February 2012 Archives by author

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Starting: Wed Feb 1 02:25:19 CST 2012
Ending: Wed Feb 29 18:43:29 CST 2012
Messages: 897

• [asterisk-users] Binding to 0.0.0.0 a security risk?   Raj Mathur ( राज माथुर )
• [asterisk-users] Script to automatically update externip. Useful for a host with dynamic public IP   Raj Mathur ( राज माथुर )
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Raj Mathur ( राज माथुर )
• [asterisk-users] Forwarding queue to remote agent over PSTN   Raj Mathur ( राज माथुर )
• [asterisk-users] Forwarding queue to remote agent over PSTN   Raj Mathur ( राज माथुर )
• [asterisk-users] T38 faxing - UDPTL creation failed   =?utf-8?B?IpronZ794e3h6e0i?=
• [asterisk-users] T38 faxing - UDPTL creation failed   =?utf-8?B?IpronZ794e3h6e0i?=
• [asterisk-users] T38 faxing - UDPTL creation failed   =?utf-8?B?IpronZ794e3h6e0i?=
• [asterisk-users] Asterisk && RTCP   Gohar Ahmed
• [asterisk-users] dial plan with hangup cause 34   ing.Achim Alexandru
• [asterisk-users] MixMonitor and ChanSpy   Carlos Alvarez
• [asterisk-users] MixMonitor and ChanSpy   Carlos Alvarez
• [asterisk-users] Headset Options   Carlos Alvarez
• [asterisk-users] SIP hardware phones   Carlos Alvarez
• [asterisk-users] SIP hardware phones   Carlos Alvarez
• [asterisk-users] Automatic Number Identification and anonymous calls   Carlos Alvarez
• [asterisk-users] checking if a phone number is UP   Carlos Alvarez
• [asterisk-users] checking if a phone number is UP   Carlos Alvarez
• [asterisk-users] SIP hardware phones   Carlos Alvarez
• [asterisk-users] "conferenced" transfers   Carlos Alvarez
• [asterisk-users] how many UDP ports is required for 1 call   Carlos Alvarez
• [asterisk-users] how many UDP ports is required for 1 call   Carlos Alvarez
• [asterisk-users] Trunking betweeb two Asterisk System   Carlos Alvarez
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Carlos Alvarez
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Carlos Alvarez
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Carlos Alvarez
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Carlos Alvarez
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Carlos Alvarez
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Carlos Alvarez
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Carlos Alvarez
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Carlos Alvarez
• [asterisk-users] SIP hardware phones   Benny Amorsen
• [asterisk-users] SIP hardware phones   Benny Amorsen
• [asterisk-users] Problem with libpri / asterisk   Andres
• [asterisk-users] Problem with libpri / asterisk   Andres
• [asterisk-users] "conferenced" transfers   Andres
• [asterisk-users] Possible bug (or feature?) in extension matching and parking feature   Sebastian Arcus
• [asterisk-users] Possible bug (or feature?) in extension matching and parking feature   Sebastian Arcus
• [asterisk-users] pstn bridge to asterisk - phones connected to pstn stop ringing when asterisk answers   Sebastian Arcus
• [asterisk-users] Subscribe Problem - Zombie Channel   Örn Arnarson
• [asterisk-users] cell mysql odbc support   Arstan
• [asterisk-users] Web and Email Chat   Goke M Aruna
• [asterisk-users] How to Send SMS on SS7 DChannel-16(Signaling Channel)   Mian Asif
• [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?   Bruce B
• [asterisk-users] Should you "ever" use nat=no?   Bruce B
• [asterisk-users] Should you "ever" use nat=no?   Bruce B
• [asterisk-users] High verbose set at console effects the logger file "Full" - Why is that?   Bruce B
• [asterisk-users] how many UDP ports is required for 1 call   Bruce B
• [asterisk-users] Best CRM for Asterisk   Bruce B
• [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?   David Backeberg
• [asterisk-users] AGI: blocking script until playback complete   Chris Bagnall
• [asterisk-users] AGI: blocking script until playback complete   Chris Bagnall
• [asterisk-users] AGI: blocking script until playback complete   Chris Bagnall
• [asterisk-users] AGI: blocking script until playback complete   Chris Bagnall
• [asterisk-users] how many UDP ports is required for 1 call   Alex Balashov
• [asterisk-users] how many UDP ports is required for 1 call   Alex Balashov
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alex Balashov
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alex Balashov
• [asterisk-users] Router that support Asterisk   Gerardo Barajas
• [asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link   Gerardo Barajas
• [asterisk-users] Matching asterisk PBX cdrs to Telco's Trunk CDR's   Nicholas Barnes
• [asterisk-users] India Pune Pri call problem   Satish Barot
• [asterisk-users] India Pune Pri call problem   Satish Barot
• [asterisk-users] Forwarding queue to remote agent over PSTN   Satish Barot
• [asterisk-users] Asterisk 1.8.x app_meetme.so   Abdul Basit
• [asterisk-users] Postgresql in Asterisk   Sergio Basurto
• [asterisk-users] Postgresql in Asterisk   Sergio Basurto
• [asterisk-users] Postgresql in Asterisk   Sergio Basurto
• [asterisk-users] Postgresql in Asterisk   Sergio Basurto
• [asterisk-users] Asterisk CLI unresponsive   Paul Belanger
• [asterisk-users] Asterisk CLI unresponsive   Paul Belanger
• [asterisk-users] Asterisk CLI unresponsive   Paul Belanger
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Paul Belanger
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Paul Belanger
• [asterisk-users] Weird IPs in Fail2ban list   Paul Belanger
• [asterisk-users] No valid transports available, falling back to 'udp'.   Paul Belanger
• [asterisk-users] dahdi and digium debian package   Paul Belanger
• [asterisk-users] dahdi and digium debian package   Paul Belanger
• [asterisk-users] app_rpt and chan_usbradio removal from trunk   Paul Belanger
• [asterisk-users] 10.2.0-rc2: permitted contact can't register.   Paul Belanger
• [asterisk-users] Inclusion of the ILBC codec starting from asterisk 1.8.10/10.2   Niccolò Belli
• [asterisk-users] Asterisk perl AGI confusing variables   Ron Bergin
• [asterisk-users] Asterisk perl AGI confusing variables   Ron Bergin
• [asterisk-users] AGI: blocking script until playback complete   Ron Bergin
• [asterisk-users] Asterisk V/s FreeSwitch   Jeff Brower
• [asterisk-users] Asterisk V/s FreeSwitch   Jeff Brower
• [asterisk-users] Streaming musiconhold via mpg123   Stephen Brown
• [asterisk-users] Streaming musiconhold via mpg123   Stephen Brown
• [asterisk-users] Streaming musiconhold via mpg123   Stephen Brown
• [asterisk-users] Question for the group   Terry Brummell
• [asterisk-users] Headset Options   Blake Burgess
• [asterisk-users] DECT with handover   Blake Burgess
• [asterisk-users] Stumped about adding a semi-colon to a variable   CDR
• [asterisk-users] Is there any way to make call fail after # of rings?   CDR
• [asterisk-users] Script to automatically update externip. Useful for a host with dynamic public IP   John Cahill
• [asterisk-users] Script to automatically update externip. Useful for a host with dynamic public IP   John Cahill
• [asterisk-users] Transmit NOA (sss) to Dialogic IMG via SIP / Transmisión de NOA hacia Dialogic IMG por SIP   Luis Alejandro Beltran Castañeda
• [asterisk-users] Problem with SIP phone outside local network   Carlos Chavez
• [asterisk-users] CDR Analyzer/Queue stats reporting   Carlos Chavez
• [asterisk-users] No valid transports available, falling back to 'udp'.   Adolphe Cher-Aime
• [asterisk-users] IDLE Features on Avaya through Asterisk   Aamir Chougule
• [asterisk-users] Why doesn't this manager.conf code work on Asterisk 1.6.2 and 1.8.9?   Frank Church
• [asterisk-users] What packages are required to get cdr_adaptive_odbc to be compiled in Asterisk?   Frank Church
• [asterisk-users] What packages are required to getcdr_adaptive_odbc to be compiled in Asterisk?   Frank Church
• [asterisk-users] Are there any ATAs that support IP6?   Frank Church
• [asterisk-users] Asterisk V/s FreeSwitch   Frank Church
• [asterisk-users] Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network?   Frank Church
• [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?   Ast Coder
• [asterisk-users] India Pune Pri call problem   Ast Coder
• [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up   Ast Coder
• [asterisk-users] Where can I find some good examples of listening to AMI events via PHP & how to listen to a specific event?   Ast Coder
• [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up   Ast Coder
• [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up   Ast Coder
• [asterisk-users] Where can I find some good examples of listening to AMI events via PHP & how to listen to a specific event?   Ast Coder
• [asterisk-users] asterisk dahdi problem.   Tzafrir Cohen
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Tzafrir Cohen
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Tzafrir Cohen
• [asterisk-users] Router that support Asterisk   Tzafrir Cohen
• [asterisk-users] error during dahdi installation on centos   Tzafrir Cohen
• [asterisk-users] app_rpt and chan_usbradio removal from trunk   Tzafrir Cohen
• [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP   Leandro Dardini
• [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP   Leandro Dardini
• [asterisk-users] SER Still recommended for large installs?   Leandro Dardini
• [asterisk-users] Phone Inventory   Leandro Dardini
• [asterisk-users] Alphanumeric DTMF !?   Matt Darnell
• [asterisk-users] Asterisk CLI unresponsive   Steve Davies
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Steve Davies
• [asterisk-users] Is there any way to make call fail after # of rings?   Alec Davis
• [asterisk-users] Problem installing B410P BRI card for asterisk   Alec Davis
• [asterisk-users] Polycom firmware 4.0.1 and paging   Jim DeVito
• [asterisk-users] Reffered By header is missing from SIP INVITE in call transfer scenarios   Deka, Rajib IN MAA SL
• [asterisk-users] asterisk MCID detection   Deka, Rajib IN MAA SL
• [asterisk-users] Calling a group of phones and force the speaker   Danny Dias
• [asterisk-users] HT286 dialtone and ring cadence   Mike Diehl
• [asterisk-users] Turning off splash ring on PAP2T   Mike Diehl
• [asterisk-users] Transfer to fax   Mike Diehl
• [asterisk-users] Transfer to fax   Mike Diehl
• [asterisk-users] Transfer to fax   Mike Diehl
• [asterisk-users] Transfer to fax   Mike Diehl
• [asterisk-users] Transfer to fax   Mike Diehl
• [asterisk-users] Quick bash tip for finding free SIP extensions from your sip.conf   Steve Edwards
• [asterisk-users] [asterisk-dev] How to play audio file in background in dialplan?   Steve Edwards
• [asterisk-users] How to play audio file in background in dialplan?   Steve Edwards
• [asterisk-users] How to play audio file in background in dialplan?   Steve Edwards
• [asterisk-users] How to play audio file in background in dialplan?   Steve Edwards
• [asterisk-users] Is this doable?   Steve Edwards
• [asterisk-users] How to play audio file in background in dialplan?   Steve Edwards
• [asterisk-users] How to play audio file in background in dialplan?   Steve Edwards
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Steve Edwards
• [asterisk-users] Question for the group   Steve Edwards
• [asterisk-users] Asterisk perl AGI confusing variables   Steve Edwards
• [asterisk-users] Asterisk perl AGI confusing variables   Steve Edwards
• [asterisk-users] asterisk network connections   Steve Edwards
• [asterisk-users] AGI: blocking script until playback complete   Steve Edwards
• [asterisk-users] AGI: blocking script until playback complete   Steve Edwards
• [asterisk-users] AGI: blocking script until playback complete   Steve Edwards
• [asterisk-users] Callmanager 4 Asterisk Malformed/Missing URL   Nigel Eldred
• [asterisk-users] Callmanager 4 Asterisk Malformed/Missing URL   Nigel Eldred
• [asterisk-users] CA Issued Certificates / TLS + SRTP   Stuart Elvish
• [asterisk-users] Hot desking and presence   Noah Engelberth
• [asterisk-users] Hot desking and presence   Noah Engelberth
• [asterisk-users] Possible bug (or feature?) in extension matching and parking feature   Noah Engelberth
• [asterisk-users] CDR Analyzer/Queue stats reporting   Noah Engelberth
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Noah Engelberth
• [asterisk-users] Set(CALLERID(name)) when incoming call is "anonymous"   Mark Engelhardt
• [asterisk-users] amd detect answering machine   Etann
• [asterisk-users] amd detect answering machine   Etann
• [asterisk-users] amd detect answering machine   Etann
• [asterisk-users] Answering machine dectection (AMD)   Etann
• [asterisk-users] Answering machine dectection (AMD)   Etann
• [asterisk-users] Router that support Asterisk   C F
• [asterisk-users] Is this doable?   C F
• [asterisk-users] externip nat audio sip trunk issue problem   C F
• [asterisk-users] externip nat audio sip trunk issue problem   C F
• [asterisk-users] Binding to 0.0.0.0 a security risk?   C F
• [asterisk-users] Is this doable?   C F
• [asterisk-users] Is this doable?   C F
• [asterisk-users] Polycom firmware 4.0.1 and paging   C F
• [asterisk-users] How to play audio file in background indialplan?   Cary Fitch
• [asterisk-users] Dynamically toggling ConfBridge recording from conference menu   Kevin P. Fleming
• [asterisk-users] Problem with DTMF in Voicemail main   Kevin P. Fleming
• [asterisk-users] Inclusion of the ILBC codec starting from asterisk 1.8.10/10.2   Kevin P. Fleming
• [asterisk-users] Automatic Number Identification and anonymous calls   Kevin P. Fleming
• [asterisk-users] Automatic Number Identification and anonymous calls   Kevin P. Fleming
• [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles   Kevin P. Fleming
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Kevin P. Fleming
• [asterisk-users] res_http_post.so questions   Kevin P. Fleming
• [asterisk-users] T38 faxing - UDPTL creation failed   Kevin P. Fleming
• [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles   Kevin P. Fleming
• [asterisk-users] Help with Codes and Polycom Phones   Kevin P. Fleming
• [asterisk-users] Asterisk V/s FreeSwitch   Kevin P. Fleming
• [asterisk-users] Call Completion   Kevin P. Fleming
• [asterisk-users] Stumped about adding a semi-colon to a variable   Kevin P. Fleming
• [asterisk-users] Stumped about adding a semi-colon to a variable   Kevin P. Fleming
• [asterisk-users] skip authentication for REGISTER   Kevin P. Fleming
• [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2   Kevin P. Fleming
• [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2   Kevin P. Fleming
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Kevin P. Fleming
• [asterisk-users] Asterisk && RTCP   Kevin P. Fleming
• [asterisk-users] Should you "ever" use nat=no?   Kevin P. Fleming
• [asterisk-users] Asterisk && RTCP   Kevin P. Fleming
• [asterisk-users] Should you "ever" use nat=no?   Kevin P. Fleming
• [asterisk-users] how many UDP ports is required for 1 call   Kevin P. Fleming
• [asterisk-users] how many UDP ports is required for 1 call   Kevin P. Fleming
• [asterisk-users] how many UDP ports is required for 1 call   Kevin P. Fleming
• [asterisk-users] how many UDP ports is required for 1 call   Kevin P. Fleming
• [asterisk-users] how many UDP ports is required for 1 call   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.x app_meetme.so   Kevin P. Fleming
• [asterisk-users] how many UDP ports is required for 1 call   Kevin P. Fleming
• [asterisk-users] Streaming musiconhold via mpg123   Kevin P. Fleming
• [asterisk-users] Transfer to fax   Kevin P. Fleming
• [asterisk-users] Asterisk NOT in the media path   Kevin P. Fleming
• [asterisk-users] Transfer to fax   Kevin P. Fleming
• [asterisk-users] Transfer to fax   Kevin P. Fleming
• [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up   Kevin P. Fleming
• [asterisk-users] Replicating SIP registration Info between active to standby   Kevin P. Fleming
• [asterisk-users] TCE400P diagnostic messages   Kevin P. Fleming
• [asterisk-users] better timing source for an asterisk gateway   Kevin P. Fleming
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Kevin P. Fleming
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Kevin P. Fleming
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Kevin P. Fleming
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Kevin P. Fleming
• [asterisk-users] DTMF forwarding and Page   Matteo Fortini
• [asterisk-users] DTMF forwarding and Page   Matteo Fortini
• [asterisk-users] DTMF forwarding and Page [SOLVED] [PATCH 1/1]   Matteo Fortini
• [asterisk-users] Dynamically toggling ConfBridge recording from conference menu   Josh Freeman
• [asterisk-users] app_rpt and chan_usbradio removal from trunk   Josh Freeman
• [asterisk-users] Presence subscription from other pbx systems   Jan Fricke
• [asterisk-users] Call queuing behavior   Phil Frost
• [asterisk-users] Call queuing behavior   Phil Frost
• [asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link   Phil Frost
• [asterisk-users] "conferenced" transfers   Phil Frost
• [asterisk-users] "conferenced" transfers   Phil Frost
• [asterisk-users] Troubleshooting realtime LDAP   Phil Frost
• [asterisk-users] "conferenced" transfers   Phil Frost
• [asterisk-users] how many UDP ports is required for 1 call   Phil Frost
• [asterisk-users] how many UDP ports is required for 1 call   Phil Frost
• [asterisk-users] "conferenced" transfers   Phil Frost
• [asterisk-users] Rejecting transfers to in-use parking spaces   Phil Frost
• [asterisk-users] Rejecting transfers to in-use parking spaces   Phil Frost
• [asterisk-users] Rejecting transfers to in-use parking spaces   Phil Frost
• [asterisk-users] Asterisk NOT in the media path   Phillip Frost
• [asterisk-users] Polycom firmware 4.0.1 and paging   Dave Fullerton
• [asterisk-users] Polycom firmware 4.0.1 and paging   Dave Fullerton
• [asterisk-users] chan_capi audio weirdness   Arik Raffael Funke
• [asterisk-users] Call holding with chan_capi   Arik Raffael Funke
• [asterisk-users] Call holding with chan_capi   Arik Raffael Funke
• [asterisk-users] Call holding with chan_capi   Arik Raffael Funke
• [asterisk-users] chan_capi audio weirdness   Arik Raffael Funke
• [asterisk-users] chan_capi audio weirdness   Arik Raffael Funke
• [asterisk-users] SIP Provider Russia, Ukraine, Poland   Christian Gansberger
• [asterisk-users] MixMonitor and ChanSpy   Tiago Geada
• [asterisk-users] Asterisk V/s FreeSwitch   Jared Geiger
• [asterisk-users] Asterisk NOT in the media path   Jared Geiger
• [asterisk-users] GSM gateway call redirect   Peter Gelencser
• [asterisk-users] No IVR audio. Jump in RTP sequence number   Dave George
• [asterisk-users] Asterisk 1.8.9.1 Now Available   Eric Germann
• [asterisk-users] Asterisk 1.8.9.2 Now Available   Eric Germann
• [asterisk-users] Router that support Asterisk   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Gilles
• [asterisk-users] Asterisk V/s FreeSwitch   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Gilles
• [asterisk-users] How to receive SMS ?   Gilles
• [asterisk-users] How to receive SMS ?   Gilles
• [asterisk-users] How to receive SMS ?   Gilles
• [asterisk-users] how many UDP ports is required for 1 call   Gilles
• [asterisk-users] how many UDP ports is required for 1 call   Gilles
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Jean-Denis Girard
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] TE410P (1st) without cables always green   Marcio Gomes
• [asterisk-users] Streaming musiconhold via mpg123   Israel Gottlieb
• [asterisk-users] Asterisk version that support Database Configuration   Sammy Govind
• [asterisk-users] Can someone tell me what is this issue ?   Sammy Govind
• [asterisk-users] Playback with noanswer in AGI   Sammy Govind
• [asterisk-users] Playback with noanswer in AGI   Sammy Govind
• [asterisk-users] Playback with noanswer in AGI   Sammy Govind
• [asterisk-users] MixMonitor and ChanSpy   Sammy Govind
• [asterisk-users] MixMonitor and ChanSpy   Sammy Govind
• [asterisk-users] Asterisk V/s FreeSwitch   Sammy Govind
• [asterisk-users] Asterisk perl AGI confusing variables   Sammy Govind
• [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?   Sammy Govind
• [asterisk-users] Asterisk perl AGI confusing variables   Sammy Govind
• [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?   Sammy Govind
• [asterisk-users] Asterisk perl AGI confusing variables   Sammy Govind
• [asterisk-users] Asterisk perl AGI confusing variables   Sammy Govind
• [asterisk-users] Asterisk && RTCP   Sammy Govind
• [asterisk-users] Asterisk && RTCP   Sammy Govind
• [asterisk-users] Asterisk && RTCP   Sammy Govind
• [asterisk-users] Capture sip Response   Sammy Govind
• [asterisk-users] Alphanumeric DTMF !?   Sammy Govind
• [asterisk-users] Alphanumeric DTMF !?   Sammy Govind
• [asterisk-users] Alphanumeric DTMF !?   Sammy Govind
• [asterisk-users] p-associated-uri in 200OK   Goyal, Amit
• [asterisk-users] Automatic Number Identification and anonymous calls   Maximilian Grobecker
• [asterisk-users] Early Media configuration doesn't seem to be working   Maximilian Grobecker
• [asterisk-users] How to associate agents - extensions?   Asterisk Guy
• [asterisk-users] How to implement outlook popup   Luke Hamburg
• [asterisk-users] High verbose set at console effects the logger file "Full" - Why is that?   Luke Hamburg
• [asterisk-users] High verbose set at console effects the logger file "Full" - Why is that?   Luke Hamburg
• [asterisk-users] Dialing multiple numbers   Matt Hamilton
• [asterisk-users] Dialing multiple numbers   Matt Hamilton
• [asterisk-users] criteria for setting registration expiration   Matt Hamilton
• [asterisk-users] skip authentication for REGISTER   Matt Hamilton
• [asterisk-users] skip authentication for REGISTER   Matt Hamilton
• [asterisk-users] AstLinux 1.0.2 Release   Darrick Hartman
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Jakob Hirsch
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Jakob Hirsch
• [asterisk-users] Problem callerid ignored by using callfiles   Thomas Hoellriegel
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Steven Howes
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Steven Howes
• [asterisk-users] Stumped about adding a semi-colon to a variable   Steven Howes
• [asterisk-users] India Pune Pri call problem   Steven Howes
• [asterisk-users] Aastra phone dial plan   Brian C. Huffman
• [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.   DHAVAL INDRODIYA
• [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.   DHAVAL INDRODIYA
• [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug   DHAVAL INDRODIYA
• [asterisk-users] Dahdi Answer a Call On ringing State.   DHAVAL INDRODIYA
• [asterisk-users] Dahdi Answer a Call On ringing State.   DHAVAL INDRODIYA
• [asterisk-users] Dahdi Answer a Call On ringing State.   DHAVAL INDRODIYA
• [asterisk-users] Dahdi Answer a Call On ringing State.   DHAVAL INDRODIYA
• [asterisk-users] Asterisk Version 1.8.9.2 Question About SIP/SRTP/TLS   DHAVAL INDRODIYA
• [asterisk-users] Virtual Server   Alejandro Imass
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alejandro Imass
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alejandro Imass
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alejandro Imass
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alejandro Imass
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alejandro Imass
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alejandro Imass
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alejandro Imass
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Alejandro Imass
• [asterisk-users] dial plan with hangup cause 34   Ioan Indreias
• [asterisk-users] asterisk distributions   Ioan Indreias
• [asterisk-users] Problem with DTMF in Voicemail main   Ira
• [asterisk-users] Problem with DTMF in Voicemail main   Ira
• [asterisk-users] Regional settings with Sipura SPA-3000   Jan Johansson
• [asterisk-users] Polycom IP331 Configuration   Mark Johnson
• [asterisk-users] Polycom IP331 Configuration   Mark Johnson
• [asterisk-users] Polycom IP331 Configuration   Mark Johnson
• [asterisk-users] Asterisk 10 and DUNDi, Extended Support?   Matthew Jordan
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Matthew Jordan
• [asterisk-users] High verbose set at console effects the logger file "Full" - Why is that?   Matthew Jordan
• [asterisk-users] Park() ignores 'r' option which should disable music on hold in favour of ringing tone   Matthew Jordan
• [asterisk-users] Park() ignores 'r' option which should disable music on hold in favour of ringing tone   Matthew Jordan
• [asterisk-users] Define custom vm-login sound file per VM context?   Matthew Jordan
• [asterisk-users] Asterisk 1.8.x app_meetme.so   Matthew Jordan
• [asterisk-users] 10.2.0-rc2: permitted contact can't register.   Matthew Jordan
• [asterisk-users] SIP Provider Russia, Ukraine, Poland   Joseph
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] res_http_post.so questions   Josh
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Josh
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] res_http_post.so questions   Josh
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] Is this doable?   Josh
• [asterisk-users] Custom extension: dial a queue   Kanuvar
• [asterisk-users] asterisk dahdi problem.   Oguzhan Kayhan
• [asterisk-users] asterisk dahdi problem.   Oğuzhan Kayhan
• [asterisk-users] MixMonitor and ChanSpy   Jonas Kellens
• [asterisk-users] Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Jonas Kellens
• [asterisk-users] MixMonitor and ChanSpy   Jonas Kellens
• [asterisk-users] MixMonitor and ChanSpy   Jonas Kellens
• [asterisk-users] asterisk network connections   Jonas Kellens
• [asterisk-users] Starting asterisk: Cannot find specified TTY   Jonas Kellens
• [asterisk-users] Asterisk NOT in the media path   Jonas Kellens
• [asterisk-users] Starting asterisk: Cannot find specified TTY   Jonas Kellens
• [asterisk-users] Asterisk NOT in the media path   Jonas Kellens
• [asterisk-users] Trunking betweeb two Asterisk System   Faraj Khasib
• [asterisk-users] Asterisk 10 and DUNDi, Extended Support?   David Klaverstyn
• [asterisk-users] Help with Codes and Polycom Phones   David Klaverstyn
• [asterisk-users] Script to automatically update externip. Useful for a host with dynamic public IP   Klaverstyn, David C
• [asterisk-users] No valid transports available, falling back to 'udp'.   Klaverstyn, David C
• [asterisk-users] Polycom IP331 Configuration   Klaverstyn, David C
• [asterisk-users] how many UDP ports is required for 1 call   John Knight
• [asterisk-users] Experience with Eicon Diva PRO 3.0?   Michelle Konzack
• [asterisk-users] FW by postmaster at datavox.co.uk : Delivery Status Notification (Failure)   Michelle Konzack
• [asterisk-users] IAX DNS SRV   Justin Korkiner
• [asterisk-users] error during dahdi installation on centos   Kamlesh Kumar
• [asterisk-users] error during dahdi installation on centos   Kamlesh Kumar
• [asterisk-users] Asterisk CLI unresponsive   Anton Kvashenkin
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Jeff LaCoursiere
• [asterisk-users] Calling SIP extension through Local/XXXX negotiates slin codec instead of gsm   Alex Villací­s Lasso
• [asterisk-users] res_http_post.so questions   Andrew Latham
• [asterisk-users] Queuemember status before calling the Queue command   Roberto Linck
• [asterisk-users] Asterisk CLI unresponsive   Mikhail Lischuk
• [asterisk-users] TE410P (1st) without cables always green   Patrick Lists
• [asterisk-users] TE410P (1st) without cables always green   Patrick Lists
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Patrick Lists
• [asterisk-users] SIP hardware phones   Patrick Lists
• [asterisk-users] Asterisk V/s FreeSwitch   Patrick Lists
• [asterisk-users] Asterisk V/s FreeSwitch   Patrick Lists
• [asterisk-users] Forwarding queue to remote agent over PSTN   Patrick Lists
• [asterisk-users] High verbose set at console effects the logger file "Full" - Why is that?   Patrick Lists
• [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up   Patrick Lists
• [asterisk-users] asterisk MCID detection   Patrick Lists
• [asterisk-users] externip nat audio sip trunk issue problem   Gabriel Ortiz Lour
• [asterisk-users] externip nat audio sip trunk issue problem   Gabriel Ortiz Lour
• [asterisk-users] finally, an open alternative to Viber   Daniel - Lumicall.org
• [asterisk-users] asterisk-users at lists.digium.com Nacha Alert ID10416   Doug Lytle
• [asterisk-users] Asterisk 1.8.x app_meetme.so   Doug Lytle
• [asterisk-users] Asterisk 1.8.x app_meetme.so   Doug Lytle
• [asterisk-users] Early Media configuration doesn't seem to be working   Ishfaq Malik
• [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.   Ishfaq Malik
• [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug   Ishfaq Malik
• [asterisk-users] Is there any way to make call fail after # of rings?   Ishfaq Malik
• [asterisk-users] How does format_mp3 work?   Ishfaq Malik
• [asterisk-users] How does format_mp3 work?   Ishfaq Malik
• [asterisk-users] Phone Inventory   Ishfaq Malik
• [asterisk-users] SIP Provider Russia, Ukraine, Poland   Markus
• [asterisk-users] checking if a phone number is UP   Markus
• [asterisk-users] checking if a phone number is UP   Markus
• [asterisk-users] pstn bridge to asterisk - phones connected to pstn stop ringing when asterisk answers   Martin
• [asterisk-users] Replicating SIP registration Info between active to standby   Takehiro Matsushima
• [asterisk-users] Replicating SIP registration Info between active to standby   Takehiro Matsushima
• [asterisk-users] outbound fax over t38 gateway can't pass   Dmitry Melekhov
• [asterisk-users] outbound fax over t38 gateway can't pass   Dmitry Melekhov
• [asterisk-users] Is this doable?   Gordon Messmer
• [asterisk-users] externip nat audio sip trunk issue problem   Gordon Messmer
• [asterisk-users] Is this doable?   Gordon Messmer
• [asterisk-users] Is this doable?   Gordon Messmer
• [asterisk-users] Is this doable?   Gordon Messmer
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Gordon Messmer
• [asterisk-users] Is this doable?   Gordon Messmer
• [asterisk-users] Is this doable?   Gordon Messmer
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] Polycom firmware 4.0.1 and paging   Mike
• [asterisk-users] High verbose set at console effects the logger file "Full" - Why is that?   Mike
• [asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link   Vladimir Mikhelson
• [asterisk-users] Streaming musiconhold via mpg123   Vladimir Mikhelson
• [asterisk-users] Capture sip Response   John Millican
• [asterisk-users] Stuck DAHDI Lines   Antonio Modesto
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   Adam Moffett
• [asterisk-users] asterisk distributions   Adam Moffett
• [asterisk-users] asterisk distributions   Adam Moffett
• [asterisk-users] Asterisk Version 1.8.9.2 Question About SIP/SRTP/TLS   Larry Moore
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Tony Mountifield
• [asterisk-users] How does format_mp3 work?   Tony Mountifield
• [asterisk-users] FXS hangup issues   Richard Mudgett
• [asterisk-users] Junghanns QuadBri install help   Richard Mudgett
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Richard Mudgett
• [asterisk-users] How to Send SMS on SS7 DChannel-16(Signaling Channel)   Richard Mudgett
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Richard Mudgett
• [asterisk-users] Block Collect Calls on ISDN trunk   Richard Mudgett
• [asterisk-users] Block Collect Calls on ISDN trunk   Richard Mudgett
• [asterisk-users] Dahdi Answer a Call On ringing State.   Richard Mudgett
• [asterisk-users] Block Collect Calls on ISDN trunk   Richard Mudgett
• [asterisk-users] Block Collect Calls on ISDN trunk   Richard Mudgett
• [asterisk-users] Park and PARKINGDYNAMIC   Richard Mudgett
• [asterisk-users] Block Collect Calls on ISDN trunk   Richard Mudgett
• [asterisk-users] Rejecting transfers to in-use parking spaces   Richard Mudgett
• [asterisk-users] Rejecting transfers to in-use parking spaces   Richard Mudgett
• [asterisk-users] Getting one way audio even NAT is configured   Ahmed Munir
• [asterisk-users] Getting one way audio even NAT is configured   Ahmed Munir
• [asterisk-users] Getting Ulimit Message after restart asterisk service   Ahmed Munir
• [asterisk-users] Getting Ulimit Message after restart asterisk service   Ahmed Munir
• [asterisk-users] Getting Ulimit Message after restart asterisk service   Ahmed Munir
• [asterisk-users] Phone Inventory   Muro, Sam
• [asterisk-users] Phone Inventory   Muro, Sam
• [asterisk-users] Replicating SIP registration Info between active to standby   Muro, Sam
• [asterisk-users] Replicating SIP registration Info between active to standby   Muro, Sam
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   Tim Nelson
• [asterisk-users] Question for the group   Tim Nelson
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   Tim Nelson
• [asterisk-users] T38 faxing - UDPTL creation failed   Danny Nicholas
• [asterisk-users] T38 faxing - UDPTL creation failed   Danny Nicholas
• [asterisk-users] T38 faxing - UDPTL creation failed   Danny Nicholas
• [asterisk-users] [asterisk-dev] How to play audio file in background in dialplan?   Danny Nicholas
• [asterisk-users] Recording the follow-me calls‏   Danny Nicholas
• [asterisk-users] Queuemember status before calling the Queue command   Danny Nicholas
• [asterisk-users] [asterisk-dev] How to play audio file in background in dialplan?   Danny Nicholas
• [asterisk-users] How to play audio file in background in dialplan?   Danny Nicholas
• [asterisk-users] How to play audio file in background in dialplan?   Danny Nicholas
• [asterisk-users] Asterisk 10 and DUNDi, Extended Support?   Danny Nicholas
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Danny Nicholas
• [asterisk-users] Playback with noanswer in AGI   Danny Nicholas
• [asterisk-users] Custom extension: dial a queue   Danny Nicholas
• [asterisk-users] Custom extension: dial a queue   Danny Nicholas
• [asterisk-users] BETTER_BACKTRACES   Danny Nicholas
• [asterisk-users] MixMonitor and ChanSpy   Danny Nicholas
• [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles   Danny Nicholas
• [asterisk-users] Asterisk V/s FreeSwitch   Danny Nicholas
• [asterisk-users] Polycom firmware 4.0.1 and paging   Danny Nicholas
• [asterisk-users] "conferenced" transfers   Danny Nicholas
• [asterisk-users] "conferenced" transfers   Danny Nicholas
• [asterisk-users] Forwarding queue to remote agent over PSTN   Danny Nicholas
• [asterisk-users] Block Collect Calls on ISDN trunk   Danny Nicholas
• [asterisk-users] Block Collect Calls on ISDN trunk   Danny Nicholas
• [asterisk-users] Block Collect Calls on ISDN trunk   Danny Nicholas
• [asterisk-users] Block Collect Calls on ISDN trunk   Danny Nicholas
• [asterisk-users] asterisk network connections   Danny Nicholas
• [asterisk-users] Park and PARKINGDYNAMIC   Danny Nicholas
• [asterisk-users] Define custom vm-login sound file per VM context?   Danny Nicholas
• [asterisk-users] Define custom vm-login sound file per VM context?   Danny Nicholas
• [asterisk-users] Define custom vm-login sound file per VM context?   Danny Nicholas
• [asterisk-users] Praking lot issues.   Danny Nicholas
• [asterisk-users] Praking lot issues.   Danny Nicholas
• [asterisk-users] AGI: blocking script until playback complete   Danny Nicholas
• [asterisk-users] AGI: blocking script until playback complete   Danny Nicholas
• [asterisk-users] codec mismatch on channel   Danny Nicholas
• [asterisk-users] codec mismatch on channel   Danny Nicholas
• [asterisk-users] AGI: blocking script until playback complete   Danny Nicholas
• [asterisk-users] Phone Inventory   Danny Nicholas
• [asterisk-users] Phone Inventory   Danny Nicholas
• [asterisk-users] Phone Inventory   Danny Nicholas
• [asterisk-users] View Uniqueid of Active Calls   Danny Nicholas
• [asterisk-users] Transfer to fax   Danny Nicholas
• [asterisk-users] CDR Analyzer/Queue stats reporting   Danny Nicholas
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Danny Nicholas
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Danny Nicholas
• [asterisk-users] Getting Ulimit Message after restart asterisk service   Danny Nicholas
• [asterisk-users] asterisk distributions   Danny Nicholas
• [asterisk-users] Getting Ulimit Message after restart asterisk service   Danny Nicholas
• [asterisk-users] asterisk distributions   Danny Nicholas
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   Darren Nickerson
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   Darren Nickerson
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   Darren Nickerson
• [asterisk-users] Phone Inventory   Dale Noll
• [asterisk-users] Phone Inventory   Dale Noll
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   John Novack
• [asterisk-users] Asterisk 10.0 Realtime   Andrew Nowrot
• [asterisk-users] asterisk 10.0 realtime   Andrew Nowrot
• [asterisk-users] codec mismatch on channel   Andrew Nowrot
• [asterisk-users] Asterisk and realtime   Andrew Nowrot
• [asterisk-users] Custom extension: dial a queue   Alejandro Cabrera Obed
• [asterisk-users] Custom extension: dial a queue   Alejandro Cabrera Obed
• [asterisk-users] Custom extension: dial a queue   Alejandro Cabrera Obed
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Olivier
• [asterisk-users] Automatic Number Identification and anonymous calls   Olivier
• [asterisk-users] SIP hardware phones   Olivier
• [asterisk-users] Automatic Number Identification and anonymous calls   Olivier
• [asterisk-users] SIP hardware phones   Olivier
• [asterisk-users] Automatic Number Identification and anonymous calls   Olivier
• [asterisk-users] Automatic Number Identification and anonymous calls   Olivier
• [asterisk-users] SIP hardware phones   Olivier
• [asterisk-users] How to implement outlook popup   Olivier
• [asterisk-users] How to implement outlook popup   Olivier
• [asterisk-users] OT - Which cheap ISDN phone   Olivier
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   Olivier
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   Olivier
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   Olivier
• [asterisk-users] How to receive SMS ?   Olivier
• [asterisk-users] How to receive SMS ?   Olivier
• [asterisk-users] R: Asterisk V/s FreeSwitch   Alexandru Oniciuc
• [asterisk-users] pstn bridge to asterisk - phones connected to pstn stop ringing when asterisk answers   Gaurav P
• [asterisk-users] Driver for TOR3E ( Govarion ).   Richard Palmeron
• [asterisk-users] [asterisk-dev] How to play audio file in background in dialplan?   Yaroslav Panych
• [asterisk-users] [asterisk-dev] How to play audio file in background in dialplan?   Yaroslav Panych
• [asterisk-users] How to play audio file in background in dialplan?   Yaroslav Panych
• [asterisk-users] How to play audio file in background in dialplan?   Yaroslav Panych
• [asterisk-users] How to play audio file in background in dialplan?   Yaroslav Panych
• [asterisk-users] How to play audio file in background in dialplan?   Yaroslav Panych
• [asterisk-users] Multiprocess Asterisk   Yaroslav Panych
• [asterisk-users] Praking lot issues.   Jason Parker
• [asterisk-users] Streaming musiconhold via mpg123   Jason Parker
• [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up   Jason Parker
• [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up   Jason Parker
• [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up   Jason Parker
• [asterisk-users] SIP hardware phones   Jason Parks
• [asterisk-users] SIP hardware phones   Jason W. Parks
• [asterisk-users] SIP hardware phones   Jason W. Parks
• [asterisk-users] SIP hardware phones   Jason W. Parks
• [asterisk-users] SIP hardware phones   Jason W. Parks
• [asterisk-users] SER Still recommended for large installs?   Jason W. Parks
• [asterisk-users] SIP hardware phones   Roberto Piola
• [asterisk-users] CA Issued Certificates / TLS + SRTP   Daniel Pocock
• [asterisk-users] CA Issued Certificates / TLS + SRTP   Daniel Pocock
• [asterisk-users] Binding to 0.0.0.0 a security risk?   Daniel Pocock
• [asterisk-users] FXS hangup issues   Ari Pollak
• [asterisk-users] Queuemember status before calling the Queue command   Nathan Pryor
• [asterisk-users] Queuemember status before calling the Queue command   Nathan Pryor
• [asterisk-users] Your buddy Edwin needs your help!   Edwin Quijada
• [asterisk-users] Playback with noanswer in AGI   Zohair Raza
• [asterisk-users] Playback with noanswer in AGI   Zohair Raza
• [asterisk-users] Playback with noanswer in AGI   Zohair Raza
• [asterisk-users] Playback with noanswer in AGI   Zohair Raza
• [asterisk-users] Playback with noanswer in AGI   Zohair Raza
• [asterisk-users] Playback with noanswer in AGI   Zohair Raza
• [asterisk-users] Asterisk V/s FreeSwitch   Zohair Raza
• [asterisk-users] Asterisk V/s FreeSwitch   Zohair Raza
• [asterisk-users] Asterisk V/s FreeSwitch   Zohair Raza
• [asterisk-users] Is there any way to make call fail after # of rings?   Zohair Raza
• [asterisk-users] how many UDP ports is required for 1 call   Zohair Raza
• [asterisk-users] AGI: blocking script until playback complete   Zohair Raza
• [asterisk-users] AGI: blocking script until playback complete   Zohair Raza
• [asterisk-users] AGI: blocking script until playback complete   Zohair Raza
• [asterisk-users] 10.2.0-rc2: permitted contact can't register.   Zohair Raza
• [asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link (Dustin fails)   Mc GRATH Ricardo
• [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug   JR Richardson
• [asterisk-users] Garbled voicemail   Dan Ritter
• [asterisk-users] Virtual Server   Robert-IPhone
• [asterisk-users] New router, registration problems   Robert-IPhone
• [asterisk-users] What packages are required to getcdr_adaptive_odbc to be compiled in Asterisk?   Rebecca Robinson
• [asterisk-users] Callmanager 4 Asterisk Malformed/Missing URL   Rebecca Robinson
• [asterisk-users] checking if a phone number is UP   Rebecca Robinson
• [asterisk-users] Dialing multiple numbers   Carlos Rojas
• [asterisk-users] Dialing multiple numbers   Carlos Rojas
• [asterisk-users] Virtual Server   Carlos Rojas
• [asterisk-users] Postgresql in Asterisk   Jonathan Rose
• [asterisk-users] Problem with libpri / asterisk   Nicolas Ross
• [asterisk-users] Problem with libpri / asterisk   Nicolas Ross
• [asterisk-users] Define custom vm-login sound file per VM context?   Todd Routhier
• [asterisk-users] Define custom vm-login sound file per VM context?   Todd Routhier
• [asterisk-users] Define custom vm-login sound file per VM context?   Todd Routhier
• [asterisk-users] Define custom vm-login sound file per VM context?   Todd Routhier
• [asterisk-users] Define custom vm-login sound file per VM context?   Todd Routhier
• [asterisk-users] Disconnect after 12 seconds w/Cisco 303g Phones   Todd Routhier
• [asterisk-users] TE410P (1st) without cables always green   Shaun Ruffell
• [asterisk-users] TE410P (1st) without cables always green   Shaun Ruffell
• [asterisk-users] TE410P (1st) without cables always green   Shaun Ruffell
• [asterisk-users] Garbled voicemail   Ruben Rögels
• [asterisk-users] DECT with handover   Paulo Santos
• [asterisk-users] Block Collect Calls on ISDN trunk   Rafael dos Santos Saraiva
• [asterisk-users] Block Collect Calls on ISDN trunk   Rafael dos Santos Saraiva
• [asterisk-users] Block Collect Calls on ISDN trunk   Rafael dos Santos Saraiva
• [asterisk-users] Block Collect Calls on ISDN trunk   Rafael dos Santos Saraiva
• [asterisk-users] Block Collect Calls on ISDN trunk   Rafael dos Santos Saraiva
• [asterisk-users] Block Collect Calls on ISDN trunk   Rafael dos Santos Saraiva
• [asterisk-users] Block Collect Calls on ISDN trunk   Rafael dos Santos Saraiva
• [asterisk-users] Fwd: Block Collect Calls on ISDN trunk   Rafael dos Santos Saraiva
• [asterisk-users] Asterisk version that support Database Configuration   C. Savinovich
• [asterisk-users] Question for the group   C. Savinovich
• [asterisk-users] chan_capi audio weirdness   Armin Schindler
• [asterisk-users] Call holding with chan_capi   Armin Schindler
• [asterisk-users] chan_capi audio weirdness   Armin Schindler
• [asterisk-users] Asterisk V/s FreeSwitch   Stefan Schmidt
• [asterisk-users] Asterisk V/s FreeSwitch   Stefan Schmidt
• [asterisk-users] Asterisk V/s FreeSwitch   Stefan Schmidt
• [asterisk-users] Asterisk V/s FreeSwitch   Stefan Schmidt
• [asterisk-users] Garbled voicemail   Stefan Schmidt
• [asterisk-users] p-associated-uri in 200OK   Stefan Schmidt
• [asterisk-users] Getting one way audio even NAT is configured   Warren Selby
• [asterisk-users] Calling a group of phones and force the speaker   Warren Selby
• [asterisk-users] Park() ignores 'r' option which should disable music on hold in favour of ringing tone   Warren Selby
• [asterisk-users] Streaming musiconhold via mpg123   Warren Selby
• [asterisk-users] Asterisk 1.8.x app_meetme.so   Warren Selby
• [asterisk-users] Quick bash tip for finding free SIP extensions from your sip.conf   Kyle Sexton
• [asterisk-users] Quick bash tip for finding free SIP extensions from your sip.conf   Kyle Sexton
• [asterisk-users] Set(CALLERID(name)) when incoming call is "anonymous"   Kevin Shanahan
• [asterisk-users] Set(CALLERID(name)) when incoming call is "anonymous"   Kevin Shanahan
• [asterisk-users] Router that support Asterisk   James Sharp
• [asterisk-users] Virtual Server   James Sharp
• [asterisk-users] OT - "T.38 unreliable on a LAN" : truth or obscurantism ?   James Sharp
• [asterisk-users] Headset Options   Justin Sherrill
• [asterisk-users] How to receive SMS ?   Andreas Sikkema
• [asterisk-users] Is this doable?   Aurimas Skirgaila
• [asterisk-users] amd detect answering machine   Aurimas Skirgaila
• [asterisk-users] checking if a phone number is UP   Aurimas Skirgaila
• [asterisk-users] checking if a phone number is UP   Aurimas Skirgaila
• [asterisk-users] Asterisk auto-dial out a SIP .   Keith Sloan
• [asterisk-users] TCE400P diagnostic messages   Chris Sohns
• [asterisk-users] India Pune Pri call problem   Unai Solaguren
• [asterisk-users] India Pune Pri call problem   Unai Solaguren
• [asterisk-users] Spurious DTMF recognition problems.   Antonio Goméz Soto
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Arthur Stanfield
• [asterisk-users] What packages are required to get cdr_adaptive_odbc to be compiled in Asterisk?   A J Stiles
• [asterisk-users] Calling a group of phones and force the speaker   A J Stiles
• [asterisk-users] Is there any way to make call fail after # of rings?   A J Stiles
• [asterisk-users] dahdi and digium debian package   A J Stiles
• [asterisk-users] dahdi and digium debian package   A J Stiles
• [asterisk-users] Trunking betweeb two Asterisk System   A J Stiles
• [asterisk-users] Asterisk auto-dial out a SIP .   A J Stiles
• [asterisk-users] Park() ignores 'r' option which should disable music on hold in favour of ringing tone   James Stocks
• [asterisk-users] Recording the follow-me calls   Torintino T
• [asterisk-users] Recording the follow-me calls‏   Torintino T
• [asterisk-users] Hangupcause on DAHDI 2.4.9-svn-r9328 channels - Asterisk 1.4.36   Administrator TOOTAI
• [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles   Administrator TOOTAI
• [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles   Administrator TOOTAI
• [asterisk-users] How to receive SMS ?   Administrator TOOTAI
• [asterisk-users] How to receive SMS ?   Administrator TOOTAI
• [asterisk-users] Multiprocess Asterisk   M Takahashi
• [asterisk-users] Multiprocess Asterisk   M Takahashi
• [asterisk-users] read digits during recording / DTMF in conference?   Kingsley Tart
• [asterisk-users] read digits during recording / DTMF in conference?   Kingsley Tart
• [asterisk-users] Asterisk 1.8.9.2 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 10.1.2 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 10.1.1 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.9.1 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.9.3 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 10.1.3 Now Available   Asterisk Development Team
• [asterisk-users] View Uniqueid of Active Calls   Alper Tekinalp
• [asterisk-users] Correct call duration when transfer a call   Alper Tekinalp
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Troy Telford
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Troy Telford
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Troy Telford
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Troy Telford
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Troy Telford
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Troy Telford
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Troy Telford
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Troy Telford
• [asterisk-users] AGI with wrong ${AGISTATUS} Value   Antônio Theóphilo
• [asterisk-users] [SPAM] - FW by postmaster at datavox.co.uk : Delivery Status Notification (Failure) - Email found in subject   Andrew Thomas
• [asterisk-users] Headset Options   Brynjolfur Thorvardsson
• [asterisk-users] Asterisk V/s FreeSwitch   Brynjolfur Thorvardsson
• [asterisk-users] Asterisk V/s FreeSwitch   Brynjolfur Thorvardsson
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   Brynjolfur Thorvardsson
• [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP   Brynjolfur Thorvardsson
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   Brynjolfur Thorvardsson
• [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP   Brynjolfur Thorvardsson
• [asterisk-users] Stumped trying to find AGI script on my server .....   Brynjolfur Thorvardsson
• [asterisk-users] Stumped trying to find AGI script on my server ..... solved (I think)   Brynjolfur Thorvardsson
• [asterisk-users] Nat=route ???   Brynjolfur Thorvardsson
• [asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?   Christoph Timm
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Steve Totaro
• [asterisk-users] Polycom firmware 4.0.1 and paging   Gord Urquhart
• [asterisk-users] SIP hardware phones   Vieri
• [asterisk-users] SIP trunk audio bad but is OK again after SIP re-registration   Vieri
• [asterisk-users] SIP hardware phones   Vieri
• [asterisk-users] SIP hardware phones   Vieri
• [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP   Vieri
• [asterisk-users] distributed queue information over several Asterisk nodes   Vieri
• [asterisk-users] Stuck DAHDI Lines   Ryan Wagoner
• [asterisk-users] Major changes between 1.4/1.6.1.8/10?   Chad Wallace
• [asterisk-users] Call queuing behavior   Chad Wallace
• [asterisk-users] T.38 Incoming Fax Problem   Ken Wells
• [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2   Karsten Wemheuer
• [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2   Karsten Wemheuer
• [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2   Karsten Wemheuer
• [asterisk-users] Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2   Karsten Wemheuer
• [asterisk-users] Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2   Karsten Wemheuer
• [asterisk-users] Calling from extension that I don't create   Eko Kukuh Wibowo
• [asterisk-users] How to play audio file in background in dialplan?   Eric Wieling
• [asterisk-users] Help with Codes and Polycom Phones   Eric Wieling
• [asterisk-users] CEL ODBC Issues Asterisk 1.8.9.2   Eric Wieling
• [asterisk-users] Polycom firmware 4.0.1 and paging   Eric Wieling
• [asterisk-users] Dahdi Answer a Call On ringing State.   Eric Wieling
• [asterisk-users] How to receive SMS ?   Eric Wieling
• [asterisk-users] Dahdi Answer a Call On ringing State.   Eric Wieling
• [asterisk-users] AGI: blocking script until playback complete   Eric Wieling
• [asterisk-users] codec mismatch on channel   Eric Wieling
• [asterisk-users] AGI: blocking script until playback complete   Eric Wieling
• [asterisk-users] Phone Inventory   Eric Wieling
• [asterisk-users] cell mysql odbc support   Eric Wieling
• [asterisk-users] Alphanumeric DTMF !?   Eric Wieling
• [asterisk-users] Same provider - IAX sounds bad, SIP sounds great   Eric Wieling
• [asterisk-users] Define custom vm-login sound file per VM context?   Johan Wilfer
• [asterisk-users] Matching asterisk PBX cdrs to Telco's Trunk CDR's   Shaun Wingrin
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Hans Witvliet
• [asterisk-users] SIP hardware phones   Hans Witvliet
• [asterisk-users] New router, registration problems   David Woodfall
• [asterisk-users] New router, registration problems   David Woodfall
• [asterisk-users] New router, registration problems   David Woodfall
• [asterisk-users] New router, registration problems   David Woodfall
• [asterisk-users] Question for the group   James Wystead
• [asterisk-users] Question for the group   James Wystead
• [asterisk-users] dahdi timing   Mert Yazgart
• [asterisk-users] No valid transports available, falling back to 'udp'.   Michael L. Young
• [asterisk-users] What packages are required to getcdr_adaptive_odbc to be compiled in Asterisk?   Bryant Zimmerman
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   Bryant Zimmerman
• [asterisk-users] BETTER_BACKTRACES   Bryant Zimmerman
• [asterisk-users] Asterisk V/s FreeSwitch   Bryant Zimmerman
• [asterisk-users] Asterisk V/s FreeSwitch   Bryant Zimmerman
• [asterisk-users] Asterisk V/s FreeSwitch   Bryant Zimmerman
• [asterisk-users] Asterisk V/s FreeSwitch   Bryant Zimmerman
• [asterisk-users] Stuck DAHDI Lines   Bryant Zimmerman
• [asterisk-users] checking if a phone number is UP   Bryant Zimmerman
• [asterisk-users] Asterisk V/s FreeSwitch   Bryant Zimmerman
• [asterisk-users] checking if a phone number is UP   Bryant Zimmerman
• [asterisk-users] checking if a phone number is UP   Bryant Zimmerman
• [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.   Bryant Zimmerman
• [asterisk-users] SIP hardware phones   Bryant Zimmerman
• [asterisk-users] Log SRC & DST IP address and PDD   Bryant Zimmerman
• [asterisk-users] Park and PARKINGDYNAMIC   Bryant Zimmerman
• [asterisk-users] Park and PARKINGDYNAMIC   Bryant Zimmerman
• [asterisk-users] Praking lot issues.   Bryant Zimmerman
• [asterisk-users] Praking lot issues.   Bryant Zimmerman
• [asterisk-users] Praking lot issues.   Bryant Zimmerman
• [asterisk-users] Praking lot issues.   Bryant Zimmerman
• [asterisk-users] Rejecting transfers to in-use parking spaces   Bryant Zimmerman
• [asterisk-users] dahdi and digium debian package   ml asterisk
• [asterisk-users] better timing source for an asterisk gateway   ml asterisk
• [asterisk-users] Call Completion   bakko
• [asterisk-users] Call Completion   bakko
• [asterisk-users] Set T38 protocol   bakko
• [asterisk-users] Trunking betweeb two Asterisk System   bakko
• [asterisk-users] read digits during recording / DTMF in conference?   virendra bhati
• [asterisk-users] Can someone tell me what is this issue ?   virendra bhati
• [asterisk-users] Asterisk V/s FreeSwitch   virendra bhati
• [asterisk-users] Asterisk V/s FreeSwitch   virendra bhati
• [asterisk-users] Asterisk V/s FreeSwitch   virendra bhati
• [asterisk-users] Asterisk V/s FreeSwitch   virendra bhati
• [asterisk-users] India Pune Pri call problem   virendra bhati
• [asterisk-users] India Pune Pri call problem   virendra bhati
• [asterisk-users] India Pune Pri call problem   virendra bhati
• [asterisk-users] how many UDP ports is required for 1 call   virendra bhati
• [asterisk-users] how many UDP ports is required for 1 call   virendra bhati
• [asterisk-users] how many UDP ports is required for 1 call   virendra bhati
• [asterisk-users] how many UDP ports is required for 1 call   virendra bhati
• [asterisk-users] how many UDP ports is required for 1 call   virendra bhati
• [asterisk-users] how many UDP ports is required for 1 call   virendra bhati
• [asterisk-users] Where can I find some good examples of listening to AMI events via PHP & how to listen to a specific event?   virendra bhati
• [asterisk-users] Best CRM for Asterisk   virendra bhati
• [asterisk-users] runtime codec selection   virendra bhati
• [asterisk-users] cell mysql odbc support   cegadsl
• [asterisk-users] cell mysql odbc support   cegadsl
• [asterisk-users] cell mysql odbc support   cegadsl
• [asterisk-users] dahdi and digium debian package   cfh
• [asterisk-users] Should you "ever" use nat=no?   sean darcy
• [asterisk-users] Should you "ever" use nat=no?   sean darcy
• [asterisk-users] Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network?   sean darcy
• [asterisk-users] 10.2.0-rc2: permitted contact can't register.   sean darcy
• [asterisk-users] 10.2.0-rc2: permitted contact can't register.   sean darcy
• [asterisk-users] Subscribe Problem - Zombie Channel   dotnetdub
• [asterisk-users] Weird IPs in Fail2ban list   dotnetdub
• [asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link   Dustin fails
• [asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link   Dustin fails
• [asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link   Dustin fails
• [asterisk-users] Router that support Asterisk   bilal ghayyad
• [asterisk-users] Asterisk version that support Database Configuration   bilal ghayyad
• [asterisk-users] Web and Email Chat   bilal ghayyad
• [asterisk-users] DECT with handover   giovanni.v
• [asterisk-users] read digits during recording / DTMF in conference?   isrlgb at gmail.com
• [asterisk-users] Fwd: Re: Asterisk CLI unresponsive   isrlgb at gmail.com
• [asterisk-users] Asterisk V/s FreeSwitch   isrlgb at gmail.com
• [asterisk-users] Reading second rdnis   isrlgb at gmail.com
• [asterisk-users] "conferenced" transfers   isrlgb at gmail.com
• [asterisk-users] Streaming musiconhold via mpg123   isrlgb at gmail.com
• [asterisk-users] Streaming musiconhold via mpg123   isrlgb at gmail.com
• [asterisk-users] Junghanns QuadBri install help   joekane at gmail.com
• [asterisk-users] Junghanns QuadBri install help   joekane at gmail.com
• [asterisk-users] Dahdi, PRI and all circuits are busy now   ik
• [asterisk-users] Calling a group of phones and force the speaker   Brian ipt
• [asterisk-users] Polycom firmware 4.0.1 and paging   Brian ipt
• [asterisk-users] Should you "ever" use nat=no?   Brian ipt
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   asterisk jobs
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   asterisk jobs
• [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?   asterisk jobs
• [asterisk-users] Question for the group   asterisk jobs
• [asterisk-users] Question for the group   asterisk jobs
• [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?   asterisk jobs
• [asterisk-users] Weird IPs in Fail2ban list   asterisk jobs
• [asterisk-users] Weird IPs in Fail2ban list   asterisk jobs
• [asterisk-users] Weird IPs in Fail2ban list   asterisk jobs
• [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?   asterisk jobs
• [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?   asterisk jobs
• [asterisk-users] India Pune Pri call problem   mahesh katta
• [asterisk-users] Best CRM for Asterisk   mahesh katta
• [asterisk-users] asterisk-users at lists.digium.com Nacha Alert ID10416   asterisk-users at lists.digium.com
• [asterisk-users] a1zax th99sh   umesh maharjan
• [asterisk-users] Asterisk-users caller ID   motty.cruz
• [asterisk-users] Asterisk auto-dial out a SIP .   upendra
• [asterisk-users] Asterisk 1.8.x app_meetme.so   [Digital^Dude] ®
• [asterisk-users] Asterisk 1.8.x app_meetme.so   [Digital^Dude] ®
• [asterisk-users] Asterisk 1.8.x app_meetme.so   [Digital^Dude] ®
• [asterisk-users] Asterisk 1.8.x app_meetme.so   [Digital^Dude] ®

Last message date: Wed Feb 29 18:43:29 CST 2012
Archived on: Wed Feb 29 18:52:32 CST 2012

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