March 2007 Archives by date

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Starting: Thu Mar 1 00:06:10 MST 2007
Ending: Sat Mar 31 23:44:06 MST 2007
Messages: 2482

• [asterisk-users] extensions.conf & sccp.conf howto call external number   Michiel van Baak
• [asterisk-users] read write or only read fields in cdr?   Bayrouni
• [asterisk-users] read write or only read fields in cdr?   Bayrouni
• [asterisk-users] DTMF not being detected with 1 provider. Works with the other provider...   Evert
• [asterisk-users] Siemens HiPATH 3700 with Asterisk   Jorge de Diego
• [asterisk-users] How to get values of local channels context   Yuan LIU
• [asterisk-users] transfer function   Denis V. Gudtsov
• [asterisk-users] about bluetooth channel   Dave Cotton
• [asterisk-users] Not registering Port with VSP   Davy Chan
• [asterisk-users] FAX using T38   Zoa
• [asterisk-users] 1.4 lost internet internal phones loose registration   Thomas Kenyon
• [asterisk-users] FAX using T38   Ricardo Carvalho
• [asterisk-users] Re: queue information into db   Tomislav Parcina
• [asterisk-users] Re: fax support   Tomislav Parcina
• [asterisk-users] Re: SIP interface status and calllimit   Tomislav Parcina
• [asterisk-users] Issue with Calling Name ID in SIP: Asterisk sets Caller ID Number as Name if NO Name   Jean-Marc Salsa
• [asterisk-users] TDM400p Loaded only once   Il Neofita
• [asterisk-users] Revolution Call Accounting Desktop   Tomislav Parcina
• [asterisk-users] voicemail advanced options problem with mysql datbase   srinivas Antarvedi
• [asterisk-users] UK SIP Gateway   --
• [asterisk-users] TDM400p Loaded only once   Tzafrir Cohen
• [asterisk-users] UK SIP Gateway   Chris Stenton
• [asterisk-users] 3 way calling independent of phone hw.   Simon Tennant
• [asterisk-users] UK SIP Gateway   Steve Kennedy
• [asterisk-users] UK SIP Gateway   Wilson Pickett
• [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI   Matt
• [asterisk-users] Cannot hear ringback music from telco   Vincent Tam
• [asterisk-users] about bluetooth channel   Steve Totaro
• [asterisk-users] Cannot hear ringback music from telco   Steve Totaro
• [asterisk-users] Cannot hear ringback music from telco   Trevor Peirce
• [asterisk-users] IAX best practices   Asterisk
• [Asterisk-Users] SIP to IAX - forcing codec pass thru   Thomas Kenyon
• [asterisk-users] Zaptel 1.4.0   Mike Hammett
• [asterisk-users] IAX best practices   Steve Totaro
• [asterisk-users] Zaptel 1.4.0   Carlos Alperin
• [asterisk-users] Snom 320 password   Mike Hammett
• [asterisk-users] Paid support offered   asterisk at 911networks.com
• [asterisk-users] blieve i my TE110P or My teleco provider ??   younss azzayani
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Zaptel 1.4.0   Ricardo Carvalho
• [asterisk-users] UK SIP Gateway   Gordon Henderson
• [asterisk-users] Testing asterisk with sipp   John Albano
• [asterisk-users] Extensions +International   Rob Schall
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Webster, Andrew
• [asterisk-users] Call connected, cannot hear or speak - $20 for fix   Supa
• [asterisk-users] Multiple simultaneous calls   stefano.totaro at transport.alstom.com
• [asterisk-users] Extensions +International   McGhee, Stefano
• [asterisk-users] About queues and multiple lines.   Nuria Fernandez
• [asterisk-users] UK SIP Gateway   Wireless
• [asterisk-users] Asterisk Realtime   Mike Hammett
• [asterisk-users] Multiple simultaneous calls   Zoa
• [asterisk-users] Fwd: Problem with TE212P   Benito Camelas
• [asterisk-users] Extensions +International   Rob Schall
• [asterisk-users] Multiple simultaneous calls   Steve Totaro
• [asterisk-users] about bluetooth channel   Facundo Ameal
• [asterisk-users] Problem with TE212P   younss azzayani
• [asterisk-users] Asterisk Realtime   Brian Capouch
• [asterisk-users] Help understanding SIP SHOW CHANNELS   Mojo with Horan & Company, LLC
• [asterisk-users] Tesco Internet Phone   Julian Lyndon-Smith
• [asterisk-users] about bluetooth channel   Iban Lopetegi Zinkunegi
• [asterisk-users] Tesco Internet Phone   Dean Collins
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Tesco Internet Phone   Julian Lyndon-Smith
• [asterisk-users] Asterisk Realtime   Philipp Kempgen
• [asterisk-users] Tesco Internet Phone   bails
• [asterisk-users] Asterisk Realtime   Mike Hammett
• [asterisk-users] Tesco Internet Phone   bails
• [asterisk-users] Asterisk Realtime   Bruce Reeves
• [asterisk-users] Voice mail is not giving unavailable or busy prompts   Stephen Bosch
• [asterisk-users] Asterisk 1.4.1   Forrest Beck
• [asterisk-users] Asterisk 1.4.1   Steve Murphy
• [asterisk-users] Asterisk 1.4.1   Paul
• [asterisk-users] build rpm fails   Devraj Mukherjee
• [asterisk-users] Re: build rpm fails   Axel Thimm
• [asterisk-users] Re: build rpm fails   Devraj Mukherjee
• [asterisk-users] Digium S101i - pickupexten doesn't work   Joseph
• [asterisk-users] gtalktovoip and Asteirsk   Klaverstyn, David C
• [asterisk-users] Polycom reject button   Jason Walker
• [asterisk-users] 2 Call locations   Jason Walker
• [asterisk-users] 2 Call locations   Yuan LIU
• [asterisk-users] TDM400p Loaded only once   Il Neofita
• [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI   C F
• [asterisk-users] RE: Polycom reject button   JR Richardson
• [asterisk-users] No Caller ID Name PRI NI2   C F
• [asterisk-users] How can I use the "GET VARIABLE variablename" in AGI   李君
• [asterisk-users] How can I use the "GET VARIABLE variablename" in AGI   Lee Jenkins
• [asterisk-users] Test   Wai Wu
• [asterisk-users] Re: build rpm fails   Devraj Mukherjee
• [asterisk-users] Newbie extensions.conf question   Chris Griffin
• [asterisk-users] How can I use the "GET VARIABLE variablename" inAGI   Yuan LIU
• [asterisk-users] Newbie extensions.conf question   Chris Griffin
• [asterisk-users] UK SIP Gateway   George Gardiner
• [asterisk-users] UK SIP Gateway   --
• [asterisk-users] Tesco Internet Phone   Alan Chandler
• [asterisk-users] Re: queue information into db   Tomislav Parcina
• [asterisk-users] Multiple simultaneous calls   stefano.totaro at transport.alstom.com
• Rif: Re: [asterisk-users] Multiple simultaneous calls   stefano.totaro at transport.alstom.com
• [asterisk-users] BLF not working with Asterisk 1.4.0   Ricardo Carvalho
• [asterisk-users] Running Fax on E1 line   younss azzayani
• [asterisk-users] Problem with TE212P   Ioan Indreias
• [asterisk-users] Re: build rpm fails   Axel Thimm
• [asterisk-users] Newbie extensions.conf question   Bruce Reeves
• [asterisk-users] Re: Sending SMS   Tomislav Parcina
• [asterisk-users] Re: Asterisk and DTMF   Carlos Barros
• [asterisk-users] Re: Sending SMS   Michiel van Baak
• [asterisk-users] Cannot hear ringback music from telco   Vincent Tam
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Webster, Andrew
• [asterisk-users] No Caller ID Name PRI NI2   Webster, Andrew
• [asterisk-users] Running Fax on E1 line   Doug Lytle
• [asterisk-users] Multiple simultaneous calls   Steve Totaro
• [asterisk-users] Re: Sending SMS   Steve Totaro
• [asterisk-users] Cannot hear ringback music from telco   Steve Totaro
• [asterisk-users] RE: Polycom reject button   Mike
• [asterisk-users] BLF not working with Asterisk 1.4.0   Andrey Solovjov
• [asterisk-users] BLF not working with Asterisk 1.4.0   Steve Langstaff
• [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI   Andrew Latham
• [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207   Hall, Eric M.
• [asterisk-users] T38   Khaled
• [asterisk-users] Problem with TE212P   Benito Camelas
• [asterisk-users] Multiple simultaneous calls   Stefano Totaro
• [asterisk-users] RE: Polycom reject button   Doug Lytle
• [asterisk-users] No Caller ID Name PRI NI2   C F
• [asterisk-users] Test   C F
• [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207   Lee Jenkins
• [asterisk-users] IAX best practices   Asterisk
• [asterisk-users] Asterisk and Fax   --[ UxBoD ]--
• [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] Asterisk and Fax   Michelle Dupuis
• [asterisk-users] IAX best practices   Michelle Dupuis
• [asterisk-users] DTMF detection problems on PRI channels?   Tony Mountifield
• [asterisk-users] Asterisk and Fax   Tzafrir Cohen
• [asterisk-users] Asterisk and Fax   Robert A. Rawlinson
• [asterisk-users] IAX best practices   Steve Totaro
• [asterisk-users] Asterisk and Fax   Steve Totaro
• [asterisk-users] IAX best practices   Henry J. Cobb
• [asterisk-users] Alec Saunders post about Mashable Telco's   Steve Totaro
• [asterisk-users] DTMF detection problems on PRI channels?   Michelle Dupuis
• [asterisk-users] WMI from Asterisk to Cisco Call Manager   Frédéric Marti
• [asterisk-users] IAX best practices   Asterisk
• [asterisk-users] RE: Polycom reject button   Jason Walker
• [asterisk-users] IAX best practices   Zoa
• [asterisk-users] Alec Saunders post about Mashable Telco's   Tim Panton
• [asterisk-users] IAX best practices   Steve Totaro
• [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] Help Voicemail to SMS using asterisk   Goke Aruna
• [asterisk-users] Asterisk and Fax   --[ UxBoD ]--
• [asterisk-users] gtalktovoip and Asteirsk   Cosmin Prund
• [asterisk-users] Voicemail to SMS using asterisk   Goke Aruna
• [asterisk-users] Re: Sending SMS   Michiel van Baak
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Remi Quezada
• [asterisk-users] DTMF from TDM400P and X100P   Yuan LIU
• [asterisk-users] Alec Saunders post about Mashable Telco's   Steve Totaro
• FW: [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] Help Voicemail to SMS using asterisk   Steve Totaro
• [asterisk-users] IAX best practices   Tim Panton
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Russell Bryant
• [asterisk-users] Alec Saunders post about Mashable Telco's   Cory Andrews
• [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] PRI progress codes.   John Bittner
• [asterisk-users] Test   Mojo with Horan & Company, LLC
• [asterisk-users] rtsavesysname not working in 1.4   David Thomas
• [asterisk-users] RE: Polycom reject button   Mojo with Horan & Company, LLC
• [asterisk-users] transfer function   Mojo with Horan & Company, LLC
• [asterisk-users] DTMF from TDM400P and X100P   Tzafrir Cohen
• [asterisk-users] IAX best practices   Tzafrir Cohen
• [asterisk-users] 1.4 - SLA   Lacy Moore - Aspendora
• Spam? Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207   Hall, Eric M.
• [asterisk-users] Help Voicemail to SMS using asterisk   Dean Collins
• [asterisk-users] Problem with TE212P   Ioan Indreias
• [asterisk-users] Alec Saunders post about Mashable Telco's   Dean Collins
• [asterisk-users] Asterisk and Fax   Andrew Joakimsen
• [asterisk-users] T38   Andrew Joakimsen
• [asterisk-users] blieve i my TE110P or My teleco provider ??   Andrew Joakimsen
• [asterisk-users] FAX using T38   Andrew Joakimsen
• [asterisk-users] multiple phones registered for the same user   Andrew Joakimsen
• [asterisk-users] PRI progress codes.   Eric "ManxPower" Wieling
• [asterisk-users] REMOTE CRASH FIX   Mike Lynchfield
• [asterisk-users] rtsavesysname not working in 1.4   Bruce Reeves
• [asterisk-users] Help Voicemail to SMS using asterisk   Goke Aruna
• [asterisk-users] Asterisk Java and Astmanproxy   Doug Garstang
• [asterisk-users] svn 1.4 - mp3 support and changing the installation directory   tzieleniewski
• [asterisk-users] IP addresses   Mike Hammett
• [asterisk-users] DTMF from TDM400P and X100P   Yuan LIU
• [asterisk-users] Asterisk Java w/ Threads   Doug Garstang
• [asterisk-users] REMOTE CRASH FIX   BJ Weschke
• [asterisk-users] REMOTE CRASH FIX   Kevin P. Fleming
• [asterisk-users] blieve i my TE110P or My teleco provider ??   younss azzayani
• [asterisk-users] DTMF from TDM400P and X100P   Tzafrir Cohen
• [asterisk-users] Asterisk Java w/ Threads   Stefan Reuter
• [asterisk-users] How to log VERBOSE statement to a file?   Larry Alkoff
• [asterisk-users] Zaptel 1.2.15 Released   Asterisk Development Team
• [asterisk-users] REMOTE CRASH FIX   Mike Lynchfield
• [asterisk-users] Asterisk 1.2.16 Released   Asterisk Development Team
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] Asterisk 1.4.1 Released   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.1   Kevin P. Fleming
• [asterisk-users] FAX using T38   Kevin P. Fleming
• [asterisk-users] svn 1.4 - mp3 support and changing the installation directory   Kevin P. Fleming
• [asterisk-users] IP addresses   Steve Totaro
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   Zeeshan Zakaria
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   Mailing Lists
• [asterisk-users] Re: Sending SMS   Al Bochter
• [asterisk-users] How to fail an AGI   Yuan LIU
• [asterisk-users] Help Voicemail to SMS using asterisk   Goke Aruna
• [asterisk-users] Help Voicemail to SMS using asterisk   Yuan LIU
• [asterisk-users] Asterisk - e164 (enum) lookup confused   Joseph
• [asterisk-users] Re: What means: Request to schedule in the past?!?!   Martin Joseph
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   Senad Jordanovic
• [asterisk-users] Somebody can help me?   Andrea Seghezzi
• [asterisk-users] How to fail an AGI   Lenz
• [asterisk-users] gtalk2voip and Asterisk   Mani Sridhar
• [asterisk-users] FAX using T38   Steve Underwood
• [asterisk-users] Re: Sending SMS   Gordon Henderson
• [asterisk-users] Asterisk Java w/ Threads   Jesus Mogollon
• [asterisk-users] Re: Sending SMS   Steve Kennedy
• [asterisk-users] Re: Sending SMS   Steve Totaro
• [asterisk-users] Asterisk Java w/ Threads   Stefan Reuter
• [asterisk-users] RE: Polycom reject button   Jason Walker
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Mike Lynchfield
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Tzafrir Cohen
• [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.   shadowym
• [asterisk-users] Re: Sending SMS   Jens Vagelpohl
• [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk   Mike Lynchfield
• Re: Re: [asterisk-users] svn 1.4 - mp3 support and changing the installation directory   tzieleniewski
• [asterisk-users] svn 1.4 - mp3 support and changing the installation directory   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4.1 Released   Michelle Dupuis
• [asterisk-users] hanging an asterisk box off of a PBX analog extension   Mike D'Ambrogia
• [asterisk-users] creating new asterisk application   Giedrius Augys
• [asterisk-users] Asterisk - e164 (enum) lookup confused   Yuan LIU
• [asterisk-users] How to fail an AGI   Yuan LIU
• [asterisk-users] hanging an asterisk box off of a PBX analog extension   Lacy Moore - Aspendora
• [asterisk-users] hanging an asterisk box off of a PBX analog extension   Gordon Henderson
• [asterisk-users] Polycom reject button   Stephen Bosch
• [asterisk-users] REMOTE CRASH FIX   Stephen Bosch
• [asterisk-users] Asterisk 1.4.1 Released   Thomas Kenyon
• [asterisk-users] hanging an asterisk box off of a PBX analog extension   Eric "ManxPower" Wieling
• [asterisk-users] dial question   Hall, Eric M.
• [asterisk-users] dial question   Alvin Austin
• [asterisk-users] dial question   C F
• [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk   Matt
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Brad Templeton
• [asterisk-users] dial question   Pavel Jezek
• [asterisk-users] Voice quality issues   Alan Chandler
• [asterisk-users] Voice quality issues   Tzafrir Cohen
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Michiel van Baak
• [asterisk-users] running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn   tzieleniewski
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Kevin P. Fleming
• [asterisk-users] PRI progress codes.   Wai Wu
• [asterisk-users] Re: Sending SMS   Al Bochter
• [asterisk-users] When does local leg in call file start?   Yuan LIU
• [asterisk-users] When does local leg in call file start?   Eric "ManxPower" Wieling
• [asterisk-users] When does local leg in call file start?   Doug Lytle
• [asterisk-users] x100p.com   Mail list
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Brad Templeton
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Kevin P. Fleming
• [asterisk-users] Real Time, sip.conf, [general]   André Santos
• [asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1   TZieleniewski
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Brad Templeton
• [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1   Joshua Colp
• [asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1   Kevin P. Fleming
• [asterisk-users] FAX using T38   Andrew Joakimsen
• [asterisk-users] Digium S101i - pickupexten doesn't work   Andrew Joakimsen
• [asterisk-users] Digium S101i - pickupexten doesn't work   Eric "ManxPower" Wieling
• [asterisk-users] running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1   TZieleniewski
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Asterisk Java w/ Threads   Jesus Mogollon
• [asterisk-users] FAX using T38   Steve Underwood
• [asterisk-users] FAX using T38   Kevin P. Fleming
• [asterisk-users] Configurations Files of TE110P   Joel Hill
• [asterisk-users] Voicemail to SMS using asterisk   Andrew Joakimsen
• [asterisk-users] Multiple simultaneous calls   Andrew Joakimsen
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Re: [A*UG] How to log VERBOSE statement to a file?   Larry Alkoff
• [asterisk-users] When does local leg in call file start?   Yuan LIU
• [asterisk-users] Is the 1.0.X branch vulnerable to the SIP issue?   Thermal Wetland
• [asterisk-users] Read() status?   Yuan LIU
• [asterisk-users] new kernel and zaptel   Giedrius Augys
• [asterisk-users] new kernel and zaptel   Tzafrir Cohen
• [asterisk-users] Configurations Files of TE110P   Ioan Indreias
• [asterisk-users] Re: Problem with TE212P   Benito Camelas
• [asterisk-users] Re: DTMF detection problems on PRI channels?   Tony Mountifield
• [asterisk-users] Is the 1.0.X branch vulnerable to the SIP issue?   Kevin P. Fleming
• [asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now Open   Praburaajan
• [asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now Open   Praburaajan
• [asterisk-users] Re: Problem with TE212P   Tzafrir Cohen
• [asterisk-users] SMS ON ASTERISK   Assis, Eduardo
• [asterisk-users] CTI   Nuria Fernandez
• [asterisk-users] Sangoma A102d and Asterisk on Debian 3.1.   Markus Monka
• [asterisk-users] Problem with TE212P   C F
• [asterisk-users] Read() status?   C F
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] Configurations Files of TE110P   Eric "ManxPower" Wieling
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Remi Quezada
• [asterisk-users] TC400B   Wai Wu
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] rtsavesysname not working in 1.4   David Thomas
• [asterisk-users] Configurations Files of TE110P   younss azzayani
• [asterisk-users] rtsavesysname not working in 1.4   Bruce Reeves
• [asterisk-users] How to disable MOH completely?   David Thomas
• [asterisk-users] rtsavesysname not working in 1.4   David Thomas
• [asterisk-users] TDM400P/FXS in a HP DL380 G5   James FitzGibbon
• [asterisk-users] Asterisk Java w/ Threads   Doug Garstang
• [asterisk-users] Asterisk Java w/ Threads   Doug Garstang
• [asterisk-users] running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn   TZieleniewski
• [asterisk-users] Read() status?   Doug Garstang
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P   younss azzayani
• [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P   Eric "ManxPower" Wieling
• [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P   Michael Collins
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P   younss azzayani
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] Re: Asterisk Java w/ Threads   Stefan Reuter
• [asterisk-users] Re: Asterisk Java w/ Threads   Eric "ManxPower" Wieling
• [asterisk-users] TDM400P/FXS in a HP DL380 G5   Hans Witvliet
• [asterisk-users] Re: TDM400P/FXS in a HP DL380 G5   James FitzGibbon
• [asterisk-users] How to disable MOH completely?   C F
• [asterisk-users] Re: Asterisk Java w/ Threads   Stefan Reuter
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Singer Wang
• [asterisk-users] How to disable MOH completely?   David Thomas
• [asterisk-users] g.729 on solaris10/x86   Juraj Bednar
• [asterisk-users] Setting Sip Headers From Dial App?   Stuart Sheldon
• [asterisk-users] Instant Messaging with SIP Softphone Eyebeam (was: SMS ON ASTERISK)   Anselm Martin Hoffmeister
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Bruce Reeves
• [asterisk-users] Voicemail question   Hall, Eric M.
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Andrew Joakimsen
• [asterisk-users] Using Asterisk as Voicemail Server on a dinosaur Meridian System   J French
• [asterisk-users] [Announce] Web-MeetMe V3.0.1 released   Dan Austin
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   David Thomas
• [asterisk-users] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???   Mr. James W. Laferriere
• [asterisk-users] Using Asterisk as Voicemail Server on a dinosaur Meridian System   Steve Totaro
• [asterisk-users] How to disable MOH completely?   C F
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Dovid B
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Andres
• [asterisk-users] Polycom Questions   »Steven Ringwald«
• [asterisk-users] How to disable MOH completely?   David Thomas
• [asterisk-users] server generated outbound conference calls?   Dean Collins
• [asterisk-users] IAX2, DTMF and x86_64.   William F. Acker WB2FLW +1-303-722-7209
• [asterisk-users] app_queue not using exit context?   Steve Edwards
• [asterisk-users] Re: Registrations, how many is too many?   Tomislav Parcina
• [asterisk-users] Re: 1.4 lost internet internal phones loose registration   Tomislav Parcina
• [asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED   Tomislav Parcina
• [asterisk-users] preventing voicemail pickup after SIP redirect ?   Louis-David Mitterrand
• [asterisk-users] Re: Asterisk Faxing Support   Tomislav Parcina
• [asterisk-users] Re: Digium cards on Vmware   Tomislav Parcina
• [asterisk-users] web based sipphone   Pezhman Lali
• [asterisk-users] Re: FAX using T38   Tomislav Parcina
• [asterisk-users] Re: Setting Sip Headers From Dial App?   Benny Amorsen
• [asterisk-users] Micros-Fidelio - billing in hotel   Tomislav Parcina
• [asterisk-users] server generated outbound conference calls?   Chris Mason (Lists)
• [asterisk-users] [asterisk_voip] asterisk and ogg files   Bayrouni
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Chris Mason (Lists)
• [asterisk-users] Digium cards on Vmware   Morten Isaksen
• [asterisk-users] Digium cards on Vmware   Massimo Nuvoli
• [asterisk-users] Using Asterisk as Voicemail Server on a dinosaurMeridian System   Steve Langstaff
• [asterisk-users] visdn, misdn and the hell   Massimo Nuvoli
• [asterisk-users] Polycom Questions   Steve Blair
• [asterisk-users] Re: 1.4 lost internet internal phones loose registration   Thomas Kenyon
• [asterisk-users] Polycom Questions   Doug Lytle
• [asterisk-users] Re: build rpm fails   Tomislav Parcina
• [asterisk-users] Re: build rpm fails   Tomislav Parcina
• [asterisk-users] Polycom Questions   Bryan M. Johns
• [asterisk-users] Ringing does not terminate on mISDN after pickup   Arik Raffael Funke
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Anselm Martin Hoffmeister
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   rehan at supertec.com
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Chris Mason (Lists)
• [asterisk-users] Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent   Asterisk Asterisk
• [asterisk-users] Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent   --[ UxBoD ]--
• [asterisk-users] Re: build rpm fails   Axel Thimm
• [asterisk-users] preventing voicemail pickup after SIP redirect ?   Eric "ManxPower" Wieling
• [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850   mazhiyong at sohu.com
• [asterisk-users] Polycom Questions   Eric "ManxPower" Wieling
• [asterisk-users] Pickup failover   René Enskat
• [asterisk-users] Re: TDM400P/FXS in a HP DL380 G5   Kevin P. Fleming
• [asterisk-users] g.729 on solaris10/x86   Kevin P. Fleming
• [asterisk-users] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???   Kevin P. Fleming
• [asterisk-users] Re: preventing voicemail pickup after SIP redirect ?   Louis-David Mitterrand
• [asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED   Jerry Jones
• [asterisk-users] chan_cellphone won't pair with phone   Earle Clubb
• [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850   Tzafrir Cohen
• [asterisk-users] Linksys PAP2 and Caller ID   Gergo Csibra
• [asterisk-users] Polycom Questions   Dave Fullerton
• [asterisk-users] Double DTMF digits sent on IAX native bridge   Remi Quezada
• [asterisk-users] Asterisk 1.2.15 chan_vpb with vpb-driver 4.0   Yifan Zhang
• [asterisk-users] Re:Problem with TE212P   Benito Camelas
• [asterisk-users] How to disable MOH completely?   Mojo with Horan & Company, LLC
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Polycom Questions   »Steven Ringwald«
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] Long term voicemail archival and synchronisation between multiple storage locations?   Dean Collins
• [asterisk-users] 1.4 - SLA   Lacy Moore - Aspendora
• [asterisk-users] Building a new voicemail system... Testers needed!   Olle E Johansson
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] running asterisk through cellphone   Mojo with Horan & Company, LLC
• [asterisk-users] 1.4 - SLA   Bill Gibbs
• [asterisk-users] How many gsm channels   Wai Wu
• [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'   Ken Williams
• [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'   Alex Robar
• [asterisk-users] How many gsm channels   Michelle Dupuis
• [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'   Ken Williams
• [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'   Tzafrir Cohen
• [asterisk-users] Compiling smsq in 1.2   Yuan LIU
• [asterisk-users] Cancelling a digit in IVR   Kuba
• [asterisk-users] Add current caller to junk-callers-database   Alvin Austin
• [asterisk-users] Cancelling a digit in IVR   Dean Collins
• [asterisk-users] Re: build rpm fails   Thomas Patterson
• [asterisk-users] 1.4 - SLA   Russell Bryant
• [asterisk-users] Long term voicemail archival and synchronisationbetween multiple storage locations?   Steve Totaro
• [asterisk-users] Polycom Questions   Dave Miller
• [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850   Ma Zhiyong
• [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850   Ma Zhiyong
• [asterisk-users] RE: Polycom reject button   Kenneth Padgett
• [asterisk-users] Polycom Questions   Marty Mastera
• [asterisk-users] GTalk/Jabber passing audio in 1.4.1!   Ronald Lewis
• [asterisk-users] Nomination for Coolest App in 2007   Steve Totaro
• [asterisk-users] gtalk2voip and Asterisk   Klaverstyn, David C
• [asterisk-users] Anybody having problems using sellvoip?   Tom Lynn
• [asterisk-users] RE: Linksys PAP2 and Caller ID   Stewart Nelson
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 21   Justin Newman
• [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Call droping issue   Vidura Senadeera
• [asterisk-users] Voicemail question   Olle E Johansson
• [asterisk-users] Real Time, sip.conf, [general]   Olle E Johansson
• [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Calldroping issue   Steve Totaro
• [asterisk-users] Asterisk 1.4.1 - Calling problem   Thomas Deillon
• [asterisk-users] capi installation problem   Giedrius Augys
• [asterisk-users] Building a new voicemail system... Testers needed!   John Marvin
• [asterisk-users] capi installation problem   Armin Schindler
• [asterisk-users] VoIP over Alvarion Wireless   Matt
• [asterisk-users] Background / Invalid Extension through cell phone   John Congdon
• [asterisk-users] Background / Invalid Extension through cell phone   Doug Lytle
• [asterisk-users] VoIP over Alvarion Wireless   Gordon Henderson
• [asterisk-users] Background / Invalid Extension through cell phone   Eric "ManxPower" Wieling
• [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Call droping issue   Eric "ManxPower" Wieling
• [asterisk-users] TC400B   Wai Wu
• [asterisk-users] Background / Invalid Extension through cell phone   Steve Totaro
• [asterisk-users] Realtime Extensions and "Include"   Peder at NetworkOblivion
• [asterisk-users] Background / Invalid Extension through cell phone   Eric "ManxPower" Wieling
• [asterisk-users] asterisk and ssl   nik600
• [asterisk-users] Realtime Extensions and "Include"   Rob Schall
• [asterisk-users] asterisk and ssl   Dean Collins
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   Steve Totaro
• [asterisk-users] asterisk and ssl   Olle E Johansson
• [asterisk-users] Background / Invalid Extension through cell phone   Steve Totaro
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   Steve Totaro
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   cb
• [asterisk-users] asterisk and ssl   Steve Totaro
• [asterisk-users] Background / Invalid Extension through cell phone   Jorge Mendoza
• [asterisk-users] queue information in mySQL   Thomas Winter
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Asterisk Auto-dial out   Phil Menico
• [asterisk-users] AsteriskNow Beta 4 - zaptel.conf / zapata.conf problems   mark.d.rowe at bt.com
• [asterisk-users] queue information in mySQL   Philipp Kempgen
• [asterisk-users] AsteriskNow Beta 4 - zaptel.conf / zapata.conf problems   demuel at thephinix.org
• [asterisk-users] Realtime Extensions and "Include"   Peder at NetworkOblivion
• [asterisk-users] queue information in mySQL   Steve Totaro
• [asterisk-users] mobility with asterisk   Alvaro Pacho
• [asterisk-users] Problem HandyTone 488 does not call transfer   Pablo Almido
• [asterisk-users] mobility with asterisk   Alex Robar
• [asterisk-users] Asterisk queue and agents   Octavio Ruiz (Ta^3)
• [asterisk-users] VoIP over Alvarion Wireless   Andrew Niemantsverdriet
• [asterisk-users] mobility with asterisk   Jorge Mendoza
• [asterisk-users] gtalk2voip and Asteris   DID 4ME
• [asterisk-users] VoIP over Alvarion Wireless   Matt
• [asterisk-users] Voicemail question   Race Vanderdecken
• [asterisk-users] Problem HandyTone 488 does not call transfer   Drew Gibson
• [asterisk-users] Asterisk Auto-dial out   Michael Collins
• [asterisk-users] Asterisk Registering to other SIP servers.   Bala Neelakantan
• [asterisk-users] Asterisk Realtime   Mike Hammett
• [asterisk-users] Re: Asterisk Realtime   Mike Hammett
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] VoIP over Alvarion Wireless   Andrew Niemantsverdriet
• [asterisk-users] Re: Asterisk Realtime   Brian Capouch
• [asterisk-users] Asterisk Registering to other SIP servers.   Olle E Johansson
• [asterisk-users] auto dialer   Hall, Eric M.
• [asterisk-users] VoIP over Alvarion Wireless   Matt
• [asterisk-users] auto dialer   Melcon Moraes
• [asterisk-users] auto dialer   Hall, Eric M.
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Problems with Zaptel Drivers   Lutgring, Sam
• [asterisk-users] Problems with Zaptel Drivers   Tzafrir Cohen
• [asterisk-users] sip show channels   Bill Gibbs
• [asterisk-users] Asterisk queue and agents   BJ Weschke
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Number of SIP messages per minute   Mark Davies
• [asterisk-users] Asterisk 1.4.1 Released   voiplist
• [asterisk-users] SIP remote crash bug   voiplist
• [asterisk-users] Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping   Vidura Senadeera
• [asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba pbx - Call droping   Vidura Senadeera
• [asterisk-users] Call recording and archiving   Voip Asterisk
• [asterisk-users] sip show channels   Olle E Johansson
• [asterisk-users] Asterisk 1.4.1 Released   Olle E Johansson
• [asterisk-users] Re: queue information into db   nik600
• [Asterisk-Users] Re: Pickup *8 with CallerID   Olivier
• [asterisk-users] Timeouts not working   Richard Trenchard
• [asterisk-users] How to handle SIP-Callerid?   Andreas Anderson
• [asterisk-users] Queue Announcements for Operators   scott
• [asterisk-users] Info: Nokia E65 working with Asterisk   Robert Jenkins
• [asterisk-users] "pritimer" parameter in zapata.conf   Vidura Senadeera
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Henry Cobb
• [asterisk-users] Re: build rpm fails   Tomislav Parcina
• [asterisk-users] Asterisk queue and agents   BJ Weschke
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Tomislav Parcina
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Info: Nokia E65 working with Asterisk   Olivier
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Re: visdn, misdn and the hell   Tomislav Parcina
• [asterisk-users] Info: Nokia E65 working with Asterisk   Jens Vagelpohl
• [asterisk-users] Asterisk queue and agents   BJ Weschke
• [asterisk-users] Asterisk queue and agents   Hall, Eric M.
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Steve Totaro
• [asterisk-users] Asterisk Auto-dial out   Phil Menico
• [asterisk-users] Hinting and Realtime   René Enskat
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Steve Totaro
• [asterisk-users] Asterisk + Panasonic pbx   Sanspareils Greenlans
• [asterisk-users] New Linksys SPA Daylight Saving Time Rule for US/Canada   Trevor G. Hammonds
• [asterisk-users] Queue Announcements for Operators   Philipp Kempgen
• [asterisk-users] Hinting and Realtime   Philipp Kempgen
• AW: [asterisk-users] Hinting and Realtime   René Enskat
• [asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping   Steve Totaro
• [asterisk-users] AMI Originate and release channels   Paulo Vicentini
• AW: [asterisk-users] Hinting and Realtime   Philipp Kempgen
• [asterisk-users] Queue Announcements for Operators   Steve Totaro
• [asterisk-users] cmd pickup Problem   jroesch at bsradio.de
• [asterisk-users] Asterisk + Panasonic pbx   Steve Totaro
• [asterisk-users] Asterisk distributed deployment   ggonzalez at telviso.com.ar
• [asterisk-users] Call recording and archiving   Matthew J. Roth
• [asterisk-users] Re: Asterisk Realtime   Mike Hammett
• [asterisk-users] Asterisk distributed deployment   Steve Totaro
• [asterisk-users] Accessing Voicemail by dialing own number   Chris Carey
• [asterisk-users] Accessing Voicemail by dialing own number   Dave Bour
• [asterisk-users] Re: Asterisk Realtime   Philipp Kempgen
• [asterisk-users] Accessing Voicemail by dialing own number   Andrew Joakimsen
• [asterisk-users] Accessing Voicemail by dialing own number   Philipp Kempgen
• [asterisk-users] Queue announcing hold sequence instead of hold time   Drew Gibson
• [asterisk-users] Packet2Packet Bridging Questions   Daryl Jurbala
• [asterisk-users] Re: Asterisk distributed deployment   ggonzalez at telviso.com.ar
• [asterisk-users] Packet2Packet Bridging Questions   Joshua Colp
• [asterisk-users] Re: Asterisk distributed deployment   Steve Totaro
• [asterisk-users] Queue announcing hold sequence instead of hold time   Rob Schall
• [asterisk-users] Sender phone ringing while recipient talking   Nathan Bell
• [asterisk-users] transfers and CDR   Rodrigo Gonzalez
• [asterisk-users] Sender phone ringing while recipient talking   Steve Totaro
• [asterisk-users] Asterisk distributed deployment   Bruce Reeves
• [asterisk-users] Number of groups?   Webster, Andrew
• [asterisk-users] outdial to phone for new VM notification   end1r
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Mojo with Horan & Company, LLC
• [asterisk-users] Sender phone ringing while recipient talking   Bill Gibbs
• [asterisk-users] auto dialer   Melcon Moraes
• [asterisk-users] Accessing Voicemail by dialing own number   Chris Carey
• [asterisk-users] Packet2Packet Bridging Questions   Daryl Jurbala
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Steve Totaro
• [asterisk-users] Coaching in asterisk   Wai Wu
• [asterisk-users] Re: Asterisk distributed deployment   shadowym
• [asterisk-users] RE: Coaching in asterisk   Dean Collins
• [asterisk-users] No application 'Prefix' for extension in1.2x, what app I have to use instead?   Rafael J. Risco G.V.
• [asterisk-users] Re: Coaching in asterisk   Steve Totaro
• [asterisk-users] Asterisk SIP to MAX TNT Gateway, Sporadic Echo   JR Richardson
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] Zap Channel Deadlocks   Ron McCarthy
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Coaching in asterisk   Dovid B
• [asterisk-users] No application 'Prefix' for extension in1.2x, what app I have to use instead?   Eric "ManxPower" Wieling
• [asterisk-users] RE: Coaching in asterisk   BJ Weschke
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] Sender phone ringing while recipient talking   Nathan Bell
• [asterisk-users] 1.4 compile issue   Wai Wu
• [asterisk-users] Re: Coaching in asterisk   Justin Newman
• [asterisk-users] Sender phone ringing while recipient talking   Steve Totaro
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   Bruce Reeves
• [asterisk-users] Newbie Question   Chris Nighswonger
• [asterisk-users] Call load balancing   Steve Edwards
• [asterisk-users] outdial to phone for new VM notification   Porier, Jeremy M.
• [asterisk-users] Newbie Question   Dovid B
• [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)   Steve Totaro
• [asterisk-users] 1.4 compile issue   Russell Bryant
• [asterisk-users] Is Allison going to be banned from foreign travel over polar bears?   Steve Prior
• [asterisk-users] 1.4 compile issue   Wai Wu
• [asterisk-users] Zap Channel Deadlocks   shadowym
• [asterisk-users] Newbie Question   Leonardo Kamache (Gmail)
• [asterisk-users] Newbie Question   Chris Nighswonger
• [asterisk-users] Is Allison going to be banned from foreign travel over polar bears?   Leo Ann Boon
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine   Jose Bertuzzi
• [asterisk-users] Zap Channel Deadlocks   Ron McCarthy
• [asterisk-users] RE: Coaching in asterisk   Doug Garstang
• [asterisk-users] RE: Coaching in asterisk   Steve Totaro
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Erick Perez
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Zeeshan Zakaria
• [asterisk-users] Which VoIP router and switch to use for medium size business   Zeeshan Zakaria
• [asterisk-users] Fwd: Can't hear any sound   Asterisk Asterisk
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Mark Davies
• [asterisk-users] When to use Echo Cancellation cards?   Zeeshan Zakaria
• [asterisk-users] Can't hear any sound (This time in plain text)   Asterisk Asterisk
• [asterisk-users] Can't hear any sound (This time in plain text)   map
• [asterisk-users] How to handle SIP-Callerid?   Olle E Johansson
• [asterisk-users] How to enter bridge_native_loop???   Santosh Raghuram
• [asterisk-users] Hinting and Realtime   Olle E Johansson
• [asterisk-users] Packet2Packet Bridging Questions   Olle E Johansson
• [asterisk-users] How to enter bridge_native_loop???   Olle E Johansson
• [asterisk-users] RE: Zaptel problem after upgrading to 1.2.16   Mark Davies
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Tzafrir Cohen
• [asterisk-users] RE: Zaptel problem after upgrading to 1.2.16   Tzafrir Cohen
• [asterisk-users] Which VoIP router and switch to use for medium size business   Gordon Henderson
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine   Tzafrir Cohen
• [asterisk-users] Is there any variable for Voicemail Password in Asterisk   jamshed zaidi
• [asterisk-users] Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping issue [ ref:00D36mPe.50032wycQ:ref ]   Vidura Senadeera
• [asterisk-users] Queue announcing hold sequence instead of hold time   Trevor G. Hammonds
• [asterisk-users] Queue announcing hold sequence instead of hold time   Rob Schall
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] disable client side hangup after dialing 911   Patrick Fortin
• [asterisk-users] disable client side hangup after dialing 911   Wai Wu
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Is Allison going to be banned from foreign travel over polar bears?   Drew Gibson
• [asterisk-users] disable client side hangup after dialing 911   Jacob Helwig
• [asterisk-users] Newbie Question   mail-lists
• [asterisk-users] Call load balancing   Steve Edwards
• [asterisk-users] disable client side hangup after dialing 911   Gergo Csibra
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Drew Gibson
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Queue announcing hold sequence instead of hold time   Drew Gibson
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine   Melcon Moraes
• [asterisk-users] sip tunnel   Pezhman Lali
• [asterisk-users] Queue announcing hold sequence instead of hold time   BJ Weschke
• [asterisk-users] YAACID and manager.conf security   Todd H
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine   Tzafrir Cohen
• [asterisk-users] sip tunnel   Vicky
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Luki
• [asterisk-users] Another Faxing Question   Rob Schall
• [asterisk-users] Call load balancing   David Ruggles
• [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same   Jose Bertuzzi
• [asterisk-users] RE: Coaching in asterisk   Stephen Bosch
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Stephen Bosch
• [asterisk-users] A New Phone Service - www.virtualphoneline.com   Stephen Bosch
• [asterisk-users] Another Faxing Question   Wes Baehr
• [asterisk-users] When to use Echo Cancellation cards?   shadowym
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] RE: Coaching in asterisk   Wai Wu
• [asterisk-users] Which hylafax client ?   Olivier
• [asterisk-users] Call load balancing   Octavio Ruiz (Ta^3)
• [asterisk-users] Which hylafax client ?   Darren Nickerson
• [asterisk-users] Newbie Question   Henry Cobb
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] When to use Echo Cancellation cards?   Lacy Moore - Aspendora
• [asterisk-users] Cdr_mysql compile question   David Ruggles
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] AEL #include file   Tzafrir Cohen
• [asterisk-users] OS X Frequent console disconnects 1.4.1   Bruce Ferrell
• [asterisk-users] Cdr_mysql compile question   David Ruggles
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] disable client side hangup after dialing 911   Olle E Johansson
• [asterisk-users] play file and action only stop if one defined key has been pressed   Thomas Winter
• [asterisk-users] How to enter bridge_native_loop???   Santosh Raghuram
• [asterisk-users] Asterisk 1.2.15 chan_vpb with vpb-driver 4.0   Stephen Bosch
• [asterisk-users] IAX2, DTMF and x86_64.   Stephen Bosch
• [asterisk-users] AEL #include file   Steve Murphy
• [asterisk-users] Recorded file processing app wanted   Steve Edwards
• [asterisk-users] Polycom call parking feature and Asterisk call parking   Stephen Bosch
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] play file and action only stop if one defined key has been pressed   Time Bandit
• [asterisk-users] RE: Coaching in asterisk   Stephen Bosch
• [asterisk-users] play file and action only stop if one defined key has been pressed   Thomas Winter
• [asterisk-users] RE: Coaching in asterisk   Stephen Bosch
• [asterisk-users] AEL #include file   Steve Murphy
• [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN   Brian Capouch
• [asterisk-users] play file and action only stop if one defined key has been pressed   Steve Murphy
• [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN   Davis Sylvester III
• [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN   Brian Capouch
• [asterisk-users] AEL #include file   Philipp Kempgen
• [asterisk-users] spandsp, app_rxfax: apps_Makefile.patch v1.2 > v1.4 = No Workie!   Doug
• [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN   Davis Sylvester III
• [asterisk-users] How to best manage my dial plans as the continue to grow, and grow, and grow....   Christopher Aloi
• [asterisk-users] Which VoIP router and switch to use for medium size business   Zeeshan Zakaria
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Stephen Bosch
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Stephen Bosch
• [asterisk-users] DTMF issue with TDM404   Al
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Mark Davies
• [asterisk-users] Zaptel problem after upgrading to 1.2.16   Mark Davies
• [asterisk-users] installation pb on debian etch   Philippe MONROUX
• [asterisk-users] How to enter bridge_native_loop???   Olle E Johansson
• [asterisk-users] play file and action only stop if one defined key has been pressed   Thomas Winter
• [asterisk-users] Which VoIP router and switch to use for medium size business   Gordon Henderson
• [asterisk-users] Which VoIP router and switch to use for medium size business   Henry Cobb
• [asterisk-users] installation pb on debian etch   Tzafrir Cohen
• [asterisk-users] Polycom call parking feature and Asterisk callparking   Bill Gibbs
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Erick Perez
• [asterisk-users] Connecting two asterisk server.   Sanspareils Greenlans
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Tzafrir Cohen
• [asterisk-users] How to access Voicemail Password in Asterisk without using VoicemailMain????????   jamshed zaidi
• [asterisk-users] How to access Voicemail Password in Asterisk without using VoicemailMain????????   Philipp Kempgen
• [asterisk-users] Is there any variable for Voicemail Password in Asterisk   Tzafrir Cohen
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   jamshed zaidi
• [asterisk-users] FastAGI Question   Lee Jenkins
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Eric "ManxPower" Wieling
• [asterisk-users] chan_misdn and Asterisk 1.4.1 - Early B3 not working any more   Gary Hawkins
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Philipp Kempgen
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Stephen Bosch
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Tzafrir Cohen
• [asterisk-users] Polycom call parking feature and Asterisk callparking   Stephen Bosch
• [asterisk-users] Polycom call parking feature and Asterisk callparking   Eric "ManxPower" Wieling
• [asterisk-users] Polycom call parking feature and Asterisk callparking   Stephen Bosch
• [asterisk-users] Issues with a Linksys SPA 2102 and asterisk   Andres
• [asterisk-users] asterisk on mini-itx   Mail Lists
• [asterisk-users] Polycom call parking feature and Asteriskcallparking   Bill Gibbs
• [asterisk-users] How to enter bridge_native_loop???   Steve Totaro
• [asterisk-users] Security and DTMF   Lee Jenkins
• [asterisk-users] Polycom 501 - Auto answer on one line appearance   Chris Mason (Lists)
• [asterisk-users] asterisk on mini-itx   Chris Mason (Lists)
• [asterisk-users] asterisk on mini-itx   Gordon Henderson
• [asterisk-users] asterisk on mini-itx   Brian Capouch
• [asterisk-users] Which VoIP router and switch to use for medium size business   Lacy Moore - Aspendora
• [asterisk-users] asterisk on mini-itx   Chris Mason (Lists)
• [asterisk-users] Which VoIP router and switch to use for medium size business   Chris Mason (Lists)
• [asterisk-users] Which VoIP router and switch to use for medium size business   Gordon Henderson
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Philipp Kempgen
• [asterisk-users] how to determine the remote IP address when dialing in ooh323   VoIP User
• [asterisk-users] asterisk on mini-itx   Mail Lists
• [asterisk-users] asterisk on mini-itx   Mail Lists
• [asterisk-users] play file and action only stop if one definedkey has been p   Yuan LIU
• [asterisk-users] IAX2 audio issues   Vulpes Velox
• [asterisk-users] IAX2 audio issues   Steve Totaro
• [asterisk-users] IAX2 audio issues   Steve Murphy
• [asterisk-users] How to best manage my dial plans as the continue to grow, and grow, and grow....   C F
• [asterisk-users] IAX2 audio issues   Vulpes Velox
• [asterisk-users] IAX2 audio issues   Vulpes Velox
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] IAX2 audio issues   Vulpes Velox
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Steve Totaro
• [Asterisk-Users] Cannot get back chan_zap.so module!??   OCOSA List Acct.
• [asterisk-users] Re: Recorded file processing app wanted   Justin Newman
• [asterisk-users] Re: Recorded file processing app wanted   Stephen Bosch
• [asterisk-users] Re: Recorded file processing app wanted   Steve Edwards
• [asterisk-users] Follow Up on Cannot get back chan_zap.so module!??   OCOSA List Acct.
• [asterisk-users] Follow Up on Cannot get back chan_zap.so module!??   Tzafrir Cohen
• [asterisk-users] Polycom 501 - Auto answer on one line appearance - Solution   Chris Mason (Lists)
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Olle E Johansson
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Julian Lyndon-Smith
• [asterisk-users] asterisk on mini-itx   Sune Kloppenborg Jeppesen
• [asterisk-users] asterisk on mini-itx   Sune Kloppenborg Jeppesen
• [asterisk-users] How to access Voicemail Password in Asteriskwithout using V   Olle E Johansson
• [asterisk-users] asterisk on mini-itx   Gordon Henderson
• [asterisk-users] How to best manage my dial plans as the continueto grow, and grow, and grow....   Steve Totaro
• [asterisk-users] Re: Recorded file processing app wanted   Steve Totaro
• [asterisk-users] Re: Recorded file processing app wanted   Tzafrir Cohen
• [asterisk-users] Which VoIP router and switch to use for medium size business   Henry Cobb
• [asterisk-users] Which VoIP router and switch to use for medium size business   Gordon Henderson
• [asterisk-users] asterisk on mini-itx   Chris Mason (Lists)
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Matt
• [asterisk-users] Re: Recorded file processing app wanted   Steve Totaro
• [asterisk-users] asterisk on mini-itx   Mail Lists
• [asterisk-users] Fix for TZ values updates for DST   Darrick Hartman
• [Asterisk-Users] Cannot get back chan_zap.so module!??   Russell Bryant
• [asterisk-users] Fix for TZ values updates for DST   Tzafrir Cohen
• [asterisk-users] Fix for TZ values updates for DST   Darrick Hartman
• [asterisk-users] Fix for TZ values updates for DST   Darrick Hartman
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Steve Totaro
• [asterisk-users] IAX2, DTMF and x86_64.   Russell Bryant
• [asterisk-users] Re: queue information into db   nik600
• [asterisk-users] asterisk on mini-itx   shadowym
• [asterisk-users] asterisk on mini-itx   Sune Kloppenborg Jeppesen
• [asterisk-users] How to enter bridge_native_loop???   Santosh Raghuram
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] asterisk on mini-itx   Ira
• [asterisk-users] asterisk on mini-itx   Gordon Henderson
• [asterisk-users] Recorded file processing app wanted   Steve Totaro
• [asterisk-users] Recorded file processing app wanted   Steve Edwards
• [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.   Henry Cobb
• [asterisk-users] Complicated callback solution   marcelobiz at comcast.net
• [asterisk-users] Complicated callback solution   Steve Edwards
• [asterisk-users] g711 -> iLBC garbled voice in 1.4?   Ray Jackson
• [asterisk-users] DST changes for the US   C F
• [asterisk-users] FAX using T38   Matt Riddell (NZ)
• [asterisk-users] FastAGI Question   Lee Jenkins
• [asterisk-users] Grandstream phones   JNA
• [asterisk-users] Re-parking (or transfer) a parked call   Barry D. Hassler
• [asterisk-users] Re-parking (or transfer) a parked call   Marc Archer
• [asterisk-users] Grandstream phones   Gerwin van de Steeg
• [asterisk-users] Complicated callback solution   marcelobiz at comcast.net
• [asterisk-users] Compiling smsq in 1.2   Yuan LIU
• [asterisk-users] Asterisk and Databases   marcelobiz at comcast.net
• [asterisk-users] Problem configuring voice conference   Asterisk Asterisk
• [asterisk-users] asterisk on mini-itx   Ioan Biris
• [asterisk-users] Noob Question   Thomas Patterson
• [asterisk-users] Noob Question   Yuan LIU
• [asterisk-users] Problem configuring voice conference   Paul Hales
• [asterisk-users] Problem configuring voice conference   John covici
• [asterisk-users] How to access Voicemail Passwordin Asteriskwithout using V   jamshed zaidi
• [asterisk-users] asterisk on mini-itx   Sune Kloppenborg Jeppesen
• [asterisk-users] Problems with Voice conferencing   John covici
• [asterisk-users] Re: DTMF not being detected with 1 provider. Works with the other provider...   Evert
• [asterisk-users] How many outgoing phone line/voip account do I need?   Kurt Kuo
• [asterisk-users] How many outgoing phone line/voip account do I need?   Dave Cotton
• [asterisk-users] Pickup group   Khaled Chehab
• [asterisk-users] Coming events in Europe   Olle E Johansson
• [asterisk-users] How many outgoing phone line/voip account do I need?   Paul Hales
• [asterisk-users] Re: DTMF not being detected with 1 provider. Works with the other provider...   Paul Hales
• [asterisk-users] How many outgoing phone line/voip account do I need?   Dave Cotton
• [asterisk-users] Problem configuring voice conference   Asterisk Asterisk
• [asterisk-users] Re: Problems with Voice conferencing   John covici
• [asterisk-users] _ALERT_INFO replacement in 1.4?   Nikhil Jogia
• [asterisk-users] Re: Help: CallerID Name not being sent   Matthew Warren
• [asterisk-users] Problem with H323   Sebastian Bozioreanu
• [asterisk-users] AMI - DBPut   Tomislav Parcina
• [asterisk-users] Citel Handset Gateway DST fix - FYI   Steven
• [asterisk-users] Rebooting ALL polycom phones   Mike
• [asterisk-users] Shoutcast music-on-hold   Jon Schøpzinsky
• [asterisk-users] Single sign on PC + phone?   Patrick
• [asterisk-users] Asterisk and Databases   nik600
• [asterisk-users] Rebooting ALL polycom phones   Julian J. M.
• [asterisk-users] New to Asterisk   NetSys Admin
• [asterisk-users] When to use Echo Cancellation cards?   Wireless
• [asterisk-users] Re: queue information into db   Tomislav Parcina
• [asterisk-users] DST changes for the US   Peder at NetworkOblivion
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   Tomislav Parcina
• [asterisk-users] Rebooting ALL polycom phones   Rob Schall
• [asterisk-users] DST changes for the US   Alex Robar
• [asterisk-users] DST changes for the US   Doug Lytle
• [asterisk-users] DST changes for the US   Dave Fullerton
• [asterisk-users] Rebooting ALL polycom phones   Dave Fullerton
• [asterisk-users] DST changes for the US   Doug Lytle
• [asterisk-users] DST changes for the US   Darryl Dunkin
• [asterisk-users] In Asterisk 1.4.x, Why Digium has two H323 Channels   Thiago Maluf
• [asterisk-users] Rebooting all Aastra phones   Matt
• [asterisk-users] In Asterisk 1.4.x, Why Digium has two H323 Channels   Pavel Jezek
• [asterisk-users] Rebooting all Aastra phones   Dave Cotton
• [asterisk-users] Rebooting all Aastra phones   Matt
• [asterisk-users] New to Asterisk   Matt
• [asterisk-users] DST 2007 Config for Cisco 7970   Gary T. Giesen
• [asterisk-users] Filter IDENT(Port 113) on Linksys router puts remote extensions to one way audio   Zeeshan Zakaria
• [asterisk-users] Rebooting all Aastra phones   Dave Cotton
• [asterisk-users] ACM question   David Ruggles
• [asterisk-users] SIP unicode support ?   Klaus Darilion
• [asterisk-users] Noob Question   Steve Murphy
• [asterisk-users] Re-parking (or transfer) a parked call   Barry D. Hassler
• [asterisk-users] ACM question   Steve Edwards
• [asterisk-users] _ALERT_INFO replacement in 1.4?   Bruce Reeves
• [asterisk-users] Rebooting all Aastra phones   Ira
• [asterisk-users] GXP-2000 DST Change   Ken Williams
• [asterisk-users] Create meetme conference rooms on the flight.   Wai Wu
• [asterisk-users] Rebooting ALL polycom phones   Mike
• [asterisk-users] Rebooting all Aastra phones   Matt
• [asterisk-users] ACM question   David Ruggles
• [asterisk-users] Rebooting all Aastra phones   shadowym
• [asterisk-users] Re-parking (or transfer) a parked call   Doug Lytle
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Brandon Comouche
• [asterisk-users] OT: Sipura DST Rules   Dave Fullerton
• [asterisk-users] Create meetme conference rooms on the flight.   Tristan
• [asterisk-users] SIP unicode support ?   Olle E Johansson
• [asterisk-users] asterisk on mini-itx   voiplist
• [asterisk-users] LIDB/CNAM STORAGE DATABASE NEEDED   Matt
• [asterisk-users] OT: Sipura DST Rules   Curt Shaffer
• [asterisk-users] Incase anyone wanted it - SNOM USA DST settings   Andrew Latham
• [asterisk-users] Re: Filter IDENT(Port 113) on Linksys router puts remote extensions to one way audio   Zeeshan Zakaria
• [asterisk-users] GXV3000 & Speakphone   Davis Sylvester III
• [asterisk-users] deprecated ALERT_INFO var andAMI's Originate command   Octavio Ruiz (Ta^3)
• [asterisk-users] Polycom: warble on registration?   Ken D'Ambrosio
• [asterisk-users] GXP-2000 DST Change   Todd H
• [asterisk-users] New to Asterisk   NetSys Admin
• [asterisk-users] deprecated ALERT_INFO var andAMI's Originate command   Olle E Johansson
• [asterisk-users] Re: DTMF not being detected with 1 provider. Works with the other provider...   Stephen Bosch
• [asterisk-users] Polycom: warble on registration?   Stephen Bosch
• [asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk   C F
• [asterisk-users] Rebooting all Aastra phones   Ioan Indreias
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Stephen Wingfield
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Rick Smith
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Bruce Reeves
• [asterisk-users] _ALERT_INFO replacement in 1.4?   Nikhil Jogia
• [asterisk-users] Fix for TZ values updates for DST   Luis Claudio Santos
• [asterisk-users] Call Back   Luis Claudio Santos
• [asterisk-users] AMI - DBPut   Lee Jenkins
• [asterisk-users] Fix for TZ values updates for DST   Alex Robar
• Spam? Re: [asterisk-users] cmd page crashes AsteriskSVN-branch-1.4-r57207   Hall, Eric M.
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Sean Bright
• Spam? Re: [asterisk-users] cmd page crashesAsteriskSVN-branch-1.4-r57207   Hall, Eric M.
• [asterisk-users] Create meetme conference rooms on the flight.   Steve Totaro
• [asterisk-users] Voicemails with occasional speeded up portions   Anthony Rodgers
• [asterisk-users] TDM-400, Polycom SIP phones, and echo problems   Stephen Bosch
• [asterisk-users] Call Back   Lee Jenkins
• [asterisk-users] Voicemails with occasional speeded up portions   Stephen Bosch
• [asterisk-users] Playback 5% Too Fast?   David Brazier
• [asterisk-users] RE: Playback 0.5% Too Fast?   David Brazier
• [asterisk-users] Nomination for Coolest App in 2007   Brad Templeton
• [asterisk-users] Call Back   Ivo Zivkov
• [asterisk-users] Call Back   Klaverstyn, David C
• [asterisk-users] Call Back   Stephen Bosch
• [asterisk-users] Call Back   Klaverstyn, David C
• [asterisk-users] Call Back   Ivo Zivkov
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] Call Back   Ivo Zivkov
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Bruce Reeves
• [asterisk-users] Single sign on PC + phone?   Trevor Peirce
• [asterisk-users] _ALERT_INFO replacement in 1.4?   Olle E Johansson
• [asterisk-users] Re: AMI - DBPut   Tomislav Parcina
• [asterisk-users] Update Asterisk 1.2.12 to 1.4.1 ?   Noc Phibee
• [asterisk-users] Update Asterisk 1.2.12 to 1.4.1 ?   Olle E Johansson
• [asterisk-users] Polycom: warble on registration?   Steven Ringwald
• [asterisk-users] Call load balancing   Tim Panton
• [asterisk-users] Re: SIP unicode support ?   Benny Amorsen
• [asterisk-users] Single sign on PC + phone?   Patrick
• [asterisk-users] MusicOnHold stops after upgrade from 1.4.0 to 1.4.1   Damian Adamski
• [asterisk-users] Re: Number of SIP messages per minute   Tomislav Parcina
• [asterisk-users] CDR and CallerID   Mike
• [asterisk-users] Re: 1.4 compile issue   Tomislav Parcina
• [asterisk-users] GTalk/Jabber passing audio in 1.4.1!   Charles Wang
• [asterisk-users] Number of SIP messages per minute   Matt
• [asterisk-users] Queue - Call delivery problems - Asterisk 1.4 - autofill=yes   equis software
• [asterisk-users] voicemail scenario   richard Coco
• [asterisk-users] French PRI channel - exact signaling used   Cedric MILLET
• [asterisk-users] How many outgoing phone line/voip account do I need?   Henry Cobb
• [asterisk-users] IAX2 Question (Asterisk 1.4 tarball)   David Ruggles
• [asterisk-users] How many outgoing phone line/voip account do I need?   Chris Bagnall
• [asterisk-users] SIP hardphones with good jitter tolerance   Chris Bagnall
• [asterisk-users] FW: Seamless Multi Office Asterisk Deployment   Brandon Comouche
• [asterisk-users] New to Asterisk   Steve Murphy
• [asterisk-users] Call Back   Stephen Bosch
• [asterisk-users] DST and VM timestamp   Damon Estep
• [asterisk-users] French PRI channel - exact signaling used   younss azzayani
• [asterisk-users] How many outgoing phone line/voip account do I need?   Henry Cobb
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Héctor Maldonado
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] DST and VM timestamp   Dave Fullerton
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Tzafrir Cohen
• [asterisk-users] How to match wild card inside a GoToIf?   Ricardo Carvalho
• [asterisk-users] IAX2 Question (Asterisk 1.4 tarball)   David Ruggles
• [asterisk-users] cisco sip firmware update for cisco 7970   Connolly, Tim
• [asterisk-users] How to match wild card inside a GoToIf?   Connolly, Tim
• [asterisk-users] DST and VM timestamp   Tzafrir Cohen
• [asterisk-users] great problem with sounds and ztdummy   Tzafrir Cohen
• [asterisk-users] Getting 7970 to update   Connolly, Tim
• [asterisk-users] How to match wild card inside a GoToIf?   Ken Williams
• [asterisk-users] How to match wild card inside a GoToIf?   Ken Williams
• [asterisk-users] DST and VM timestamp   Damon Estep
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Matt Florell
• [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 48   David Cook
• [asterisk-users] Number of SIP messages per minute   Eric "ManxPower" Wieling
• [asterisk-users] asterisk 1.2.15 fax   Khaled Chehab
• [asterisk-users] Getting 7970 to update   Connolly, Tim
• [asterisk-users] How to match wild card inside a GoToIf?   Eric "ManxPower" Wieling
• [asterisk-users] RE: In Asterisk 1.4.x, Why Digium has two H323 channels?   Thiago Maluf
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Héctor Maldonado
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Matt Florell
• [asterisk-users] voicemail scenario   richard Coco
• [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?   Héctor Maldonado
• [asterisk-users] Asterisknow with video and X-Lite not quite working   Benedikt Franz
• [asterisk-users] RE: In Asterisk 1.4.x, Why Digium has two H323 channels?   Pavel Jezek
• [asterisk-users] Number of SIP messages per minute   Luki
• [asterisk-users] great problem with sounds and ztdummy   Germán Aracil Boned
• [asterisk-users] Press quotes needed from ENUM users on Asterisk   John Todd
• [asterisk-users] Number of SIP messages per minute   Matt
• [asterisk-users] Call Back   Samy Antoun
• [asterisk-users] Call Back   Vernier Umali
• [asterisk-users] voicemail scenario   Dovid B
• [asterisk-users] Digium S101i - Adapter DTMF works perfeclty   Joseph
• [asterisk-users] Re: Asterisknow with video and X-Lite not quite working   dave cantera
• [asterisk-users] Strange issue SIP URI to follow me busy signal   George A. Roberts IV
• [asterisk-users] Call Back   Stephen Bosch
• [asterisk-users] T1 Integrator Birch   John Schmerold
• [asterisk-users] Re: Asterisknow with video and X-Lite not quite working   Benedikt Franz
• [asterisk-users] Asterisknow with video and X-Lite not quite working   Matt Riddell (NZ)
• [asterisk-users] strange things on call transfer   Christoph Fürstaller
• [asterisk-users] Re: Asterisknow with video and X-Lite not quiteworking   Biju
• [asterisk-users] voicemail scenario   richard Coco
• [asterisk-users] TDM-400, Polycom SIP phones, and echo problems   Peter Doyle
• [asterisk-users] strange things on call transfer   MIVOC Systemes SAS
• [asterisk-users] beronet BN4S0   Thomas Stein
• [asterisk-users] beronet BN4S0   Massimo Nuvoli
• [asterisk-users] ChanSpy with Record() : Doesnt seem to work for an unbridged SIP call