterisk-users] How to disable CDR adaptative logging on asterisk 1.8 ?   Olivier
• [asterisk-users] OT - How to localize Freepbx 2.10 or 2.9 ?   Olivier
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Olivier
• [asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE   Olivier
• [asterisk-users] OT - Which Sugarcm 6.2 plugin to add click-to-dial (with asterisk) ?   Olivier
• [asterisk-users] DAHDI 2.6.1 - What does " Build OSLEC EC if in the tree" feature means ?   Olivier
• [asterisk-users] Invite + decreasing sequence number => 500 Error?   Benoit Panizzon
• [asterisk-users] Advice on Asterisk Conference   Tim Panton
• [asterisk-users] Transcoding degradation G711<->iLBC   Tim Panton
• [asterisk-users] Advice on Asterisk Conference   Tim Panton
• [asterisk-users] Incoming SIP call is rejected always.   Yaroslav Panych
• [asterisk-users] Incoming SIP call is rejected always.   Yaroslav Panych
• [asterisk-users] Incoming SIP call is rejected always.   Yaroslav Panych
• [asterisk-users] Flashphoner   Jason Parker
• [asterisk-users] Dahdi QSIG with Tadiran Coral - not working   Eduardo Pimenta
• [asterisk-users] E & M signalling and Dahdi   Eduardo Pimenta
• [asterisk-users] Peer SIP authentication with Taqua switch   Philip Prindeville
• [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)   p070075 Muhammad Atif Ramzan
• [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)   p070075 Muhammad Atif Ramzan
• [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)   p070075 Muhammad Atif Ramzan
• [asterisk-users] upgrading from asterisk 1.4 to 1.6   p070075 Muhammad Atif Ramzan
• [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)   Zohair Raza
• [asterisk-users] Asterisk 1.8.10 getaddrinfo   Zohair Raza
• [asterisk-users] dahdi cannot make simaltaneous calls   Mc GRATH Ricardo
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Mc GRATH Ricardo
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Mc GRATH Ricardo
• [asterisk-users] Asterisk - Nortel transfer problem   Mc GRATH Ricardo
• [asterisk-users] Asterisk - Nortel transfer problem   Mc GRATH Ricardo
• [asterisk-users] concurrent channels limit   Matt Riddell
• [asterisk-users] extending fallback numbers   Matt Riddell
• [asterisk-users] Personal queue with one agent: add calls to extension   Roland
• [asterisk-users] Personal queue with one agent: add calls to extension   Roland
• [asterisk-users] Ringing detection ?   Nicolas Ross
• [asterisk-users] Open source replacement for AudioCodes nCite 1000 SBC   Matthew J. Roth
• [asterisk-users] Mute DTMF   Todd Routhier
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Todd Routhier
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Todd Routhier
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Todd Routhier
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Todd Routhier
• [asterisk-users] Run AGI while agent ringing instead of only when connected   Todd Routhier
• [asterisk-users] issue with Digium TDM410P   Shaun Ruffell
• [asterisk-users] issue with Digium TDM410P   Shaun Ruffell
• [asterisk-users] issue with Digium TDM410P   Shaun Ruffell
• [asterisk-users] VMWI DAHDI   Shaun Ruffell
• [asterisk-users] Dahdi-2.4.0+2.4.0 means ??   Shaun Ruffell
• [asterisk-users] chan_sip.so module not loading   Shaun Ruffell
• [asterisk-users] DAHDI FXO Call Issues / Indication Types   Shaun Ruffell
• [asterisk-users] deleting: res_ldap.conf   Shaun Ruffell
• [asterisk-users] deleting: res_ldap.conf   Shaun Ruffell
• [asterisk-users] dahdi versions before 2.5 compilation error and ubuntu   Shaun Ruffell
• [asterisk-users] DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available   Shaun Ruffell
• [asterisk-users] DAHDI 2.6.1 - What does " Build OSLEC EC if in the tree" feature means ?   Shaun Ruffell
• [asterisk-users] Asterisk 1.8 and DeadAGI   SamyGo
• [asterisk-users] Asterisk 1.8 and DeadAGI   SamyGo
• [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)   SamyGo
• [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)   SamyGo
• [asterisk-users] No extension found ?   SamyGo
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call   SamyGo
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call   SamyGo
• [asterisk-users] No extension found ?   SamyGo
• [asterisk-users] No extension found ?   SamyGo
• [asterisk-users] No extension found ?   SamyGo
• [asterisk-users] Strange problem on ougoing call   SamyGo
• [asterisk-users] No UDPTL ports remaining   SamyGo
• [asterisk-users] Music as ringtone   SamyGo
• [asterisk-users] Question for a Jira bug marshal   Stefan Schmidt
• [asterisk-users] Mac OS X sip client with Video support   Stefan Schmidt
• [asterisk-users] No UDPTL ports remaining   Stefan Schmidt
• [asterisk-users] Asterisk ACL   Warren Selby
• [asterisk-users] extending fallback numbers   Warren Selby
• [asterisk-users] sip pregi net account registration   Warren Selby
• [asterisk-users] Does Cisco 79XX with SIP firmware support asterisk's BLF ?   Warren Selby
• [asterisk-users] Dial Plan - Routing via Caller ID   Warren Selby
• [asterisk-users] Dial Plan - Routing via Caller ID   Warren Selby
• [asterisk-users] Dial Plan - Routing via Caller ID   Warren Selby
• [asterisk-users] Dial Plan - Routing via Caller ID   Warren Selby
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6: chan_dahdi? dahdi?   Warren Selby
• [asterisk-users] Set variables from one asterisk ta a second.   Warren Selby
• [asterisk-users] hints and server-side DND (do not disturb)   Warren Selby
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)   Warren Selby
• [asterisk-users] unsubscribe   Zeeshan Ali Shah
• [asterisk-users] Call Deflection with DAHDISendCallreroutingFacility   Mehdi Shirazi
• [asterisk-users] Call Deflection with DAHDISendCallreroutingFacility   Mehdi Shirazi
• [asterisk-users] Open source speech recognition engine?   Nickolay V. Shmyrev
• [asterisk-users] DAHDI inter-digit timeout = 0   Chris Sohns
• [asterisk-users] DAHDI inter-digit timeout = 0   Chris Sohns
• [asterisk-users] Google TTS - Asterisk   Sriram
• [asterisk-users] cross ivr is comming in my ivr system   Arthur Stanfield
• [asterisk-users] cross ivr is comming in my ivr system   Arthur Stanfield
• [asterisk-users] Caller ID problem   Arthur Stanfield
• [asterisk-users] Experience with virtual servers?   Arthur Stanfield
• [asterisk-users] Fwd: Flashphoner   Arthur Stanfield
• [asterisk-users] Asterisk ACL   A J Stiles
• [asterisk-users] Voicemail crashs asterisk   A J Stiles
• [asterisk-users] cross ivr is comming in my ivr system   A J Stiles
• [asterisk-users] g729 freezes 1.8   A J Stiles
• [asterisk-users] g729 freezes 1.8   A J Stiles
• [asterisk-users] g729 freezes 1.8   A J Stiles
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call   A J Stiles
• [asterisk-users] HELP!! Caller ID "unknown" for all inbound call   A J Stiles
• [asterisk-users] Cannot send mail from System command   A J Stiles
• [asterisk-users] chan_mobile with Nokia 6021 - incoming SMS causes call to drop   James Stocks
• [asterisk-users] chan_sip.so module not loading   Roi Stork
• [asterisk-users] FollowMe and billsec field of the CDRs   Sunny
• [asterisk-users] extending fallback numbers   Paolo Supino
• [asterisk-users] extending fallback numbers   Paolo Supino
• [asterisk-users] extending fallback numbers   Paolo Supino
• [asterisk-users] Mac OS X sip client with Video support   Paolo Supino
• [asterisk-users] Mac OS X sip client with Video support   Paolo Supino
• [asterisk-users] Mac OS X sip client with Video support   Paolo Supino
• [asterisk-users] concurrent channels limit   Syco
• [asterisk-users] Limit Call ?   Syco
• [asterisk-users] concurrent channels limit   Syco
• [asterisk-users] AMI Originate double call   Syco
• [asterisk-users] Set variables from one asterisk ta a second.   Stuart Elvish - IP Exchange Systems
• [asterisk-users] Experience with virtual servers?   Stuart Elvish - IP Exchange Systems
• [asterisk-users] Advice on Asterisk Conference   Stuart Elvish - IP Exchange Systems
• [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp   Administrator TOOTAI
• [asterisk-users] Set variables from one asterisk ta a second.   Administrator TOOTAI
• [asterisk-users] Set variables from one asterisk ta a second.   Administrator TOOTAI
• [asterisk-users] Dial Local doesn't honore the channel language setting   Administrator TOOTAI
• [asterisk-users] Question for a Jira bug marshal   Administrator TOOTAI
• [asterisk-users] No extension found ?   Administrator TOOTAI
• [asterisk-users] Question for a Jira bug marshal   Administrator TOOTAI
• [asterisk-users] No extension found ?   Administrator TOOTAI
• [asterisk-users] Asterisk - Nortel transfer problem   Jonn Taylor
• [asterisk-users] Asterisk - Nortel transfer problem   Jonn Taylor
• [asterisk-users] DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.24, 1.8.11.1, 10.3.1 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] Scheduled Maintenance for Asterisk Project community services   Asterisk Development Team
• [asterisk-users] AST-2012-004: Asterisk Manager User Unauthorized Shell Access   Asterisk Security Team
• [asterisk-users] AST-2012-005: Heap Buffer Overflow in Skinny Channel Driver   Asterisk Security Team
• [asterisk-users] AST-2012-006: Remote Crash Vulnerability in SIP Channel Driver   Asterisk Security Team
• [asterisk-users] FXO -> GSM Gateway Problem   Tech
• [asterisk-users] FXO -> GSM Gateway Problem   Tech
• [asterisk-users] FXO -> GSM Gateway Problem   Tech
• [asterisk-users] issue with Digium TDM410P   Mathieu Therrien
• [asterisk-users] issue with Digium TDM410P   Mathieu Therrien
• [asterisk-users] VMWI DAHDI   Mathieu Therrien
• [asterisk-users] Experience with virtual servers?   Brynjolfur Thorvardsson
• [asterisk-users] cross ivr is comming in my ivr system   Jagadish Thoutam
• [asterisk-users] FXO -> GSM Gateway Problem   Duncan Turnbull
• [asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??   Duncan Turnbull
• [asterisk-users] Transcoding degradation G711<->iLBC   Steve Underwood
• [asterisk-users] E & M signalling and Dahdi   Steve Underwood
• [asterisk-users] No extension found ?   Michel Verbraak
• [asterisk-users] red5sip SIP ua can't register   Vieri
• [asterisk-users] ExtensionStatus event   Vieri
• [asterisk-users] hints and server-side DND (do not disturb)   Vieri
• [asterisk-users] hints and server-side DND (do not disturb)   Vieri
• [asterisk-users] Call recovery feature   Kristijan Vrban
• [asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE   Ben WIlliams
• [asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE   Ben WIlliams
• [asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE [SOLVED]   Ben WIlliams
• [asterisk-users] Dial Plan - Routing via Caller ID   Chad Wallace
• [asterisk-users] GXP1400   Chad Wallace
• [asterisk-users] priorityjumping - asterisk 1.8   Chad Wallace
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   Chad Wallace
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   Chad Wallace
• [asterisk-users] MessageSend, SIP, and call files   Roger Burton West
• [asterisk-users] MessageSend, SIP, and call files   Roger Burton West
• [asterisk-users] Asterisk 1.8.12.0-rc1   Roger Burton West
• [asterisk-users] Asterisk 1.8.12.0-rc1   Roger Burton West
• [asterisk-users] Voicemail crashs asterisk   Eric Wieling
• [asterisk-users] Dial Plan - Routing via Caller ID   Eric Wieling
• [asterisk-users] Dial Plan - Routing via Caller ID   Eric Wieling
• [asterisk-users] Dahdi-2.4.0+2.4.0 means ??   Eric Wieling
• [asterisk-users] priorityjumping - asterisk 1.8   Eric Wieling
• [asterisk-users] priorityjumping - asterisk 1.8   Eric Wieling
• [asterisk-users] priorityjumping - asterisk 1.8   Eric Wieling
• [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)   Eric Wieling
• [asterisk-users] BUSY vs. CONGESTION   Eric Wieling
• [asterisk-users] Delete "Session timer" ?   Eric Wieling
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   Eric Wieling
• [asterisk-users] Hangup Cause and SIP Response Code   Eric Wieling
• [asterisk-users] Set SIP peer state busy   Eric Wieling
• [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp   Johan Wilfer
• [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp   Johan Wilfer
• [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp   Johan Wilfer
• [asterisk-users] Process a variable in a string.   Bryant Zimmerman
• [asterisk-users] Experience with virtual servers?   Bryant Zimmerman
• [asterisk-users] GXP1400   Bryant Zimmerman
• [asterisk-users] Grandstream 1.0.3.30 BETA Firmware   Bryant Zimmerman
• [asterisk-users] Hangup Cause and SIP Response Code   Bryant Zimmerman
• [asterisk-users] Master Registrations?   Bryant Zimmerman
• [asterisk-users] how do I catch the audio stream in real-time for another application?   andre
• [asterisk-users] process_sdp: Multiple audio streams are not supported   cjwstudios
• [asterisk-users] Account code script needed.   cjwstudios
• [asterisk-users] Google TTS - Asterisk   asterisk at ck-lee.com
• [asterisk-users] 10.3.0: gtalk_request: No XMPP client to talk to, us (partial JID)   sean darcy
• [asterisk-users] Unable to access the running directory (Permission denied).   sean darcy
• [asterisk-users] Unable to access the running directory (Permission denied).   sean darcy
• [asterisk-users] another non-root problem: unable to set utime ??   sean darcy
• [asterisk-users] syntax error from digium fax manual ??   sean darcy
• [asterisk-users] syntax error from digium fax manual ??   sean darcy
• [asterisk-users] 10.3 : sip loses registration ?   sean darcy
• [asterisk-users] 10.3 : sip loses registration ?   sean darcy
• [asterisk-users] DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available   sean darcy
• [asterisk-users] DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available   sean darcy
• [asterisk-users] Realtime asterisk 10.3.0   abc def
• [asterisk-users] Fwd: Realtime asterisk 10.3.0   abc def
• [asterisk-users] Telephony Card: GSM slots + Analoge   bilal ghayyad
• [asterisk-users] Asterisk 1.8 and DeadAGI   bilal ghayyad
• [asterisk-users] dahdi versions before 2.5 compilation error and ubuntu   bilal ghayyad
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6: chan_dahdi? dahdi?   bilal ghayyad
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   bilal ghayyad
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   bilal ghayyad
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6 on Ubuntu   bilal ghayyad
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   bilal ghayyad
• [asterisk-users] asterisk 1.4.39 and dahdi 2.6   bilal ghayyad
• [asterisk-users] hints and server-side DND (do not disturb)   isrlgb at gmail.com
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   isrlgb at gmail.com
• [asterisk-users] Asterisk + Phones behind different Nat Firewalls   isrlgb at gmail.com
• [asterisk-users] Dial Plan - Routing via Caller ID   lists65 at gmail.com
• [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN   lists65 at gmail.com
• [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN   lists65 at gmail.com
• [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN   lists65 at gmail.com
• [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN   lists65 at gmail.com
• [asterisk-users] Dial Plan - Routing via Caller ID   lists65 at gmail.com
• [asterisk-users] Dial Plan - Routing via Caller ID   lists65 at gmail.com
• [asterisk-users] Dial Plan - Routing via Caller ID   lists65 at gmail.com
• [asterisk-users] Dial Plan - Routing via Caller ID   lists65 at gmail.com
• [asterisk-users] Dial Plan - Routing via Caller ID   lists65 at gmail.com
• [asterisk-users] Dial Plan - Routing via Caller ID   lists65 at gmail.com
• [asterisk-users] Call recovery feature   shayne.alone at gmail.com
• [asterisk-users] Playing Multiple File ( simultaneously ) on Channel   shayne.alone at gmail.com
• [asterisk-users] Playing Multiple File ( simultaneously ) on Channel   shayne.alone at gmail.com
• [asterisk-users] Playing Multiple File ( simultaneously ) on Channel   shayne.alone at gmail.com
• [asterisk-users] Playing Multiple File ( simultaneously ) on Channel   shayne.alone at gmail.com
• [asterisk-users] Flashphoner   shayne.alone at gmail.com
• [asterisk-users] Asterisk 1.8.12.0-rc1   motty.cruz
• [asterisk-users] Asterisk 1.8.12.0-rc1   motty.cruz
• [asterisk-users] Asterisk 1.8.10 getaddrinfo   motty.cruz
• [asterisk-users] Auto answer Asterisk ; Unable to create channel of type   motty.cruz
• [asterisk-users] Asterisk as sip client Unable to create channel of type 'Console' (cause 0 - Unknown)   motty.cruz
• [asterisk-users] When CALL-ID were same , I could hijack another session   nakaji
• [asterisk-users] g729 freezes 1.8   samuel
• [asterisk-users] g729 freezes 1.8   samuel
• [asterisk-users] g729 freezes 1.8   samuel
• [asterisk-users] g729 freezes 1.8   samuel
• [asterisk-users] g729 freezes 1.8   samuel
• [asterisk-users] dahdi cannot make simaltaneous calls   rosli sukri
• [asterisk-users] Dahdi-2.4.0+2.4.0 means ??   upendra
• [asterisk-users] Dahdi-2.4.0+2.4.0 means ??   upendra
• [asterisk-users] Simple Gateway to setup calls bewteen two or 3 asterisk systems .   upendra
• [asterisk-users] Hangup Cause and SIP Response Code   BryantZ at zktech.com
• [asterisk-users] Asterisk CDRs   [Digital^Dude] ®

Last message date: Mon Apr 30 13:26:02 CDT 2012
Archived on: Mon Apr 30 13:35:34 CDT 2012

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