November 2007 Archives by date

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Starting: Thu Nov 1 00:48:57 CDT 2007
Ending: Fri Nov 30 23:42:43 CST 2007
Messages: 1778

• [asterisk-users] hostname in MySQL CDR records   Tilghman Lesher
• [asterisk-users] hostname in MySQL CDR records   Steve Edwards
• [asterisk-users] G.729 required for IP<--->TDM<--->IP   satish patel
• [asterisk-users] flooded by "Maximum trunk data space exceeded" messages   Louis-David Mitterrand
• [asterisk-users] Druid   Alex Epshteyn
• [asterisk-users] SER with Asterisk intergration   satish patel
• [asterisk-users] Spam Filter News   Thomas Kenyon
• [asterisk-users] SparkLan WVPR-100 Wireless Handset.   Thomas Kenyon
• [asterisk-users] SER with Asterisk intergration   Arun Kumar
• [asterisk-users] Druid   Alan Lord
• [asterisk-users] Autodialing   Bhrugu Mehta
• [asterisk-users] Call Failed   Doug Lytle
• [asterisk-users] Video Call   voip Server asterisk
• [asterisk-users] Druid   Dean Collins
• [asterisk-users] SER with Asterisk intergration   satish patel
• [asterisk-users] SER with Asterisk intergration   J. Oquendo
• [asterisk-users] Mobile phone codecs ...   Benny Amorsen
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   Antoine Megalla
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   Antoine Megalla
• [asterisk-users] Mobile phone codecs ...   Steve Kennedy
• [asterisk-users] Spam Filter News   Jared Smith
• [asterisk-users] Connection astrisk to a RAS (portmaster)   Jared Smith
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   satish patel
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   satish patel
• [asterisk-users] Druid   Juan Sandro
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   SIP
• [asterisk-users] Connection astrisk to a RAS (portmaster)   Nicolas Ross
• [asterisk-users] Help   Jarga Jallow
• [asterisk-users] AEL2 and Callbacks   Richard Lyman
• [asterisk-users] Parking speed   Kelly Opal
• [asterisk-users] RTP Read too short   John Faubion
• [asterisk-users] Help   Chris 'Xenon' Hanson
• [asterisk-users] Help   Jarga Jallow
• [asterisk-users] libpri & tie line vs trunk   Michelle Dupuis
• [asterisk-users] Digium Vs sangoma Hradware   Stephen Bosch
• [asterisk-users] PRI over T1 calls dropping, cause 100   Matthew Fredrickson
• [asterisk-users] PRI commands missing...   Matthew Fredrickson
• [asterisk-users] issues with downloads.digium.com   Matthew Fredrickson
• [asterisk-users] DST   Joe Acquisto
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   ram
• [asterisk-users] AEL2 and Callbacks   Douglas Garstang
• [asterisk-users] DST   Kevin P. Fleming
• [asterisk-users] DST   Philipp Kempgen
• [asterisk-users] DST   Dave Fullerton
• [asterisk-users] DST   BJ Weschke
• [asterisk-users] PRI over T1 calls dropping, cause 100   Michelle Dupuis
• [asterisk-users] PRI commands missing...   Tzafrir Cohen
• [asterisk-users] Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Outgoing PRI CID?   mail-lists
• [asterisk-users] AsteriskNOW and TDM800P   Rafael Canchola
• [asterisk-users] Outgoing PRI CID?   Anciso, Roy
• [asterisk-users] Chanspy attaching to a caller ID entry?   Martin Smith
• [asterisk-users] DST   Joe Acquisto
• [asterisk-users] Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Intercom with Snom phones   Zaheer Master
• [asterisk-users] DST   Turbo Fredriksson
• [asterisk-users] DST   Martin Smith
• [asterisk-users] AEL2 and Callbacks   Sean Bright
• [asterisk-users] DST   Joe Acquisto
• [asterisk-users] DST   BJ Weschke
• [asterisk-users] Outgoing PRI CID?   Eric "ManxPower" Wieling
• [asterisk-users] PRI debuging shows 'Ext: 0' (Was: Outgoing PRI CID?)   Turbo Fredriksson
• [asterisk-users] PRI debuging shows 'Ext: 0' (Was: Outgoing PRI CID?)   Eric "ManxPower" Wieling
• [asterisk-users] Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Costa Dinoteli
• [asterisk-users] Solved: Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Philipp Kempgen
• [asterisk-users] ring group containing external 10-digit numbers   Ryan Stille
• [asterisk-users] Polycom Park Button   Kelly Opal
• [asterisk-users] OpenSER for Asterisk Load balance   Edgar Guadamuz
• [asterisk-users] ring group containing external 10-digit numbers   Don Pobanz
• [asterisk-users] Two PRI setup questions   Lutgring, Sam
• [asterisk-users] Two PRI setup questions   Doug Lytle
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   Luki
• [asterisk-users] Intercom with Snom phones   Doug
• [asterisk-users] Two PRI setup questions   Erik Anderson
• [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?   Arpit Mehta
• [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?   Philipp Kempgen
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   ram
• [asterisk-users] OpenSER for Asterisk Load balance   ram
• [asterisk-users] __sip_xmit problem   Rilawich Ango
• [asterisk-users] zaptel.conf missing   kitti jaisong
• [asterisk-users] zaptel.conf missing   Rilawich Ango
• [asterisk-users] Two PRI setup questions   Tilghman Lesher
• [asterisk-users] zaptel.conf missing   Tzafrir Cohen
• [asterisk-users] asterisk as a gateway   Bincy K. Philip
• [asterisk-users] issues with downloads.digium.com   Tony Mountifield
• [asterisk-users] issues with downloads.digium.com   Michiel van Baak
• [asterisk-users] issues with downloads.digium.com   Atis Lezdins
• [asterisk-users] issues with downloads.digium.com   Jason White
• [asterisk-users] issues with downloads.digium.com   Anselm Martin Hoffmeister
• [asterisk-users] AEX800 (TDM800 Express) - not detected   Mark J Elkins
• [asterisk-users] asterisk as a gateway   Rizwan Hisham
• [asterisk-users] ring group containing external 10-digit numbers   Ryan Stille
• [asterisk-users] __sip_xmit problem   Rizwan Hisham
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Rizwan Hisham
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Rizwan Hisham
• [asterisk-users] res_mysql versus res_odbc   Tony Plack
• [asterisk-users] ring group containing external 10-digit numbers   David Gomillion
• [asterisk-users] ring group containing external 10-digit numbers   rennes
• [asterisk-users] Druid   Anthony Francis
• [asterisk-users] Polycom Park Button   Kyle Sexton
• [asterisk-users] ring group containing external 10-digit numbers   Ryan Stille
• [asterisk-users] Route an incoming call by ANI*DNIS   Dan Casey
• [asterisk-users] Route an incoming call by ANI*DNIS   BJ Weschke
• [asterisk-users] Druid   Dean Collins
• [asterisk-users] res_mysql versus res_odbc   Tilghman Lesher
• [asterisk-users] Asterisk as a SIP to SIP Gateway   Tomasz Zieleniewski
• [asterisk-users] res_mysql versus res_odbc   Tony Plack
• [asterisk-users] Jitterbuffer issues   Tony Plack
• [asterisk-users] SIP Channels   Asterisk
• [asterisk-users] RealTime register for iax DID   Alexandre: Nault
• [asterisk-users] Route an incoming call by ANI*DNIS   Dan Casey
• [asterisk-users] Route an incoming call by ANI*DNIS   Eric "ManxPower" Wieling
• [asterisk-users] Two PRI setup questions   Matthew Fredrickson
• [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?   Arpit Mehta
• [asterisk-users] steps for installing on Debian Etch   Baji Panchumarti
• [asterisk-users] Route an incoming call by ANI*DNIS   Tilghman Lesher
• [asterisk-users] ztdummy and BackGround   Tony Plack
• [asterisk-users] Route an incoming call by ANI*DNIS   Eric "ManxPower" Wieling
• [asterisk-users] ztdummy and BackGround   Eric "ManxPower" Wieling
• [asterisk-users] Off-Topic: add GSM codec to X-Lite   Alejandro Cabrera Obed
• [asterisk-users] Off-Topic: add GSM codec to X-Lite   Julio Tejera
• [asterisk-users] Off-Topic: add GSM codec to X-Lite   SIP
• [asterisk-users] Off-Topic: add GSM codec to X-Lite   Alejandro Cabrera Obed
• [asterisk-users] ztdummy and BackGround   Tony Plack
• [asterisk-users] ztdummy and BackGround   Atis Lezdins
• [asterisk-users] ztdummy and BackGround   Eric "ManxPower" Wieling
• [asterisk-users] sip show peers in 1.4.13   Jerry Geis
• [asterisk-users] ztdummy and BackGround   Jason Parker
• [asterisk-users] one way RTP using NAT   Hector Quiroz
• [asterisk-users] use dial plan passed arg value in C agi code   Arpit Mehta
• [asterisk-users] use dial plan passed arg value in C agi code   Eric "ManxPower" Wieling
• [asterisk-users] sip show peers in 1.4.13   Doug Lytle
• [asterisk-users] ztdummy and BackGround   Tony Plack
• [asterisk-users] sip show peers in 1.4.13   Tony Plack
• [asterisk-users] ztdummy and BackGround   Tony Plack
• [asterisk-users] use dial plan passed arg value in C agi code   James FitzGibbon
• [asterisk-users] Druid   Alex Epshteyn
• [asterisk-users] use dial plan passed arg value in C agi code   Eric "ManxPower" Wieling
• [asterisk-users] Druid   Dean Collins
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Costa Dinoteli
• [asterisk-users] sip show peers in 1.4.13   Tilghman Lesher
• [asterisk-users] use dial plan passed arg value in C agi code   James FitzGibbon
• [asterisk-users] Druid   Dean Collins
• [asterisk-users] use dial plan passed arg value in C agi code   Eric "ManxPower" Wieling
• [asterisk-users] res_mysql versus res_odbc   Philipp Kempgen
• [asterisk-users] use dial plan passed arg value in C agi code   Steve Edwards
• [asterisk-users] Druid   Alan Lord
• [asterisk-users] Everyone is busy/congested: IP Trunk   bilal ghayyad
• [asterisk-users] Asterisk SIP Channels Bridge   Asterisk
• [asterisk-users] RTP Read too short   John Faubion
• [asterisk-users] Asterisk SIP Channels Bridge   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk SIP Channels Bridge   Asterisk
• [asterisk-users] 3pcc/click to dial accounting   CSB
• [asterisk-users] one way RTP using NAT   srgqwerty
• [asterisk-users] Polycom Park Button   Lorentz Hinrichsen
• [asterisk-users] SparkLan WVPR-100 Wireless Handset.   Thomas Kenyon
• [asterisk-users] Jitterbuffer issues   Volker Sauer
• [asterisk-users] OT: Snom 300 losing config?   Mike Dent
• [asterisk-users] [Fwd: voicemail locked up Asterisk 1.4.13]   Lyle Giese
• [asterisk-users] Wifi handover/roaming   Luis Antonio Prata Barbosa
• [asterisk-users] Grandstream 488 and DTMF with a call file   Jerry Geis
• [asterisk-users] AEL2 and Callbacks   Sean Bright
• [asterisk-users] OpenSER for Asterisk Load balance   Edgar Guadamuz
• [asterisk-users] F3000 not connecting to Netgear routing   Dovid B
• [asterisk-users] Fax Problems with SpanDSP   Dovid B
• [asterisk-users] issues with downloads.digium.com   Dovid B
• [asterisk-users] Video Call   Bert Haverkamp
• [asterisk-users] F3000 not connecting to Netgear router   Dovid B
• [asterisk-users] Asterisk versions and H323   bilal ghayyad
• [asterisk-users] Video Call   Yann JOUANIN
• [asterisk-users] Asterisk as a SIP to SIP Gateway   Dovid B
• [asterisk-users] Asterisk versions and H323   Dovid B
• [asterisk-users] ztdummy and BackGround   Tzafrir Cohen
• [asterisk-users] Asterisk as a SIP to SIP Gateway   Tomasz Zieleniewski
• [asterisk-users] Restart when convenient   Doug Lytle
• [asterisk-users] Off-Topic: add GSM codec to X-Lite   Jaswinder Singh
• [asterisk-users] Restart when convenient   Eric "ManxPower" Wieling
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Jean-Yves Avenard
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Eric "ManxPower" Wieling
• [asterisk-users] Restart when convenient   Doug Lytle
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Jean-Yves Avenard
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Eric "ManxPower" Wieling
• [asterisk-users] Restart when convenient   Eric "ManxPower" Wieling
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Tzafrir Cohen
• [asterisk-users] Restart when convenient   Eric "ManxPower" Wieling
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Jean-Yves Avenard
• [asterisk-users] Restart when convenient   Turbo Fredriksson
• [asterisk-users] 7960 Queue Issue   Nick Brown
• [asterisk-users] Need Reference sites   Michael Davidson
• [asterisk-users] 7960 Queue Issue   Paul Hales
• [asterisk-users] Are the ATAs which can allow multiple extensions from one network connection?   Frank Church
• [asterisk-users] 7960 Queue Issue   Eric Merkel
• [asterisk-users] 7960 Queue Issue   Nick Brown
• [asterisk-users] Are the ATAs which can allow multiple extensions from one network connection?   Dave Bour
• [asterisk-users] 7960 Queue Issue   jadams at clearcasetechnology.com
• [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card - Update   Joseph Begumisa
• [asterisk-users] 7960 Queue Issue   Duncan Turnbull
• [asterisk-users] Fax Problems with SpanDSP   joakimsen at gmail.com
• [asterisk-users] asterisk-users Digest, Vol 40, Issue 5   Bincy K. Philip
• [asterisk-users] Dynamic Queue Members - Auto Logoff   Nick Brown
• [asterisk-users] crash   Rilawich Ango
• [asterisk-users] AGI connected to meetme conf gives Failed to write frame error   Chris Nestrud
• [asterisk-users] TE220 PCI express performnace   satish patel
• [asterisk-users] Video Call   voip Server asterisk
• [asterisk-users] Not Hearing hello-world Play   Jeng Yu
• [asterisk-users] TE220 PCI express performnace   Tzafrir Cohen
• [asterisk-users] Need Reference sites   Bhrugu Mehta
• [asterisk-users] RTP Read too short   Steve Davies
• [asterisk-users] asterisk-users Digest, Vol 40, Issue 5   Rizwan Hisham
• [asterisk-users] asterisk as a gateway   Bincy K. Philip
• [asterisk-users] How to disable Asterisk 407 Proxy Authentication Required Challenge response   Tomasz Zieleniewski
• [asterisk-users] How to delete voice mail messages?   voip crazy
• [asterisk-users] How to delete voice mail messages?   Michiel van Baak
• [asterisk-users] Need Reference sites   Steven
• [asterisk-users] Compatibility Issues with dell poweredge 195and TE110P card   Steven
• [asterisk-users] Which Variable???   Jeng Yu
• [asterisk-users] Dynamic Queue Members - Auto Logoff   Jason Adams
• [asterisk-users] Which Variable???   Eric "ManxPower" Wieling
• [asterisk-users] Which Variable???   Dave Fullerton
• [asterisk-users] Dynamic Queue Members - Auto Logoff   James FitzGibbon
• [asterisk-users] Kirk IP600/3 Wireless Server SIP config   Remco Barendse
• [asterisk-users] OT: Which SIP method to use for this specific behaviour ?   Olivier
• [asterisk-users] Parameters effect on the success registeration   bilal ghayyad
• [asterisk-users] OT: Which SIP method to use for this specificbehaviour ?   Steve Langstaff
• [asterisk-users] OT: Which SIP method to use for this specificbehaviour ?   Olivier
• [asterisk-users] Asterisk versions and H323   bilal ghayyad
• [asterisk-users] Which Variable???   Gordon Henderson
• [asterisk-users] Meetme - how to protect the conference?   Ondrej Valousek
• [asterisk-users] Problem with CDR userfield not being set   James Moore
• [asterisk-users] Meetme - how to protect the conference?   Darryl Dunkin
• [asterisk-users] Problem with CDR userfield not being set   Carlos Chavez
• [asterisk-users] Which Variable???   C F
• [asterisk-users] Free T1 Card?   Michael Collins
• [asterisk-users] Testcall   sistemas at zimax.com.ar
• [asterisk-users] Help: Static and dropped calls   Jarga Jallow
• [asterisk-users] Arbitrary limit on length of email address?   Alan Lord
• [asterisk-users] AEX800 (TDM800 Express) - not detected   Kevin P. Fleming
• [asterisk-users] Free T1 Card?   Kristian Kielhofner
• [asterisk-users] Arbitrary limit on length of email address?   Per Jessen
• [asterisk-users] Arbitrary limit on length of email address?   Alan Lord
• [asterisk-users] Problem with CDR userfield not being set   James Moore
• [asterisk-users] Meetme - how to protect the conference?   Atis Lezdins
• [asterisk-users] RTP Read too short   Drew Gibson
• [asterisk-users] Testcall   Moises Silva
• [asterisk-users] PRI dialout problem with some numbers...   Carlos Chavez
• [asterisk-users] Free T1 Card?   Michael Collins
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   cfh
• [asterisk-users] OT: Which SIP method to use for thisspecificbehaviour ?   Steve Langstaff
• [asterisk-users] Free T1 Card?   Guillermo Salas M.
• [asterisk-users] Free T1 Card?   Michael Joyner
• [asterisk-users] Queue Statistics reporting   Bob Pierce
• [asterisk-users] Please explain the correct LED color for B410P   asterisk-users at rogg.is
• [asterisk-users] Two B410P cards in one machine   asterisk-users at rogg.is
• [asterisk-users] Kirk IP600/3 Wireless Server SIP config   Luis Antonio Prata Barbosa
• [asterisk-users] Asterisk & OpenVZ   JR Richardson
• [asterisk-users] 1.4 SIP Jitter Buffer   Gregory Boehnlein
• [asterisk-users] 1.4 SIP Jitter Buffer   Luc Moreira
• [asterisk-users] PRI dialout problem with some numbers...   Alejandro Kauffmann
• [asterisk-users] Asterisk & OpenVZ   George Pajari
• [asterisk-users] Asterisk & OpenVZ   Tzafrir Cohen
• [asterisk-users] Linksys SPA-941 Unavailable   Kim Joung-il
• [asterisk-users] Asterisk and Grandstream both behind different NAT   Vivek Shrivastava
• [asterisk-users] Recording just first part of call?   Tony Mountifield
• [asterisk-users] Kirk IP600/3 Wireless Server SIP config   Benny Amorsen
• [asterisk-users] Recording just first part of call?   Anselm Martin Hoffmeister
• [asterisk-users] Asterisk and Grandstream both behind different NAT   ram
• [asterisk-users] Linksys SPA-941 Unavailable   ram
• [asterisk-users] Recording just first part of call?   Martin Smith
• [asterisk-users] 7960 Queue Issue   Robert Lister
• [asterisk-users] How to delete voice mail messages?   Robert Lister
• [asterisk-users] Linksys SPA-941 Unavailable   joakimsen at gmail.com
• [asterisk-users] Testcall   sistemas at zimax.com.ar
• [asterisk-users] Queue Statistics reporting   Lenz
• [asterisk-users] MeetMe CPU resources   Carles Pina i Estany
• [asterisk-users] MeetMe CPU resources   Doug Lytle
• [asterisk-users] Testcall   Moises Silva
• [asterisk-users] MeetMe CPU resources   Tony Mountifield
• [asterisk-users] Queue Statistics Reporting   Russell Eden
• [asterisk-users] Everyone is busy/congested: IP Trunk   Baji Panchumarti
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Hans Feringa
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Eric "ManxPower" Wieling
• [asterisk-users] Testcall   sistemas at zimax.com.ar
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Jared Smith
• [asterisk-users] Asterisk 1.4 + Presence   Alejandro Cabrera Obed
• [asterisk-users] Help: Asterisk info   Jarga Jallow
• [asterisk-users] MeetMe CPU resources   Steve Edwards
• [asterisk-users] Sangoma S200 and Digium TDM400P together   Paulo Garcia
• [asterisk-users] Asterisk Help   Jarga Jallow
• [asterisk-users] MeetMe CPU resources   Steve Edwards
• [asterisk-users] 1.4 SIP Jitter Buffer   Gregory Boehnlein
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Anciso, Roy
• [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID   Stephen Bosch
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Dave Fullerton
• [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID   James FitzGibbon
• [asterisk-users] Asterisk Help   Doug
• [asterisk-users] Please explain the correct LED color for B410P   Matthew Fredrickson
• [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID   Stephen Bosch
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Alan Lord
• [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID   James FitzGibbon
• [asterisk-users] use dial plan passed arg value in C agi code   Arpit Mehta
• [asterisk-users] Pickup Command not working   Lutgring, Sam
• [asterisk-users] Mystery phone!   Administrator TOOTAI
• [asterisk-users] dtmf / misdn   Hans Witvliet
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   joakimsen at gmail.com
• [asterisk-users] MeetMe CPU resources   Carles Pina i Estany
• [asterisk-users] Mystery phone!   Jim Houser
• [asterisk-users] Help: Asterisk info   Lyle Giese
• [asterisk-users] Extracting custom headers from SIP REFER   CSB
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Anciso, Roy
• [asterisk-users] Everyone is busy/congested: IP Trunk   Vivek Shrivastava
• [asterisk-users] Sangoma S200 and Digium TDM400P together   joakimsen at gmail.com
• [asterisk-users] Sangoma S200 and Digium TDM400P together   joakimsen at gmail.com
• [asterisk-users] MeetMe CPU resources   joakimsen at gmail.com
• [asterisk-users] Mystery phone!   Peter Lindquist
• [asterisk-users] use dial plan passed arg value in C agi code   Steve Edwards
• [asterisk-users] wifi   Michael Graves
• [asterisk-users] dtmf / misdn   Josh Richards
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Hans Feringa
• [asterisk-users] Sangoma S200 and Digium TDM400P together   Tzafrir Cohen
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Tzafrir Cohen
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Tzafrir Cohen
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Tzafrir Cohen
• [asterisk-users] detecting voltage on fxo   Paradise Dove
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Roi Stork
• [asterisk-users] grandstream troubles   Per Jessen
• [asterisk-users] Determination of billsec   CSB
• [asterisk-users] Call terminated with error message logged   Jason White
• [asterisk-users] MeetMe CPU resources   Tony Mountifield
• [asterisk-users] Extracting custom headers from SIP REFER   Johansson Olle E
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] Linksys SPA-941 Unavailable   Kim Joung-il
• [asterisk-users] Board configuration - specification or recommendation   Kim Joung-il
• [asterisk-users] What do you do to keep asterisk alive?   Tzafrir Cohen
• [asterisk-users] What do you do to keep asterisk alive?   Andrea Spadaccini
• [asterisk-users] Linksys SPA-941 Unavailable   Steve Davies
• [asterisk-users] What do you do to keep asterisk alive?   Tony Mountifield
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   J. Oquendo
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Andrea Spadaccini
• [asterisk-users] What do you do to keep asterisk alive?   Sajith T S
• [asterisk-users] What do you do to keep asterisk alive?   Tzafrir Cohen
• [asterisk-users] Linksys SPA-941 Unavailable   Patrick
• [asterisk-users] What do you do to keep asterisk alive?   J. Oquendo
• [asterisk-users] Audiocodes over Sat link. and delay   Justin Case
• [asterisk-users] MeetMe CPU resources   Carles Pina i Estany
• [asterisk-users] wifi   Luis Antonio Prata Barbosa
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] SIP: "To:" header?   Tony Mountifield
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Jared Smith
• [asterisk-users] Call terminated with error message logged   Jared Smith
• [asterisk-users] Linksys SPA-941 Unavailable   Kim Joung-il
• [asterisk-users] Linksys SPA-941 Unavailable   Kim Joung-il
• [asterisk-users] wifi   Baji Panchumarti
• [asterisk-users] Pickup Command not working   Robert Lister
• [asterisk-users] CDR on channel not posted   Thomas Stein
• [asterisk-users] Linksys SPA-941 Unavailable   Per Jessen
• [asterisk-users] CDR on channel not posted   Atis Lezdins
• [asterisk-users] What do you do to keep asterisk alive?   C F
• [asterisk-users] Pickup Command not working   Lutgring, Sam
• [asterisk-users] What do you do to keep asterisk alive?   Chris Bagnall
• [asterisk-users] detecting voltage on fxo   Eric "ManxPower" Wieling
• [asterisk-users] Polycom SoundStation VTX 1000 with Asterisk?   Alvin Austin
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] Determination of billsec   Steve Murphy
• [asterisk-users] accessing variables when using SIP vs. IAX   Jason Wolfe
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   joakimsen at gmail.com
• [asterisk-users] Linksys SPA-941 Unavailable   joakimsen at gmail.com
• [asterisk-users] wifi   joakimsen at gmail.com
• [asterisk-users] grandstream troubles   joakimsen at gmail.com
• [asterisk-users] Cisco phone 7911g restarts   Paul Lacatus
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   joakimsen at gmail.com
• [asterisk-users] grandstream troubles   Per Jessen
• [asterisk-users] Polycom SoundStation VTX 1000 with Asterisk?   Don Pobanz
• [asterisk-users] Determination of billsec   Doug
• [asterisk-users] Determination of billsec   CSB
• [asterisk-users] wifi   Michael Graves
• [asterisk-users] Video Call   Marek B
• [asterisk-users] What do you do to keep asterisk alive?   Matthew J. Roth
• [asterisk-users] Video Call   joakimsen at gmail.com
• [asterisk-users] Determination of billsec   joakimsen at gmail.com
• [asterisk-users] Little OT: Compilation of EICON driver, fails with capi errors   Stefan Guenther
• [asterisk-users] Board configuration - specification or recommendation   Josh Richards
• [asterisk-users] Determination of billsec   CSB
• [asterisk-users] Connection astrisk to a RAS (portmaster)   Nicolas Ross
• [asterisk-users] Pickup Command not working   Baji Panchumarti
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] What do you do to keep asterisk alive?   Kyle Sexton
• [asterisk-users] Pickup Command not working   Lutgring, Sam
• [asterisk-users] Autodialing   Dovid B
• [asterisk-users] Video Call   Gordon Henderson
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Michelle Dupuis
• [asterisk-users] Call terminated with error message logged   Jason White
• [asterisk-users] Pickup Command not working   Eric "ManxPower" Wieling
• [asterisk-users] Pickup Command not working   Baji Panchumarti
• [asterisk-users] Determination of billsec   CSB
• [asterisk-users] Pickup Command not working   Lacy Moore
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Roi Stork
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Michelle Dupuis
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Roi Stork
• [asterisk-users] DeStar finalist in Les Trophées du Libre 2007   Juan Manuel
• [asterisk-users] extensions.conf pattern match info   Eric "ManxPower" Wieling
• [asterisk-users] What do you do to keep asterisk alive?   Luki
• [asterisk-users] weird 185 secs timeout call problem   Tilghman Lesher
• [asterisk-users] weird 185 secs timeout call problem   Andre Quintaes
• [asterisk-users] Polycom IP601 (mac)-directory.xml changes don't update phone   Alvin Austin
• [asterisk-users] Polycom IP601 call parking   Alvin Austin
• [asterisk-users] Client lost on skinny   Paul Lacatus
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] extensions.conf pattern match info   Jason White
• [asterisk-users] Snom 320 with TDM02B and echo problems   voip crazy
• [asterisk-users] make h323 native transfer on stablished call   lumen
• [asterisk-users] dtmfmode RFC2833 and inband   Jerry Geis
• [asterisk-users] asterisk and installing chan_h323.so rpm   Bincy K. Philip
• [asterisk-users] Channel variables, any difference with SIP vs. IAX?   Jason Wolfe
• [asterisk-users] weird 185 secs timeout call problem   Steve Totaro
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Michelle Dupuis
• [asterisk-users] time on polycom 501   Jerry Geis
• [asterisk-users] 'a' extension   Peder at NetworkOblivion
• [asterisk-users] time on polycom 501   Alex Robar
• [asterisk-users] time on polycom 501   David Gomillion
• [asterisk-users] 'a' extension   James FitzGibbon
• [asterisk-users] 'a' extension   BJ Weschke
• [asterisk-users] 'a' extension   Vivek Shrivastava
• [asterisk-users] Channel variables, any difference with SIP vs. IAX?   Josh Richards
• [asterisk-users] SIP: "To:" header?   Johansson Olle E
• [asterisk-users] __sip_xmit problem   Dovid B
• [asterisk-users] Wifi handover/roaming   joakimsen at gmail.com
• [asterisk-users] Client lost on skinny   Dan Austin
• [asterisk-users] 7960 Queue Issue   joakimsen at gmail.com
• [asterisk-users] Asterisk 1.4 + Presence   joakimsen at gmail.com
• [asterisk-users] dtmf / misdn   Hans Witvliet
• [asterisk-users] Asterisk Help   joakimsen at gmail.com
• [asterisk-users] Client lost on skinny   Paul Lacatus
• [asterisk-users] Cisco IP Communicator with Asterisk   Anciso, Roy
• [asterisk-users] Asterisk and OBDC   Gilberto Nunes
• [asterisk-users] SIP: "To:" header?   Michael Joyner
• [asterisk-users] Client lost on skinny   Dan Austin
• [asterisk-users] Switchvox Space Requirements   Andres
• [asterisk-users] AST-2007-024 - Fallacious security advisory spread on the Internet involving buffer overflow in Zaptel's sethdlc application   The Asterisk Development Team
• [asterisk-users] Asterisk and OBDC   Maxi Belino
• [asterisk-users] AsteriskNOW - how to open SIP ports?   Zaheer K. Master
• [asterisk-users] Asterisk and OBDC   Gilberto Nunes Ferreira
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Eric Chamberlain
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Geoff Jacobs
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Philip Prindeville
• [asterisk-users] If caller id is null set to a specific number   Jon Weisman
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Mik Cheez
• [asterisk-users] If caller id is null set to a specific number   Michelle Dupuis
• [asterisk-users] Route an incoming call by ANI*DNIS   Jon Weisman
• [asterisk-users] If caller id is null set to a specific number   Doug Lytle
• [asterisk-users] If caller id is null set to a specific number   Jon Weisman
• [asterisk-users] Snom 320 with TDM02B and echo problems   Jason White
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Roi Stork
• [asterisk-users] Snom 320 with TDM02B and echo problems   Philipp Kempgen
• [asterisk-users] Snom 320 with TDM02B and echo problems   Paul Hales
• [asterisk-users] If caller id is null set to a specific number   Eric "ManxPower" Wieling
• [asterisk-users] Snom 320 with TDM02B and echo problems   Michael J. Liberatore
• [asterisk-users] Kernel Native PCIE Network Cards?   Michael J. Liberatore
• [asterisk-users] Your favorite desktop wifi sip hardphone ?   Olivier
• [asterisk-users] __sip_xmit problem   Johansson Olle E
• [asterisk-users] Asterisk 1.4 + Presence   Johansson Olle E
• [asterisk-users] SIP: "To:" header?   Johansson Olle E
• [asterisk-users] __sip_xmit problem   Steve Davies
• [asterisk-users] SIP: "To:" header?   Tony Mountifield
• [asterisk-users] What can I do with Jabber?   Vincent
• [asterisk-users] Snom 320 with TDM02B and echo problems   Tim Koehler
• [asterisk-users] If caller id is null set to a specific number   Doug Lytle
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Philippe Sultan
• [asterisk-users] How to get ten-digit number?   Vincent
• [asterisk-users] How to get ten-digit number?   Tilghman Lesher
• [asterisk-users] Switchvox Space Requirements   Kevin P. Fleming
• [asterisk-users] How to get ten-digit number?   Baji Panchumarti
• [asterisk-users] Asterisk 1.4 + Presence   Eric "ManxPower" Wieling
• [asterisk-users] __sip_xmit problem   Eric "ManxPower" Wieling
• [asterisk-users] If caller id is null set to a specific number   Eric "ManxPower" Wieling
• [asterisk-users] svn+chan_mobile+Asterisk+Siemens FXO no voice   Mandeep Singh Bhabha
• [asterisk-users] Switchvox Space Requirements   Gordon Henderson
• [asterisk-users] Mystery phone!   Mandeep Singh Bhabha
• [asterisk-users] Your favorite desktop wifi sip hardphone ?   mgraves at mstvp.com
• [asterisk-users] Switchvox Space Requirements   shadowym
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Eric Chamberlain
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Eric Chamberlain
• [asterisk-users] How to get ten-digit number?   Vincent
• [asterisk-users] Kernel Native PCIE Network Cards?   Jared Smith
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   joakimsen at gmail.com
• [asterisk-users] Cisco IP Communicator with Asterisk   Kai-Uwe Jensen
• [asterisk-users] __sip_xmit problem   Steve Davies
• [asterisk-users] Cisco IP Communicator with Asterisk   Anciso, Roy
• [asterisk-users] Weekly Conference Reminder: Friday Nov 9th 12:30 PM Eastern Time   randulo
• [asterisk-users] asterisk ODBC dependencies   Robert McNaught
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Steve Edwards
• [asterisk-users] asterisk ODBC dependencies   Jason Parker
• [asterisk-users] Everyone is busy/congested: IP Trunk   bilal ghayyad
• [asterisk-users] Everyone is busy/congested: IP Trunk   Baji Panchumarti
• [asterisk-users] How to get ten-digit number?   Tilghman Lesher
• [asterisk-users] Asterisk on Zonbu, impact of transcoding   Steve Edwards
• [asterisk-users] asterisk ODBC dependencies   Tilghman Lesher
• [asterisk-users] Asterisk and OBDC   Tilghman Lesher
• [asterisk-users] ztdummy, zttest   Tony Plack
• [asterisk-users] Asterisk and OBDC   Tony Plack
• [asterisk-users] Kernel Native PCIE Network Cards?   Michael J. Liberatore
• [asterisk-users] If caller id is null set to a specific number   Jon Weisman
• [asterisk-users] H323 registeration and routing the calls   bilal ghayyad
• [asterisk-users] Kernel Native PCIE Network Cards?   Doug Lytle
• [asterisk-users] Need Reference sites   Jon Weisman
• [asterisk-users] Asterisk Help   Jarga Jallow
• [asterisk-users] Copying the needed configuration files to be used on new installation   bilal ghayyad
• [asterisk-users] ztdummy, zttest   Tony Plack
• [asterisk-users] Cisco IP Communicator with Asterisk   Kai-Uwe Jensen
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Alan Lord
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Philip Prindeville
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Darryl Dunkin
• [asterisk-users] ztdummy, zttest   Tzafrir Cohen
• [asterisk-users] Asterisk Help   ram
• [asterisk-users] asterisk 1.4 prereq   Mark Quitoriano
• [asterisk-users] PHP - Queues - etc.   Marcelo Freitas
• [asterisk-users] asterisk 1.4 prereq   Lenz
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Alan Lord
• [asterisk-users] Asterisk direct dialing   Gopal krishnan
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Vivek Shrivastava
• [asterisk-users] Everyone is busy/congested: IP Trunk   Vivek Shrivastava
• [asterisk-users] 'Traditional' Faxing   Greg Cockburn
• [asterisk-users] sidetone   Todd
• [asterisk-users] sidetone   Gopal krishnan
• [asterisk-users] 'Traditional' Faxing   Gopal krishnan
• [asterisk-users] PHP - Queues - etc.   Atis Lezdins
• [asterisk-users] 'Traditional' Faxing   Jonn R Taylor
• [asterisk-users] Asterisk on Zonbu, impact of transcoding   JR Richardson
• [asterisk-users] Strange asterisk message "Remote host can't match request NOTIFY..."   Bruce Komito
• [asterisk-users] mpg123 on Thecus N2100   Thomas Winter
• [asterisk-users] mpg123 on Thecus N2100   Eric "ManxPower" Wieling
• [asterisk-users] asterisk 1.4 prereq   Eric "ManxPower" Wieling
• [asterisk-users] r2 multiframe error   arkon at nast.pl
• [asterisk-users] asterisk 1.4 prereq   Baji Panchumarti
• [asterisk-users] r2 multiframe error   Moises Silva
• [asterisk-users] mpg123 on Thecus N2100   Thomas Winter
• [asterisk-users] Kernel Native PCIE Network Cards?   Benny Amorsen
• [asterisk-users] r2 multiframe error   Jakub Syrek
• [asterisk-users] Record() : How to get filename created with %d?   Vincent
• [asterisk-users] Kernel Native PCIE Network Cards?   Tzafrir Cohen
• [asterisk-users] mpg123 on Thecus N2100   Tzafrir Cohen
• [asterisk-users] asterisk 1.4 prereq   Tzafrir Cohen
• [asterisk-users] Record() : How to get filename created with %d?   Baji Panchumarti
• [asterisk-users] RTP traffic not being forwarded   Ryan Newington
• [asterisk-users] RTP traffic not being forwarded   Luki
• [asterisk-users] Kernel Native PCIE Network Cards?   joakimsen at gmail.com
• [asterisk-users] RTP traffic not being forwarded   Ryan Newington
• [asterisk-users] Record() : How to get filename created with %d?   Vincent
• [asterisk-users] Record() : How to get filename created with %d?   Eric "ManxPower" Wieling
• [asterisk-users] r2 multiframe error   Moises Silva
• [asterisk-users] sangoma asterisk patches   Tzafrir Cohen
• [asterisk-users] RTP traffic not being forwarded   Vivek Shrivastava
• [asterisk-users] sangoma zaptel patches   Tzafrir Cohen
• [asterisk-users] RTP traffic not being forwarded   Ryan Newington
• [asterisk-users] detect asterisk pbx via sip   Giedrius Augys
• [asterisk-users] 'Traditional' Faxing   Per Jessen
• [asterisk-users] Record() : How to get filename created with %d?   Vincent
• [asterisk-users] detect asterisk pbx via sip   Yann JOUANIN
• [asterisk-users] detect asterisk pbx via sip   Giedrius Augys
• [asterisk-users] detect asterisk pbx via sip   Gopal krishnan
• [asterisk-users] RTP traffic not being forwarded   Vivek Shrivastava
• [asterisk-users] mpg123 on Thecus N2100   Thomas Winter
• [asterisk-users] detect asterisk pbx via sip   Giedrius Augys
• [asterisk-users] multiple PBXs in one box   Joao Pereira
• [asterisk-users] sangoma zaptel patches   Steve Totaro
• [asterisk-users] Record() : How to get filename created with %d?   Baji Panchumarti
• [asterisk-users] Record() : How to get filename created with %d?   Eric "ManxPower" Wieling
• [asterisk-users] IMAP Voicemail -- HELP! Asterisk not playing Greeting!   Michael Schwartz
• [asterisk-users] sangoma zaptel patches   Baji Panchumarti
• [asterisk-users] sangoma zaptel patches   Baji Panchumarti
• [asterisk-users] sangoma zaptel patches   Tzafrir Cohen
• [asterisk-users] Playback() clicking sound at the end of the prompt   dave cantera
• [asterisk-users] [not about sangoma/zaptel] Playback() clicking sound at the end of the prompt   Tzafrir Cohen
• [asterisk-users] IMAP Voicemail -- HELP! Asterisk not playing Greeting!   Vivek Shrivastava
• [asterisk-users] sangoma zaptel patches   Tilghman Lesher
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] ztdummy, zttest   Tzafrir Cohen
• [asterisk-users] sangoma zaptel patches   Steve Totaro
• [asterisk-users] __sip_xmit problem   Rilawich Ango
• [asterisk-users] RTP traffic not being forwarded   Ryan Newington
• [asterisk-users] RTP traffic not being forwarded   Vivek Shrivastava
• [asterisk-users] 'Traditional' Faxing   joakimsen at gmail.com
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   randulo
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Tzafrir Cohen
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   randulo
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   randulo
• [asterisk-users] Record() : How to get filename created with %d?   Tony Mountifield
• [asterisk-users] Grandstream GXP2020 + Asterisk 1.4.11   Erik Wartusch
• [asterisk-users] detect asterisk pbx via sip   Mindaugas Kuprys
• [asterisk-users] 'h' extension on call-out   Artifex Maximus
• [asterisk-users] Video Call   Marek B
• [asterisk-users] 'Traditional' Faxing   Dave Fullerton
• [asterisk-users] Internal CallerID problem   Mark Bell
• [asterisk-users] 'h' extension on call-out   Tony Mountifield
• [asterisk-users] sip_chan - how to use value of the SIP 'To:' header field for extension logic   Tomasz Zieleniewski
• [asterisk-users] 'Traditional' Faxing   Doug Lytle
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Andrew Furey
• [asterisk-users] 'h' extension on call-out   Eric "ManxPower" Wieling
• [asterisk-users] sip_chan - how to use value of the SIP 'To:' header field for extension logic   Tony Mountifield
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   randulo
• [Asterisk-Users] Re: Unable to use voicemail(Please suggestion)   CW_ASN - Gus
• [asterisk-users] ztdummy, zttest   Tony Plack
• [asterisk-users] Polycom Speakerphone   Eric Jacksch
• [asterisk-users] No sound from playback and voicemail   Stefan Guenther
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   SIP
• [asterisk-users] Polycom Speakerphone   John Millican
• [asterisk-users] Polycom Speakerphone   Doug Lytle
• [asterisk-users] Polycom Speakerphone   Michael Graves
• [asterisk-users] 'h' extension on call-out   Artifex Maximus
• [asterisk-users] 'h' extension on call-out   Artifex Maximus
• [asterisk-users] Grandstream GXP2020 + Asterisk 1.4.11   John Faubion
• [asterisk-users] 'h' extension on call-out   Tony Mountifield
• [asterisk-users] No sound from playback and voicemail   Carlos Chavez
• [asterisk-users] Polycom Speakerphone   Alan Lord
• [asterisk-users] No sound from playback and voicemail   Atis Lezdins
• [asterisk-users] 'Traditional' Faxing   Andrew Kohlsmith
• [asterisk-users] No sound from playback and voicemail   Forrest Beck
• [asterisk-users] No sound from playback and voicemail   Tzafrir Cohen
• [asterisk-users] No sound from playback and voicemail (Carlos Chavez)   Stefan Guenther
• [asterisk-users] Polycom Speakerphone   Doug
• [asterisk-users] No sound from playback and voicemail (Atis Lezdins)   Stefan Guenther
• [asterisk-users] Polycom Speakerphone   John Millican
• [asterisk-users] Polycom Speakerphone   David Gomillion
• [asterisk-users] VoiceMail hangup   Il Neofita
• [asterisk-users] VoiceMail hangup   Il Neofita
• [asterisk-users] Polycom Speakerphone   Doug
• [asterisk-users] Polycom Speakerphone   F6HQZ
• [asterisk-users] VoiceMail hangup   Anselm Martin Hoffmeister
• [asterisk-users] VoiceMail hangup   Il Neofita
• [asterisk-users] ACD Queue LOG RINGNOANSWER Content 0   asterisk
• [asterisk-users] ACD Queue LOG RINGNOANSWER Content 0   BJ Weschke
• [asterisk-users] 'h' extension on call-out   Artifex Maximus
• [asterisk-users] ztdummy and BackGround   Carles Pina i Estany
• [asterisk-users] 'h' extension on call-out   Tony Mountifield
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] ODBC connection to Microsoft SQL Server   Robert McNaught
• [asterisk-users] Record() : How to get filename created with %d?   Vincent
• [asterisk-users] Record() : How to get filename created with %d?   Vincent
• [asterisk-users] Record() : How to get filename created with %d?   Vincent
• [asterisk-users] sangoma zaptel patches   Dovid B
• [asterisk-users] 'Traditional' Faxing   Dovid B
• [asterisk-users] 'Traditional' Faxing   Klaverstyn, David C
• [asterisk-users] asterisk and installing chan_h323.so rpm   Dovid B
• [asterisk-users] Asterisk direct dialing   Dovid B
• [asterisk-users] Help: Static and dropped calls   Dovid B
• [asterisk-users] How to delete voice mail messages?   Dovid B
• [asterisk-users] asterisk and installing chan_h323.so rpm   David Boyd
• [asterisk-users] ztdummy and BackGround   Tony Plack
• [asterisk-users] MOH Codec Issue   Nick Brown
• [asterisk-users] RTP traffic not being forwarded   Ryan Newington
• [asterisk-users] MOH Codec Issue   Paul Hales
• [asterisk-users] Chatterbug   Robert Goodyear
• [asterisk-users] MOH Codec Issue   Nick Brown
• [asterisk-users] Chatterbug   Paul Hales
• [asterisk-users] MOH Codec Issue   Paul Hales
• [asterisk-users] Stress-Testing Asterisk   Jeng Yu
• [asterisk-users] Stress-Testing Asterisk   Suity Zsolt
• [asterisk-users] Stress-Testing Asterisk   Tzafrir Cohen
• [asterisk-users] Toshiba DK - Asterisk Integration   Indika Wasala
• [asterisk-users] Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11   Erik Wartusch
• [asterisk-users] Stress-Testing Asterisk   Atis Lezdins
• [asterisk-users] Record() : How to get filename created with %d?   Tony Mountifield
• [asterisk-users] chan_alsa issue   Mohammad Shokuie
• [asterisk-users] Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11   marcotasto
• [asterisk-users] Default mohmp3 : free of rights ?   Sébastien Mortier
• [asterisk-users] Default mohmp3 : free of rights ?   Doug Lytle
• [asterisk-users] VoiceMail hangup   Il Neofita
• [asterisk-users] VoiceMail hangup   Doug Lytle
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] ACD Queue LOG RINGNOANSWER Content 0   asterisk
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   mail-lists
• [asterisk-users] Toshiba DK - Asterisk Integration   Tony Plack
• [asterisk-users] [Fwd: Re: VoiceMail hangup]   marcotasto
• [asterisk-users] How to play Asterisk .raw file   Gary
• [asterisk-users] [Fwd: Re: VoiceMail hangup]   Il Neofita
• [asterisk-users] Conference rooms   Fabio Cappelletti
• [asterisk-users] Record() : How to get filename created with %d?   Steve Murphy
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] ACD Queue LOG RINGNOANSWER Content 0   James FitzGibbon
• [asterisk-users] How to integrate Asterisk with Avaya   Costa Dinoteli
• [asterisk-users] Conference rooms   map
• [asterisk-users] [Fwd: Re: VoiceMail hangup]   Anselm Martin Hoffmeister
• [asterisk-users] Install Scripts for CentOS 4   Jonn R Taylor
• [asterisk-users] How to play Asterisk .raw file   Sean Bright
• [asterisk-users] How to play Asterisk .raw file   BJ Weschke
• [asterisk-users] How to play Asterisk .raw file   Baji Panchumarti
• [asterisk-users] RTP traffic not being forwarded   Carlos Chavez
• [asterisk-users] Install Scripts for CentOS 4   Zaheer K. Master
• [asterisk-users] ACD Queue LOG RINGNOANSWER Content 0   asterisk
• [asterisk-users] sangoma zaptel patches   Steve Totaro
• [asterisk-users] route INVITE sip:s at sip.test.fr   Marc LEURENT
• [asterisk-users] Install Scripts for CentOS 4   Jonn R Taylor
• [asterisk-users] Can I connect device on FXS of Sipura 3000 to internet virtually ? - it can only call ISPs numbers on POTS line   Robert Rozman
• [asterisk-users] Install Scripts for CentOS 4   Zaheer K. Master
• [asterisk-users] Install Scripts for CentOS 4   SIP
• [asterisk-users] Install Scripts for CentOS 4   Baji Panchumarti
• [asterisk-users] Install Scripts for CentOS 4   Patrick
• [asterisk-users] sangoma zaptel patches   Matthew Fredrickson
• [asterisk-users] Install Scripts for CentOS 4   Kyle Sexton
• [asterisk-users] Install Scripts for CentOS 4   Jonn R Taylor
• [asterisk-users] Call Forward on SIP unreachable (network failure)   Antoine Megalla
• [asterisk-users] Install Scripts for CentOS 4   Tzafrir Cohen
• [asterisk-users] Install Scripts for CentOS 4   Baji Panchumarti
• [asterisk-users] ztdummy, zttest   Tony Plack
• [asterisk-users] Install Scripts for CentOS 4   Tony Plack
• [asterisk-users] Install Scripts for CentOS 4   Baji Panchumarti
• [asterisk-users] sangoma zaptel patches   Andres
• [asterisk-users] route INVITE sip:s at sip.test.fr   Marco Mouta
• [asterisk-users] Call Forward on SIP unreachable (network failure)   Marco Mouta
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Marco Mouta
• [asterisk-users] function voicemailmain   Rilawich Ango
• [asterisk-users] Nortel digital FXO channel bank? Exists?   Michelle Dupuis
• [asterisk-users] How to pay for libpri development   Michelle Dupuis
• [asterisk-users] Nortel digital FXO channel bank? Exists?   Jon Pounder
• [asterisk-users] How to pay for libpri development   Jon Pounder
• [asterisk-users] asterisk-users Digest, Vol 40, Issue 37   jeerawan at starsystems.co.th
• [asterisk-users] OT: Re: How to pay for libpri development   Richard Lyman
• [asterisk-users] MOH Codec Issue - Fixed   Nick Brown
• [asterisk-users] What is wrong with this mailing list   Mohammad Shokuie
• [asterisk-users] function voicemailmain   Tilghman Lesher
• [asterisk-users] What is wrong with this mailing list   Erik Anderson
• [asterisk-users] ODBC connection to Microsoft SQL Server   Tilghman Lesher
• [asterisk-users] What is wrong with this mailing list   Baji Panchumarti
• [asterisk-users] What is wrong with this mailing list   Mohammad Shokuie
• [asterisk-users] What is wrong with this mailing list   Erik Anderson
• [asterisk-users] What is wrong with this mailing list   Eric "ManxPower" Wieling
• [asterisk-users] function voicemailmain   joakimsen at gmail.com
• [asterisk-users] MOH Codec Issue - Fixed   Paul Hales
• [asterisk-users] What is wrong with this mailing list   Baji Panchumarti
• [asterisk-users] What is wrong with this mailing list   Mohammad Shokuie
• [asterisk-users] Asterisk trunk and manager redirect problem   Mohammad Shokuie
• [asterisk-users] function voicemailmain   Rilawich Ango
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Greg Oliver
• [asterisk-users] Chatterbug   Robert Goodyear
• [asterisk-users] Toshiba DK - Asterisk Integration   Indika Wasala
• [asterisk-users] Asterisk ignoring manager events when busy   Nick Adams
• [asterisk-users] asterisk and installing chan_h323.so rpm   Bincy K. Philip
• [asterisk-users] Asterisk ignoring manager events when busy   Doug Lytle
• [asterisk-users] Linksys 942 Call Transfer   Jon Farmer
• [asterisk-users] Nortel digital FXO channel bank? Exists?   Rupert Utteridge - Digital Techniques (Asia) Limited
• [asterisk-users] Using php exec() in agi script   Andre Courchesne
• [asterisk-users] What is wrong with this mailing list   Baji Panchumarti
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] Problem with AGI Script   Brett Crapser
• [asterisk-users] Nortel digital FXO channel bank? Exists?   Michelle Dupuis
• [asterisk-users] Nortel digital FXO channel bank? Exists?   Jon Pounder
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] route INVITE sip:s at sip.test.fr   Marc LEURENT
• [asterisk-users] Using php exec() in agi script   Moises Silva
• [asterisk-users] Problem with AGI Script   Moises Silva
• [asterisk-users] IVR Tree Best Practices   Matt
• [asterisk-users] "Whats New at Digium the Asterisk Company" -- Junk?   Philipp Kempgen
• [asterisk-users] Problem with AGI Script   Mindaugas Kezys
• [asterisk-users] "Whats New at Digium the Asterisk Company" -- Junk?   Jesse Molina
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] PBX Testing Framework   IQ Labs VoIP Team
• [asterisk-users] Help in getting a dialplan to produce the right CDR info   Robert Moskowitz
• [asterisk-users] Error: inserting return line in dialing strings   Alejandro Lengua
• [asterisk-users] route INVITE sip:s at sip.test.fr   Marco Mouta
• [asterisk-users] Routing Anonymous Callerid   C. Duncan Hudson
• [asterisk-users] Asterisk ignoring manager events when busy   Nick Adams
• [asterisk-users] Real Time CDR   Tony Plack
• [asterisk-users] Problem with AGI Script   Steve Edwards
• [asterisk-users] pip tones in Monitor or MixMonitor   Tony Plack
• [asterisk-users] pip tones in Monitor or MixMonitor   Mindaugas Kezys
• [asterisk-users] asterisk-stat problem   Richard Cahilig
• [asterisk-users] pip tones in Monitor or MixMonitor   Tony Plack
• [asterisk-users] asterisk-stat problem   Anthony Messina
• [asterisk-users] asterisk-stat problem   Erik Anderson
• [asterisk-users] function voicemailmain   joakimsen at gmail.com
• [asterisk-users] Linksys 942 Call Transfer   joakimsen at gmail.com
• [asterisk-users] function voicemailmain   Tzafrir Cohen
• [asterisk-users] function voicemailmain   Baji Panchumarti
• [asterisk-users] asterisk integration with panasonic analog pbx   jorain
• [asterisk-users] asterisk-stat problem   Paul Hales
• [asterisk-users] Integration of Asterisk with MS Dynamics CRM   Robert Roach
• [asterisk-users] Problem with AGI Script   Benjamin Jacob
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Patrick
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Olivier
• [asterisk-users] Queue   Bhrugu Mehta
• [asterisk-users] pip tones in Monitor or MixMonitor   Mindaugas Kezys
• [asterisk-users] asterisk integration with panasonic analog pbx   C F
• [asterisk-users] Dialogic   asterisk at nicox.org
• [asterisk-users] How to play Asterisk .raw file   Lenz
• [asterisk-users] pipemedia   Pezhman Lali
• [asterisk-users] r2 multiframe error   Jakub Syrek
• [asterisk-users] Call Forward on SIP unreachable (network failure)   Johansson Olle E
• [asterisk-users] route INVITE sip:s at sip.test.fr   Johansson Olle E
• [asterisk-users] Problems with positions in callqueues   Jan-Hendrik Palic
• [asterisk-users] Friday Conference reminder: AGI example Nov 16th at 12:30 PM EST   randulo
• [asterisk-users] Friday Conference reminder: AGI example Nov 16th at 12:30 PM EST   Lenz
• [asterisk-users] DTMF Problem   Arun Kumar
• [asterisk-users] Friday Conference reminder: AGI example Nov 16th at 12:30 PM EST   randulo
• [asterisk-users] DTMF Problem   木木
• [asterisk-users] TE220 PCI express performnace   satish patel
• [asterisk-users] TE210P Vs TE220P difference   satish patel
• [asterisk-users] reload command   Jerry Geis
• [asterisk-users] TE210P Vs TE220P difference   William Moore
• [asterisk-users] Shared gsm files   David Ruggles
• [asterisk-users] T-mobile Blackberry 8320 & Asterisk   Jon Weisman
• [asterisk-users] pip tones in Monitor or MixMonitor   Tony Plack
• [asterisk-users] Problem with AGI Script   Steve Edwards
• [asterisk-users] Dialogic   Steve Totaro
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Greg Oliver
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Greg Oliver
• [asterisk-users] What is wrong with this mailing list   Anthony Francis
• [asterisk-users] DTMF Problem   Doug
• [asterisk-users] Dialing time-out   Jim Houser
• [asterisk-users] Centos 5 Asterisk 1.4 FreePBX Install script   Jonn R Taylor
• [asterisk-users] Shared gsm files   Tilghman Lesher
• [asterisk-users] Dialing time-out   Mindaugas Kezys
• [asterisk-users] Centos 5 Asterisk 1.4 FreePBX Install script   Jonn R Taylor
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Olivier
• [asterisk-users] Lists dead?   Philipp Kempgen
• [asterisk-users] Asterisk Program Closes   Ryan M. Colbert
• [asterisk-users] SPA-2100 into Paging System "Hangs"   Peder at NetworkOblivion
• [asterisk-users] make config update-rc.d   Philipp Kempgen
• [asterisk-users] Pass CallerID when call forwards to PSTN?   Russell Horn
• [asterisk-users] Lists dead?   Baji Panchumarti
• [asterisk-users] Asterisk Program Closes   Tilghman Lesher
• [asterisk-users] OT - best policy for logs   Olivier
• [asterisk-users] Music on Hold -- Error   Ryan M. Colbert
• [asterisk-users] make config update-rc.d   Tilghman Lesher
• [asterisk-users] Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI   Jacob Lefkowitz
• [asterisk-users] OT - best policy for logs   Jay R. Ashworth
• [asterisk-users] OT - best policy for logs   Erik Anderson
• [asterisk-users] Asterisk Program Closes   Doug
• [asterisk-users] OT - best policy for logs   Steve Edwards
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] Building an Asterisk 1.4 RPM.   Douglas Garstang
• [asterisk-users] Help on strange problem...   Stuart Sheldon
• [asterisk-users] DTMF Problem   木木
• [asterisk-users] Asterisk 1.4 with LDAP   Pepo
• [asterisk-users] Music on Hold -- Error   Nick Brown
• [asterisk-users] reload command   Moises Silva
• [asterisk-users] Asterisk 1.4 with LDAP   Anthony Francis
• [asterisk-users] reload command   Jerry Geis
• [asterisk-users] dtmf detection   Rilawich Ango
• [asterisk-users] reload command   Moises Silva
• [asterisk-users] DTMF Problem   Benjamin Jacob
• [asterisk-users] Problem with AGI Script   Benjamin Jacob
• [asterisk-users] Toshiba DK - Asterisk Integration   Indika Wasala
• [asterisk-users] Dialogic   asterisk at nicox.org
• [asterisk-users] Which files to be copied   bilal ghayyad
• [asterisk-users] DTMF Problem   木木
• [asterisk-users] DTMF Problem   Rilawich Ango
• [asterisk-users] Dialogic   Baji Panchumarti
• [asterisk-users] Which files to be copied   Alex Balashov
• [asterisk-users] r2 multiframe error - continue   Jakub Syrek
• [asterisk-users] Change the Voice promps in asterisk 1.4   voip crazy
• [asterisk-users] channels to destroy   Carles Pina i Estany
• [asterisk-users] Changing audio message to text message   Anthony Chapellier
• [asterisk-users] Changing audio message to text message   EdPimentl
• [asterisk-users] reload command   Tony Plack
• [asterisk-users] reload command   Julian Lyndon-Smith
• [asterisk-users] r2 multiframe error - continue   Moises Silva
• [asterisk-users] Changing audio message to text message   Anthony Chapellier
• [asterisk-users] Help with Polycom 320   Jarga Jallow
• [asterisk-users] Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI   Jared Smith
• [asterisk-users] Which files to be copied   bilal ghayyad
• [asterisk-users] Music on Hold -- Error   Ryan M. Colbert
• [asterisk-users] reload command   Tony Plack
• [asterisk-users] Polycom softkey transfer issue   Wayne P. Hill
• [asterisk-users] r2 multiframe error - continue   Jakub Syrek
• [asterisk-users] make config update-rc.d   Tzafrir Cohen
• [asterisk-users] r2 multiframe error - continue   Luis Antonio Prata Barbosa
• [asterisk-users] Help with Polycom 320   Bruce Reeves
• [asterisk-users] r2 multiframe error - continue   Moises Silva
• [asterisk-users] Dumb AGI question   David Ruggles
• [asterisk-users] Dumb AGI question   Steve Edwards
• [asterisk-users] r2 multiframe error - continue   Jakub Syrek
• [asterisk-users] 1.4 SIP Jitter Buffer   Mehdi chouikh
• [asterisk-users] Which files to be copied   Mojo with Horan & Company, LLC
• [asterisk-users] Dumb AGI question   David Ruggles
• [asterisk-users] r2 multiframe error - continue   Moises Silva
• [asterisk-users] r2 multiframe error - continue   Jakub Syrek
• [asterisk-users] Dialing time-out   Jared Smith
• [asterisk-users] Asterisk 1.4.14 Released   The Asterisk Development Team
• [asterisk-users] modifying a dialed exension before dialplan processing   Brian J. Murrell
• [asterisk-users] r2 multiframe error - continue   Steve Underwood
• [asterisk-users] modifying a dialed exension before dialplan processing   Baji Panchumarti
• [asterisk-users] modifying a dialed exension before dialplan processing   Eric "ManxPower" Wieling
• [asterisk-users] The call does not disconnect at the softphone when caller hangup the mobile   bilal ghayyad
• [asterisk-users] Change the Voice promps in asterisk 1.4   Per Jessen
• [asterisk-users] California based PSTN connections   Adrian Marsh
• [asterisk-users] Blackberry MVS and Asterisk.   Mike Dent
• [asterisk-users] Astmanproxy Yahoo Group   Mamadou Lamine KA
• [asterisk-users] Change the Voice promps in asterisk 1.4   Baji Panchumarti
• [asterisk-users] chan_ss7 0.10   marek cervenka
• [asterisk-users] r2 multiframe error - continue   Jakub Syrek
• [asterisk-users] modifying a dialed exension before dialplan processing   Brian J. Murrell
• [asterisk-users] modifying a dialed exension before dialplan processing   Tilghman Lesher
• [asterisk-users] modifying a dialed exension before dialplan processing   Brian J. Murrell
• [asterisk-users] Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI   Matthew Fredrickson
• [asterisk-users] modifying a dialed exension before dialplan processing   Eric "ManxPower" Wieling
• [asterisk-users] Page Command   Anciso, Roy
• [asterisk-users] Changing audio message to text message   Robert Lister
• [asterisk-users] Pass CallerID when call forwards to PSTN?   Robert Lister
• [asterisk-users] Multiple B410P's in one machine   asterisk-users at rogg.is
• [asterisk-users] Building and running mISDN for B410P on Ubuntu 7.04   asterisk-users at rogg.is
• [asterisk-users] Polycom Provisioning Tool Source Code Released   Michael Munger
• [asterisk-users] Multiple B410P's in one machine   Kevin P. Fleming
• [asterisk-users] Connecting Ericsson 4422 or similar set to Asterisk ?   sanal adam
• [asterisk-users] Multiple B410P's in one machine   asterisk-users at rogg.is
• [asterisk-users] modifying a dialed exension before dialplan processing   Brian J. Murrell
• [asterisk-users] p2p t1 with sangoma hw   Michael J. Liberatore
• [asterisk-users] modifying a dialed exension before dialplan processing   Baji Panchumarti
• [asterisk-users] p2p t1 with sangoma hw   Jesse Molina
• [asterisk-users] modifying a dialed exension before dialplan processing   Brian J. Murrell
• [asterisk-users] p2p t1 with sangoma hw   Michael J. Liberatore
• [asterisk-users] modifying a dialed exension before dialplan processing   Baji Panchumarti
• [asterisk-users] problem with tdm2400p configuration   Mark Quitoriano
• [asterisk-users] problem with tdm2400p configuration   Eric "ManxPower" Wieling
• [asterisk-users] problem with tdm2400p configuration   Mark Quitoriano
• [asterisk-users] Trouble with asterisk-users mailman   Jesse Molina
• [asterisk-users] AsteriskNOW and TDM800P   dave cantera
• [asterisk-users] Trouble with asterisk-users mailman   Jesse Molina
• [asterisk-users] Trouble with asterisk-users mailman   Baji Panchumarti
• [asterisk-users] problem with tdm2400p configuration   Tzafrir Cohen
• [asterisk-users] Trouble with asterisk-users mailman   Baji Panchumarti
• [asterisk-users] p2p t1 with sangoma hw   Michael J. Liberatore
• [asterisk-users] p2p t1 with sangoma hw   Erik Anderson
• [asterisk-users] p2p t1 with sangoma hw   Michael J. Liberatore
• [asterisk-users] sip + jitter buffer   satish patel
• [asterisk-users] Building and running mISDN for B410P on Ubuntu 7.04   Per Jessen
• [asterisk-users] problem with tdm2400p configuration   Mark Quitoriano
• [asterisk-users] facilityenable in zapata.conf   Michelle Dupuis
• [asterisk-users] [IAX] Does the client have to use UDP4569 as source port?   Vincent
• [asterisk-users] sip + jitter buffer   Patrick
• [asterisk-users] problem with tdm2400p configuration   Tzafrir Cohen
• [asterisk-users] Problem with AGI Script   didier
• [asterisk-users] Problem with AGI Script   Tilghman Lesher
• [asterisk-users] Conference Call Dial-Out to a participant   broadband Voice
• [asterisk-users] problem with tdm2400p configuration   Mark Quitoriano
• [asterisk-users] Problem with AGI Script   Eric "ManxPower" Wieling
• [asterisk-users] [IAX] Does the client have to use UDP4569 as source port?   Eric "ManxPower" Wieling
• [asterisk-users] Conference Call Dial-Out to a participant   Tilghman Lesher
• [asterisk-users] problem with tdm2400p configuration   Tzafrir Cohen
• [asterisk-users] Conference Call Dial-Out to a participant   broadband Voice
• [asterisk-users] Conference Call Dial-Out to a participant   Yann JOUANIN
• [asterisk-users] Conference Call Dial-Out to a participant   Brett Crapser
• [asterisk-users] Conference Call Dial-Out to a participant   Eric "ManxPower" Wieling
• [asterisk-users] Conference Call Dial-Out to a participant   Tzafrir Cohen
• [asterisk-users] [IAX] Does the client have to use UDP4569 as source port?   Vincent
• [asterisk-users] Asterisk on Pcengines Alix board   Giuseppe Barichello
• [asterisk-users] Trouble with asterisk-users mailman   Jesse Molina
• [asterisk-users] problem with tdm2400p configuration   Mark Quitoriano
• [asterisk-users] Trouble with asterisk-users mailman   Philip Prindeville
• [asterisk-users] Help: How to configure SIP domain on SPA942   Philip Prindeville
• [asterisk-users] problem with tdm2400p configuration   Eric "ManxPower" Wieling
• [asterisk-users] Help: How to configure SIP domain on SPA942   Eric "ManxPower" Wieling
• [asterisk-users] Trouble with asterisk-users mailman   Baji Panchumarti
• [asterisk-users] Trouble with asterisk-users mailman   Baji Panchumarti
• [asterisk-users] problem with tdm2400p configuration   Mark Quitoriano
• [asterisk-users] Help: How to configure SIP domain on SPA942   Ira
• [asterisk-users] Help on strange problem...   Johansson Olle E
• [asterisk-users] channels to destroy   Johansson Olle E
• [asterisk-users] reload command   Johansson Olle E
• [asterisk-users] Asterisk Sound File   Abdul
• [asterisk-users] Changing audio message to text message   Anthony Chapellier
• [asterisk-users] Registration problem: UA -> SER -> Asterisk   Stefano Capitanio
• [asterisk-users] Problem with AGI Script   Chris Blunt
• [asterisk-users] Gigaset S450ip and simultaneous calls   Olivier
• [asterisk-users] Asterisk Sound File   Baji Panchumarti
• [asterisk-users] Asterisk 1.4 with LDAP   Alejandro Cabrera Obed
• [asterisk-users] How to enable res_config_mysql   Tomasz Zieleniewski
• [asterisk-users] How to enable res_config_mysql   Atis Lezdins
• [asterisk-users] Gigaset S450ip and simultaneous calls   Per Jessen
• [asterisk-users] How to enable res_config_mysql   Philipp Kempgen
• [asterisk-users] Asterisk on Pcengines Alix board   Matthew Rubenstein
• [asterisk-users] blind transfer dumping calls   Brian J. Murrell
• [asterisk-users] How to enable res_config_mysql   Tomasz Zieleniewski
• [asterisk-users] blind transfer dumping calls   Atis Lezdins
• [asterisk-users] Asterisk Sound File   Tilghman Lesher
• [asterisk-users] How to enable res_config_mysql   Tony Plack
• [asterisk-users] How to enable res_config_mysql   Tomasz Zieleniewski
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] How to enable res_config_mysql   Tony Plack
• [asterisk-users] Asterisk Sound File   Atis Lezdins
• [asterisk-users] Gigaset S450ip and simultaneous calls   Anselm Martin Hoffmeister
• [asterisk-users] How to enable res_config_mysql   Tomasz Zieleniewski
• [asterisk-users] Asterisk Sound File   Baji Panchumarti
• [asterisk-users] Gigaset S450ip and simultaneous calls &nb