December 2007 Archives by date

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Starting: Sat Dec 1 00:17:07 CST 2007
Ending: Mon Dec 31 23:50:27 CST 2007
Messages: 1646

• [asterisk-users] Off-Topic: Avaya   Jesse Molina
• [asterisk-users] REFER mesage extraction using SIP_HEADER   Norman W. Franke
• [asterisk-users] Using existing extensions.conf macros, and co-habitation   Anselm Martin Hoffmeister
• [asterisk-users] Copy or Make + Make Install   Philipp Kempgen
• [asterisk-users] REFER mesage extraction using SIP_HEADER   Arpit Mehta
• [asterisk-users] Outgoing PSTN calls , unusable voice quality   Joanna Liza Mariazeta
• [asterisk-users] IAX complaints? What are they?   Thomas Kenyon
• [asterisk-users] Only call me once   Joanna Liza Mariazeta
• [asterisk-users] Consulting/Integration Services Non-US & US *u   dave cantera
• [asterisk-users] Asterisk 1.4.15 Voicemail   Il Neofita
• [asterisk-users] Consulting/Integration Services Non-US & US *u   Steve Edwards
• [asterisk-users] cdr_pgsql error in 1.4.15   equis software
• [asterisk-users] Consulting/Integration Services Non-US & US *u   dave cantera
• [asterisk-users] Only call me once   Anthony Francis
• [asterisk-users] asterisk as non-root/best practices   Anthony Francis
• [asterisk-users] Using existing extensions.conf macros, and co-habitation   Anthony Francis
• [asterisk-users] Cross-compiling asterisk-1.4 for Debian on a slug   Fabiano Sidler
• [asterisk-users] Increasing the voice volume from the diguim cards   bilal ghayyad
• [asterisk-users] Do While loop   Vincent
• [asterisk-users] Realtime SIP & BLF   Daniel Hazelbaker
• [asterisk-users] Cross-compiling asterisk-1.4 for Debian on a slug   Philipp Kempgen
• [asterisk-users] Increasing the voice volume from the diguim cards   Matthew Fredrickson
• [asterisk-users] IAX complaints? What are they?   Andrew Kohlsmith
• [asterisk-users] IAX complaints? What are they?   Philipp Kempgen
• [asterisk-users] Outgoing PSTN calls , unusable voice quality   Salvatore Giudice
• [asterisk-users] Using existing extensions.conf macros, and co-habitation   Philip Prindeville
• [asterisk-users] Only call me once   Adam Moffett
• [asterisk-users] Asterisk 1.4.15 crash without generating core file   Matt Riddell
• [asterisk-users] Registration state: Failed   Matt Riddell
• [asterisk-users] Requiring a login to a phone   Steve Johnson
• [asterisk-users] Off-Topic: Avaya   Salvatore Giudice
• [asterisk-users] Requiring a login to a phone   Alex Balashov
• [asterisk-users] cdr_pgsql error in 1.4.15   Tilghman Lesher
• [asterisk-users] Asterisk on Pcengines Alix board   Vincent
• [asterisk-users] Requiring a login to a phone   Tilghman Lesher
• [asterisk-users] Requiring a login to a phone   Josh Richards
• [asterisk-users] Answer Machine/Fax/modem detection   Tong
• [asterisk-users] Asterisk on Pcengines Alix board   John Faubion
• [asterisk-users] Answer Machine/Fax/modem detection   Mindaugas Kezys
• [asterisk-users] DID provider   srgqwerty
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] Asterisk on Solaris   Mike Clark
• [asterisk-users] Outgoing PSTN calls , unusable voice quality   Veselin Kantsev
• [asterisk-users] get SIP extension status without calling it   dave cantera
• [asterisk-users] Answering Machine Detection   dave cantera
• [asterisk-users] Asterisk on Solaris   Vivek Shrivastava
• [asterisk-users] Cross-compiling asterisk-1.4 for Debian on a slug   Fabiano Sidler
• [asterisk-users] get SIP extension status without calling it   Tilghman Lesher
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] Answering Machine Detection   Tong
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] Dictaphone Freedom interface to Asterisk ABE   R. Paul Warriner
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] What is the status and future of chan_mobile   Robert Moskowitz
• [asterisk-users] Cross-compiling asterisk-1.4 for Debian on a slug   Philipp Kempgen
• [asterisk-users] T1 Timing Troubleshooting   Jonathan C. Bailey
• [asterisk-users] What is the status and future of chan_mobile   Tilghman Lesher
• [asterisk-users] When calling in via AGI, gsm sound file plays but sometimes drops out   Dominic Son
• [asterisk-users] Asterisk install beta testing/config help   James Cox
• [asterisk-users] Asterisk install beta testing/config help   Bryan M. Johns
• [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX   John Constalgie
• [asterisk-users] IAX complaints? What are they?   Vincent
• [asterisk-users] get SIP extension status without calling it   Raj Jain
• [asterisk-users] Softswitch digim   Carlos Rojas
• [asterisk-users] IAX complaints? What are they?   Zoa
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] setting up two asterisk server as ss7 back to back.   Goke Aruna
• [asterisk-users] get SIP extension status without calling it   Richard Revels
• [asterisk-users] Softswitch digim   Bill Hackensack
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] IAX complaints? What are they?   SIP
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] get SIP extension status without calling it   Richard Revels
• [asterisk-users] Subject: Newb Question   Vidura Senadeera
• [asterisk-users] get SIP extension status without calling it   Vieri
• [asterisk-users] Asterisk 1.4.15 sip.conf register   Joe Morsbach
• [asterisk-users] Softswitch digim   ram
• [asterisk-users] How to setup redundant SIP peers   ram
• [asterisk-users] Oracle and asterisk   Bhrugu Mehta
• [asterisk-users] New VICIDIAL astGUIclient Release: 2.0.4   Matt Florell
• [asterisk-users] Oracle and asterisk   Tilghman Lesher
• [asterisk-users] Oracle and asterisk   Bhrugu Mehta
• [asterisk-users] OT - How to add a new TAPI driver on an XP system ?   Olivier
• [asterisk-users] MeetMe Conference on Asterisk-1.4.13   GNUbie
• [asterisk-users] Nov 28, 2007 Asterisk Poll Results   randulo
• [asterisk-users] get REMOTE SIP extension status without calling it   Abel Molina Landrián
• [asterisk-users] Underground Asterisk Command Set?   Jeng Yu
• [asterisk-users] Attended transfer to Queue   Steve Davies
• [asterisk-users] Attended transfer to Queue   Steve Davies
• [asterisk-users] SPA-3102 Registration Failed .. need advise   Newbie
• [asterisk-users] Problem: Using timelimit (L) and Macro (M) in Dial from AGI   Rajeev Natarajan
• [asterisk-users] Problem: Using timelimit (L) and Macro (M) in Dial from AGI   Rajeev Natarajan
• [asterisk-users] SLA: Handling of errors in outgoing call   Steve Langstaff
• [asterisk-users] Asterisk 1.4.15 crash without generating core file   Tzafrir Cohen
• [asterisk-users] MeetMe Conference on Asterisk-1.4.13   Tzafrir Cohen
• [asterisk-users] How to setup redundant SIP peers   Thomas Balsfulland
• [asterisk-users] Increasing the voice volume from the diguim cards   Mohammad Shokuie
• [asterisk-users] MeetMe Conference on Asterisk-1.4.13   GNUbie
• [asterisk-users] Asterisk install beta testing/config help   Tzafrir Cohen
• [asterisk-users] MeetMe Conference on Asterisk-1.4.13   Tzafrir Cohen
• [asterisk-users] SPA-3102 Registration Failed .. need advise   Vincent
• [asterisk-users] Asterisk 1.4.15 crash without generating core file   equis software
• [asterisk-users] cdr_pgsql error in 1.4.15   equis software
• [asterisk-users] MeetMe Conference on Asterisk-1.4.13   GNUbie
• [asterisk-users] Underground Asterisk Command Set?   Jared Smith
• [asterisk-users] Underground Asterisk Command Set?   Richard Revels
• [asterisk-users] Replacing Skype with Asterisk Peering Servers - and Security   Siju George
• [asterisk-users] Replacing Skype with Asterisk Peering Servers - and Security   Philipp Kempgen
• [asterisk-users] Problem: Using timelimit (L) and Macro (M) in Dial from AGI   Mark Michelson
• [asterisk-users] MeetMe Conference on Asterisk-1.4.13   Tzafrir Cohen
• [asterisk-users] Red Alarm TE420 with E1s - R2   Roger C. Beraldi Martins
• [asterisk-users] Replacing Skype with Asterisk Peering Servers - and Security   zoa
• [asterisk-users] Problem: Using timelimit (L) and Macro (M) in Dial from AGI   Rajeev Natarajan
• [asterisk-users] Red Alarm TE420 with E1s - R2   Richard Lyman
• [asterisk-users] Click2call from Thunderbird   Olivier
• [asterisk-users] Underground Asterisk Command Set?   Anthony Francis
• [asterisk-users] Do While loop   Mojo with Horan & Company, LLC
• [asterisk-users] Replacing Skype with Asterisk Peering Servers - and Security   Alan Lord
• [asterisk-users] Anyone here using JUNGHANNS.net douBRI 2.0 ISDN ?   Elijah
• [asterisk-users] Underground Asterisk Command Set?   Tomás Laureano Peralta Tormey
• [asterisk-users] Red Alarm TE420 with E1s - R2   Roger C. Beraldi Martins
• [asterisk-users] Red Alarm TE420 with E1s - R2   Carlos Chavez
• [asterisk-users] Outgoing PSTN calls , unusable voice quality   Salvatore Giudice
• [asterisk-users] Anyone here using JUNGHANNS.net douBRI 2.0 ISDN ?   Stelios Koroneos
• [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??   Stefan Guenther
• [asterisk-users] Red Alarm TE420 with E1s - R2   Richard Lyman
• [asterisk-users] MWI error   Marc LEURENT
• [asterisk-users] Hoteling   Rob Schall
• [asterisk-users] Anyone here using JUNGHANNS.net douBRI 2.0 ISDN ?   Dave Cotton
• [asterisk-users] Hoteling   Philipp Kempgen
• [asterisk-users] Anyone here using JUNGHANNS.net douBRI 2.0 ISDN ?   Tzafrir Cohen
• [asterisk-users] Red Alarm TE420 with E1s - R2   Roger C. Beraldi Martins
• [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??   Lacy Moore
• [asterisk-users] MWI error   Alex Balashov
• [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX   Edwin Lam
• [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??   Jason Parker
• [asterisk-users] Red Alarm TE420 with E1s - R2   Tilghman Lesher
• [asterisk-users] Hoteling   Rob Schall
• [asterisk-users] Underground Asterisk Command Set?   Tilghman Lesher
• [asterisk-users] Adhearsion Install Fails.   Douglas Garstang
• [asterisk-users] IAX complaints? What are they?   Vincent
• [asterisk-users] Do While loop   Vincent
• [asterisk-users] Hoteling   Gregory Malsack
• [asterisk-users] Red Alarm TE420 with E1s - R2   Philipp Kempgen
• [asterisk-users] Underground Asterisk Command Set?   Philipp Kempgen
• [asterisk-users] Hoteling   Philipp Kempgen
• [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX   John Constalgie
• [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??   Mojo with Horan & Company, LLC
• [asterisk-users] Red Alarm TE420 with E1s - R2   Richard Lyman
• [asterisk-users] Red Alarm TE420 with E1s - R2   Philipp Kempgen
• [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??   Stefan Guenther
• [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??   Matthew Yingling
• [asterisk-users] Asterisk and Ekiga Chat   Alan WN Hanley
• [asterisk-users] Hoteling   Philipp von Klitzing
• [asterisk-users] Only call me once   Paul Hales
• [asterisk-users] Red Alarm TE420 with E1s - R2   Richard Lyman
• [asterisk-users] Shared line appearance phones?   shadowym
• [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX   Edwin Lam
• [asterisk-users] Cepstral's Allison is having troublespeaking clearly   Shaun Ewing
• [asterisk-users] IBM x3400 w/ Digium TE220   Edwin Lam
• [asterisk-users] MeetMe Conference on Asterisk-1.4.13   GNUbie
• [asterisk-users] Phone with public address functionality   Doug Meredith
• [asterisk-users] Queue App - crash (1.4.15)   equis software
• [asterisk-users] Phone with public address functionality   Paul Hales
• [asterisk-users] Queue App - (1.4.15) free agents with callers waiting   equis software
• [asterisk-users] Phone with public address functionality   Doug
• [asterisk-users] Anyone here using JUNGHANNS.net douBRI 2.0 ISDN ?   Elijah
• [asterisk-users] Problem forwarding voicemail messages   Pepo
• [asterisk-users] Red Alarm TE420 with E1s - R2   Tzafrir Cohen
• [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)   Nick Seraphin
• [asterisk-users] Asterisk on multi-homed systems   Shlomo Dubrowin
• [asterisk-users] Red Alarm TE420 with E1s - R2   Tzafrir Cohen
• [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??   Tzafrir Cohen
• [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX   John Constalgie
• [asterisk-users] Red Alarm TE420 with E1s - R2   Roger C. Beraldi Martins
• [asterisk-users] Explain AGI and AMI   Olivier
• [asterisk-users] Gtalk callerID   Abel Molina Landrián
• [asterisk-users] Red Alarm TE420 with E1s - R2   Tzafrir Cohen
• [asterisk-users] My AsteriskNo unable to registration   Guillermo Rodriguez
• [asterisk-users] My AsteriskNo unable to registration   Guillermo Rodriguez
• [asterisk-users] My AsteriskNo unable to registration   Newbie
• [asterisk-users] Red Alarm TE420 with E1s - R2   Roger C. Beraldi Martins
• [asterisk-users] enable eyeBeam to accept only one call   Joao Pereira
• [asterisk-users] enable eyeBeam to accept only one call   Steve Davies
• [asterisk-users] [SOLVED] Re: Soundcard necessary on an asterisk server to get output of playback()??   Stefan Guenther
• [asterisk-users] enable eyeBeam to accept only one call   Terry Wade
• [asterisk-users] enable eyeBeam to accept only one call   Yehavi Bourvine +972-8-9489444
• [asterisk-users] pstn call waiting and zap   Patricio Valarezo Lozano
• [asterisk-users] Explain AGI and AMI   Steve Edwards
• [asterisk-users] [SOLVED] Re: Soundcard necessary on an asterisk server to get output of playback()??   Steve Edwards
• [asterisk-users] Red Alarm TE420 with E1s - R2   Philipp Kempgen
• [asterisk-users] Digium Support Comparison with Sangoma   Kashif Naeem
• [asterisk-users] System Specifications for 60 Extensions   Kashif Naeem
• [asterisk-users] Red Alarm TE420 with E1s - R2   Philipp Kempgen
• [asterisk-users] Filtering duplicate RTP packets   Boris Bakchiev
• [asterisk-users] Problem forwarding voicemail messages   Atis Lezdins
• [asterisk-users] Asterisk on multi-homed systems   Freddi Hansen
• [asterisk-users] Voicemail box   Lees, James (UK)
• [asterisk-users] Soundcard necessary on an asterisk server toget output of playback()??   Stefan Guenther
• [asterisk-users] enable eyeBeam to accept only one call   Alex Balashov
• [asterisk-users] Voicemail box   Gordon Henderson
• [asterisk-users] MWI error   Marc LEURENT
• [asterisk-users] New User   ticket john
• [asterisk-users] MWI error   Jared Smith
• [asterisk-users] New User   Jared Smith
• [asterisk-users] IRVs Asterisk example configuration   dadsadsadf dsadasdsa
• [asterisk-users] Explain AGI and AMI   Olivier
• [asterisk-users] Phone with public address functionality   Doug Meredith
• [asterisk-users] Echo cancellation and DTMF from the Asterisk console?   Nikhil Nair
• [asterisk-users] New User   Mojo with Horan & Company, LLC
• [asterisk-users] Echo cancellation and DTMF from the Asterisk console?   Mojo with Horan & Company, LLC
• [asterisk-users] Echo cancellation and DTMF from the Asterisk console?   Mojo with Horan & Company, LLC
• [asterisk-users] Echo cancellation and DTMF from the Asterisk console?   Mojo with Horan & Company, LLC
• [asterisk-users] Echo cancellation and DTMF from the Asterisk console?   Alan Lord
• [asterisk-users] IRVs Asterisk example configuration   Vincent
• [asterisk-users] Call center scenario   Jay Moore
• [asterisk-users] Fax on asterisk   Everton Goularth
• [asterisk-users] IBM x3400 w/ Digium TE220   Matthew Fredrickson
• [asterisk-users] Which rackable server for TDM2400 ?   Olivier
• [asterisk-users] IBM x3400 w/ Digium TE220   Edwin Lam
• [asterisk-users] Echo cancellation and DTMF from the Asterisk console?   Nikhil Nair
• [asterisk-users] pstn call waiting and zap   C F
• [asterisk-users] Filtering duplicate RTP packets   Boris Bakchiev
• [asterisk-users] pstn call waiting and zap   John covici
• [asterisk-users] Fax on asterisk   Alex Balashov
• [asterisk-users] Melbourne Asterisk meetup - again   Paul Hales
• [asterisk-users] [Fwd: load test zap channels (in and out)]   Benjamin Jacob
• [asterisk-users] New feature: calling all bug marshals   Philip Prindeville
• [asterisk-users] New feature: calling all bug marshals   Steve Edwards
• [asterisk-users] Multiple contacts.   Alex Balashov
• [asterisk-users] New feature: calling all bug marshals   Philip Prindeville
• [asterisk-users] IRVs Asterisk example configuration   Gordon Henderson
• [asterisk-users] Disturbance "noise" in the background for digium card   bilal ghayyad
• [asterisk-users] [HELP] Problems with VOIP organization   Григорий Никоноров
• [asterisk-users] MWI error   Marc LEURENT
• [asterisk-users] Text-To-Speech synthesizer--help required   srinivas Antarvedi
• [asterisk-users] Increasing the voice volume from the digium card   bilal ghayyad
• [asterisk-users] My AsteriskNo unable to registration   Guillermo Rodriguez
• [asterisk-users] New feature: calling all bug marshals   Ryan Burke
• [asterisk-users] [HELP] Problems with VOIP organization   Bruce Reeves
• [asterisk-users] Multiple contacts.   Steve Totaro
• [asterisk-users] Multiple contacts.   SIP
• [asterisk-users] New feature: calling all bug marshals   Per Jessen
• [asterisk-users] New feature: calling all bug marshals   Tony Mountifield
• [asterisk-users] Bad behaviour between X-Lite 3.0 and Asterisk   Benoît Mérouze
• [asterisk-users] Multiple contacts.   Steve Totaro
• [asterisk-users] SIP-Realtime and sip reload   Henrik Buchholz
• [asterisk-users] New feature: calling all bug marshals   Trevor G. Hammonds
• [asterisk-users] New feature: calling all bug marshals   Ryan Burke
• [asterisk-users] redirected call failure   Mostyn Surname
• [asterisk-users] Adtran supervision problems   Jordan Novak
• [asterisk-users] Adtran supervision problems   Steve Totaro
• [asterisk-users] Multiple contacts.   Alex Balashov
• [asterisk-users] Use of "slin" as a codec   Whisker, Peter
• [asterisk-users] Multiple contacts.   SIP
• [asterisk-users] Use of "slin" as a codec   Whisker, Peter
• [asterisk-users] Multiple contacts.   Alex Balashov
• [asterisk-users] Multiple contacts.   Lutgring, Sam
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Michael Melia Jr.
• [asterisk-users] Popcorn ( was Re: New feature: calling all bug marshals )   John Novack
• [asterisk-users] Multiple contacts.   Tzafrir Cohen
• [asterisk-users] Multiple contacts.   Alex Balashov
• [asterisk-users] Disturbance "noise" in the background for digium card   Russell Bryant
• [asterisk-users] pstn call waiting and zap   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Jared Smith
• [asterisk-users] New feature: calling all bug marshals   Steve Edwards
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Guillermo Salas M.
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Rob Schall
• [asterisk-users] Softswitch digim   Mojo with Horan & Company, LLC
• [asterisk-users] New feature: calling all bug marshals   John Novack
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Tzafrir Cohen
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Olivier
• [asterisk-users] New feature: calling all bug marshals   Philip Prindeville
• [asterisk-users] Asterisk and TDM400P   Gustavo Gonzalez
• [asterisk-users] No timezone in Voicemail email?   Jason Martin
• [asterisk-users] No timezone in Voicemail email?   Jason Parker
• [asterisk-users] No timezone in Voicemail email?   Anthony Messina
• [asterisk-users] Text-To-Speech synthesizer--help required   Doug
• [asterisk-users] No timezone in Voicemail email?   Anthony Messina
• [asterisk-users] New feature: calling all bug marshals   Ira
• [asterisk-users] G729/MOH Quality   Darryl Dunkin
• [asterisk-users] Asterisk server and DSCP QOS   Steve Johnson
• [asterisk-users] Asterisk server and DSCP QOS   Darryl Dunkin
• [asterisk-users] New feature: calling all bug marshals   Philip Prindeville
• [asterisk-users] pstn call waiting and zap   Patricio Valarezo Lozano
• [asterisk-users] Multiple contacts.   Steve Totaro
• [asterisk-users] Re: Asterisk server and DSCP QOS   Steve Johnson
• [asterisk-users] Multiple contacts.   Steve Totaro
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Stefan Guenther
• [asterisk-users] Asterisk server and DSCP QOS   Darryl Dunkin
• [asterisk-users] Cisco 7960 to 2 SIP servers?   Shawn Laemmrich
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Michael Melia Jr.
• [asterisk-users] SIP-Realtime and sip reload   JR Richardson
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Marco Mouta
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Dean Collins
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   shadowym
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D   Josué Conti
• [asterisk-users] My AsteriskNo unable to registration   Newbie
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR   Josué Conti
• [asterisk-users] Polycom Soundpoint (NO LINE)   Ricardo Melendez
• [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)   joakimsen at gmail.com
• [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)   Paul Hales
• [asterisk-users] Multiple contacts.   Lacy Moore
• [asterisk-users] Multiple contacts.   Alex Balashov
• [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)   Jon Pounder
• [asterisk-users] s, CDR and NoCDR in v1.4.10.1   Peder at NetworkOblivion
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Darrick Hartman
• [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)   Paul Hales
• [asterisk-users] TDm804B   Jerry Geis
• [asterisk-users] Door Intercom? (was: Re: Phonewith public address functionality)   Dean Collins
• [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)   C F
• [asterisk-users] Running AGI script if condition met?   Vincent
• [asterisk-users] Polycom Soundpoint (NO LINE)   Doug
• [asterisk-users] Running AGI script if condition met?   Philipp Kempgen
• [asterisk-users] Running AGI script if condition met?   Vincent
• [asterisk-users] Asterisk server and DSCP QOS   Igor A. Goncharovsky
• [asterisk-users] Can Asterix seperate the signalling and the media ip's with Quintum   Shaun Wingrin
• [asterisk-users] G729 on wrong bus   broadband Voice
• [asterisk-users] Multiple contacts.   Lacy Moore
• [asterisk-users] Remote Office, Centrally Shared Voicemail   Anthony Messina
• [asterisk-users] Multiple contacts.   Andreas Sikkema
• [asterisk-users] C&W ISDN110   Geraint
• [asterisk-users] asterisk performance   jorain
• [asterisk-users] GSM and CDMA Gateways   broadband Voice
• [asterisk-users] Polycom call drops   YT Lim
• [asterisk-users] Multiple contacts.   Benny Amorsen
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Tzafrir Cohen
• [asterisk-users] DeadAgi   Bhrugu Mehta
• [asterisk-users] s, CDR and NoCDR in v1.4.10.1   Atis Lezdins
• [asterisk-users] Increasing the voice volume from the digium card   bilal ghayyad
• [asterisk-users] Fax on asterisk   Dinesh Nair
• [asterisk-users] Multiple contacts.   Hans Witvliet
• [asterisk-users] Multiple contacts.   Benny Amorsen
• [asterisk-users] IVR problem (Trixbox)   Osvaldo Holanda Peixoto
• [asterisk-users] Increasing the voice volume from the digium card   Mohammad Shokuie
• [asterisk-users] Play Beep instead of MOH   Anciso, Roy
• [asterisk-users] Play Beep instead of MOH   Doug Lytle
• [asterisk-users] Play Beep instead of MOH   Steve Davies
• [asterisk-users] Asterisk & Cisco calling Name   Dinesh Nair
• [asterisk-users] Call Center Scenario -- take 2   Jay Moore
• [asterisk-users] Increasing the voice volume for digium card   bilal ghayyad
• [asterisk-users] Logging in and off sessions in the dialplan   Ricardo Carvalho
• [asterisk-users] Call Center Scenario -- take 2   Nigel Kendrick
• [asterisk-users] DeadAgi   Steve Edwards
• [asterisk-users] Cisco power injector with GXP2000 phones   Ricardo Carvalho
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR   Moises Silva
• [asterisk-users] Logging in and off sessions in the dialplan   Gordon Henderson
• [asterisk-users] Asterisk & Cisco calling Name   John Bittner
• [asterisk-users] Setting custom field in CDR   Mike
• [asterisk-users] Cisco power injector with GXP2000 phones   Salvatore Giudice
• [asterisk-users] Call Center Scenario -- take 2   Alex Balashov
• [asterisk-users] Cisco power injector with GXP2000 phones   Tony Plack
• [asterisk-users] Cisco power injector with GXP2000 phones   Jon Pounder
• [asterisk-users] Logging in and off sessions in the dialplan   Ricardo Carvalho
• [asterisk-users] Cisco power injector with GXP2000 phones   Jon Pounder
• [asterisk-users] Cisco 7960 to 2 SIP servers?   asterisk
• [asterisk-users] Cisco power injector with GXP2000 phones   Ricardo Carvalho
• [asterisk-users] Cisco power injector with GXP2000 phones   Salvatore Giudice
• [asterisk-users] SIP-Realtime and sip reload   Henrik Buchholz
• [asterisk-users] Cisco power injector with GXP2000 phones   Jon Pounder
• [asterisk-users] Setting custom field in CDR   Mike
• [asterisk-users] Setting custom field in CDR   Atis Lezdins
• [asterisk-users] Cisco power injector with GXP2000 phones   Salvatore Giudice
• [asterisk-users] Setting custom field in CDR   Carlos Chavez
• [asterisk-users] Cisco 7960 to 2 SIP servers?   Salvatore Giudice
• [asterisk-users] Setting custom field in CDR   Tilghman Lesher
• [asterisk-users] Setting custom field in CDR   Steve Edwards
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR   Josué Conti
• [asterisk-users] Dial() Macro option error in 1.4.15   Anthony Messina
• [asterisk-users] Setting Multiple Values via func_odbc ...?   Douglas Garstang
• [asterisk-users] CDR Function in Hangup Channel   Joshua Colp
• [asterisk-users] Setting Multiple Values via func_odbc ...?   Alex Balashov
• [asterisk-users] Setting Multiple Values via func_odbc ...?   Douglas Garstang
• [asterisk-users] Dial() Macro option error in 1.4.15   Mark Michelson
• [asterisk-users] Setting Multiple Values via func_odbc ...?   Tilghman Lesher
• [asterisk-users] Dial() Macro option error in 1.4.15   Anthony Messina
• [asterisk-users] G729 on wrong bus   joakimsen at gmail.com
• [asterisk-users] Call Center Scenario -- take 2   Jay Moore
• [asterisk-users] Probems receiving 200ok message   Frederico Madeira
• [asterisk-users] SIP-Realtime and sip reload   SIP
• [asterisk-users] G729 on wrong bus   broadband Voice
• [asterisk-users] Dial() Macro option error in 1.4.15   Mark Michelson
• [asterisk-users] G729 on wrong bus   Darryl Dunkin
• [asterisk-users] Call Center Scenario -- take 2   Mojo with Horan & Company, LLC
• [asterisk-users] CDR Function in Hangup Channel   Douglas Garstang
• [asterisk-users] SIP-Realtime and sip reload   Torbjörn Abrahamsson
• [asterisk-users] Cisco power injector with GXP2000 phones   Tony Plack
• [asterisk-users] VOIP Users Conference for Friday Dec 7 @ 12 Noon EST   randulo
• [asterisk-users] CDR Function in Hangup Channel   Douglas Garstang
• [asterisk-users] Cisco power injector with GXP2000 phones   Kristian Kielhofner
• [asterisk-users] Dial() Macro option error in 1.4.15   Anthony Messina
• [asterisk-users] Voicemail Question   Lutgring, Sam
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR   Moises Silva
• [asterisk-users] CDR Function in Hangup Channel   Tilghman Lesher
• [asterisk-users] Cisco power injector with GXP2000 phones   Jon Pounder
• [asterisk-users] CDR Function in Hangup Channel   Douglas Garstang
• [asterisk-users] "Happy Birthday Asterisk"   Dean Collins
• [asterisk-users] Perl FastAGI service port.   Alex Balashov
• [asterisk-users] CDR Function in Hangup Channel   Steve Edwards
• [asterisk-users] CDR Function in Hangup Channel   Steve Edwards
• [asterisk-users] New feature: calling all bug marshals   Joshua Colp
• [asterisk-users] Asterisk & Cisco calling Name   Greg Oliver
• [asterisk-users] CDR Function in Hangup Channel   Douglas Garstang
• [asterisk-users] Print CALLERID in CLI during "pri debug "   Arpit Mehta
• [asterisk-users] scrubbing a call list to remove cell phone numbers   Jason Wolfe
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR   Steve Underwood
• [asterisk-users] PRI: calling an Unallocated Number   Jorge Mendoza
• [asterisk-users] GSM and CDMA Gateways   Cameron Hissey
• [asterisk-users] "Happy Birthday Asterisk"   Steve Totaro
• [asterisk-users] scrubbing a call list to remove cell phone numbers   Steve Totaro
• [asterisk-users] GSM and CDMA Gateways   Dovid B
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR   Moises Silva
• [asterisk-users] "Happy Birthday Asterisk"   Philip Prindeville
• [asterisk-users] New feature: calling all bug marshals   Philip Prindeville
• [asterisk-users] Voicemail Question   C F
• [asterisk-users] Probems receiving 200ok message   C F
• [asterisk-users] Print CALLERID in CLI during "pri debug "   Arpit Mehta
• [asterisk-users] scrubbing a call list to remove cell phone numbers   Chris Bagnall
• [asterisk-users] Where does the call go in the dialplan after a call disconnects   Arpit Mehta
• [asterisk-users] New feature: calling all bug marshals   Philip Prindeville
• [asterisk-users] 7960 Won't Register Yet Multiple Attempts?   robert.norton at sophmedia.com
• [asterisk-users] Where does the call go in the dialplan after a call disconnects   Igor A. Goncharovsky
• [asterisk-users] 7960 Won't Register Yet Multiple Attempts?   Darryl Dunkin
• [asterisk-users] Asterisk is not adding Via field   Sandeep Ammaladinna
• [asterisk-users] PHP AGI script   Nicholas Blasgen
• [asterisk-users] 7960 Won't Register Yet Multiple Attempts?   Richard Revels
• [asterisk-users] Pickup cmd   Rilawich Ango
• [asterisk-users] SIP-Realtime and sip reload   Benny Amorsen
• [asterisk-users] Any idea how making Asterisk "transparent"?   Artifex Maximus
• [asterisk-users] PHP AGI script   Philipp Kempgen
• [asterisk-users] Any idea how making Asterisk "transparent"?   Philipp Kempgen
• [asterisk-users] dtmf detection not working on sip trunks using asterisk-1.4.15   Andreas Brodmann
• [asterisk-users] Any idea how making Asterisk "transparent"?   Artifex Maximus
• [asterisk-users] PHP AGI script   Andreas Brodmann
• [asterisk-users] Any idea how making Asterisk "transparent"?   Olivier
• [asterisk-users] Any idea how making Asterisk "transparent"?   dave cantera
• [asterisk-users] Cisco power injector with GXP2000 phones   Karsten Wemheuer
• [asterisk-users] Probems receiving 200ok message   Frederico Madeira
• [asterisk-users] PHP AGI script   Tilghman Lesher
• [asterisk-users] Perspective on Asterisk   Steve Org
• [asterisk-users] Any idea how making Asterisk "transparent"?   Tilghman Lesher
• [asterisk-users] "Happy Birthday Asterisk"   Jason Parker
• [asterisk-users] SIP-Realtime and sip reload   Henrik Buchholz
• [asterisk-users] PHP AGI script   Steve Edwards
• [asterisk-users] asterisk performance   C F
• [asterisk-users] "Happy Birthday Asterisk"   Bill Andersen
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR   Josué Conti
• [asterisk-users] Sidetone with Snom 370   Zaheer K. Master
• [asterisk-users] Open Asterisk Exchange Project   Michael Munger
• [asterisk-users] AMQP Support for Asterisk?   Henry J. Cobb
• [asterisk-users] Polycom 601 stops ringing   Joe Acquisto
• [asterisk-users] asterisk performance   Giovanni Miano
• [asterisk-users] [Asterisk-users] Show calls in progress   Steve Johnson
• [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR   Moises Silva
• [asterisk-users] [Asterisk-users] Show calls in progress   Tony Mountifield
• [asterisk-users] Limit participants in Meetme...   Carlos Chavez
• [asterisk-users] Limit participants in Meetme...   Tilghman Lesher
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Stefan Guenther
• [asterisk-users] "Happy Birthday Asterisk"   Tilghman Lesher
• [asterisk-users] Polycom 601 stops ringing   Doug
• [asterisk-users] Function vs. Application?   Vincent
• [asterisk-users] "Happy Birthday Asterisk"   Philip Prindeville
• [asterisk-users] Function vs. Application?   Jared Smith
• [asterisk-users] "Happy Birthday Asterisk"   Philip Prindeville
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Armin Schindler
• [asterisk-users] Problem with the ring timeout in dial command for local extensions   tloginbr-asterisk at yahoo.com.br
• [asterisk-users] Function vs. Application?   Vincent
• [asterisk-users] Function vs. Application?   Tilghman Lesher
• [asterisk-users] Polycom 601 stops ringing   Joe Acquisto
• [asterisk-users] Print CALLERID in CLI during "pri debug "   Arpit Mehta
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Philip Prindeville
• [asterisk-users] Print CALLERID in CLI during "pri debug "   joakimsen at gmail.com
• [asterisk-users] Print CALLERID in CLI during "pri debug "   Arpit Mehta
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Darryl Dunkin
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Philip Prindeville
• [asterisk-users] asterisk performance   Michael Graves
• [asterisk-users] CDR Function in Hangup Channel   Steve Murphy
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Tilghman Lesher
• [asterisk-users] Playback file and detect a key press   Bob Smither
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Philip Prindeville
• [asterisk-users] Problem with the ring timeout in dial command for local extensions   Anselm Martin Hoffmeister
• [asterisk-users] using modem with asterisk   Rizwan Hisham
• [asterisk-users] Can't listen to voicemail message   Turbo Fredriksson
• [asterisk-users] Direct Inward Dialing How To   broadband Voice
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Ryan Burke
• [asterisk-users] using modem with asterisk   Jonn R Taylor
• [asterisk-users] using modem with asterisk   Doug Lytle
• [asterisk-users] using modem with asterisk   Doug Lytle
• [asterisk-users] using modem with asterisk   Luis Carlos Martos Ratías
• [asterisk-users] using modem with asterisk   Tzafrir Cohen
• [asterisk-users] Asterisk CDR Variable   Abdul
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Dovid B
• [asterisk-users] DID provider   Dovid B
• [asterisk-users] Probems receiving 200ok message   Dovid B
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Tilghman Lesher
• [asterisk-users] Playback file and detect a key press   Moises Silva
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Michael Graves
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Per Jessen
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Tzafrir Cohen
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Philip Prindeville
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Philip Prindeville
• [asterisk-users] Probems receiving 200ok message   Frederico Madeira
• [asterisk-users] Direct Inward Dialing How To   C F
• [asterisk-users] Where does the call go in the dialplan after acall disconnects   Dovid B
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Jared Smith
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Philip Prindeville
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Tzafrir Cohen
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Philip Prindeville
• [asterisk-users] asterisk performance   jorain
• [asterisk-users] Open Asterisk Exchange Project   Olivier
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Olivier
• [asterisk-users] Pickup cmd   F6HQZ
• [asterisk-users] Installing/configuring TE120P debian way   Andres Jimenez
• [asterisk-users] Installing/configuring TE120P debian way   Tzafrir Cohen
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Martin Smith
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Tzafrir Cohen
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   joakimsen at gmail.com
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Martin Smith
• [asterisk-users] Open Asterisk Exchange Project   joakimsen at gmail.com
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Kristian Kielhofner
• [asterisk-users] Open Asterisk Exchange Project   Michelle Dupuis
• [asterisk-users] Don't enter a queue if no one is logged in   Peter Pauly
• [asterisk-users] Open Asterisk Exchange Project   Tzafrir Cohen
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Richard Revels
• [asterisk-users] [DB] Using SQLite instead of AST?   Vincent
• [asterisk-users] Function vs. Application?   Vincent
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   joakimsen at gmail.com
• [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.   Philip Prindeville
• [asterisk-users] Dual-home Wifi/GSM phones for North America   Philip Prindeville
• [asterisk-users] Open Asterisk Exchange Project   Dan Austin
• [asterisk-users] [DB] Using SQLite instead of AST?   Tilghman Lesher
• [asterisk-users] One server, multiple companies   Eric C.
• [asterisk-users] One server, multiple companies   Philip Prindeville
• [asterisk-users] T.38 fax solution, opinions?   arkda
• [asterisk-users] Open Asterisk Exchange Project   Olivier
• [asterisk-users] Sidetone with Snom 370   Steve Davies
• [asterisk-users] Don't enter a queue if no one is logged in   Atis Lezdins
• [asterisk-users] Asterisk + Cisco Call Manager Express   Антон Шуленин
• [asterisk-users] Catalyst 2950 series with Asterisk + FAX   Josué Conti
• [asterisk-users] Sidetone with Snom 370   Mark J Elkins
• [asterisk-users] Play Beep instead of MOH   Atis Lezdins
• [asterisk-users] CAPI didn't get a frame | avoiding initial deadlock | multiple instances of Asterisk   Marco Mouta
• [asterisk-users] Multiple contacts.   Atis Lezdins
• [asterisk-users] Using Asterisk to connect 2 locations with legacy PBX   Kovář Jan
• [asterisk-users] Graceful Asterisk Shutdown   Jeng Yu
• [asterisk-users] Graceful Asterisk Shutdown   Alex Robar
• [asterisk-users] Graceful Asterisk Shutdown   Atis Lezdins
• [asterisk-users] Gateway doesn't ring   Hans-Peter Straub
• [asterisk-users] Graceful Asterisk Shutdown   J. Oquendo
• [asterisk-users] Graceful Asterisk Shutdown   J. Oquendo
• [asterisk-users] asterisk linkedin group   BerkHolz, Steven
• [asterisk-users] Sidetone with Snom 370   Michael Legart
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Michael Melia Jr.
• [asterisk-users] Using Asterisk to connect 2 locations with legacy PBX   Marco Mouta
• [asterisk-users] Multiple contacts.   Philipp Kempgen
• [asterisk-users] asterisk linkedin group   Dean Collins
• [asterisk-users] Using Asterisk to connect 2 locations with legacy PBX   Philipp Kempgen
• [asterisk-users] SIP 7960 soft key customization?   Peter Pauly
• [asterisk-users] Graceful Asterisk Shutdown   Jeng Yu
• [asterisk-users] Suppressing certain queue announcement voice prompts   Peter Pauly
• [asterisk-users] Play Beep instead of MOH   John Novack
• [asterisk-users] diferents events between ast1.2 & ast1.4 ??   Josep Bort Sancho
• [asterisk-users] text to speech   Dean Collins
• [asterisk-users] Dynamically change sip.conf properties.   Alex Balashov
• [asterisk-users] foneBRIDGE2 vs. foneBRIDGE2-EC   Kevin DeGraaf
• [asterisk-users] asterisk 1.4 with around 230 SIP connections   Jerry Geis
• [asterisk-users] Don't enter a queue if no one is logged in   Darryl Dunkin
• [asterisk-users] SIP 7960 soft key customization?   Darryl Dunkin
• [asterisk-users] Cisco 7911g Poor Audio Quality w/ Asterisk Voicemail and MOH   Anciso, Roy
• [asterisk-users] asterisk 1.4 with around 230 SIP connections   BJ Weschke
• [asterisk-users] Checking Dial Status   Joe Morsbach
• [asterisk-users] asterisk 1.4 with around 230 SIP connections   Kristian Kielhofner
• [asterisk-users] Graceful Asterisk Shutdown   joakimsen at gmail.com
• [asterisk-users] One server, multiple companies   Eric C.
• [asterisk-users] T.38 fax solution, opinions?   Matt Watson
• [asterisk-users] Sidetone with Snom 370   Zaheer K. Master
• [asterisk-users] One server, multiple companies   Matthew Mackes
• [asterisk-users] T.38 fax solution, opinions?   Robert Augustyn
• [asterisk-users] One server, multiple companies   George Pajari
• [asterisk-users] T.38 fax solution, opinions?   Adam Moffett
• [asterisk-users] Pickup re-invite   Tim St. Pierre
• [asterisk-users] T.38 fax solution, opinions?   Jonn R Taylor
• [asterisk-users] Didnt get a frame from Channel and call gets disconnected   Jai Rangi
• [asterisk-users] foneBRIDGE2 vs. foneBRIDGE2-EC   Patrick
• [asterisk-users] T.38 fax solution, opinions?   Robert Augustyn
• [asterisk-users] foneBRIDGE2 vs. foneBRIDGE2-EC   Tzafrir Cohen
• [asterisk-users] Open Asterisk Exchange Project   CunningPike
• [asterisk-users] Didnt get a frame from Channel and call gets disconnected   Jared Smith
• [asterisk-users] Didnt get a frame from Channel and call gets disconnected   Jai Rangi
• [asterisk-users] SMS gateway recommendation   Robert McNaught
• [asterisk-users] T.38 fax solution, opinions?   arkda
• [asterisk-users] line cut   Rilawich Ango
• [asterisk-users] Fw: asterisk performance   jorain
• [asterisk-users] Appending two voice files   Bart Fisher
• [asterisk-users] Pickup cmd   Rilawich Ango
• [asterisk-users] using modem with asterisk   Rizwan Hisham
• [asterisk-users] Pickup cmd   dave cantera
• [asterisk-users] Didnt get a frame from Channel and call gets disconnected   Jai Rangi
• [asterisk-users] Appending two voice files   dave cantera
• [asterisk-users] asterisk 1.4 with around 230 SIP connections   dave cantera
• [asterisk-users] text management   dave cantera
• [asterisk-users] Pickup cmd   Rilawich Ango
• [asterisk-users] text management   Dave Cotton
• [asterisk-users] Pickup re-invite   dave cantera
• [asterisk-users] Pickup over IAX   dave cantera
• [asterisk-users] asterisk 1.4 with around 230 SIP connections   Kristian Kielhofner
• [asterisk-users] [DB] Using SQLite instead of AST?   Vincent
• [asterisk-users] Monitoring Asterisk 1.4.15   Pepo
• [asterisk-users] Video Conference Or Server   bilal ghayyad
• [asterisk-users] Asterisk and NAT   bilal ghayyad
• [asterisk-users] VPN Client with the IP Phone, and what its VPN Server   bilal ghayyad
• [asterisk-users] X100P Fxo card headaches   Chris Boczko
• [asterisk-users] hi   sandeep.s
• [asterisk-users] SIP 7960 soft key customization?   Robert Lister
• [asterisk-users] rollback procedure requirements before asterisk upgrade   Marco Mouta
• [asterisk-users] VPN Client with the IP Phone, and what its VPN Server   Olivier
• [asterisk-users] Iax and ZAP   Joe Acquisto
• [asterisk-users] [Fwd: load test zap channels (in and out)]   Atis Lezdins
• [asterisk-users] hi   Philipp Kempgen
• [asterisk-users] Unicall protocol error. Cause 32776   Roger C. Beraldi Martins
• [asterisk-users] Pickup re-invite   Tim St. Pierre
• [asterisk-users] Recorded calls skipping   Jay Moore
• [asterisk-users] SIP 7960 soft key customization?   Michael Graves
• [asterisk-users] foneBRIDGE2 vs. foneBRIDGE2-EC   Noah Miller
• [asterisk-users] Dynamically change sip.conf properties.   Noah Miller
• [asterisk-users] Dynamically change sip.conf properties.   asterisk
• [asterisk-users] T.38 fax solution, opinions?   Noah Miller
• [asterisk-users] RFC3389 message   Tony Mountifield
• [asterisk-users] merge gsm files   Rilawich Ango
• [asterisk-users] Unicall protocol error. Cause 32776   Moises Silva
• [asterisk-users] Dynamically change sip.conf properties.   Simon Elliston Ball
• [asterisk-users] merge gsm files   Tzafrir Cohen
• [asterisk-users] hi   Alex Balashov
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   Jerry Jones
• [asterisk-users] hi   CunningPike
• [asterisk-users] new Asterisk installation with openvox 1.2 or 1.4?   nik600
• [asterisk-users] Unicall protocol error. Cause 32776   Roger C. Beraldi Martins
• [asterisk-users] Asterisk and NAT   Carlos Chavez
• [asterisk-users] asterisk-users Digest, Vol 41, Issue 35   Daniel M. Baker
• [asterisk-users] T.38 fax solution, opinions?   arkda
• [asterisk-users] OT - Fax and anti-spam   Olivier
• [asterisk-users] OT - Fax and anti-spam   John Beaman
• [asterisk-users] OT - Fax and anti-spam   Tzafrir Cohen
• [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?   Vincent
• [asterisk-users] RFC3389 message   Kevin P. Fleming
• [asterisk-users] X100P Fxo card headaches   Vincent
• [asterisk-users] Asterisk SIP Microsoft Outlook Integration   mail-lists
• [asterisk-users] VPN Client with the IP Phone and what its VPN Server   bilal ghayyad
• [asterisk-users] X100P Fxo card headaches   Drew Gibson
• [asterisk-users] Didnt get a frame from Channel and call gets disconnected   Jai Rangi
• [asterisk-users] Asterisk not sending 200 OK   Rob Schall
• [asterisk-users] VPN Client with the IP Phone and what its VPNServer   Christian Stredicke
• [asterisk-users] X100P Fxo card headaches   Chris Boczko
• [asterisk-users] Any phone capable of displaying real time queue statistics?   Peter Pauly
• [asterisk-users] RFC3389 message   Tony Mountifield
• [asterisk-users] X100P Fxo card headaches   Jonn R Taylor
• [asterisk-users] Check if SIP user is available or not ?   Arpit Mehta
• [asterisk-users] Asterisk and NAT   Vincent
• [asterisk-users] X100P Fxo card headaches   Tzafrir Cohen
• [asterisk-users] Check if SIP user is available or not ?   Henk Dick
• [asterisk-users] X100P Fxo card headaches   John Millican
• [asterisk-users] Check if SIP user is available or not ?   Alex Balashov
• [asterisk-users] Unicall protocol error. Cause 32776   Moises Silva
• [asterisk-users] hi   Luis Carlos Martos Ratías
• [asterisk-users] Aastra 480i CT   Jeremy Mann
• [asterisk-users] Aastra 480i CT   Cory Andrews
• [asterisk-users] Aastra 480i CT   Mike Clark
• [asterisk-users] X100P Fxo card headaches   Tzafrir Cohen
• [asterisk-users] foneBRIDGE2 vs. foneBRIDGE2-EC   George Pajari
• [asterisk-users] Bribane bases contractor....   Paul Hales
• [asterisk-users] Most Stable version of Asterisk   Jai Rangi
• [asterisk-users] Bribane bases contractor....   Steve Edwards
• [asterisk-users] SMS gateway recommendation   Greg Oliver
• [asterisk-users] SMS gateway recommendation   Robert McNaught
• [asterisk-users] Most Stable version of Asterisk   C F
• [asterisk-users] Asterisk not sending 200 OK   C F
• [asterisk-users] Most Stable version of Asterisk   Jai Rangi
• [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?   Michael Graves
• [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?   Vincent
• [asterisk-users] Most Stable version of Asterisk   C F
• [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?   Daniel Cole
• [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?   Michael Graves
• [asterisk-users] Call Quality Issues With 2 Trixbox's - RouterIssue?   Alexander Lopez
• [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?   Andres
• [asterisk-users] Call Quality Issues With 2 Trixbox's - RouterIssue?   Daniel Cole
• [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?   Paul Hales
• [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?   Daniel Cole
• [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?   Daniel Cole
• [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?   Paul Hales
• [asterisk-users] Load Balancing over 2 E1 Lines   Eric Delaporte
• [asterisk-users] asterisk-users Digest, Vol 41, Issue 38   Daniel M. Baker
• [asterisk-users] Aastra 480i CT   Ira
• [asterisk-users] asterisk performance   Gordon Henderson
• [asterisk-users] Load Balancing over 2 E1 Lines   Andres Jimenez
• [asterisk-users] OT - Fax and anti-spam   lotusscript
• [asterisk-users] OT - Fax and anti-spam   Tzafrir Cohen
• [asterisk-users] Call Center Setup on asterisk   satish patel
• [asterisk-users] OT - Fax and anti-spam   dave cantera
• [asterisk-users] OT - Callto:// tags options   dave cantera
• [asterisk-users] Load Balancing over 2 E1 Lines   Marco Mouta
• [asterisk-users] Asterisk B2BUA and Site to Site transfers   Chris Bennett
• [asterisk-users] X100P Fxo card headaches   Alan Lord
• [asterisk-users] rollback procedure requirements before asterisk upgrade   dave cantera
• [asterisk-users] Enable/Disable Sip without registration   equis software
• [asterisk-users] Video Conference Or Server   dave cantera
• [asterisk-users] Call Center Setup on asterisk   Godson Gera
• [asterisk-users] asterisk linkedin group   dave cantera
• [asterisk-users] Call Center Setup on asterisk   Senad Jordanovic
• [asterisk-users] Video Conference Or Server   Dean Collins
• [asterisk-users] VPN Client with the IP Phone, and what its VPN Server   bilal ghayyad
• [asterisk-users] Iax and ZAP   Joe Acquisto
• [asterisk-users] Enable/Disable Sip without registration   ram
• [asterisk-users] Didnt get a frame from Channel and call gets disconnected   ram
• [asterisk-users] Fw: asterisk performance   Benny Amorsen
• [asterisk-users] Playback file and detect a key press   Bob Smither
• [asterisk-users] Asterisk not sending 200 OK   Rob Schall
• [asterisk-users] Enable/Disable Sip without registration   equis software
• [asterisk-users] Recorded calls skipping   Jay Moore
• [asterisk-users] Caller ID Issue   Lutgring, Sam
• [asterisk-users] Video Conference Or Server   Patrick
• [asterisk-users] asterisk 1.4 with around 230 SIP connections   Russell Bryant
• [asterisk-users] Video Conference Or Server   Dean Collins
• [asterisk-users] [Fwd: Request for testing SIP TCP/TLS]   Russell Bryant
• [asterisk-users] Dynamically change sip.conf properties.   asterisk
• [asterisk-users] Polycom Paging   Michael Munger
• [asterisk-users] Most Stable version of Asterisk   shadowym
• [asterisk-users] Most Stable version of Asterisk   Jai Rangi
• [asterisk-users] Polycom Paging   Alex Balashov
• [asterisk-users] Account codes in CDR   Glenn Cobb
• [asterisk-users] asterisk 1.4 with around 230 SIP connections   Glenn Cobb
• [asterisk-users] Polycom Paging   Michael Munger
• [asterisk-users] Disable VAD on Polycom 330 or 301   Ed Nuñez
• [asterisk-users] Polycom Paging   Alex Balashov
• [asterisk-users] Polycom Paging   arkda
• [asterisk-users] Linksys SPA962 with SPA932 unexpected reboots   Marty Mastera
• [asterisk-users] Call Center Setup on asterisk   broadband Voice
• [asterisk-users] Can Local channels inhibit an Answer() until it is satisfied with the endpoint?   Shane Spencer
• [asterisk-users] TDM400 hangup issue in China   Steven O'Reilly
• [asterisk-users] asterisk 1.4 with around 230 SIP connections   Kristian Kielhofner
• [asterisk-users] Fwd: re: Iax and ZAP   Joe Acquisto
• [asterisk-users] Caller ID Issue   Anselm Martin Hoffmeister
• [asterisk-users] Sip Version   sanjay.rajdev at featherstoneinformatics.com
• [asterisk-users] Linksys SPA962 with SPA932 unexpected reboots   Paul Hales
• [asterisk-users] OT - Callto:// tags options   Mojo with Horan & Company, LLC
• [asterisk-users] Linksys SPA962 with SPA932 unexpected reboots   Jared Smith
• [asterisk-users] Farward calls between 2 sip servers   Juklingos
• [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?   Vincent
• [asterisk-users] Sipura provisioning   Philip Prindeville
• [asterisk-users] Didnt get a frame from Channel and call gets   Vidura Senadeera
• [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?   Michael Graves
• [asterisk-users] Linksys SPA962 with SPA932 unexpected reboots   Marty Mastera
• [asterisk-users] Linksys SPA962 with SPA932 unexpected reboots   Marty Mastera
• [asterisk-users] calls are getting disconnected automatically   sandeep.s
• [asterisk-users] calls are getting disconnected automatically   Doug
• [asterisk-users] Any phone capable of displaying real time queuestatistics?   Dovid B
• [asterisk-users] Fwd: Re: Re: calls are getting dis   sandeep.s
• [asterisk-users] Account codes in CDR   Paul Hales
• [asterisk-users] chan_mobile problems   Rob
• [asterisk-users] Linksys SPA962 with SPA932 unexpected reboots   Fons van der Beek
• [asterisk-users] Farward calls between 2 sip servers   Philipp Kempgen
• [asterisk-users] Linksys SPA962 with SPA932 unexpected reboots   Fons van der Beek
• [asterisk-users] Most Stable version of Asterisk   satish patel
• [asterisk-users] Sipura provisioning   Igor A. Goncharovsky
• [asterisk-users] TDM400 hangup issue in China   gincantalupo
• [asterisk-users] Unicall protocol error. Cause 32776   Roger C. Beraldi Martins
• [asterisk-users] TDM400 hangup issue in China   Tzafrir Cohen
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Stefan Guenther
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Armin Schindler
• [asterisk-users] Any phone capable of displaying real time queuestatistics?   Peter Pauly
• [asterisk-users] Any phone capable of displaying real timequeuestatistics?   Dean Collins
• [asterisk-users] Unicall protocol error. Cause 32776   Steve Underwood
• [asterisk-users] How do I do this?   Ade Vickers
• [asterisk-users] How do I do this?   Steve Totaro
• [asterisk-users] Unicall protocol error. Cause 32776   Moises Silva
• [asterisk-users] How do I do this?   Ade Vickers
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Stefan Guenther
• [asterisk-users] How do I do this?   Steve Totaro
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Steve Totaro
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Tzafrir Cohen
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Steve Totaro
• [asterisk-users] DID in Cape Town South Africa   Eric Smith
• [asterisk-users] Unicall protocol error. Cause 32776   Luis Antonio Prata Barbosa
• [asterisk-users] How do I do this?   Ade Vickers
• [asterisk-users] Unicall protocol error. Cause 32776   Luis Antonio Prata Barbosa
• [asterisk-users] CallManager sip trunk - callerid name?   Peter Pauly
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Armin Schindler
• [asterisk-users] Asterisk B2BUA and Site to Site transfers   mail-lists
• [asterisk-users] Cell Phone SMS   Matt
• [asterisk-users] T.38 fax solution, opinions?   Matt Watson
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Stefan Guenther
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Armin Schindler
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Armin Schindler
• [asterisk-users] Cell Phone SMS   Dean Collins
• [asterisk-users] Sip Version   Keshav K.
• [asterisk-users] Cell Phone SMS   Matt
• [asterisk-users] TDM400 hangup issue in China   Steven O'Reilly
• [asterisk-users] How do I do this?   Mojo with Horan & Company, LLC
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Stefan Guenther
• [asterisk-users] Cell Phone SMS   Steve Totaro
• [asterisk-users] Strange ISDN-problem with incoming calls out of the same city   Armin Schindler
• [asterisk-users] Cell Phone SMS   Dean Collins
• [asterisk-users] Asterisk 1.2.18 and Polycom phones not forwarding anymore   Mike
• [asterisk-users] chan_mobile problems   Rob
• [asterisk-users] asterisk-users Digest, Vol 41, Issue 44   Daniel M. Baker
• [asterisk-users] asterisk-users Digest, Vol 41, Issue 44   Steve Edwards
• [asterisk-users] Libpri 1.2.7 and 1.4.3 released   The Asterisk Development Team
• [asterisk-users] Zaptel 1.2.22.1 and 1.4.7.1 released   The Asterisk Development Team
• [asterisk-users] One server, multiple companies   Diego Andrés Asenjo González
• [asterisk-users] asterisk-users Digest, Vol 41, Issue 44   Doug Lytle
• [asterisk-users] Problem with TE205P with TeleWest in the UK   Matt King
• [asterisk-users] [Zaptel] Why no port to Windos?   Vincent
• [asterisk-users] TDM400 hangup issue in China   Rilawich Ango
• [asterisk-users] Asterisk 1.2.18 and Polycom phones not forwarding anymore   Noah Miller
• [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?   Vincent
• [asterisk-users] Asterisk 1.2.18 and Polycom phones notforwarding anymore   Mike
• [asterisk-users] Asterisk on IBM Netvista 2800 8364-EXX?   Michael Graves
• [asterisk-users] TDM400 hangup issue in China   Rupert Utteridge - Digital Techniques (Asia) Limited
• [asterisk-users] Asterisk 1.2.18 and Polycom phones notforwarding anymore - found the problem   Mike
• [asterisk-users] Asterisk 1.2.18 and Polycom phones notforwarding anymore   Noah Miller
• [asterisk-users] [Zaptel] Why no port to Windos?   Doug
• [asterisk-users] [Zaptel] Why no port to Windos?   thg at interoil.com
• [asterisk-users] [Zaptel] Why no port to Windos?   Philip Prindeville
• [asterisk-users] [Zaptel] Why no port to Windos?   Tilghman Lesher
• [asterisk-users] [Zaptel] Why no port to Windos?   Vincent
• [asterisk-users] [Zaptel] Why no port to Windos?   Philip Prindeville
• [asterisk-users] [Zaptel] Why no port to Windos?   Paul Hales
• [asterisk-users] [Zaptel] Why no port to Windos?   thg at interoil.com
• [asterisk-users] [Zaptel] Why no port to Windos?   Vincent
• [asterisk-users] chan_mobile problems   Rob
• [asterisk-users] Poor gsm playback   John Fawcett
• [asterisk-users] Account codes in CDR   Dumpolid Exeplish
• [asterisk-users] G729 on PS3 Cell   Jon Schøpzinsky
• [asterisk-users] ZRTP + asterisk and Best Security Practice   Andres Gomez
• [asterisk-users] [Zaptel] Why no port to Windos?   Tzafrir Cohen
• [asterisk-users] [Zaptel] Why no port to Windos?   Gergo Csibra
• [asterisk-users] asterisk-users Digest, Vol 41, Issue 46   Daniel M. Baker
• [asterisk-users] Unicall protocol error. Cause 32776   Roger C. Beraldi Martins
• [asterisk-users] Stange pause between extensions commands.   Catalin S.
• [asterisk-users] Stange pause between extensions commands.   Atis Lezdins
• [asterisk-users] Asterisk Qeueu with static agent   satish patel
• [asterisk-users] Poor gsm playback   Doug Lytle
• [asterisk-users] Poor gsm playback   Tilghman Lesher
• [asterisk-users] Stange pause between extensions commands.   Catalin S.
• [asterisk-users] [Zaptel] Why no port to Windos?   Michael Graves
• [asterisk-users] Gtalk callerID   Philippe Sultan
• [asterisk-users] TDM400 hangup issue in China   bilal ghayyad
• [asterisk-users] One server, multiple companies   Jerry Jones
• [asterisk-users] Poor gsm playback   John Fawcett
• [asterisk-users] Monitor a queue   dadsadsadf dsadasdsa
• [asterisk-users] [Zaptel] Why no port to Windos?   Lee Jenkins
• [asterisk-users] Stange pause between extensions commands.   Tony Plack
• [asterisk-users] asterisk-users Digest, Vol 41, Issue 44   Tony Plack
• [asterisk-users] chan_h323 compilation   bilal ghayyad
• [asterisk-users] Cell Phone SMS   Steve Totaro
• [asterisk-users] [Zaptel] Why no port to Windos?   mgraves at mstvp.com
• [asterisk-users] Stange pause between extensions commands.   Atis Lezdins
• [asterisk-users] GUI for Asterisk: Call Flow   bilal ghayyad
• [asterisk-users] chan_h323 compilation   Kevin P. Fleming
• [asterisk-users] Unicall protocol error. Cause 32776   Roger C. Beraldi Martins
• [asterisk-users] Unicall protocol error. Cause 32776   Moises Silva
• [asterisk-users] [Zaptel] Why no port to Windos?   Vincent
• [asterisk-users] [Zaptel] Why no port to Windos?   Vincent
• [asterisk-users] [Zaptel] Why no port to Windos?   Vincent
• [asterisk-users] SIP fails to register   Joe Acquisto
• [asterisk-users] [Zaptel] Why no port to Windos?   Philipp Kempgen
• [asterisk-users] [Zaptel] Why no port to Windos?   Ron Joffe
• [asterisk-users] Asterisk to make multiple extensions simultaneous calls on a single telephone line   Vincent Li
• [asterisk-users] Asterisk to make multiple extensions simultaneous calls on a single telephone line   robert boardman
• [asterisk-users] Delay in processing dial string   Ryan M. Colbert
• [asterisk-users] SIP fails to register   Zaheer K. Master
• [asterisk-users] SIP fails to register   Joe Acquisto
• [asterisk-users] Delay in processing dial string   Andres
• [asterisk-users] [Zaptel] Why no port to Windos?   Hans Witvliet
• [asterisk-users] [Zaptel] Why no port to Windos?   Hans Witvliet
• [asterisk-users] [Zaptel] Why no port to Windos?   Michael Graves
• [asterisk-users] [Zaptel] Why no port to Windos?   Tzafrir Cohen
• [asterisk-users] TDM400 hangup issue in China   Tzafrir Cohen
• [asterisk-users] Unicall protocol error. Cause 32776   Moises Silva
• [asterisk-users] TDM400 hangup issue in China   Rilawich Ango
• [asterisk-users] GUI for Asterisk: Call Flow   Bhrugu Mehta
• [asterisk-users] GUI for Asterisk: Call Flow   dave cantera
• [asterisk-users] [Zaptel] Why no port to Windos?   Vincent
• [asterisk-users] [Zaptel] Why no port to Windos?   randulo
• [asterisk-users] ZRTP + asterisk and Best Security Practice   Olle E Johansson
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Johansson Olle E
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Alessio Focardi
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Mindaugas Kezys
• [asterisk-users] OpenVox B800P and asterisk 1.4/ mISDN-1_1_7   nik600
• [asterisk-users] Open ITU G.107 Implementation to measure voice quality   Andre Gustavo Lomonaco
• [asterisk-users] ZRTP + asterisk and Best Security Practice   Olivier
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Steve Totaro
• [asterisk-users] ZRTP + asterisk and Best Security Practice   Olle E Johansson
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Olle E Johansson
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Rob Hillis
• [asterisk-users] [Zaptel] Why no port to Windos?   Dovid B
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Tilghman Lesher
• [asterisk-users] [Zaptel] Why no port to Windos?   Philipp Kempgen
• [asterisk-users] [Zaptel] Why no port to Windos?   Michael Graves
• [asterisk-users] Call Center Setup on asterisk   Dovid B
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Philipp Kempgen
• [asterisk-users] DNS broken for www.voip-info.org ??   Steve Johnson
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Olle E Johansson
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Chris Bagnall
• [asterisk-users] DNS broken for www.voip-info.org ??   Philipp Kempgen
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   David Boyd
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Tzafrir Cohen
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Ira
• [asterisk-users] DNS broken for www.voip-info.org ??   James H Thompson
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Mr. James W. Laferriere
• [asterisk-users] chan_h323 compilation   bilal ghayyad
• [asterisk-users] Upgrade to Asterisk 1.4 - from a 1.0 style configuration   Olle E Johansson
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Dovid B
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Dovid B
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Philipp Kempgen
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Ron Arts
• [asterisk-users] [Zaptel] Why no port to Windos?   Steve Thomas
• [asterisk-users] [Zaptel] Why no port to Windos?   Time Bandit
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Steve Totaro
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Tilghman Lesher
• [asterisk-users] One touch with Polycom Phones.   Matthew Mackes
• [asterisk-users] Reputable company for SIP/IAX2 trunking   Steve Finkelstein
• [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me   Philip Prindeville
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Lyle Giese
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Rob Hillis
• [asterisk-users] Trixbox Arbitrary Command Execution Vulnerability   Than Taro
• [asterisk-users] Asterisk B2BUA and Site to Site transfers   Chris Bennett
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Olle E Johansson
• [asterisk-users] LDAPget question, usage   arkda
• [asterisk-users] Calling Party Category Field   Carlos Chavez
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Ira
• [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me   Anselm Martin Hoffmeister
• [asterisk-users] stanaphone issues. can someone verify my config?   Richard
• [asterisk-users] stanaphone issues. can someone verify my config?   Richard
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Duncan Turnbull
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Benny Amorsen
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Benny Amorsen
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Hans Witvliet
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Olle E Johansson
• [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me   Philipp Kempgen
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Tilghman Lesher
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Mike
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Kevin P. Fleming
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Patrick
• [asterisk-users] Reputable company for SIP/IAX2 trunking   Luki
• [asterisk-users] Reputable company for SIP/IAX2 trunking   Steve Finkelstein
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   JR Richardson
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Tilghman Lesher
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Philipp Kempgen
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   David Boyd
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Ira
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Philipp Kempgen
• [asterisk-users] [Zaptel] Why no port to Windos?   Paul Hales
• [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me   Philip Prindeville
• [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me   Philipp Kempgen
• [asterisk-users] Trixbox Phones Home   Matthew Rubenstein
• [asterisk-users] dial, answered and then hangup   Rilawich Ango
• [asterisk-users] dial, answered and then hangup   dave cantera
• [asterisk-users] dial, answered and then hangup   Rilawich Ango
• [asterisk-users] Trixbox Phones Home   Than Taro
• [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!   Tilghman Lesher
• [asterisk-users] VoIP service providers/PSTN termination points   Benjamin Jacob
• [asterisk-users] [Zap