September 2010 Archives by date

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Starting: Wed Sep 1 03:23:50 CDT 2010
Ending: Thu Sep 30 15:44:31 CDT 2010
Messages: 1240

• [asterisk-users] Digest Username/auth name mismatch   t. k
• [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny   Nikhil Nair
• [asterisk-users] Asterisk routing to SoftSwitch   Pratik Shrestha
• [asterisk-users] Asterisk routing to SoftSwitch   Steve Howes
• [asterisk-users] Logging the CID from the Privacy Manager   Jaap Winius
• [asterisk-users] Freepbx + Asterisk problem - NEED HELP   Mehmet Kuzulugil
• [asterisk-users] Freepbx + Asterisk problem - NEED HELP   Alex Ferrara
• [asterisk-users] Freepbx + Asterisk problem - NEED HELP   Tzafrir Cohen
• [asterisk-users] 3Com 3102 Phones   Barry Fawthrop
• [asterisk-users] ChanSpy getting piled up   Rushikesh
• [asterisk-users] ChanSpy getting piled up   Jim Dickenson
• [asterisk-users] 3Com 3102 Phones   Kyle Kienapfel
• [asterisk-users] ChanSpy getting piled up   Rushikesh
• [asterisk-users] * and mj   Jeff Jones
• [asterisk-users] ChanSpy getting piled up   Jim Dickenson
• [asterisk-users] * and mj   Danny Nicholas
• [asterisk-users] * and mj   Infra
• [asterisk-users] MOH in the middle of the call   Dario Quiroz
• [asterisk-users] help with dialplan   Steve Murphy
• [asterisk-users] MOH in the middle of the call   Danny Nicholas
• [asterisk-users] libpri 1.4.11.4 Now Available   Asterisk Development Team
• [asterisk-users] ITSP with DDIs (or DIDs) from India   Jamie A. Stapleton
• [asterisk-users] MOH in the middle of the call   Stefan Schmidt
• [asterisk-users] Asterisk routing to SoftSwitch   Pratik Shrestha
• [asterisk-users] NCS - Cablemodem   Protectix - IT Solutions
• [asterisk-users] asterisk-1.8 problem with one-way audio with no nat   covici at ccs.covici.com
• [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?   bruce bruce
• [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?   Tzafrir Cohen
• [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?   bruce bruce
• [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?   bruce bruce
• [asterisk-users] agi playback to execute say.conf settings   asteriskguru asteriskguru
• [asterisk-users] Dahdi issue on sangoma A200   asteriskguru asteriskguru
• [asterisk-users] Call Recording Questions   Dan Journo
• [asterisk-users] Call Recording Questions   Dan Journo
• [asterisk-users] Call Recording Questions   Gareth Blades
• [asterisk-users] Call Recording Questions   Gareth Blades
• [asterisk-users] Call Recording Questions   Antonio Berrios
• [asterisk-users] Call Recording Questions   Ishfaq Malik
• [asterisk-users] Call Recording Questions   Dan Journo
• [asterisk-users] Voicemail - disable * 0 and #   Paddy Grice
• [asterisk-users] Voicemail - disable * 0 and #   Paddy Grice
• [asterisk-users] Call Recording Questions   Ishfaq Malik
• [asterisk-users] Voicemail - disable * 0 and #   Paul Belanger
• [asterisk-users] IAX2 calls getting rejected without aCAUSE CODE. How to debug this?   Danny Nicholas
• [asterisk-users] agi playback to execute say.conf settings   Danny Nicholas
• [asterisk-users] Call Recording Questions   Antonio Berrios
• [asterisk-users] Call Recording Questions   Dan Journo
• [asterisk-users] asterisk-1.8 problem with one-way audio with no nat   Tilghman Lesher
• [asterisk-users] MOH in the middle of the call   Fabiano Carlos Heringer
• [asterisk-users] Google Voice-like feature.   Ken D'Ambrosio
• [asterisk-users] Google Voice-like feature.   Gareth Blades
• [asterisk-users] Google Voice-like feature.   Danny Nicholas
• [asterisk-users] Google Voice-like feature.   Lonnie Abelbeck
• [asterisk-users] Google Voice-like feature.   A J Stiles
• [asterisk-users] Freepbx + Asterisk problem - NEED HELP   Mehmet Kuzulugil
• [asterisk-users] Freepbx + Asterisk problem - NEED HELP   Mehmet Kuzulugil
• [asterisk-users] Google Voice-like feature.   Ken D'Ambrosio
• [asterisk-users] Freepbx + Asterisk problem - NEED HELP   Mehmet Kuzulugil
• [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?   bruce bruce
• [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?   Miguel Molina
• [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?   Don Kelly
• [asterisk-users] How to finish an AGI   Danny Dias
• [asterisk-users] How to finish an AGI   Steve Edwards
• [asterisk-users] How to create a coredump for Asterisk   Thorolf Godawa
• [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- What are their current cost?   Jeff Brower
• [asterisk-users] How to create a coredump for Asterisk   Danny Nicholas
• [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- What are their current cost?   Danny Nicholas
• [asterisk-users] asterisk 1.6.2.11 freezes the server   George
• [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost?   Don Kelly
• [asterisk-users] Channel Signalling   Arnaldo Giacomitti Junior
• [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost?   bruce bruce
• [asterisk-users] How to finish an AGI   Danny Dias
• [asterisk-users] How to finish an AGI   Danny Dias
• [asterisk-users] How to finish an AGI   Danny Nicholas
• [asterisk-users] How to finish an AGI   Danny Dias
• [asterisk-users] How to finish an AGI   Danny Nicholas
• [asterisk-users] How to finish an AGI   Danny Dias
• [asterisk-users] How to finish an AGI   Steve Edwards
• [asterisk-users] Asterisk processing URI's   Sascha Ferley
• [asterisk-users] How to create a coredump for Asterisk   Luki
• [asterisk-users] Asterisk failing when recording calls   Carlos Chavez
• [asterisk-users] Polycom 670 with Extension Module | Busy Lamp Field | Directed Pickup | Speed Dial | etc   Positively Optimistic
• [asterisk-users] Call Recording Questions   Prince Singh
• [asterisk-users] not succeeding to hide callerid with outbound calls   Joost Kuif | Mobillion
• [asterisk-users] How to finish an AGI   Danny Dias
• [asterisk-users] Wanted: UK-specific hardware recommendations (FXO and FXS)   Roger Burton West
• [asterisk-users] Wanted: UK-specific hardware recommendations (FXO and FXS)   Steve Howes
• [asterisk-users] openvz   mattias
• [asterisk-users] openvz   Danny Nicholas
• [asterisk-users] How to use MYSQL(Set timeout x)   F B
• [asterisk-users] openvz   Tzafrir Cohen
• [asterisk-users] [draft] DAHDI-linux & DAHDI-tools 2.4.0 Release Announcement   Asterisk Development Team
• [asterisk-users] openvz   mattias
• [asterisk-users] Faxes   dave george
• [asterisk-users] openvz   Faris Raouf
• [asterisk-users] [SOLVED ]How to finish an AGI   Danny Dias
• [asterisk-users] Faxes   Steve Totaro
• [asterisk-users] Wanted: UK-specific hardware recommendations (FXOand FXS)   Ade Vickers
• [asterisk-users] Faxes   dave george
• [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?   Alex Bell
• [asterisk-users] openvz   bruce bruce
• [asterisk-users] openvz   Miguel Molina
• [asterisk-users] How to create a coredump for Asterisk   Paul Belanger
• [asterisk-users] Faxes   David Backeberg
• [asterisk-users] Faxes   dave george
• [asterisk-users] Faxes   Danny Nicholas
• [asterisk-users] Faxes   Kevin P. Fleming
• [asterisk-users] Faxes   dave george
• [asterisk-users] How to tell if there is a transfer from CDR?   Carlos Chavez
• [asterisk-users] Faxes   Joel Maslak
• [asterisk-users] openvz   Zeeshan Zakaria
• [asterisk-users] Faxes   Nasir Iqbal
• [asterisk-users] Early media and IAX2   Russ Dill
• [asterisk-users] Wanted: UK-specific hardware recommendations (FXOand FXS)   --[ UxBoD ]--
• [asterisk-users] Snom phones recommended firmware   John Taylor
• [asterisk-users] Freepbx + Asterisk problem - NEED HELP   Mehmet Kuzulugil
• [asterisk-users] Snom phones recommended firmware   Philipp von Klitzing
• [asterisk-users] Manuplating Queue   Timothy Smith
• [asterisk-users] Manuplating Queue   Hoggins!
• [asterisk-users] fast busy out?   Thomas Perron
• [asterisk-users] Global Outage?   Matt Desbiens
• [asterisk-users] Manuplating Queue   Timothy Smith
• [asterisk-users] Vitelity offline?   Roger Marquis
• [asterisk-users] fast busy out?   Ondrej Škopek
• [asterisk-users] fast busy out?   Ondrej Škopek
• [asterisk-users] fast busy out?   Anton Raharja
• [asterisk-users] Vitelity offline?   Kyle Kienapfel
• [asterisk-users] Vitelity offline?   Matt Desbiens
• [asterisk-users] fast busy out?   Ondrej Škopek
• [asterisk-users] Vitelity offline?   Roderick A. Anderson
• [asterisk-users] Vitelity offline?   Matt Desbiens
• [asterisk-users] fast busy out?   Thomas Perron
• [asterisk-users] How to finish an AGI   Edwin Quijada
• [asterisk-users] Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?   Vikram Ragukumar
• [asterisk-users] How to tell if there is a transfer from CDR?   C F
• [asterisk-users] Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?   Steve Underwood
• [asterisk-users] How to tell if there is a transfer from CDR?   Nic Colledge
• [asterisk-users] How to tell if there is a transfer from CDR?   Bryant Zimmerman
• [asterisk-users] How to tell if there is a transfer from CDR?   Bryant Zimmerman
• [asterisk-users] Registering and initiating a SIP call without a SIP client   Gautam Desai
• [asterisk-users] Registering and initiating a SIP call without a SIP client   Thomas Perron
• [asterisk-users] Registering and initiating a SIP call without a SIP client   Stefan Schmidt
• [asterisk-users] Registering and initiating a SIP call without a SIP client   Bruce Ferrell
• [asterisk-users] evil disconnect of call with cisco 1760   Jeremy Kister
• [asterisk-users] How to tell if there is a transfer from CDR?   Nic Colledge
• [asterisk-users] agi playback to execute say.conf settings   Ashik Ali
• [asterisk-users] Asterisk 1.8.0-beta4 Now Available   Ira
• [asterisk-users] 3Com 3102 Phones   Barry Fawthrop
• [asterisk-users] Asterisk Fax   Andrew Nowrot
• [asterisk-users] SMS and fixed land lines   Olivier
• [asterisk-users] Is it possible to keep both call legs live after Dial()   Barry O'Donovan
• [asterisk-users] SMS and fixed land lines   Philipp von Klitzing
• [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting   Jonas Kellens
• [asterisk-users] SMS and fixed land lines   Olivier
• [asterisk-users] Possible malformed G729B - SID (VAD/DTX)frames from carrier endpoint ?   Steve Underwood
• [asterisk-users] Is it possible to keep both call legs live after Dial()   Paul Belanger
• [asterisk-users] Possible malformed G729B - SID (VAD/DTX)frames from carrier endpoint ?   Jeff Brower
• [asterisk-users] SMS and fixed land lines   Administrator TOOTAI
• [asterisk-users] Dial timeout and SIP 302 Moved Temporarily   Olivier
• [asterisk-users] Going to go out on a limb here - regarding Vonage   GlenM
• [asterisk-users] SMS and fixed land lines   Olivier
• [asterisk-users] SMS and fixed land lines   Philipp von Klitzing
• [asterisk-users] SMS and fixed land lines   Administrator TOOTAI
• [asterisk-users] Possible malformed G729B - SID (VAD/DTX)framesfrom carrier endpoint ?   Jeff Brower
• [asterisk-users] Asterisk stops processing calls...   Carlos Chavez
• [asterisk-users] How are shared variables destroyed ?   Olivier
• [asterisk-users] SMS and fixed land lines   Randy R
• [asterisk-users] Asterisk Fax   Kevin P. Fleming
• [asterisk-users] SMS and fixed land lines   Administrator TOOTAI
• [asterisk-users] What can make G.729a codec hostid change?   Barry Miller
• [asterisk-users] MeetMe errorhandling   Daniel Knoll
• [asterisk-users] MeetMe errorhandling   Kai-Uwe Jensen
• [asterisk-users] asterisk-1.8 problem with one-way audio with no nat   covici at ccs.covici.com
• [asterisk-users] asterisk-1.8 problem with one-way audio with no nat   covici at ccs.covici.com
• [asterisk-users] Is it possible to keep both call legs live after Dial()   C F
• [asterisk-users] No audio on call forward after upgrade fromAsterisk 1.4 to 1.6   Alex Ferrara
• [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6   Alex Ferrara
• [asterisk-users] MeetMe errorhandling   Daniel Knoll
• [asterisk-users] Is it possible to keep both call legs live after Dial()   Barry O'Donovan
• [asterisk-users] not succeeding to hide callerid with outbound calls   Joost Kuif | Mobillion
• [asterisk-users] What can make G.729a codec hostid change?   Kyle Kienapfel
• [asterisk-users] MeetMe errorhandling   Paul Belanger
• [asterisk-users] voice mail system   Frenette, Rob
• [asterisk-users] voice mail system   Zoa
• [asterisk-users] voice mail system   Danny Nicholas
• [asterisk-users] MeetMe errorhandling   Daniel Knoll
• [asterisk-users] Dial timeout and SIP 302 Moved Temporarily   Kevin P. Fleming
• [asterisk-users] Possible malformed G729B - SID (VAD/DTX)framesfrom carrier endpoint ?   Steve Underwood
• [asterisk-users] What can make G.729a codec hostid change?   Barry Miller
• [asterisk-users] Is it possible to keep both call legs live after Dial()   Tilghman Lesher
• [asterisk-users] Dial timeout and SIP 302 Moved Temporarily   Olivier
• [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting   Jonas Kellens
• [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting   Philipp von Klitzing
• [asterisk-users] SPA3102 FAX not working   Gopalakrishnan A.N
• [asterisk-users] What can make G.729a codec hostid change?   Dave Platt
• [asterisk-users] What can make G.729a codec hostid change?   Roger Burton West
• [asterisk-users] What can make G.729a codec hostid change?   Tiago Geada
• [asterisk-users] 5-7 second connection delay in outgoing FXO calls   Frank Tarczynski
• [asterisk-users] Call Center: scripting for call routing, reporting, login and logout, CTI   bilal ghayyad
• [asterisk-users] What can make G.729a codec hostid change?   Mike
• [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting   Jonas Kellens
• [asterisk-users] Losing first DTMF digit (with ASR)   Richard Kenner
• [asterisk-users] 5-7 second connection delay in outgoing FXO calls   Danny Nicholas
• [asterisk-users] Losing first DTMF digit (with ASR)   Danny Nicholas
• [asterisk-users] What can make G.729a codec hostid change?   Barry Miller
• [asterisk-users] Losing first DTMF digit (with ASR)   Richard Kenner
• [asterisk-users] What can make G.729a codec hostid change?   Gordon Henderson
• [asterisk-users] Losing first DTMF digit (with ASR)   Danny Nicholas
• [asterisk-users] What can make G.729a codec hostid change?   Kevin P. Fleming
• [asterisk-users] Losing first DTMF digit (with ASR)   Bryant Zimmerman
• [asterisk-users] Losing first DTMF digit (with ASR)   Richard Kenner
• [asterisk-users] Solving the CDR mess of attended transfers   Fabiano Carlos Heringer
• [asterisk-users] Losing first DTMF digit (with ASR)   Richard Kenner
• [asterisk-users] Losing first DTMF digit (with ASR)   Danny Nicholas
• [asterisk-users] Losing first DTMF digit (with ASR)   Bryant Zimmerman
• [asterisk-users] Losing first DTMF digit (with ASR)   Richard Kenner
• [asterisk-users] Solving the CDR mess of attended transfers   Miguel Molina
• [asterisk-users] Losing first DTMF digit (with ASR)   Danny Nicholas
• [asterisk-users] Losing first DTMF digit (with ASR)   Richard Kenner
• [asterisk-users] Losing first DTMF digit (with ASR)   Danny Nicholas
• [asterisk-users] Losing first DTMF digit (with ASR)   Richard Kenner
• [asterisk-users] Call Center: scripting for call routing, reporting, login and logout, CTI   Paul Belanger
• [asterisk-users] What can make G.729a codec hostid change?   Steve Underwood
• [asterisk-users] SOLVED: What can make G.729a codec hostid change?   Barry Miller
• [asterisk-users] Solving the CDR mess of attended transfers   Fabiano Carlos Heringer
• [asterisk-users] Is it possible to keep both call legs live after Dial()   C F
• [asterisk-users] voice mail system   C F
• [asterisk-users] 5-7 second connection delay in outgoing FXO calls   Frank Tarczynski
• [asterisk-users] Channel Signalling   Matt Riddell
• [asterisk-users] rtcp to cdr for calls from dahdi to sip   Dmitry Melekhov
• [asterisk-users] How to Set Callerid Of Originate a call?   Zhang Shukun
• [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting   Jonas Kellens
• [asterisk-users] Requirement or just Best Practice   Danny Nicholas
• [asterisk-users] asterisk 1.8 Calendar   Adrià Vidal
• [asterisk-users] Call Center: scripting for call routing, reporting, login and logout, CTI   Matt Florell
• [asterisk-users] asterisk 1.8 Calendar   Danny Nicholas
• [asterisk-users] asterisk 1.8 Calendar   Adrià Vidal
• [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql   Jonas Kellens
• [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql   Gareth Blades
• [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql   Jonas Kellens
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Steve Davies
• [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql   Gareth Blades
• [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql   Jonas Kellens
• [asterisk-users] Problem with new AEX800 card dying because of interrupt problems   Christian Weeks
• [asterisk-users] Problem with new AEX800 card dying because of interrupt problems   Shaun Ruffell
• [asterisk-users] IPSec on asterisk   Deepika Nijhawan
• [asterisk-users] Sip real problem   Maxim Balabaev
• [asterisk-users] Asterisk 1.8.0-beta5 Now Available   Asterisk Development Team
• [asterisk-users] Sangnoma + Digium Bridging   Joel Maslak
• [asterisk-users] IPSec on asterisk   Paul Belanger
• [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting   Jonas Kellens
• [asterisk-users] openvz   CunningPike
• [asterisk-users] Thailand DID   Eric Smith
• [asterisk-users] Thailand DID   Danny Nicholas
• [asterisk-users] Asterisk 1.6 and fax   Stanislav Korsei
• [asterisk-users] Problem with new AEX800 card dying because of interrupt problems   Christian Weeks
• [asterisk-users] Asterisk 1.6 and fax   David Backeberg
• [asterisk-users] Max TDM calls per asterisk box   Adolphe Cher-Aime
• [asterisk-users] Problem with new AEX800 card dying because of interrupt problems   Shaun Ruffell
• [asterisk-users] Sangnoma + Digium Bridging   Joel Maslak
• [asterisk-users] Max TDM calls per asterisk box   Paul Belanger
• [asterisk-users] Max TDM calls per asterisk box   Adolphe Cher-Aime
• [asterisk-users] Max TDM calls per asterisk box   Steve Edwards
• [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..   dashy dude
• [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..   Philipp von Klitzing
• [asterisk-users] agi playback to execute say.conf settings   Ashik Ali
• [asterisk-users] IPSec on asterisk   Deepika Nijhawan
• [asterisk-users] IPSec on asterisk   Rob Hillis
• [asterisk-users] asterisk 1.8 Calendar   Adrià Vidal
• [asterisk-users] syntax error, unexpected '<token>'   Jonas Kellens
• [asterisk-users] syntax error, unexpected '<token>'   Gareth Blades
• [asterisk-users] How to avoid interruptions with DIGIUM   Danny Dias
• [asterisk-users] IPSec on asterisk   Paul Belanger
• [asterisk-users] SPA3102 FAX not working   Gopalakrishnan A.N
• [asterisk-users] How to avoid interruptions with DIGIUM   Danny Nicholas
• [asterisk-users] Mirroring or other arangement to secure *   hbk
• [asterisk-users] How to avoid interruptions with DIGIUM   Andrew Latham
• [asterisk-users] Mirroring or other arangement to secure *   Matthew J. Roth
• [asterisk-users] agi playback to execute say.conf settings   Danny Nicholas
• [asterisk-users] Set channel variable from within other channel   Jonas Kellens
• [asterisk-users] SPA3102 FAX not working   Tim Nelson
• [asterisk-users] How to avoid interruptions with DIGIUM   Tim Nelson
• [asterisk-users] info about application not available asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Set channel variable from within other channel   Danny Nicholas
• [asterisk-users] How to avoid interruptions with DIGIUM   Kevin P. Fleming
• [asterisk-users] SPA3102 FAX not working   Gopalakrishnan A.N
• [asterisk-users] Set channel variable from within other channel   Jonas Kellens
• [asterisk-users] Set channel variable from within other channel   Danny Nicholas
• [asterisk-users] SPA3102 FAX not working   Gergo Csibra
• [asterisk-users] How to avoid interruptions with DIGIUM   Danny Dias
• [asterisk-users] vegastream 50 BRI-s latest firmware ?   mancyborg at gmail.com
• [asterisk-users] SPA3102 FAX not working   Gopalakrishnan A.N
• [asterisk-users] Issues with in-call DTMF using Broadvox and Level 3   Bryant Zimmerman
• [asterisk-users] info about application not available asterisk 1.6.2.11   Paul Belanger
• [asterisk-users] info about application not available asterisk 1.6.2.11   Paul Belanger
• [asterisk-users] 3Com 3102 Phones   Barry Fawthrop
• [asterisk-users] info about application not available asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] 3Com 3102 Phones   Kyle Kienapfel
• [asterisk-users] info about application not available asterisk 1.6.2.11   Paul Belanger
• [asterisk-users] info about application not available asterisk1.6.2.11   Danny Nicholas
• [asterisk-users] VoIP friendly Internet providers in Dallas and Philadelphia   bruce bruce
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Antonio Berrios
• [asterisk-users] Call Center: scripting for call routing, reporting, login and logout, CTI   Antonio Berrios
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Steve Davies
• [asterisk-users] Archive of security advisories?   Carlos Chavez
• [asterisk-users] Archive of security advisories?   Kyle Kienapfel
• [asterisk-users] Archive of security advisories?   Danny Nicholas
• [asterisk-users] Archive of security advisories?   Barry Miller
• [asterisk-users] VoIP friendly Internet providers in Dallas and Philadelphia   Peder
• [asterisk-users] Mirroring or other arangement to secure *   Greg Woods
• [asterisk-users] Cisco 7975g running 8.3.4   Jamie A. Stapleton
• [asterisk-users] Archive of security advisories?   Tilghman Lesher
• [asterisk-users] Curious what 'early media' is in terms of Answer()   Hose
• [asterisk-users] DAHDI fxstest?   Tim Nelson
• [asterisk-users] Curious what 'early media' is in terms of Answer()   Paul Belanger
• [asterisk-users] DAHDI fxstest?   Paul Belanger
• [asterisk-users] DAHDI fxstest?   Edwin Lam
• [asterisk-users] How to avoid interruptions with DIGIUM   Moises Silva
• [asterisk-users] Asterisk SIP woes   Jason Hayer
• [asterisk-users] How to avoid interruptions with DIGIUM   Danny Dias
• [asterisk-users] 1.6.2.11 realtime sip registrations disappear from DB   Jonas Kellens
• [asterisk-users] How to avoid interruptions with DIGIUM   Miguel Molina
• [asterisk-users] problem with iax call (chan unavailable)   iscario at free.fr
• [asterisk-users] How to avoid interruptions with DIGIUM   Danny Dias
• [asterisk-users] Anyone can share their experience about Thomson TG784 wireless router/ATA?   bruce bruce
• [asterisk-users] Cisco or Linksys WRP400 reliability?   bruce bruce
• [asterisk-users] A way to check against a list of numbers?   Hose
• [asterisk-users] A way to check against a list of numbers?   Danny Nicholas
• [asterisk-users] A way to check against a list of numbers?   Roger Burton West
• [asterisk-users] A way to check against a list of numbers?   Steve Edwards
• [asterisk-users] A way to check against a list of numbers?   Steve Edwards
• [asterisk-users] A way to check against a list of numbers?   Jose P. Espinal
• [asterisk-users] Polycom dhcp boot   colin mcdermott
• [asterisk-users] Polycom dhcp boot   Joel Maslak
• [asterisk-users] SIP softphones answer but do not connect...   Carlos Chavez
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   bruce bruce
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Zeeshan Zakaria
• [asterisk-users] A way to check against a list of numbers?   Olivier
• [asterisk-users] A way to check against a list of numbers?   Faisal Hanif
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Faisal Hanif
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   John Novack
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Paul Belanger
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Zeeshan Zakaria
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Jeff LaCoursiere
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Jeff LaCoursiere
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Antonio Berrios
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Zeeshan Zakaria
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Zeeshan Zakaria
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Paul Belanger
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Zeeshan Zakaria
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Jeff LaCoursiere
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Zeeshan Zakaria
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Jeff LaCoursiere
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Warren Selby
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Rob Hillis
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Paul Belanger
• [asterisk-users] voicemail not working for all extensions in same way   AMARDEEP SINGH
• [asterisk-users] Asterisk voicemail server - gsm notifications   AMARDEEP SINGH
• [asterisk-users] asterisk as POC(push to talk) server?   AMARDEEP SINGH
• [asterisk-users] Polycom dhcp boot   colin mcdermott
• [asterisk-users] username mismatch with 1.6.2.11   Jonas Kellens
• [asterisk-users] Moving from DSL to T1   Richard Stuppi
• [asterisk-users] Moving from DSL to T1   Kyle Kienapfel
• [asterisk-users] Moving from DSL to T1   jon pounder
• [asterisk-users] username mismatch with 1.6.2.11   Kyle Kienapfel
• [asterisk-users] First boot asterisk -vvvvvgcn segfaults   Tom Lohmuller
• [asterisk-users] Synway cards   Anita Hall
• [asterisk-users] First boot asterisk -vvvvvgcn segfaults   Paul Belanger
• [asterisk-users] Polycom dhcp boot   Mark Deneen
• [asterisk-users] Moving from DSL to T1   Kevin Keane
• [asterisk-users] How to create a coredump for Asterisk   Thorolf Godawa
• [asterisk-users] How to create a coredump for Asterisk   dotnetdub
• [asterisk-users] SIP softphones answer but do not connect...   Matt Riddell
• [asterisk-users] Moving from DSL to T1   Steve Edwards
• [asterisk-users] problem with iax call (chan unavailable)   Matt Riddell
• [asterisk-users] Moving from DSL to T1   Kevin Keane
• [asterisk-users] doing dnsmgr_lookup   Jonas Kellens
• [asterisk-users] Moving from DSL to T1   Gordon Henderson
• [asterisk-users] Moving from DSL to T1   Hans Witvliet
• [asterisk-users] Moving from DSL to T1   Kevin Keane
• [asterisk-users] Correct queue agi syntax in 1.6.2.11   Jonas Kellens
• [asterisk-users] How to send SMS to Gigaset phones ?   Olivier
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Steve Davies
• [asterisk-users] High volume BLF - Suggestions?   Steve Davies
• [asterisk-users] Which voicemail file format is the most widely understood ?   Olivier
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Antonio Berrios
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Steve Davies
• [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtimemysql   Andrew Thomas
• [asterisk-users] High volume BLF - Suggestions?   Ron Arts
• [asterisk-users] Changing voicemail.conf file format list   Olivier
• [asterisk-users] High volume BLF - Suggestions?   Olivier
• [asterisk-users] Which voicemail file format is the most widely understood ?   Sebastian
• [asterisk-users] High volume BLF - Suggestions?   Steve Davies
• [asterisk-users] Changing voicemail.conf file format list   Sebastian
• [asterisk-users] High volume BLF - Suggestions?   Stefan Schmidt
• [asterisk-users] High volume BLF - Suggestions?   Steve Davies
• [asterisk-users] High volume BLF - Suggestions?   Andrew Thomas
• [asterisk-users] High volume BLF - Suggestions?   Matt Riddell
• [asterisk-users] How to send SMS to Gigaset phones ?   Gordon Henderson
• [asterisk-users] Moving from DSL to T1   Joel Maslak
• [asterisk-users] How to send SMS to Gigaset phones ?   Randy R
• [asterisk-users] How to send SMS to Gigaset phones ?   Olivier
• [asterisk-users] Problem with new AEX800 card dying because of interrupt problems   Benny Amorsen
• [asterisk-users] Force ip disconnect after register?   Bryant Zimmerman
• [asterisk-users] Changing voicemail.conf file format list   Tilghman Lesher
• [asterisk-users] High volume BLF - Suggestions?   Bryant Zimmerman
• [asterisk-users] Force ip disconnect after register?   Paul Belanger
• [asterisk-users] Force ip disconnect after register?   Roger Burton West
• [asterisk-users] SIP softphones answer but do not connect...   Carlos Chavez
• [asterisk-users] PostgreSQL is asterisk friendly with it?   Bryant Zimmerman
• [asterisk-users] Correct queue agi syntax in 1.6.2.11   Jonas Kellens
• [asterisk-users] SIP softphones answer but do not connect...   Mike
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Carlos Chavez
• [asterisk-users] PostgreSQL is asterisk friendly with it?   Danny Nicholas
• [asterisk-users] SIP softphones answer but do not connect...   Tarek Sawah
• [asterisk-users] Correct queue agi syntax in 1.6.2.11   Carlos Chavez
• [asterisk-users] PostgreSQL is asterisk friendly with it?   Vince Vielhaber
• [asterisk-users] Queue/Dial Recording - Capture answering channel name.   Steve Davies
• [asterisk-users] Force ip disconnect after register?   Barry Miller
• [asterisk-users] PostgreSQL is asterisk friendly with it?   Steve Kennedy
• [asterisk-users] doing dnsmgr_lookup   Ira
• [asterisk-users] PostgreSQL is asterisk friendly with it?   Sherwood McGowan
• [asterisk-users] How to send SMS to Gigaset phones ?   Anselm Martin Hoffmeister
• [asterisk-users] SIP softphones answer but do not connect...   Carlos Chavez
• [asterisk-users] High volume BLF - Suggestions?   Cassius Smith
• [asterisk-users] Correct queue agi syntax in 1.6.2.11   Jonas Kellens
• [asterisk-users] A way to check against a list of numbers?   Benny Amorsen
• [asterisk-users] PostgreSQL is asterisk friendly with it?   Benny Amorsen
• [asterisk-users] Correct queue agi syntax in 1.6.2.11   Roger Burton West
• [asterisk-users] Correct queue agi syntax in 1.6.2.11   Jonas Kellens
• [asterisk-users] doing dnsmgr_lookup   Jonas Kellens
• [asterisk-users] A way to check against a list of numbers?   Tarek Sawah
• [asterisk-users] Moving from DSL to T1   Hans Witvliet
• [asterisk-users] doing dnsmgr_lookup   Paul Belanger
• [asterisk-users] Moving from DSL to T1   Joel Maslak
• [asterisk-users] Asterisk 1.6 and fax   Stanislav Korsei
• [asterisk-users] Asterisk 1.6 and fax   David Backeberg
• [asterisk-users] Force ip disconnect after register?   Kevin P. Fleming
• [asterisk-users] Moving from DSL to T1   Kevin Keane
• [asterisk-users] Asterisk 1.6 and fax   Steve Underwood
• [asterisk-users] Moving from DSL to T1   Steve Underwood
• [asterisk-users] Which 1.6 subversion is Stable one?   Nikhil
• [asterisk-users] Polycom dhcp boot   Sebastien Thomas
• [asterisk-users] Speech To Text on linux with asterisk   DHAVAL INDRODIYA
• [asterisk-users] Speech To Text on linux with asterisk   Paul Belanger
• [asterisk-users] Speech To Text on linux with asterisk   DHAVAL INDRODIYA
• [asterisk-users] Speech To Text on linux with asterisk   Zeeshan Zakaria
• [asterisk-users] OT - Gigaset C470IP - How to access SMS settings   Olivier
• [asterisk-users] 3xx redirect response list Noop and capture   Adrian Estrada
• [asterisk-users] agi playback to execute say.conf settings   Ashik Ali
• [asterisk-users] Speech To Text on linux with asterisk   Nickolay V. Shmyrev
• [asterisk-users] Force ip disconnect after register?   Benny Amorsen
• [asterisk-users] Speech To Text on linux with asterisk   DHAVAL INDRODIYA
• [asterisk-users] Speech To Text on linux with asterisk   Nickolay V. Shmyrev
• [asterisk-users] can asterisk accept "anonymous" register ?   zhou tianjun
• [asterisk-users] Speech To Text on linux with asterisk   Zeeshan Zakaria
• [asterisk-users] Speech To Text on linux with asterisk   DHAVAL INDRODIYA
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] SPA3102 FAX not working   Gopalakrishnan A.N
• [asterisk-users] Random File Name   Dan Journo
• [asterisk-users] Speech To Text on linux with asterisk   Danny Nicholas
• [asterisk-users] Random File Name   Gareth Blades
• [asterisk-users] Which 1.6 subversion is Stable one?   Danny Nicholas
• [asterisk-users] agi playback to execute say.conf settings   Danny Nicholas
• [asterisk-users] Speech To Text on linux with asterisk   Zeeshan Zakaria
• [asterisk-users] SIP 800 Origination/Termination - International   Joe Freeman
• [asterisk-users] Random File Name   Zeeshan Zakaria
• [asterisk-users] Random File Name   Steve Edwards
• [asterisk-users] Random File Name   Steve Edwards
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   Zeeshan Zakaria
• [asterisk-users] Speech To Text on linux with asterisk   Paul Belanger
• [asterisk-users] question on asterisk 1.8 meetme   Jerry Geis
• [asterisk-users] High volume BLF - Suggestions?   Steve Davies
• [asterisk-users] can asterisk accept "anonymous" register ?   Paul Belanger
• [asterisk-users] PostgreSQL is asterisk friendly with it?   Edwin Quijada
• [asterisk-users] Random File Name   Dan Journo
• [asterisk-users] Random File Name   Dan Journo
• [asterisk-users] Random File Name   Zeeshan Zakaria
• [asterisk-users] Random File Name   Danny Nicholas
• [asterisk-users] sip show channels   Dan Journo
• [asterisk-users] Random File Name   Zeeshan Zakaria
• [asterisk-users] sip show channels   Danny Nicholas
• [asterisk-users] sip show channels   Dan Journo
• [asterisk-users] sip show channels   Zeeshan Zakaria
• [asterisk-users] sip show channels   Steve Howes
• [asterisk-users] sip show channels   Zeeshan Zakaria
• [asterisk-users] sip show channels   Danny Nicholas
• [asterisk-users] sip show channels   Dan Journo
• [asterisk-users] DTMF   Dan Journo
• [asterisk-users] Call Recording Questions   Dan Journo
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Steve Howes
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Paul Belanger
• [asterisk-users] DTMF   Paul Belanger
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Carlos Chavez
• [asterisk-users] DTMF   Dan Journo
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Danny Nicholas
• [asterisk-users] conf checkout   Danny Nicholas
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Danny Nicholas
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Tim Nelson
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Danny Nicholas
• [asterisk-users] conf checkout   Steve Edwards
• [asterisk-users] How different is implementing Cisco based system than Asterisk based system?   Zeeshan Zakaria
• [asterisk-users] How different is implementing Cisco based system than Asterisk based system?   David Backeberg
• [asterisk-users] How different is implementing Cisco based system than Asterisk based system?   Peder
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] How different is implementing Cisco based system than Asterisk based system?   David Backeberg
• [asterisk-users] How different is implementing Cisco based system than Asterisk based system?   Zeeshan Zakaria
• [asterisk-users] How different is implementing Cisco based system than Asterisk based system?   Dean Hoover
• [asterisk-users] How different is implementing Cisco based system than Asterisk based system?   Zeeshan Zakaria
• [asterisk-users] conf checkout   Shaun Ruffell
• [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?   bruce bruce
• [asterisk-users] How different is implementing Cisco based system than Asterisk based system?   Carlos Chavez
• [asterisk-users] OT - Gigaset C470IP - How to access SMS settings   asterisk asterisk
• [asterisk-users] about yahoo messager work with asterisk   qingquan luo
• [asterisk-users] Digest Username/auth name mismatch‏   t. k
• [asterisk-users] Speech To Text on linux with asterisk   DHAVAL INDRODIYA
• [asterisk-users] OT - Gigaset C470IP - How to access SMS settings   Olivier
• [asterisk-users] agi playback to execute say.conf settings   Ashik Ali
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] OT - Gigaset C470IP - How to access SMS settings   Randy R
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Skip Busy Agents/Channels from Queue   Shariq Khan
• [asterisk-users] One way audio when overlapdial is set to yes   leonimar cape
• [asterisk-users] Skip Busy Agents/Channels from Queue   Gareth Blades
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Philipp von Klitzing
• [asterisk-users] Skip Busy Agents/Channels from Queue   Shariq Khan
• [asterisk-users] Skip Busy Agents/Channels from Queue   Gareth Blades
• [asterisk-users] One way audio when overlapdial is set to yes   leonimar cape
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Gareth Blades
• [asterisk-users] Skip Busy Agents/Channels from Queue   Tarek Sawah
• [asterisk-users] Help me Out!!!!   Rob Fugina
• [asterisk-users] setting up phones   Gopalakrishnan A.N
• [asterisk-users] Help me Out!!!!   Pete
• [asterisk-users] Help me Out!!!!   Gareth Blades
• [asterisk-users] OT - Gigaset C470IP - How to access SMS settings   Olivier
• [asterisk-users] Help me Out!!!!   Rob Fugina
• [asterisk-users] Help me Out!!!!   --[ UxBoD ]--
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Help me Out!!!!   Pete
• [asterisk-users] Skip Busy Agents/Channels from Queue   Shariq Khan
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Philipp von Klitzing
• [asterisk-users] Synway cards   Shariq Khan
• [asterisk-users] Help me Out!!!!   Dan Journo
• [asterisk-users] Help me Out!!!!   Don Kelly
• [asterisk-users] Speech To Text on linux with asterisk   Nickolay V. Shmyrev
• [asterisk-users] Skip Busy Agents/Channels from Queue   Gareth Blades
• [asterisk-users] incoming call FXO   Flavio Miranda
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Help me Out!!!!   Doug Lytle
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Steve Howes
• [asterisk-users] incoming call FXO   Kevin P. Fleming
• [asterisk-users] SIP 800 Origination/Termination - International   Jeff LaCoursiere
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Philipp von Klitzing
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] OT - Gigaset C470IP - How to access SMS settings   Randy R
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Bruce Ferrell
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Zeeshan Zakaria
• [asterisk-users] incoming call FXO   Zeeshan Zakaria
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Help me Out!!!!   Cassius Smith
• [asterisk-users] OT - Gigaset C470IP - How to access SMS settings   asterisk asterisk
• [asterisk-users] Dual WAN with load balancing   asterisk asterisk
• [asterisk-users] Problems with audio   Danny Dias
• [asterisk-users] changing from zap to DAHDI   Jerry Geis
• [asterisk-users] Error loading skype_for_asterisk   Richard Kenner
• [asterisk-users] changing from zap to DAHDI   Danny Nicholas
• [asterisk-users] changing from zap to DAHDI   Shaun Ruffell
• [asterisk-users] Error loading skype_for_asterisk   Kevin P. Fleming
• [asterisk-users] Asterisk 1.4.36 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.12 Now Available   Asterisk Development Team
• [asterisk-users] changing from zap to DAHDI   Jerry Geis
• [asterisk-users] Problems with audio   Ishfaq Malik
• [asterisk-users] changing from zap to DAHDI   Jerry Geis
• [asterisk-users] changing from zap to DAHDI   Tzafrir Cohen
• [asterisk-users] Asterisk 1.6.2.12 Download   Ryan Wagoner
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Paul Belanger
• [asterisk-users] Dual WAN with load balancing   Luki
• [asterisk-users] Problems with audio   Danny Dias
• [asterisk-users] Asterisk 1.6.2.12 Download   Paul Belanger
• [asterisk-users] Asterisk not working with Festival   Mark G. Thomas
• [asterisk-users] SIP 800 Origination/Termination - International   Alex Bradley
• [asterisk-users] Problems with audio   Adrià Vidal
• [asterisk-users] changing from zap to DAHDI   Shaun Ruffell
• [asterisk-users] changing from zap to DAHDI   Shaun Ruffell
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Leif Madsen
• [asterisk-users] Problems with audio   Sebastian
• [asterisk-users] Asterisk 1.6.2.12 Download   Leif Madsen
• [asterisk-users] changing from zap to DAHDI   Jerry Geis
• [asterisk-users] Problems with audio   Danny Dias
• [asterisk-users] Problems with audio   Danny Dias
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   jon pounder
• [asterisk-users] Digest Username/auth name mismatch‏   Sebastian
• [asterisk-users] DTMF   Sebastian
• [asterisk-users] incoming call FXO   Flavio Miranda
• [asterisk-users] changing from zap to DAHDI   Shaun Ruffell
• [asterisk-users] Skip Busy Agents/Channels from Queue   Shariq Khan
• [asterisk-users] Skip Busy Agents/Channels from Queue   Danny Nicholas
• [asterisk-users] SPA3102 FAX not working   Gopalakrishnan A.N
• [asterisk-users] Asterisk 1.6.2.13 Now Available (Re-Releast of 1.6.2.12)   Asterisk Development Team
• [asterisk-users] changing from zap to DAHDI   Jerry Geis
• [asterisk-users] Skip Busy Agents/Channels from Queue   Philipp von Klitzing
• [asterisk-users] SIP 800 Origination/Termination - International   Jamie A. Stapleton
• [asterisk-users] Queue member status not changing   Justin Sherrill
• [asterisk-users] Queue member status not changing   Danny Nicholas
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] SIP 800 Origination/Termination - International   Kyle Kienapfel
• [asterisk-users] Bug with Realtime?   Danny Nicholas
• [asterisk-users] changing from zap to DAHDI   Shaun Ruffell
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Bug with Realtime?   Jonas Kellens
• [asterisk-users] Call Recording Questions   Dan Journo
• [asterisk-users] Bug with Realtime?   Leif Madsen
• [asterisk-users] Call Recording Questions   Sebastian
• [asterisk-users] SIP 800 Origination/Termination - International   Jeff LaCoursiere
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Alec Davis
• [asterisk-users] Echo on Sangoma A400 and background noise   Al lists
• [asterisk-users] Echo on Sangoma A400 and background noise   Tim Nelson
• [asterisk-users] Bug with Realtime?   Zeeshan Zakaria
• [asterisk-users] Purpose of qualify=yes   Chris Owen
• [asterisk-users] a2billing   Flavio Miranda
• [asterisk-users] Purpose of qualify=yes   Steve Totaro
• [asterisk-users] Echo on Sangoma A400 and background noise   cb
• [asterisk-users] Speech To Text on linux with asterisk   DHAVAL INDRODIYA
• [asterisk-users] Speech To Text on linux with asterisk   Nickolay V. Shmyrev
• [asterisk-users] asterisk 1.6 and BLF   Jonas Kellens
• [asterisk-users] Speech To Text on linux with asterisk   DHAVAL INDRODIYA
• [asterisk-users] Speech To Text on linux with asterisk   Nickolay V. Shmyrev
• [asterisk-users] a2billing   César Pinto Magán
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Configure Asterisk with openssl   Nikhil
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Philipp von Klitzing
• [asterisk-users] Configure Asterisk with openssl   A J Stiles
• [asterisk-users] a2billing   Vardan Harutyunyan
• [asterisk-users] Realtime semi-colon   Andrew Thomas
• [asterisk-users] Configure Asterisk with openssl   Nikhil
• [asterisk-users] How to Understand a pri intense debug span X   Danny Dias
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Configure Asterisk with openssl   Nikhil
• [asterisk-users] Realtime semi-colon   Steve Howes
• [asterisk-users] DTMF tones too long, for once   Justin Sherrill
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Dual WAN with load balancing   asterisk asterisk
• [asterisk-users] changing from zap to DAHDI   Jerry Geis
• [asterisk-users] Bug with Realtime?   Danny Nicholas
• [asterisk-users] Bug with Realtime?   Peder
• [asterisk-users] changing from zap to DAHDI   Jerry Geis
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Bug with Realtime?   John Novack
• [asterisk-users] Indications and tonelist on a SIP channel..   Carlos C.
• [asterisk-users] Bug with Realtime?   Danny Nicholas
• [asterisk-users] Bug with Realtime?   Danny Nicholas
• [asterisk-users] Echo on Sangoma A400 and background noise   Moises Silva
• [asterisk-users] Purpose of qualify=yes   Benny Amorsen
• [asterisk-users] Bug with Realtime?   Zeeshan Zakaria
• [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?   Tim Nelson
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Purpose of qualify=yes   Steve Totaro
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Paul Belanger
• [asterisk-users] Purpose of qualify=yes   Zeeshan Zakaria
• [asterisk-users] Bug with Realtime?   Danny Nicholas
• [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?   Danny Nicholas
• [asterisk-users] Purpose of qualify=yes   Chris Owen
• [asterisk-users] Purpose of qualify=yes   Chris Owen
• [asterisk-users] Purpose of qualify=yes   jon pounder
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Bug with Realtime?   Peder
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Purpose of qualify=yes   Peder
• [asterisk-users] Bug with Realtime?   Danny Nicholas
• [asterisk-users] Purpose of qualify=yes   Gareth Blades
• [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?   A J Stiles
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Paul Belanger
• [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?   Paul Belanger
• [asterisk-users] Help!! Call waiting issue   carem gyssell nieto
• [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?   Tim Nelson
• [asterisk-users] Realtime semi-colon   Tilghman Lesher
• [asterisk-users] Bug with Realtime?   Tilghman Lesher
• [asterisk-users] Purpose of qualify=yes   Steve Totaro
• [asterisk-users] Help!! Call waiting issue   Paul Belanger
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Paul Belanger
• [asterisk-users] one way audio for xlite clients behind NAT   Thomas Johnson
• [asterisk-users] Bug with Realtime?   Leif Madsen
• [asterisk-users] one way audio for xlite clients behind NAT   Sebastian
• [asterisk-users] AGI Delimiter in 1.6   Jon Farmer
• [asterisk-users] AGI Delimiter in 1.6   Danny Nicholas
• [asterisk-users] one way audio for xlite clients behind NAT   Thomas Johnson
• [asterisk-users] Purpose of qualify=yes   Sebastian
• [asterisk-users] AGI Delimiter in 1.6   Jon Farmer
• [asterisk-users] AGI Delimiter in 1.6   Danny Nicholas
• [asterisk-users] AGI Delimiter in 1.6   Jon Farmer
• [asterisk-users] one way audio for xlite clients behind NAT   Sebastian
• [asterisk-users] AGI Delimiter in 1.6   James A. Shigley
• [asterisk-users] AGI Delimiter in 1.6   Barry Miller
• [asterisk-users] one way audio for xlite clients behind NAT   Thomas Johnson
• [asterisk-users] one way audio for xlite clients behind NAT   Paul Belanger
• [asterisk-users] one way audio for xlite clients behind NAT   Thomas Johnson
• [asterisk-users] one way audio for xlite clients behind NAT   Paul Belanger
• [asterisk-users] one way audio for xlite clients behind NAT   Flavio Miranda
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] Deadlock rendering sip useless   Ingmar Steen
• [asterisk-users] changing from zap to DAHDI   Tzafrir Cohen
• [asterisk-users] Attended Transfer does not release channels   Wolfgang Pichler
• [asterisk-users] Issue with transfer (sip)   Benoit
• [asterisk-users] Sangoma A108 PCIe 2.0   Anita Hall
• [asterisk-users] Attended Transfer does not release channels   Olivier
• [asterisk-users] Issue with transfer (sip)   Olivier
• [asterisk-users] Sangoma A108 PCIe V2.0   Anita Hall
• [asterisk-users] Call restriction for particular extension   Gopalakrishnan A.N
• [asterisk-users] Determine busy state   unserossi at aol.com
• [asterisk-users] Sangoma A108 PCIe V2.0   John Novack
• [asterisk-users] Sangoma A108 PCIe V2.0   Geraint Lee
• [asterisk-users] Attended Transfer does not release channels   Wolfgang Pichler
• [asterisk-users] Realtime semi-colon   Andrew Thomas
• [asterisk-users] How to Understand a pri intense debug span X   Danny Dias
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Call restriction for particular extension   Danny Nicholas
• [asterisk-users] Not able to join conference   khalid touati
• [asterisk-users] Sangoma A108 PCIe V2.0   John Novack
• [asterisk-users] Initial Audio Cut off   Ujjval Karihaloo
• [asterisk-users] Initial Audio Cut off   Danny Nicholas
• [asterisk-users] Sangoma A108 PCIe V2.0   Nyamul Hassan
• [asterisk-users] ISDN BRI call disconnection issue   Gopalakrishnan A.N
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Mark Deneen
• [asterisk-users] need help with IVR dialplan   haloha
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Mark Deneen
• [asterisk-users] Not able to join conference   Paul Belanger
• [asterisk-users] Asterisk 1.8 and CEL logging   Bryant Zimmerman
• [asterisk-users] Not able to join conference   Danny Nicholas
• [asterisk-users] CallerId: behavior changed between 1.4.25.1 and 1.4.36 with .call files   Antonio Moragues
• [asterisk-users] do carriers detect unusual / unauthorized VoIP calling patterns?   Jeff Brower
• [asterisk-users] Rotary phone on Asterisk   Joel Maslak
• [asterisk-users] Rotary phone on Asterisk   Danny Nicholas
• [asterisk-users] do carriers detect unusual / unauthorized VoIP calling patterns?   C F
• [asterisk-users] Initial Audio Cut off   C F
• [asterisk-users] Initial Audio Cut off   Danny Nicholas
• [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11   Jonas Kellens
• [asterisk-users] 3rd party app store   Dean Collins
• [asterisk-users] 5-7 second delay in connecting outgoing FXO calls   Frank Tarczynski
• [asterisk-users] Initial Audio Cut off   Ujjval Karihaloo
• [asterisk-users] Initial Audio Cut off   Ujjval Karihaloo
• [asterisk-users] Initial Audio Cut off   Lyle McKarns
• [asterisk-users] quick 1.8 question on console/dsp   Jerry Geis
• [asterisk-users] Rotary phone on Asterisk   John Novack
• [asterisk-users] Not able to join conference   khalid touati
• [asterisk-users] 3rd party app store   Tilghman Lesher
• [asterisk-users] Rotary phone on Asterisk   Joel Maslak
• [asterisk-users] Registration attempts   dave george
• [asterisk-users] Registration attempts   Fred Posner
• [asterisk-users] 3rd party app store   Dean Collins
• [asterisk-users] Initial Audio Cut off   Ujjval Karihaloo
• [asterisk-users] Registration attempts   Zeeshan Zakaria
• [asterisk-users] 3rd party app store   Tilghman Lesher
• [asterisk-users] externip/localnet   dotnetdub
• [asterisk-users] 3rd party app store   Dean Collins
• [asterisk-users] Asterisk sip attack   bayardo.sanchez at gmail.com
• [asterisk-users] 3rd party app store   Tilghman Lesher
• [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?   Olivier CALVANO
• [asterisk-users] Determine busy state   unserossi at aol.com
• [asterisk-users] Sangoma A108 PCIe V2.0   Moises Silva
• [asterisk-users] Registration attempts   dave george
• [asterisk-users] Sangoma A108 PCIe V2.0   Nyamul Hassan
• [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?   Paul Belanger
• [asterisk-users] Determine busy state   Paul Belanger
• [asterisk-users] 3rd party app store   Mark Deneen
• [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?   Olivier CALVANO
• [asterisk-users] 3rd party app store   Darren Nickerson
• [asterisk-users] externip/localnet   Carlos Chavez
• [asterisk-users] Sangoma A108 PCIe V2.0   Moises Silva
• [asterisk-users] 3rd party app store   Steve Underwood
• [asterisk-users] 3rd party app store   Kevin P. Fleming
• [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..   dashy dude
• [asterisk-users] Asterisk sip attack   Gareth Blades
• [asterisk-users] Not able to join conference   Andrew Thomas
• [asterisk-users] Confused about notifyringing in sip.conf   Jonas Kellens
• [asterisk-users] Confused about notifyringing in sip.conf   Philipp von Klitzing
• [asterisk-users] Extension continues ringing after caller hanged up   Arie Skliarouk
• [asterisk-users] Confused about notifyringing in sip.conf   unserossi at aol.com
• [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..   dotnetdub
• [asterisk-users] Playing Audio To One Channel   Jon Farmer
• [asterisk-users] Not able to join conference   khalid touati
• [asterisk-users] Playing Audio To One Channel   Jim Dickenson
• [asterisk-users] Playing Audio To One Channel   Danny Nicholas
• [asterisk-users] Playing Audio To One Channel   Jon Farmer
• [asterisk-users] Extension continues ringing after caller hanged up   Zeeshan Zakaria
• [asterisk-users] Playing Audio To One Channel   Jon Farmer
• [asterisk-users] Extension continues ringing after caller hanged up   Arie Skliarouk
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?   Olivier CALVANO
• [asterisk-users] Extension continues ringing after caller hanged up   Zeeshan Zakaria
• [asterisk-users] Bug with Realtime?   Peder
• [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?   Paul Belanger
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Setting 'fname_base' variable doesn't affect 'automon' result file.   Jose P. Espinal
• [asterisk-users] Extension continues ringing after caller hanged up   Paul Belanger
• [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?   Tim Nelson
• [asterisk-users] Asterisk stops processing SIP UDP messages   Daniel Tryba
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Authentication best practice   Roger Burton West
• [asterisk-users] Bug with Realtime?   Roger Burton West
• [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?   Jonathan Thurman
• [asterisk-users] Authentication best practice   Marino Punturieri
• [asterisk-users] Setting 'fname_base' variable doesn't affect 'automon' result file.   Tilghman Lesher
• [asterisk-users] Commands needed via AMI to find callerid of inbound call to extension   Gavin Henry
• [asterisk-users] Extension continues ringing after caller hanged up   Arie Skliarouk
• [asterisk-users] 3rd party app store   Rod Montgomery
• [asterisk-users] Setting 'fname_base' variable doesn't affect 'automon' result file.   Jose P. Espinal
• [asterisk-users] changing from zap to DAHDI   dotnetdub
• [asterisk-users] 3rd party app store   Dean Collins
• [asterisk-users] 3rd party app store   Matt Riddell
• [asterisk-users] Asterisk News Accepting Submissions   Matt Riddell
• [asterisk-users] 3rd party app store   Rod Montgomery
• [asterisk-users] Digest Username/auth name mismatch‏   t. k
• [asterisk-users] 3rd party app store   Paul Belanger
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] Dialplan extension pattern matching for '/' character   RAJNIKANT VANZA
• [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..   dashy dude
• [asterisk-users] Not able to join conference   Andrew Thomas
• [asterisk-users] Not able to join conference   khalid touati
• [asterisk-users] 3rd party app store   Sebastian
• [asterisk-users] Digest Username/auth name mismatch‏   Sebastian
• [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..   Ondrej Škopek
• [asterisk-users] 3rd party app store   Lyle McKarns
• [asterisk-users] Unexplained message in 1.6.2   CDR
• [asterisk-users] Unexplained message in 1.6.2   Danny Nicholas
• [asterisk-users] AGI Delimiter in 1.6   Jon Farmer
• [asterisk-users] AGI Delimiter in 1.6   Jonas Kellens
• [asterisk-users] Unexplained message in 1.6.2   Tilghman Lesher
• [asterisk-users] func SHARED, how to use?   Philipp von Klitzing
• [asterisk-users] 3rd party app store   Cassius Smith
• [asterisk-users] random hangups on RBS T1   Jeff LaCoursiere
• [asterisk-users] Unexplained message in 1.6.2   CDR
• [asterisk-users] random hangups on RBS T1   Shaun Ruffell
• [asterisk-users] Polycom dhcp boot   Thomas Mullins
• [asterisk-users] random hangups on RBS T1   Jeff LaCoursiere
• [asterisk-users] digits in chan_dahdi   Marcus Vinicius
• [asterisk-users] digits in chan_dahdi   Richard Kenner
• [asterisk-users] Mixing ISDN and R2 in the same card...   Carlos Chavez
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] Res: digits in chan_dahdi   Marcus Vinicius
• [asterisk-users] Res: digits in chan_dahdi   Richard Kenner
• [asterisk-users] Res: digits in chan_dahdi   Shaun Ruffell
• [asterisk-users] Bug with Realtime?   Carlos Chavez
• [asterisk-users] Bug with Realtime?   Dan Journo
• [asterisk-users] AGI Delimiter in 1.6   Jon Farmer
• [asterisk-users] Solving the CDR mess of attended transfers   Steve Murphy
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   bruce bruce
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   Roger Burton West
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] Cross compile Asterisk for mipsel-linux   Nikhil
• [asterisk-users] Unable to open vm-INBOXs   Jonas Kellens
• [asterisk-users] Unable to open vm-INBOXs   Roger Burton West
• [asterisk-users] func SHARED, how to use?   Philipp von Klitzing
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] func SHARED, how to use?   Andrea Cristofanini
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] Unable to open vm-INBOXs   Watkins, Bradley
• [asterisk-users] T38 and codecs negotiation   federico cabiddu
• [asterisk-users] func SHARED, how to use?   Philipp von Klitzing
• [asterisk-users] Unable to open vm-INBOXs   Jonas Kellens
• [asterisk-users] Unable to open vm-INBOXs   Philipp von Klitzing
• [asterisk-users] Unable to open vm-INBOXs   Watkins, Bradley
• [asterisk-users] Unable to open vm-INBOXs   Jonas Kellens
• [asterisk-users] Unable to open vm-INBOXs   Watkins, Bradley
• [asterisk-users] Solving the CDR mess of attended transfers   Steve Murphy
• [asterisk-users] Can't cross compile asterisk 1.6.2.13 on arm using ltib   IMS
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   bruce bruce
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   Paul Belanger
• [asterisk-users] Cross compile Asterisk for mipsel-linux   Paul Belanger
• [asterisk-users] Can't cross compile asterisk 1.6.2.13 on arm using ltib   Paul Belanger
• [asterisk-users] T38 and codecs negotiation   Paul Belanger
• [asterisk-users] Asterisk T38   Adam Moffett
• [asterisk-users] Costa Rica Hangup Detection   Gustavo A. Gonzalez
• [asterisk-users] Costa Rica Hangup Detection   Paul Belanger
• [asterisk-users] Asterisk T38   Kevin P. Fleming
• [asterisk-users] Asterisk T38   David Backeberg
• [asterisk-users] TLS re-negotiation attack on SIP/TLS of Asterisk?   Fabio Pietrosanti (naif)
• [asterisk-users] Asterisk T38   Adam Moffett
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   Carlos Chavez
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Klaus Darilion
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Paul Belanger
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Steve Howes
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Gareth Blades
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Kevin P. Fleming
• [asterisk-users] Solving the CDR mess of attended transfers   Lenz Emilitri
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Jose P. Espinal
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Steve Edwards
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Kevin P. Fleming
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Jose P. Espinal
• [asterisk-users] Asterisk as a distributed paging system   Matteo Fortini
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Steve Edwards
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Leif Madsen
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   bruce bruce
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   bruce bruce
• [asterisk-users] http://www.asterisk.org/downloads naming schema   Barry Miller
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   Steve Edwards
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   bruce bruce
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   Paul Belanger
• [asterisk-users] Sangoma A500 NT BRI PTMP without woomera on asterisk 1.6   Marco Kühnel
• [asterisk-users] Asterisk as a distributed paging system   Gordon Henderson
• [asterisk-users] Asterisk as a distributed paging system   Danny Nicholas
• [asterisk-users] Asterisk as a distributed paging system   Philipp von Klitzing
• [asterisk-users] Asterisk- speech to text(Voicemail to text message)   amit salunkhe
• [asterisk-users] Asterisk- speech to text(Voicemail to text message)   Danny Nicholas
• [asterisk-users] Recording maximum time and stop on silence   David Cunningham
• [asterisk-users] Recording maximum time and stop on silence   Danny Nicholas
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?   bruce bruce
• [asterisk-users] Installing Asterisk + FreePBX from Repsitory spits out some warnings and errors for ever   bruce bruce
• [asterisk-users] Calls stuck in the queue even when ext's are available   das sandesh
• [asterisk-users] Record() Cmd and My SQL   Govind, Mahesh (NSN - IN/Bangalore)
• [asterisk-users] Can't cross compile asterisk 1.6.2.13 on arm using ltib   IMS
• [asterisk-users] Recording maximum time and stop on silence   David Cunningham
• [asterisk-users] CDR display in minute   Mickael MONSIEUR
• [asterisk-users] CDR display in minute   Faisal Hanif
• [asterisk-users] CDR display in minute   Mickael MONSIEUR
• [asterisk-users] realm: security issue   bilal ghayyad
• [asterisk-users] Asterisk and Digium TC400B   Tarek Sawah
• [asterisk-users] func SHARED, how to use?   Philipp von Klitzing
• [asterisk-users] Record() Cmd and My SQL   Gopalakrishnan A.N
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] func SHARED, how to use?   Philipp von Klitzing
• [asterisk-users] Record() Cmd and My SQL   Govind, Mahesh (NSN - IN/Bangalore)
• [asterisk-users] Record() Cmd and My SQL   Gopalakrishnan A.N
• [asterisk-users] Record() Cmd and My SQL   Danny Nicholas
• [asterisk-users] Asterisk and Digium TC400B   Tim Nelson
• [asterisk-users] Asterisk T38   Matt Watson
• [asterisk-users] Net2Phone SIP trunk problem   Alejandro Cabrera Obed
• [asterisk-users] Can't turn debug on in a 1.2 box   khalid touati
• [asterisk-users] Net2Phone SIP trunk problem   Gopalakrishnan A.N
• [asterisk-users] Record() Cmd and My SQL   Govind, Mahesh (NSN - IN/Bangalore)
• [asterisk-users] Sip from ip address   Geraint Lee
• [asterisk-users] realm: security issue   Zeeshan Zakaria
• [asterisk-users] Asterisk and Digium TC400B   Bryant Zimmerman
• [asterisk-users] Asterisk and Digium TC400B   Shaun Ruffell
• [asterisk-users] Record() Cmd and My SQL   Danny Nicholas
• [asterisk-users] Forking a call   Mike
• [asterisk-users] Record() Cmd and My SQL   David Backeberg
• [asterisk-users] Asterisk Transfer/call patching support   Dan Cropp
• [asterisk-users] Digest Username/auth name mismatch‏   t. k
• [asterisk-users] Digest Username/auth name mismatch‏   Steve Howes
• [asterisk-users] Asterisk- speech to text(Voicemail to text message)   Justin Sherrill
• [asterisk-users] Asterisk and Digium TC400B   Bryant Zimmerman
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)   Dave Platt
• [asterisk-users] Asterisk- speech to text(Voicemail totext message)   Danny Nicholas
• [asterisk-users] Unable to make outgoing call on E1   Adolphe Cher-aime
• [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available   Asterisk Development Team
• [asterisk-users] realm: security issue   bilal ghayyad
• [asterisk-users] realm: security issue   Tarek Sawah
• [asterisk-users] rtp problem with 1.8.0-rdc1   covici at ccs.covici.com
• [asterisk-users] Asterisk 1.6.2.13 - have asterisk reply from same IP address   Mike
• [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)   bruce bruce
• [asterisk-users] Record() Cmd and My SQL   Govind, Mahesh (NSN - IN/Bangalore)
• [asterisk-users] Record() Cmd and My SQL   John Taylor
• [asterisk-users] How to test BIG traffic through DAHDI/WANPIPE interfaces   Danny Dias
• [asterisk-users] tcpdump auto stats script   John Taylor
• [asterisk-users] RDNIS not passed from one box to another with BRI access   Olivier
• [asterisk-users] Record() Cmd and My SQL   Govind, Mahesh (NSN - IN/Bangalore)
• [asterisk-users] tcpdump auto stats script   Philipp von Klitzing
• [asterisk-users] Fax On Demand - Asterisk 1.4.29   Zoel Hairi - Yahoo
• [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces   Ingmar Steen
• [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces   Danny Dias
• [asterisk-users] Fax On Demand - Asterisk 1.4.29   Tarek Sawah
• [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces   Gareth Blades
• [asterisk-users] rtp problem with 1.8.0-rdc1   Leif Madsen
• [asterisk-users] Asterisk 1.6.2.13 - have asterisk reply from same IP address   Leif Madsen
• [asterisk-users] How to test BIG traffic throughDAHDI/WANPIPEinterfaces   Ingmar Steen
• [asterisk-users] rtp problem with 1.8.0-rdc1   Benny Amorsen
• [asterisk-users] Redirecting a Channel more than three times...   Yves A.
• [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces   Danny Dias
• [asterisk-users] rtp problem with 1.8.0-rdc1   covici at ccs.covici.com
• [asterisk-users] rtp problem with 1.8.0-rdc1   covici at ccs.covici.com
• [asterisk-users] Redirecting a Channel more than three times...   Danny Nicholas
• [asterisk-users] Fwd: Can't cross compile asterisk 1.6.2.13 on arm using ltib   IMS
• [asterisk-users] Fax On Demand - Asterisk 1.4.29   Zoel Hairi - Yahoo
• [asterisk-users] differential billing   Abdul Basit
• [asterisk-users] differential billing   Danny Nicholas
• [asterisk-users] [asterisk-pakistan] differential billing   A.R. Nasir Qureshi
• [asterisk-users] Fwd: Can't cross compile asterisk 1.6.2.13 on arm using ltib   Paul Belanger
• [asterisk-users] rtp problem with 1.8.0-rdc1   Lyle Giese
• [asterisk-users] differential billing   Abdul Basit
• [asterisk-users] Fwd: Can't cross compile asterisk 1.6.2.13 on arm using ltib   IMS
• [asterisk-users] Redirecting a Channel more than three times...   Yves A.
• [asterisk-users] rtp problem with 1.8.0-rdc1   covici at ccs.covici.com
• [asterisk-users] differential billing   Danny Nicholas
• [asterisk-users] rtp problem with 1.8.0-rdc1   Philipp von Klitzing
• [asterisk-users] rtp problem with 1.8.0-rdc1   covici at ccs.covici.com
• [asterisk-users] Debug compile fails   Daniel Tryba
• [asterisk-users] should trixbox system hang when ISP drops connection?   Robert P. J. Day
• [asterisk-users] should trixbox system hang when ISP dropsconnection?   Danny Nicholas
• [asterisk-users] should trixbox system hang when ISP drops connection?   Warren Selby
• [asterisk-users] should trixbox system hang when ISP dropsconnection?   Steve Howes
• [asterisk-users] should trixbox system hang when ISP drops connection?   Zeeshan Zakaria
• [asterisk-users] should trixbox system hang when ISP drops connection?   Robert P. J. Day
• [asterisk-users] Record() Cmd and My SQL   David Backeberg
• [asterisk-users] differential billing   Tarek Sawah
• [asterisk-users] differential billing   Tarek Sawah
• [asterisk-users] best format for playback/generation   Danny Nicholas
• [asterisk-users] best format for playback/generation   Gareth Blades
• [asterisk-users] Record() Cmd and My SQL   Don Kelly
• [asterisk-users] best format for playback/generation   Zeeshan Zakaria
• [asterisk-users] Losing local SIP phones when internet goes down?   Gopalakrishnan A.N
• [asterisk-users] Record() Cmd and My SQL   David Backeberg
• [asterisk-users] Losing local SIP phones when internet goes down?   Zeeshan Zakaria
• [asterisk-users] Record() Cmd and My SQL   Danny Nicholas
• [asterisk-users] Record() Cmd and My SQL   Don Kelly
• [asterisk-users] Debug compile fails   Paul Belanger
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Mike
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Danny Nicholas
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Andrew Latham
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Mike
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Danny Nicholas
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Mike
• [asterisk-users] rtp problem with 1.8.0-rdc1   Benny Amorsen
• [asterisk-users] Can't turn debug on in a 1.2 box   Paul Belanger
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Shaun Ruffell
• [asterisk-users] Can't turn debug on in a 1.2 box   Steve Edwards
• [asterisk-users] rtp problem with 1.8.0-rdc1   covici at ccs.covici.com
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Mike
• [asterisk-users] can call internal branch , but can not call external numbers with avaya , always get return message : Q931IncompatibleDestination   Thomas Liu
• [asterisk-users] Asterisk- speech to text(Voicemail totext message)   Nickolay V. Shmyrev
• [asterisk-users] Losing local SIP phones when internet goes down?   Gopalakrishnan A.N
• [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available   Ira
• [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available   Barry Miller
• [asterisk-users] asterisk-users at lists.digium.com September 67% OFF   VIAGRA ® Online
• [asterisk-users] Asterisk Cluster Scenario   Stefano Sasso
• [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available   Andrew Latham
• [asterisk-users] differential billing   Abdul Basit
• [asterisk-users] differential billing   Tarek Sawah
• [asterisk-users] differential billing   Abdul Basit
• [asterisk-users] differential billing   Tarek Sawah
• [asterisk-users] differential billing   Don Kelly
• [asterisk-users] differential billing   Faisal Hanif
• [asterisk-users] Asterisk Redundancy   Dan Journo
• [asterisk-users] Asterisk Redundancy   Michelle Dupuis
• [asterisk-users] Asterisk Redundancy   Adolphe Cher-Aime
• [asterisk-users] 1.4, 1.6, 1.8 versions and the H323 gatekeeper functionality   bilal ghayyad
• [asterisk-users] Downloading the Asterisk as tar.gz file   bilal ghayyad
• [asterisk-users] Downloading the Asterisk as tar.gz file   Tilghman Lesher
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   bruce bruce
• [asterisk-users] Asterisk ODBC Insert issue   Neeraj Chand
• [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available   Ira
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   dotnetdub
• [asterisk-users] Asterisk ODBC Insert issue   Tarek Sawah
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   Dmitry Nedospasov
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   Dmitry Nedospasov
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   Steve Totaro
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   Hans Witvliet
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Shaun Ruffell
• [asterisk-users] A2billing   Flavio Miranda
• [asterisk-users] Record() Cmd and My SQL   Govind, Mahesh (NSN - IN/Bangalore)
• [asterisk-users] misc newbie VoIP questions   Rogelio
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   Tilghman Lesher
• [asterisk-users] Asterisk ODBC Insert issue   Tilghman Lesher
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] misc newbie VoIP questions   Randy R
• [asterisk-users] PSTN to SMS and SMS to PSTN   Mian Asif
• [asterisk-users] RFC3329 support in Asterisk   sijan ahamed
• [asterisk-users] A2billing   Vardan Harutyunyan
• [asterisk-users] Debug compile fails   Daniel Tryba
• [asterisk-users] A2billing   Flavio Miranda
• [asterisk-users] func SHARED, how to use?   Philipp von Klitzing
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   Roberto Piola
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   Michael Graves
• [asterisk-users] Asterisk and dahdi on Arch linux   Christian
• [asterisk-users] Asterisk Redundancy   Benny Amorsen
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   Benny Amorsen
• [asterisk-users] RFC3329 support in Asterisk   Paul Belanger
• [asterisk-users] Record() Cmd and My SQL   David Backeberg
• [asterisk-users] groupcount - show usage   marek cervenka
• [asterisk-users] Asterisk Redundancy   Michelle Dupuis
• [asterisk-users] Asterisk as a distributed paging system   Matteo Fortini
• [asterisk-users] propagate sip reinvites with directrtpsetup=yes   Eugene Oden
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] propagate sip reinvites with directrtpsetup=yes   Kevin P. Fleming
• [asterisk-users] Problems compiling Asterisk on Debian   Dean Hoover
• [asterisk-users] Problems compiling Asterisk on Debian   Daniel Tryba
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] Problems compiling Asterisk on Debian   Roger Burton West
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] propagate sip reinvites with directrtpsetup=yes   Eugene Oden
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] Problems compiling Asterisk on Debian   Daniel Tryba
• [asterisk-users] Problems compiling Asterisk on Debian   Paul Belanger
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] Asterisk Redundancy   Vahan Yerkanian
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] Problems compiling Asterisk on Debian   Paul Belanger
• [asterisk-users] Asterisk Redundancy   Michelle Dupuis
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Dave Platt
• [asterisk-users] Asterisk Redundancy   Fred Posner
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes   Mike
• [asterisk-users] SCCP (skinny) phone behind NAT: RTP dest addr wrong   Infra
• [asterisk-users] How to pick a codec on the fly   Danny Nicholas
• [asterisk-users] How to pick a codec on the fly   Daniel Tryba
• [asterisk-users] How to pick a codec on the fly   Danny Nicholas
• [asterisk-users] Asterisk Redundancy   Tarek Sawah
• [asterisk-users] How to pick a codec on the fly   Tarek Sawah
• [asterisk-users] How to pick a codec on the fly   Danny Nicholas
• [asterisk-users] Can't turn debug on in a 1.2 box   khalid touati
• [asterisk-users] How to pick a codec on the fly   Danny Nicholas
• [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...   bruce bruce
• [asterisk-users] 1.4, 1.6, 1.8 versions and the H323 gatekeeper functionality   Leif Madsen
• [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available   Leif Madsen
• [asterisk-users] Asterisk as a distributed paging system   Sebastian
• [asterisk-users] Asterisk Redundancy   Dan Journo
• [asterisk-users] Asterisk Redundancy   Dan Journo
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] Problems compiling Asterisk on Debian   Jim Dickenson
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] Problems compiling Asterisk on Debian   Paul Belanger
• [asterisk-users] Problems compiling Asterisk on Debian   Paul Belanger
• [asterisk-users] Problems compiling Asterisk on Debian   Jim Dickenson
• [asterisk-users] Problems compiling Asterisk on Debian   Paul Belanger
• [asterisk-users] NAT issue (i think?)   Ron
• [asterisk-users] SCCP (skinny) phone behind NAT: RTP dest addr wrong   Infra
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] AstLinux 0.7.3 released   Darrick Hartman
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - FIXED...?   Mike
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Mike
• [asterisk-users] 1.4, 1.6, 1.8 versions and the H323 Gatekeeper functionality   bilal ghayyad
• [asterisk-users] 1.6 and 1.8 version & A2Billing   bilal ghayyad
• [asterisk-users] Problems compiling Asterisk on Debian   Danny Dias
• [asterisk-users] Cisco 9971   Damian Turburville
• [asterisk-users] NAT issue (i think?)   Daniel Tryba
• [asterisk-users] ISDN - Busy signal on 3rd call   Paulo Santos
• [asterisk-users] NAT issue (i think?)   Ron
• [asterisk-users] Inbound calls from TRUNK   Khaled W. Chehab
• [asterisk-users] NAT issue (i think?)   Danny Dias
• [asterisk-users] func SHARED, how to use?   Philipp von Klitzing
• [asterisk-users] What's the meaning of this?   Danny Dias
• [asterisk-users] SIP X.25   Daviramos Roussenq Fortunato
• [asterisk-users] NAT issue (i think?)   Daniel Tryba
• [asterisk-users] SIP X.25   Daviramos Roussenq Fortunato
• [asterisk-users] NAT issue (i think?)   Ron
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] SIP X.25   Kevin P. Fleming
• [asterisk-users] func SHARED, how to use?   Dmitry Melekhov
• [asterisk-users] E1 check with nagios, how to?   Dario Quiroz
• [asterisk-users] What's the meaning of this?   Doug Lytle
• [asterisk-users] Inbound calls from TRUNK   Khaled W. Chehab
• [asterisk-users] E1 check with nagios, how to?   Aurimas Skirgaila
• [asterisk-users] E1 check with nagios, how to?   Mark Deneen
• [asterisk-users] E1 check with nagios, how to?   Joel Maslak
• [asterisk-users] E1 check with nagios, how to?   Diego
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Shaun Ruffell
• [asterisk-users] TELUS British Columbia PRI Settings   Jeremy.Hellstrom at synovate.com
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Mike
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Mike
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Shaun Ruffell
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Mike
• [asterisk-users] SIP X.25   Hans Witvliet
• [asterisk-users] TELUS British Columbia PRI Settings   Tzafrir Cohen
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Mike
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Shaun Ruffell
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Mike
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Shaun Ruffell
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Luis Antonio Prata Barbosa
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Mike
• [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED   Mike
• [asterisk-users] NAT issue (i think?)   Danny Dias
• [asterisk-users] Use modprobe to find E1/T1 jumper setting on PRI card   Lee, John (Sydney)
• [asterisk-users] 1.6 and 1.8 version & A2Billing   Vardan Harutyunyan
• [asterisk-users] Use modprobe to find E1/T1 jumper setting on PRI card   Carlos Chavez
• [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card   Lee, John (Sydney)
• [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card   A J Stiles
• [asterisk-users] ISDN - Busy signal on 3rd call   Paulo Santos
• [asterisk-users] DAHDI FXO port only recognizes the "S" extension‏   Songtao Yu
• [asterisk-users] Weird Behavior with DAHDI   Danny Dias
• [asterisk-users] Jin.   jeff jones
• [asterisk-users] (no subject)   jeff jones
• [asterisk-users] Alert-Info advice   Ishfaq Malik
• [asterisk-users] DAHDI FXO port only recognizes the "S" extension‏   Warren Selby
• [asterisk-users] TELUS British Columbia PRI Settings   Jeremy.Hellstrom at synovate.com
• [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card   Shaun Ruffell
• [asterisk-users] Alert-Info advice   Philipp von Klitzing
• [asterisk-users] DAHDI FXO port only recognizes the "S" extension‏   Carlos Chavez
• [asterisk-users] asterisk > cisco gateway > westell > isdx   Damian Turburville
• [asterisk-users] Weird Behavior with DAHDI   Andrew Thomas
• [asterisk-users] Alert-Info advice   Ishfaq Malik
• [asterisk-users] DAHDI FXO port only recognizes the "S" extension?   Andrew Thomas
• [asterisk-users] DAHDI FXO port only recognizes the "S" extension?   Tilghman Lesher
• [asterisk-users] DAHDI FXO port only recognizes the "S"extension?   Danny Nicholas
• [asterisk-users] DAHDI FXO port only recognizes the "S" extension‏   A J Stiles
• [asterisk-users] DAHDI FXO port only recognizes the "S" extension‏   Jim Dickenson
• [asterisk-users] Alert-Info advice   Philipp von Klitzing
• [asterisk-users] Successive Dial apps give hang up within 30s!!   khalid touati
• [asterisk-users] can't get libpri/PRI to work, missing PRI commands   mishka at efro.us
• [asterisk-users] SCCP (skinny) phone behind NAT: RTP dest addr wrong   Infra
• [asterisk-users] Go from *100* to just 100   Jonas Kellens
• [asterisk-users] SIP Registrations   Alexandru Oniciuc
• [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card   Lee, John (Sydney)
• [asterisk-users] can't get libpri/PRI to work, missing PRI commands   Lee, John (Sydney)
• [asterisk-users] Go from *100* to just 100   Andrew Thomas
• [asterisk-users] Go from *100* to just 100   Gordon Henderson
• [asterisk-users] Kernel Panic When restarting the server   Danny Dias
• [asterisk-users] Asterisk 1.6.2.10 Internal timing   Jonas Kellens
• [asterisk-users] Kernel Panic When restarting the server   mahesh katta
• [asterisk-users] Kernel Panic When restarting the server   A J Stiles
• [asterisk-users] Asterisk 1.6.2.13 Audio Prompts Stopping   Bryant Zimmerman
• [asterisk-users] Go from *100* to just 100   Richard Kenner
• [asterisk-users] Kernel Panic When restarting the server   Tim Nelson
• [asterisk-users] Asterisk 1.6.2.10 Internal timing   Jonas Kellens
• [asterisk-users] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI   Захаров Антон
• [asterisk-users] Asterisk 1.6.2.13 Audio Prompts Stopping   Danny Nicholas
• [asterisk-users] Kernel Panic When restarting the server   Danny Dias
• [asterisk-users] Unscheduled service outage for various Asterisk community services   Asterisk Development Team
• [asterisk-users] Intercom with Dial() works, but not with Page()   Jonas Kellens
• [asterisk-users] Intercom with Dial() works, but not with Page()   Philipp von Klitzing
• [asterisk-users] Kernel Panic When restarting the server   Tim Nelson
• [asterisk-users] Intercom with Dial() works, but not with Page()   Danny Nicholas
• [asterisk-users] Unable to load fax modules   khalid touati
• [asterisk-users] Unable to load fax modules   David Backeberg
• [asterisk-users] Asterisk 1.6.2.10 Internal timing   Tilghman Lesher
• [asterisk-users] Unable to load fax modules   Kevin P. Fleming
• [asterisk-users] Unable to load fax modules   khalid touati
• [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available   Ishfaq Malik
• [asterisk-users] Unable to load fax modules   Kevin P. Fleming
• [asterisk-users] Same extension on multiple servers confusion   Matteo Fortini
• [asterisk-users] Unable to load fax modules   David Backeberg
• [asterisk-users] Friday 12 Noon EDT: VoIP Abuse Project   Randy R
• [asterisk-users] Unable to load fax modules   khalid touati
• [asterisk-users] a2billing   Flavio Miranda
• [asterisk-users] a2billing   Danny Nicholas
• [asterisk-users] a2billing   Flavio Miranda
• [asterisk-users] a2billing   Danny Nicholas

Last message date: Thu Sep 30 15:44:31 CDT 2010
Archived on: Thu Sep 30 15:44:40 CDT 2010

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