January 2008 Archives by date

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Starting: Tue Jan 1 00:44:37 CST 2008
Ending: Thu Jan 31 23:52:09 CST 2008
Messages: 1935

• [asterisk-users] Asterisk 1.4 Fax   Rob Hillis
• [asterisk-users] Asterisk 1.4 Fax   Andrew Joakimsen
• [asterisk-users] Asterisk 1.4 Fax   Steve Underwood
• [asterisk-users] One Way Delay in Audio Over Analog   MatsK
• [asterisk-users] Asterisk 1.4 Fax   Rob Hillis
• [asterisk-users] Asterisk 1.4 Fax   Steve Underwood
• [asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone   Glenn Gillen
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Vincent
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Godson Gera
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Vincent
• [asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone   Darrick Hartman (lists)
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   MatsK
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Godson Gera
• [asterisk-users] Asterisk + SIP + cisco phone confrance problem   satish patel
• [asterisk-users] Asterisk 1.4 Fax   Al lists
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   dave cantera
• [asterisk-users] (no subject)   lists65
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Adam Moffett
• [asterisk-users] With rtcachefriends=yes, when do realtime changes take effect?   Adam Moffett
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Tzafrir Cohen
• [asterisk-users] With rtcachefriends=yes, when do realtime changes take effect?   Anthony Francis
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Vieri
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Tzafrir Cohen
• [asterisk-users] (no subject)   Andrew Joakimsen
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Vieri
• [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan   Andrew Joakimsen
• [asterisk-users] Asterisk 1.4 Fax   Rob Hillis
• [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan   John Novack
• [asterisk-users] One Way Delay in Audio Over Analog   shadowym
• [asterisk-users] (no subject)   Doug Lytle
• [asterisk-users] Asterisk 1.4 Fax   Jonn R Taylor
• [asterisk-users] Asterisk access Postgres for Realtime Configuration   Mehdi chouikh
• [asterisk-users] Asterisk 1.4 Fax   Doug Lytle
• [asterisk-users] Asterisk 1.4 Fax   Jonn R Taylor
• [asterisk-users] Asterisk 1.4 Fax   Doug Lytle
• [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan   Tilghman Lesher
• [asterisk-users] Asterisk 1.4 Fax   Guillermo Salas M.
• [asterisk-users] Asterisk 1.4 Fax   Jonn R Taylor
• [asterisk-users] Asterisk 1.4 Fax   Doug Lytle
• [asterisk-users] Asterisk 1.4 Fax   Lee Howard
• [asterisk-users] Asterisk 1.4 Fax   Jonn R Taylor
• [asterisk-users] Asterisk 1.4 Fax   Jonn R Taylor
• [asterisk-users] Asterisk 1.4 Fax   Rob Hillis
• [asterisk-users] Asterisk 1.4 Fax   Jonn R Taylor
• [asterisk-users] Asterisk 1.4 Fax   Doug Lytle
• [asterisk-users] Asterisk 1.4 Fax   Steve Underwood
• [asterisk-users] Asterisk 1.4 Fax   Al lists
• [asterisk-users] Asterisk 1.4 Fax   Jonn R Taylor
• [asterisk-users] How does Asterisk scale to 500-1000 phones?   Bryan M. Johns
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   dave cantera
• [asterisk-users] Password protect a queue from callers?   Caza Henha
• [asterisk-users] Fwd: Gotoiftime help   troxlinux
• [asterisk-users] Password protect a queue from callers?   Paul Hales
• [asterisk-users] Fwd: Gotoiftime help   Tilghman Lesher
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Vincent
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Vincent
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Vincent
• [asterisk-users] Asterisk 1.4 Fax   Rob Hillis
• [asterisk-users] OT: Is Cisco 7960 SIP firmware same as 7940 SIP firmware?   Sean Dennis
• [asterisk-users] Trixbox and mail2fax   Daniel
• [asterisk-users] Trixbox and mail2fax   Bill Hackensack
• [asterisk-users] Trixbox and mail2fax   Giedrius Augys
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   bilal ghayyad
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Vieri
• [asterisk-users] Trixbox and mail2fax   Rob Hillis
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   MatsK
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Stefan Guenther
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Tzafrir Cohen
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   Rob Hillis
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Vieri
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Vieri
• [asterisk-users] auto dial and IVR   Rilawich Ango
• [asterisk-users] Asterisk dialplan date and time operations   Erik Wartusch
• [asterisk-users] Invalid extensions   Gilberto Nunes Ferreira
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Tzafrir Cohen
• [asterisk-users] Asterisk dialplan date and time operations   Doug Lytle
• [asterisk-users] auto dial and IVR   Doug Lytle
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Vieri
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Tzafrir Cohen
• [asterisk-users] How to setup redundant SIP peers   Dovid B
• [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan   Dovid B
• [asterisk-users] Looking for PSTN provider with unlimitedinbound/outbound plan   Dovid B
• [asterisk-users] Asterisk dialplan date and time operations   Erik Wartusch
• [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan   Senad Jordanovic
• [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init   Vieri
• [asterisk-users] Lamps on Snom phones   Phil Knighton
• [asterisk-users] Lamps on Snom phones   Stefan Guenther
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Tim Panton
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   Tim Panton
• [asterisk-users] With rtcachefriends=yes, when do realtime changes take effect?   Peder at NetworkOblivion
• [asterisk-users] One Way Delay in Audio Over Analog   Brian Alexander
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Vincent
• [asterisk-users] Lamps on Snom phones   Dr. Michael J. Chudobiak
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Tzafrir Cohen
• [asterisk-users] Invalid extensions   dave cantera
• [asterisk-users] With rtcachefriends=yes, when do realtime changes take effect?   Adam Moffett
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   dave cantera
• [asterisk-users] Asterisk dialplan date and time operations   dave cantera
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   Rob Hillis
• [asterisk-users] Invalid extensions   Jared Smith
• [asterisk-users] Asterisk E1/T1 Card configuration   Jeng Yu
• [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan   broadband Voice
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   bilal ghayyad
• [asterisk-users] Invalid extensions   dave cantera
• [asterisk-users] Asterisk dialplan date and time operations   Erik Wartusch
• [asterisk-users] Asterisk dialplan date and time operations   Tilghman Lesher
• [asterisk-users] Asterisk 1.4 Fax   Norman Franke
• [asterisk-users] Asterisk dialplan date and time operations   dave cantera
• [asterisk-users] Incoming Calls   Paulo Pinheiro
• [asterisk-users] Invalid extensions   Gilberto Nunes Ferreira
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   Tim Panton
• [asterisk-users] Incoming Calls   Doug Lytle
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   Tim Panton
• [asterisk-users] AGI stream file   Timothy Legge
• [asterisk-users] Incoming Calls   Jose P. Espinal
• [asterisk-users] Incoming Calls   Zaheer K. Master
• [asterisk-users] Incoming Calls   Tony Plack
• [asterisk-users] AGI stream file   dave cantera
• [asterisk-users] Incoming Calls   Paulo Pinheiro
• [asterisk-users] Incoming Calls   Jonn R Taylor
• [asterisk-users] Incoming Calls   Jonn R Taylor
• [asterisk-users] Lamps on Snom phones   Phil Knighton
• [asterisk-users] Polycom VLAN   Jeremy Mann
• [asterisk-users] Polycom VLAN   Watkins, Bradley
• [asterisk-users] Lamps on Snom phones   Michiel van Baak
• [asterisk-users] AGI stream file   Goke Aruna
• [asterisk-users] Asterisk dialplan date and time operations   Tilghman Lesher
• [asterisk-users] Polycom VLAN   Alex Balashov
• [asterisk-users] AGI stream file   Timothy Legge
• [asterisk-users] Lamps on Snom phones   Jared Smith
• [asterisk-users] Incoming Calls   Paulo Pinheiro
• [asterisk-users] Incoming Calls   Tony Plack
• [asterisk-users] Toroid: open source middleware   Brian McManus
• [asterisk-users] Lamps on Snom phones   Russell Brown
• [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken   Russell Brown
• [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken   Tzafrir Cohen
• [asterisk-users] How to stop the update of astdb?   Aryanto Rachmad
• [asterisk-users] Missing "zap" command in Asterisk 1.4.16   Raúl Gómez C.
• [asterisk-users] How to stop the update of astdb?   Jared Smith
• [asterisk-users] Missing "zap" command in Asterisk 1.4.16   Tzafrir Cohen
• [asterisk-users] Missing "zap" command in Asterisk 1.4.16   Jared Smith
• [asterisk-users] Missing "zap" command in Asterisk 1.4.16   Tomás Laureano Peralta Tormey
• [asterisk-users] Missing "zap" command in Asterisk 1.4.16   Brett Crapser
• [asterisk-users] Missing "zap" command in Asterisk 1.4.16   Raúl Gómez C.
• [asterisk-users] Missing "zap" command in Asterisk 1.4.16   Raúl Gómez C.
• [asterisk-users] How to setup redundant SIP peers   Nicholas Blasgen
• [asterisk-users] Asterisk 1.4.17 Released   The Asterisk Development Team
• [asterisk-users] AST-2008-001: Crash from transfer using BYE with Also header   Asterisk Security Team
• [asterisk-users] is Power fail transfer possible with asterisk?   John covici
• [asterisk-users] is Power fail transfer possible with asterisk?   John Novack
• [asterisk-users] is Power fail transfer possible with asterisk?   Tilghman Lesher
• [asterisk-users] is Power fail transfer possible with asterisk?   John covici
• [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken   robert boardman
• [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken   Tzafrir Cohen
• [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk   Andres Paglayan
• [asterisk-users] is Power fail transfer possible with asterisk?   MatsK
• [asterisk-users] is Power fail transfer possible with asterisk?   Douglas Garstang
• [asterisk-users] Asterisk E1/T1 Card configuration   Dovid B
• [asterisk-users] PRI Crapping Out Regularly   Dovid B
• [asterisk-users] Probems receiving 200ok message   Dovid B
• [asterisk-users] Recorded calls skipping   Dovid B
• [asterisk-users] setting up two asterisk server as ss7 back to back.   Dovid B
• [asterisk-users] sip.conf & realtime   Dovid B
• [asterisk-users] Recorded calls skipping   Steve Totaro
• [asterisk-users] How to automaticaly close calls whenAsterisk didn't receive the bye request ?   Dovid B
• [asterisk-users] Snom 370 buton Recordings   Dovid B
• [asterisk-users] Softphone to be installed on the Mobile   Dovid B
• [asterisk-users] Snom 370 buton Recordings   Christian Stredicke
• [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?   Vincent
• [asterisk-users] auto dial and IVR   Bhrugu Mehta
• [asterisk-users] How does Asterisk scale to 500-1000 phones?   Jesse Molina
• [asterisk-users] Asterisk dialplan date and time operations   Erik Wartusch
• [asterisk-users] How to automaticaly close callswhenAsterisk didn't receive the bye request ?   Steve Langstaff
• [asterisk-users] How to automaticaly close calls whenAsterisk didn't receive the bye request ?   Raj Jain
• [asterisk-users] Lamps on Snom phones   Stefan Guenther
• [asterisk-users] Asterisk 1.4 Fax   Olivier
• [asterisk-users] Polycom VLAN   Wojciech Tryc
• [asterisk-users] GSM Gateway behind SIP ATA?   Remco Barendse
• [asterisk-users] Lamps on Snom phones   Phil Knighton
• [asterisk-users] GSM Gateway behind SIP ATA?   Benchev
• [asterisk-users] Recorded calls skipping   Jay Moore
• [asterisk-users] BLF trouble   Lars Bensmann
• [asterisk-users] GSM Gateway behind SIP ATA?   Steven
• [asterisk-users] Missing "zap" command in Asterisk 1.4.16   Steven
• [asterisk-users] GSM Gateway behind SIP ATA?   Remco Barendse
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] HFC-S zap channels always busy   Tzafrir Cohen
• [asterisk-users] GSM Gateway behind SIP ATA?   Atis Lezdins
• [asterisk-users] Lamps on Snom phones   Stefan Guenther
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] Right timing for a queue call   Andrea Spadaccini
• [asterisk-users] GSM Gateway behind SIP ATA?   Michiel van Baak
• [asterisk-users] Recorded calls skipping   Steve Totaro
• [asterisk-users] Bad Link on Website...   Stuart Sheldon
• [asterisk-users] Lamps on Snom phones   Russell Brown
• [asterisk-users] Recorded calls skipping   Tzafrir Cohen
• [asterisk-users] Lamps on Snom phones   Carlos Chavez
• [asterisk-users] Bad Link on Website...   Russell Bryant
• [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken   Russell Brown
• [asterisk-users] Right timing for a queue call   Atis Lezdins
• [asterisk-users] Lamps on Snom phones   Shanon Swafford
• [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken   Jared Smith
• [asterisk-users] A thougt   Fredrik Söderlund
• [asterisk-users] Unable to retrieve my voice mail ... (password incorrect)   William Herrera
• [asterisk-users] A thougt   Dean Collins
• [asterisk-users] 1.4.17 - Breaks park announce?   Brent Torrenga
• [asterisk-users] Unable to retrieve my voice mail ... (password incorrect)   Benchev
• [asterisk-users] 1.4.17 - Breaks park announce?   Russell Bryant
• [asterisk-users] OT - GEOPRIV and location based SIP services   Olivier
• [asterisk-users] GSM Gateway behind SIP ATA?   Benchev
• [asterisk-users] BLF trouble   Dovid B
• [asterisk-users] How to automaticaly closecallswhenAsterisk didn't receive the bye request ?   Dovid B
• [asterisk-users] Unable to retrieve my voice mail ... (password incorrect)   William Herrera
• [asterisk-users] OT - GEOPRIV and location based SIP services   Dean Collins
• [asterisk-users] Unable to retrieve my voice mail ... (password incorrect)   Paul Hales
• [asterisk-users] Digium Asterisk Appliance voicemail & logs   Gregory Malsack
• [asterisk-users] Digium Asterisk Appliance voicemail & logs   Gregory Malsack
• [asterisk-users] OT - GEOPRIV and location based SIP services   MatsK
• [asterisk-users] Right timing for a queue call   Andrea Spadaccini
• [asterisk-users] Using Asterisc for Taking Calls for Radio   Shane D
• [asterisk-users] OT - GEOPRIV and location based SIP services   BJ Weschke
• [asterisk-users] automatic call marking an extension   troxlinux
• [asterisk-users] G.278 RTP conversation capture, please.   Kerry S
• [asterisk-users] Using Asterisc for Taking Calls for Radio   Tilghman Lesher
• [asterisk-users] G.278 RTP conversation capture, please.   Paul Hales
• [asterisk-users] Registration from sip failed for ACL error (permit/deny)   Doug
• [asterisk-users] Registration from sip failed for ACL error (permit/deny)   Doug
• [asterisk-users] A thougt   dave cantera
• [asterisk-users] Agents and AddQueueMember   Rajkumar S
• [asterisk-users] Agents and AddQueueMember   Rajkumar S
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Christophorus Laube
• [asterisk-users] Using Asterisc for Taking Calls for Radio   Don Fanning
• [asterisk-users] OT - GEOPRIV and location based SIP services   Olivier
• [asterisk-users] OT - GEOPRIV and location based SIP services   Olivier
• [asterisk-users] Agents and AddQueueMember   Alexandre Snarskii
• [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily   Tomasz Zieleniewski
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Patrick
• [asterisk-users] ip phone suggestion for Asia?   Ian FREISLICH
• [asterisk-users] Right timing for a queue call   Andrea Spadaccini
• [asterisk-users] Using Asterisc for Taking Calls for Radio   Shane D
• [asterisk-users] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC   randulo
• [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily   Benchev
• [asterisk-users] Agents and AddQueueMember   BJ Weschke
• [asterisk-users] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC   Shane D
• [asterisk-users] Digium Asterisk Appliance voicemail & logs   Atis Lezdins
• [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily   Johansson Olle E
• [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily   Johansson Olle E
• [asterisk-users] OT - GEOPRIV and location based SIP services   Tzafrir Cohen
• [asterisk-users] A thougt   Dean Collins
• [asterisk-users] OT - GEOPRIV and location based SIP services   dave cantera
• [asterisk-users] OT - GEOPRIV and location based SIP services   Dean Collins
• [asterisk-users] Using Asterisc for Taking Calls for Radio   dave cantera
• [asterisk-users] OT - GEOPRIV and location based SIP services   Tzafrir Cohen
• [asterisk-users] BLF trouble   Lars Bensmann
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Anciso, Roy
• [asterisk-users] GSM Gateway behind SIP ATA?   Remco Barendse
• [asterisk-users] GSM Gateway behind SIP ATA?   EdPimentl
• [asterisk-users] BLF trouble   Benny Amorsen
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Glenn Cobb
• [asterisk-users] ip phone suggestion for Asia?   d tbsky
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Christophorus Laube
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Glenn Cobb
• [asterisk-users] automatic call marking an extension   Anselm Martin Hoffmeister
• [asterisk-users] Agents and AddQueueMember   Rajkumar S
• [asterisk-users] Asterisk content @ OSCON 2008?   Martin Smith
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Anciso, Roy
• [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily   Tomasz Zieleniewski
• [asterisk-users] x100p wcfxo hangup on outgoing calss   Miguel A Felipe Rodríguez
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   John Novack
• [asterisk-users] Using Asterisc for Taking Calls for Radio   Shane D
• [asterisk-users] GSM Gateway behind SIP ATA?   Remco Barendse
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Chad Osmond
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Christophorus Laube
• [asterisk-users] 2 firewalls, different INVITES   Robert Moskowitz
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Anciso, Roy
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   John Novack
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Anciso, Roy
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Christophorus Laube
• [asterisk-users] A thougt   Tim Panton
• [asterisk-users] HFC-S zap channels always busy   Tzafrir Cohen
• [asterisk-users] x100p wcfxo hangup on outgoing calss   Tzafrir Cohen
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Glenn Cobb
• [asterisk-users] BLF trouble   Axel Thimm
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   John Novack
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Glenn Cobb
• [asterisk-users] Polycom IP4000 - Device does not match ACL   Kevin DeGraaf
• [asterisk-users] automatic call marking an extension   troxlinux
• [asterisk-users] HFC-S zap channels always busy   Ron Wellsted
• [asterisk-users] BLF trouble   Lars Bensmann
• [asterisk-users] Using Asterisc for Taking Calls for Radio   Troy Ayers
• [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan   Steve Finkelstein
• [asterisk-users] Using Asterisc for Taking Calls for Radio   Shane D
• [asterisk-users] Cisco 79xx XML services   Edwin Lam
• [asterisk-users] VOIP Provider wooes   Gregory Malsack
• [asterisk-users] A thougt   MatsK
• [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan   Senad Jordanovic
• [asterisk-users] x100p wcfxo hangup on outgoing calss   Miguel A Felipe Rodríguez
• [asterisk-users] Remote hold on PRI   Gaëtan Minet
• [asterisk-users] X100P Woes   Bob Smither
• [asterisk-users] b2bua   ameel
• [asterisk-users] Digium Asterisk Appliance voicemail & logs   Kevin P. Fleming
• [asterisk-users] GSM Gateway behind SIP ATA?   Michiel van Baak
• [asterisk-users] X100P Woes   Tzafrir Cohen
• [asterisk-users] HFC-S zap channels always busy   Michiel van Baak
• [asterisk-users] asterisk as sip server   ameel
• [asterisk-users] Conditional Dial   bilal ghayyad
• [asterisk-users] GotoIf: OR, AND   bilal ghayyad
• [asterisk-users] G723 Codec and Asterisk   bilal ghayyad
• [asterisk-users] Asterisk content @ OSCON 2008?   Brian Capouch
• [asterisk-users] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC   randulo
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] HFC-S zap channels always busy   Tzafrir Cohen
• [asterisk-users] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC   Shane D
• [asterisk-users] G.278 RTP conversation capture, please.   Kerry S
• [asterisk-users] VOIP Provider wooes   Dave Miller
• [asterisk-users] Asterisk content @ OSCON 2008?   dave cantera
• [asterisk-users] b2bua   Tilghman Lesher
• [asterisk-users] Conditional Dial   Tilghman Lesher
• [asterisk-users] GotoIf: OR, AND   Tilghman Lesher
• [asterisk-users] G723 Codec and Asterisk   Tilghman Lesher
• [asterisk-users] how to block spammer calls   ram
• [asterisk-users] how to block spammer calls   Trevor Peirce
• [asterisk-users] b2bua   Benchev
• [asterisk-users] asterisk as sip server   ram
• [asterisk-users] b2bua   Yehavi Bourvine +972-8-9489444
• [asterisk-users] b2bua   Tilghman Lesher
• [asterisk-users] how to block spammer calls   ram
• [asterisk-users] How to setup redundant SIP peers   Matt Riddell
• [asterisk-users] b2bua   Raj Jain
• [asterisk-users] ASTERISK cd-rom   Bhrugu Mehta
• [asterisk-users] ASTERISK cd-rom   obn4uticus
• [asterisk-users] G729A Install Problems   broadband Voice
• [asterisk-users] BLF trouble   Axel Thimm
• [asterisk-users] iP0020 Phone busy signal all the time.   William Herrera
• [asterisk-users] how to block spammer calls   Anselm Martin Hoffmeister
• [asterisk-users] iP0020 Phone busy signal all the time.   Anselm Martin Hoffmeister
• [asterisk-users] ASTERISK cd-rom   Tzafrir Cohen
• [asterisk-users] how to block spammer calls   Tzafrir Cohen
• [asterisk-users] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC   randulo
• [asterisk-users] how to block spammer calls   Anselm Martin Hoffmeister
• [asterisk-users] G729A Install Problems   Kevin P. Fleming
• [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken   robert boardman
• [asterisk-users] G729A Install Problems   broadband Voice
• [asterisk-users] Cisco 79xx XML services   Matthew Rubenstein
• [asterisk-users] iP0020 Phone busy signal all the time.   Administrator TOOTAI
• [asterisk-users] HFC-S zap channels always busy   Patrick
• [asterisk-users] iP0020 Phone busy signal all the time.   William Herrera
• [asterisk-users] Polycom IP phones that are brick'd   dave cantera
• [asterisk-users] iP0020 Phone busy signal all the time.   Administrator TOOTAI
• [asterisk-users] G729A Install Problems   Terry Wilson
• [asterisk-users] iP0020 Phone busy signal all the time.   Rob Hillis
• [asterisk-users] Detailed Instructions   Shane D
• [asterisk-users] Detailed Instructions   Tzafrir Cohen
• [asterisk-users] iP0020 Phone busy signal all the time.   William Herrera
• [asterisk-users] iP0020 Phone busy signal all the time.   William Herrera
• [asterisk-users] Detailed Instructions   Shane D
• [asterisk-users] Detailed Instructions   Tzafrir Cohen
• [asterisk-users] G729A Install Problems   broadband Voice
• [asterisk-users] asterisk on Hp servers   Gres +
• [asterisk-users] iP0020 Phone busy signal all the time.   Rob Hillis
• [asterisk-users] 1.4.17 - Breaks park announce?   Anthony Messina
• [asterisk-users] Detailed Instructions   Shane D
• [asterisk-users] asterisk on Hp servers   Anselm Martin Hoffmeister
• [asterisk-users] Polycom IP phones that are brick'd   Dean Collins
• [asterisk-users] 1.4.17 - Breaks park announce?   Gaëtan Minet
• [asterisk-users] ASTERISK cd-rom   Al lists
• [asterisk-users] asterisk on Hp servers   Steve Edwards
• [asterisk-users] Detailed Instructions   Steve Edwards
• [asterisk-users] Detailed Instructions   Tzafrir Cohen
• [asterisk-users] asterisk on Hp servers   Hans Witvliet
• [asterisk-users] Newbie Q: Good link to configuring NAT with Sipura ATA's & hardphones   Philip Prindeville
• [asterisk-users] asterisk on Hp servers   MatsK
• [asterisk-users] iP0020 Phone busy signal all the time.   William Herrera
• [asterisk-users] Detailed Instructions   MatsK
• [asterisk-users] asterisk on Hp servers   Tzafrir Cohen
• [asterisk-users] Zap with SIP   Faisal Ashraf
• [asterisk-users] asterisk on Hp servers   MatsK
• [asterisk-users] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC   Steve Prior
• [asterisk-users] iP0020 Phone busy signal all the time.   Timothy Legge
• [asterisk-users] iP0020 Phone busy signal all the time.   William Herrera
• [asterisk-users] iP0020 Phone busy signal all the time.   Timothy Legge
• [asterisk-users] asterisk on Hp servers   Eric S López
• [asterisk-users] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC   randulo
• [asterisk-users] Conditional Dial   Anthony Messina
• [asterisk-users] [FreeBSD 6.2] Error compiling Zaptel from Ports?   Vincent
• [asterisk-users] New site for feature wish-list: Asteriskideas.org   Johansson Olle E
• [asterisk-users] iP0020 Phone busy signal all the time.   Administrator TOOTAI
• [asterisk-users] X100P Woes   Vincent
• [asterisk-users] Cisco 79xx XML services   satish patel
• [asterisk-users] Which IP Phone is really the best?   William Herrera
• [asterisk-users] Which IP Phone is really the best?   Tim Connolly
• [asterisk-users] asterisk on Hp servers   Carlos Rojas
• [asterisk-users] Which IP Phone is really the best?   Michael Graves
• [asterisk-users] Asterisk High Availability and Clustering   JR Richardson
• [asterisk-users] Which IP Phone is really the best?   Dovid B
• [asterisk-users] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC   Steve Prior
• [asterisk-users] ip phone suggestion for Asia?   Andrew Joakimsen
• [asterisk-users] Which IP Phone is really the best?   Bill Hackensack
• [asterisk-users] RANT (was Re: Which IP Phone is really the best?)   RE Kushner List Account
• [asterisk-users] Detailed Instructions   Hans Witvliet
• [asterisk-users] Which IP Phone is really the best?   Paul Hales
• [asterisk-users] Detailed Instructions   Tzafrir Cohen
• [asterisk-users] Detailed Instructions   Philipp Kempgen
• [asterisk-users] Detailed Instructions   C F
• [asterisk-users] G.278 RTP conversation capture, please.   Paul Hales
• [asterisk-users] Which IP Phone is really the best?   Dovid B
• [asterisk-users] ip phone suggestion for Asia?   d tbsky
• [asterisk-users] Change Default Voicemail Message   Daniel Cole
• [asterisk-users] Detailed Instructions   Shane D
• [asterisk-users] RANT (was Re: Which IP Phone is really the best?)   Alejandro Kauffmann
• [asterisk-users] Change Default Voicemail Message   Paul Hales
• [asterisk-users] Which IP Phone is really the best?   David Cook
• [asterisk-users] Change Default Voicemail Message   Trevor G. Hammonds
• [asterisk-users] Which IP Phone is really the best?   William Herrera
• [asterisk-users] Which IP Phone is really the best?   Tim Connolly
• [asterisk-users] Which IP Phone is really the best?   Kev S
• [asterisk-users] Which IP Phone is really the best?   Matt Riddell
• [asterisk-users] zaptel programming   Bhrugu Mehta
• [asterisk-users] zaptel programming   Philipp Kempgen
• [asterisk-users] RANT (was Re: Which IP Phone is really the best?)   Anselm Martin Hoffmeister
• [asterisk-users] Increase Volume - SIP   marcelocbf at comcast.net
• [asterisk-users] Increase Volume - SIP   Philipp Kempgen
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Christophorus Laube
• [asterisk-users] zaptel programming   Lee Jenkins
• [asterisk-users] no outgoing calls with Digium B410P   daniele visaggio
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Christophorus Laube
• [asterisk-users] zaptel programming   Tzafrir Cohen
• [asterisk-users] Presentation Restricted h.323-SIP issue   Lucian Gheorghe
• [asterisk-users] no outgoing calls with Digium B410P   map
• [asterisk-users] Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?   Len
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] extension.conf with mysql   Gopal krishnan
• [asterisk-users] extension.conf with mysql   Doug Lytle
• [asterisk-users] HPEC   Kevin P. Fleming
• [asterisk-users] zaptel programming   Philipp Kempgen
• [asterisk-users] no outgoing calls with Digium B410P   daniele visaggio
• [asterisk-users] no outgoing calls with Digium B410P   Doug Lytle
• [asterisk-users] no outgoing calls with Digium B410P   map
• [asterisk-users] no outgoing calls with Digium B410P   Tzafrir Cohen
• [asterisk-users] How to check if a SIP phone is forwarded without ringing it ?   Olivier
• [asterisk-users] How to check if a SIP phone is forwarded without ringing it ?   Kevin P. Fleming
• [asterisk-users] no outgoing calls with Digium B410P   daniele visaggio
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] no outgoing calls with Digium B410P   John Novack
• [asterisk-users] no outgoing calls with Digium B410P   map
• [asterisk-users] Multi-SPAN (4xE1) Zap Group (Outbound)   Mike Trest - Personal
• [asterisk-users] Which IP Phone is really the best?   C F
• [asterisk-users] pickup application failed   Rilawich Ango
• [asterisk-users] no outgoing calls with Digium B410P   Tzafrir Cohen
• [asterisk-users] no outgoing calls with Digium B410P   daniele visaggio
• [asterisk-users] service provider connection problem   srinivas Antarvedi
• [asterisk-users] FWD and IPCall   Huw Richards
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   Patrick
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] Detailed Instructions   Hans Witvliet
• [asterisk-users] FWD and IPCall   Benchev
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] Media gateways and video   Olivier
• [asterisk-users] [Asterisk 1.2 + TDM FXO] Incoming call not detected   Vincent
• [asterisk-users] asterisk CLI and no such command "stop"   Vieri
• [asterisk-users] asterisk CLI and no such command "stop"   Tomás Laureano Peralta Tormey
• [asterisk-users] no outgoing calls with Digium B410P   Andres Paglayan
• [asterisk-users] Remote hold on PRI   Gaëtan Minet
• [asterisk-users] GotoIf() help   Glenn Cobb
• [asterisk-users] Remote hold on PRI   Shane Spencer
• [asterisk-users] Multi-SPAN (4xE1) Zap Group (Outbound)   Alejandro Kauffmann
• [asterisk-users] asterisk CLI and no such command "stop"   Vieri
• [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee   John Novack
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] Cisco 79xx XML services   Anciso, Roy
• [asterisk-users] FWD and IPCall   Troy Ayers
• [asterisk-users] asterisk CLI and no such command "stop"   MatsK
• [asterisk-users] FWD and IPCall   Huw Richards
• [asterisk-users] chan_mobile and W300i   Emmanuel Favre-Nicolin
• [asterisk-users] FWD and IPCall   www.IPKall.com
• [asterisk-users] asterisk-users] Increase Volume - SIP   bilal ghayyad
• [asterisk-users] Change Default Voicemail Message   Daniel Cole
• [asterisk-users] Increase Volume - SIP   bilal ghayyad
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] asterisk CLI and no such command "stop"   Vieri
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] Background Noise Elimination   Norman Franke
• [asterisk-users] Media gateways and video   Paul Hales
• [asterisk-users] :POSSIBLE SPAM: conferencing help   Nhadie
• [asterisk-users] Background Noise Elimination   Matt Riddell
• [asterisk-users] FWD and IPCall   Huw Richards
• [asterisk-users] :POSSIBLE SPAM: conferencing help   Shane D
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] FWD and IPCall   Huw Richards
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] FWD and IPCall   Matt Riddell
• [asterisk-users] Media gateways and video   Matt Riddell
• [asterisk-users] :POSSIBLE SPAM: conferencing help   Matt Riddell
• [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help   Nhadie
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] FWD and IPCall   Huw Richards
• [asterisk-users] asterisk CLI and no such command "stop"   Vieri
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] FWD and IPCall   Huw Richards
• [asterisk-users] pickup application failed   Rilawich Ango
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] FWD and IPCall   Matt Riddell
• [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help   Matt Riddell
• [asterisk-users] FWD and IPCall   Huw Richards
• [asterisk-users] no outgoing calls with Digium B410P   Lyle Giese
• [asterisk-users] FWD and IPCall   Huw Richards
• [asterisk-users] FWD and IPCall   Shane D
• [asterisk-users] no outgoing calls with Digium B410P   Tzafrir Cohen
• [asterisk-users] help need   pgck nirukshitha
• [asterisk-users] conferencing help   Nhadie
• [asterisk-users] Polycom IP4000 - Device does not match ACL   CunningPike
• [asterisk-users] app_rxfax.c and app_txxfax.c where?   Klaverstyn, David C
• [asterisk-users] help need   Alex Balashov
• [asterisk-users] conferencing help   Matt Riddell
• [asterisk-users] conferencing help   Paul Hales
• [asterisk-users] help need   Matt Riddell
• [asterisk-users] Distorted audio over Eicon Diva Server BRI   CSB
• [asterisk-users] pickup application failed   dave cantera
• [asterisk-users] Early media support for Asterisk behind NAT   Mayur
• [asterisk-users] app_rxfax.c and app_txxfax.c where?   Doug
• [asterisk-users] Remote hold on PRI   Gaëtan Minet
• [asterisk-users] conferencing help   Tzafrir Cohen
• [asterisk-users] app_rxfax.c and app_txxfax.c where?   Tzafrir Cohen
• [asterisk-users] Distorted audio over Eicon Diva Server BRI   Matt Riddell
• [asterisk-users] conferencing help   Matt Riddell
• [asterisk-users] Distorted audio over Eicon Diva Server BRI   Armin Schindler
• [asterisk-users] HPEC   clive.chan(Atn)
• [asterisk-users] no outoging calls with B410P   Daniele
• [asterisk-users] Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?   Len
• [asterisk-users] chan_h323 and asterisk 1.2   Vieri
• [asterisk-users] Distorted audio over Eicon Diva Server BRI   Tim Panton
• [asterisk-users] communicating SMS messages in asterisk   saqib butt
• [asterisk-users] disable call waiting by default   nik600
• [asterisk-users] Limiting number of simultaneous calls in E1 line   Rajkumar S
• [asterisk-users] Asterisk Nokia   Arun Kumar
• [asterisk-users] Multi-SPAN (4xE1) Zap Group (Outbound)   Mike Trest - Personal
• [asterisk-users] How to check if a SIP phone is forwarded without ringing it ?   Olivier
• [asterisk-users] app_rxfax.c and app_txxfax.c where?   Jonn R Taylor
• [asterisk-users] Limiting number of simultaneous calls in E1 line   Christian Victor
• [asterisk-users] Prevent Asterisk from rebuiling DTMF tones   Morten Isaksen
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Steve Langstaff
• [asterisk-users] Background Noise Elimination   Jerry Jones
• [asterisk-users] Bugs??   Abdul
• [asterisk-users] disable call waiting by default   Yehavi Bourvine +972-8-9489444
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Olivier
• [asterisk-users] Bugs??   Mike Trest - Personal
• [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?   Steve Langstaff
• [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help   dave cantera
• [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help   Steve Totaro
• [asterisk-users] disable call waiting by default   Don Pobanz
• [asterisk-users] GotoIf() help   dave cantera
• [asterisk-users] Early media support for Asterisk behind NAT   Johansson Olle E
• [asterisk-users] :POSSIBLE SPAM: Re: conferencing help   Nhadie
• [asterisk-users] :POSSIBLE SPAM: Re: conferencing help   Steve Edwards
• [asterisk-users] Lamps on Snom phones   Phil Knighton
• [asterisk-users] [Zaptel] Checking that TDM card works?   Vincent
• [asterisk-users] Bugs??   Michiel van Baak
• [asterisk-users] [Zaptel] Checking that TDM card works?   Tzafrir Cohen
• [asterisk-users] [Zaptel] Checking that TDM card works?   Jared Smith
• [asterisk-users] What's the best ztdummy?   Steve Edwards
• [asterisk-users] Bugs??   Abdul
• [asterisk-users] Distorted audio over Eicon Diva Server BRI   CSB
• [asterisk-users] Bugs??   Abdul
• [asterisk-users] BLF trouble   Lars Bensmann
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Jean-Louis curty
• [asterisk-users] Simultaneous Callback?!   Douglas Garstang
• [asterisk-users] BLF trouble   Johansson Olle E
• [asterisk-users] Bugs??   Steve Edwards
• [asterisk-users] BLF trouble   Benny Amorsen
• [asterisk-users] Bugs??   Steve Edwards
• [asterisk-users] Distorted audio over Eicon Diva Server BRI   Matt Riddell
• [asterisk-users] CallerID Number incorrect in SIP packet   Lutgring, Sam
• [asterisk-users] debugging bluetooth communication using chan_mobile   Emmanuel Favre-Nicolin
• [asterisk-users] Simultaneous Callback?!   Tim H. Panton
• [asterisk-users] CallerID Number incorrect in SIP packet   Adam Moffett
• [asterisk-users] tale of two firewalls   Robert Moskowitz
• [asterisk-users] get_data   Charlie Farinella
• [asterisk-users] Linksys SPA-9xx Audio Issues   Andrew Joakimsen
• [asterisk-users] Linksys SPA-9xx Audio Issues   Jared Smith
• [asterisk-users] Linksys SPA-9xx Audio Issues   Andrew Joakimsen
• [asterisk-users] Linksys SPA-9xx Audio Issues   Daniel Cole
• [asterisk-users] Distorted audio over Eicon Diva Server BRI   Matt Riddell
• [asterisk-users] Linksys SPA-9xx Audio Issues   Daniel Cole
• [asterisk-users] BLF trouble   Lars Bensmann
• [asterisk-users] txfax_exec: Transmission loop error   Roger Schreiter
• [asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ...   Douglas Garstang
• [asterisk-users] Linksys SPA-9xx Audio Issues   Kev S
• [asterisk-users] Dialplan Recordings   Shane D
• [asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ...   Douglas Garstang
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Raj Jain
• [asterisk-users] Linksys SPA-9xx Audio Issues   Daniel Cole
• [asterisk-users] tale of two firewalls   dave cantera
• [asterisk-users] Linksys SPA-9xx Audio Issues   Kev S
• [asterisk-users] :POSSIBLE SPAM: Re: conferencing help   dave cantera
• [asterisk-users] Dialplan Recordings   Tilghman Lesher
• [asterisk-users] conferencing help   Nhadie
• [asterisk-users] conferencing help   Matt Riddell
• [asterisk-users] conferencing help   gary at intele-com.com
• [asterisk-users] Linksys SPA-9xx Audio Issues   Daniel Cole
• [asterisk-users] Register source port   Al lists
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Olivier
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Olivier
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Johansson Olle E
• [asterisk-users] Register source port   Johansson Olle E
• [asterisk-users] conferencing help   Nhadie
• [asterisk-users] Set CDR userfield in a realtime dialplan   Yves Räber
• [asterisk-users] conferencing help   Matt Riddell
• [asterisk-users] conferencing help   Tzafrir Cohen
• [asterisk-users] Set CDR userfield in a realtime dialplan   Benchev
• [asterisk-users] Set CDR userfield in a realtime dialplan   Andrea Cristofanini
• [asterisk-users] [Zaptel] Checking that TDM card works?   Vincent
• [asterisk-users] [Zaptel] Checking that TDM card works?   Vincent
• [asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ...   Steve Davies
• [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?   Steve Langstaff
• [asterisk-users] remote Snom 360: no voice passing thru   gincantalupo
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Benny Amorsen
• [asterisk-users] Set CDR userfield in a realtime dialplan   Yves Räber
• [asterisk-users] [Zaptel] Checking that TDM card works?   Tzafrir Cohen
• [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?   Raj Jain
• [asterisk-users] conferencing help   Nhadie
• [asterisk-users] What's the best ztdummy?   Tzafrir Cohen
• [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?   Raj Jain
• [asterisk-users] [Zaptel] Checking that TDM card works?   Vincent
• [asterisk-users] [Zaptel] Checking that TDM card works?   Tzafrir Cohen
• [asterisk-users] Linksys SPA-9xx Audio Issues   Chris Bagnall
• [asterisk-users] [Zaptel] Checking that TDM card works?   Darrick Hartman (lists)
• [asterisk-users] Newbie: confusion with the new FXO/FXS card   Vytenis Sabaliauskas
• [asterisk-users] [Zaptel] Checking that TDM card works?   Vincent
• [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?   Johansson Olle E
• [asterisk-users] Newbie: confusion with the new FXO/FXS card   Glenn Cobb
• [asterisk-users] What's the best ztdummy?   Thomas Stein
• [asterisk-users] [Zaptel] Checking that TDM card works?   Darrick Hartman (lists)
• [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?   Steve Langstaff
• [asterisk-users] [asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers   Atis Lezdins
• [asterisk-users] Zaptel FXS Cards - Station Distance   Tim Nelson
• [asterisk-users] Busy notification with call limiting by GROUP_COUNT()   Peter Galiovsky
• [asterisk-users] Zaptel FXS Cards - Station Distance   Kevin P. Fleming
• [asterisk-users] Zaptel FXS Cards - Station Distance   Tilghman Lesher
• [asterisk-users] Zaptel FXS Cards - Station Distance   John Novack
• [asterisk-users] Broken calls   mccoy silva
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Olivier
• [asterisk-users] WaitExten and Macros   Tony Plack
• [asterisk-users] WaitExten and Macros   Tony Plack
• [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?   Olivier
• [asterisk-users] WaitExten and Macros   Tilghman Lesher
• [asterisk-users] Intercom & Paging with Polycoms   Rob Schall
• [asterisk-users] WaitExten and Macros   Rob Schall
• [asterisk-users] WaitExten and Macros   Jared Smith
• [asterisk-users] Subscriptions, Firewalls, 489 "Bad Event" and Bug 7608   Tony Plack
• [asterisk-users] WaitExten and Macros   Tony Plack
• [asterisk-users] [Zaptel] Checking that TDM card works?   Vincent
• [asterisk-users] WaitExten and Macros   Tony Plack
• [asterisk-users] Polycom 550 IP SoundStation Fuzzy Voice Quality   Mike Coakley
• [asterisk-users] Linksys SPA-9xx Audio Issues   Andrew Joakimsen
• [asterisk-users] IAXy ringing   Adam Moffett
• [asterisk-users] Busy notification with call limiting byGROUP_COUNT()   Don Pobanz
• [asterisk-users] WaitExten and Macros   Tony Plack
• [asterisk-users] Limiting number of simultaneous calls in E1line   Dovid B
• [asterisk-users] Busy notification with call limiting byGROUP_COUNT()   Ira
• [asterisk-users] Newbie: confusion with the new FXO/FXS card   Andrew Stewart
• [asterisk-users] Busy notification with call limiting byGROUP_COUNT()   Atis Lezdins
• [asterisk-users] FXOTUNE update   Matthew Fredrickson
• [asterisk-users] Intercom & Paging with Polycoms   C F
• [asterisk-users] Two Asterisk Boxes Playing Together   Shane D
• [asterisk-users] Two Asterisk Boxes Playing Together   Alex Balashov
• [asterisk-users] Two Asterisk Boxes Playing Together   Matt Riddell
• [asterisk-users] Two Asterisk Boxes Playing Together   Rob Hillis
• [asterisk-users] x100p wcfxo hangup on outgoing calss   Jonathan GF
• [asterisk-users] Polycom 550 IP SoundStation Fuzzy Voice Quality   Doug Lytle
• [asterisk-users] Two lines for outgoing calls   Jonathan GF
• [asterisk-users] Polycom 550 IP SoundStation Fuzzy Voice Quality   Paul Hales
• [asterisk-users] OT: Traffic Shaping   Matt Riddell
• [asterisk-users] OT: Traffic Shaping   Erik Anderson
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   Jaap Winius
• [asterisk-users] OT: Traffic Shaping   Matt Riddell
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   Matt Riddell
• [asterisk-users] OT: Traffic Shaping   Erik Anderson
• [asterisk-users] IEEE 802.1x capable sip phones   Jeronimo Romero
• [asterisk-users] OT: Traffic Shaping   Matt Riddell
• [asterisk-users] IEEE 802.1x capable sip phones   Robert Moskowitz
• [asterisk-users] IEEE 802.1x capable sip phones   Kev S
• [asterisk-users] IEEE 802.1x capable sip phones   Jeronimo Romero
• [asterisk-users] forward call intended for another domain   Randall Smith
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   stoffell
• [asterisk-users] Which IP Phone is really the best?   stoffell
• [asterisk-users] Which IP Phone is really the best?   randulo
• [asterisk-users] HFC-S zap channels always busy   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   IT-Connect
• [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?   Benny Amorsen
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Benny Amorsen
• [asterisk-users] Kirk and asterisk   Fons van der Beek
• [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices   Armin Krämer
• [asterisk-users] Simultaneous Callback?!   CSB
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Jean-Louis curty
• [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?   Olivier
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Olivier
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Olivier
• [asterisk-users] IEEE 802.1x capable sip phones   Olivier
• [asterisk-users] IEEE 802.1x capable sip phones   Olivier
• [asterisk-users] OT - Is handover included in DECT GAP ?   Olivier
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   Michiel van Baak
• [asterisk-users] OT: Traffic Shaping   Dovid B
• [asterisk-users] OT - Is handover included in DECT GAP ?   Michiel van Baak
• [asterisk-users] OT - Is handover included in DECT GAP ?   Robert Lister
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Jean-Louis curty
• [asterisk-users] OT - Is handover included in DECT GAP ?   Michiel van Baak
• [asterisk-users] IEEE 802.1x capable sip phones   Robert Moskowitz
• [asterisk-users] OT - Is handover included in DECT GAP ?   Anselm Martin Hoffmeister
• [asterisk-users] IEEE 802.1x capable sip phones   Robert Moskowitz
• [asterisk-users] problem about TDM400P ringback detection   Fabio Antonini
• [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices   Robert Moskowitz
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] OT - Is handover included in DECT GAP ?   Olivier
• [asterisk-users] OT - Is handover included in DECT GAP ?   Olivier
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Olivier
• [asterisk-users] IEEE 802.1x capable sip phones   Olivier
• [asterisk-users] IEEE 802.1x capable sip phones   Olivier
• [asterisk-users] HFC-S zap channels always busy   Tzafrir Cohen
• [asterisk-users] OT - Is handover included in DECT GAP ?   Robert Lister
• [asterisk-users] OT: Traffic Shaping   Drew Gibson
• [asterisk-users] SIP URI question and NATs   Robert Moskowitz
• [asterisk-users] IEEE 802.1x capable sip phones   Robert Moskowitz
• [asterisk-users] IEEE 802.1x capable sip phones   Robert Moskowitz
• [asterisk-users] OT - Is handover included in DECT GAP ?   Robert Lister
• [asterisk-users] OT - Is handover included in DECT GAP ?   Michiel van Baak
• [asterisk-users] OT: Traffic Shaping   Erik Anderson
• [asterisk-users] IEEE 802.1x capable sip phones   Olivier
• [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices   Jonn R Taylor
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Jean-Louis curty
• [asterisk-users] IEEE 802.1x capable sip phones   Robert Moskowitz
• [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices   Robert Moskowitz
• [asterisk-users] WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"unknown" <sip:unknown at xxx.xxx.xxx.xxx>   gincantalupo
• [asterisk-users] OT: Traffic Shaping   Kristian Kielhofner
• [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices   Jonn R Taylor
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Olivier
• [asterisk-users] Intercom & Paging with Polycoms   Rob Schall
• [asterisk-users] Polycom 330 beep on new VM   Kevin Kiely
• [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?   Robert Moskowitz
• [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices   Robert Moskowitz
• [asterisk-users] OT: Traffic Shaping   Matt Riddell
• [asterisk-users] Sip calls drop one leg after about 2 minutes   John Millican
• [asterisk-users] Polycom 550 IP SoundStation Fuzzy Voice Quality   Mike Coakley
• [asterisk-users] Sip calls drop one leg after about 2 minutes   Doug
• [asterisk-users] No NAT, but firewall mangles Register SDP   Robert Moskowitz
• [asterisk-users] Asterisk Realtime unixODBC timeout?   Norman Franke
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] Multiple fax extensions   Ronald Wiplinger
• [asterisk-users] HFC-S zap channels always busy   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   Patrick
• [asterisk-users] Multiple fax extensions   Doug Lytle
• [asterisk-users] Congestion/Forbidden issue with new carrier   Kev S
• [asterisk-users] PRI Down but zaptel lets calls through   Michael J. Liberatore
• [asterisk-users] Congestion/Forbidden issue with new carrier   Paul Hales
• [asterisk-users] SIP URI question and NATs   Johansson Olle E
• [asterisk-users] WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"unknown" <sip:unknown at xxx.xxx.xxx.xxx>   Johansson Olle E
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   Olivier
• [asterisk-users] interconnecting an asterisk server with an old alcatel PBX through a Digium B410P   Daniele
• [asterisk-users] OT - Where do most email2fax errors come from ?   Olivier
• [asterisk-users] Congestion/Forbidden issue with new carrier   Trevor G. Hammonds
• [asterisk-users] Developing Help   Bhrugu Mehta
• [asterisk-users] Congestion/Forbidden issue with new carrier   Kevin S
• [asterisk-users] Lest we forget: Friday 12 Noon EST - VoIP Users Conference   randulo
• [asterisk-users] 15% Off from New Cyber-Telecom.net Website   Sam Tam
• [asterisk-users] Is rfc4662 (SIP Resource Lists notification) support planned ?   Olivier
• [asterisk-users] Dealy while taking   pgck nirukshitha
• [asterisk-users] Sip calls drop one leg after about 2 minutes   John Millican
• [asterisk-users] Dealy while taking   Max Weltz
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] More detalis: Re: SIP URI question and NATs   Robert Moskowitz
• [asterisk-users] [OT] Call for speakers: BOB 2.0   Johansson Olle E
• [asterisk-users] Soundcard necessary on an asterisk server toget output of playback()?? -> Next step   Stefan Guenther
• [asterisk-users] Question about queues and the definition of agents   Stefan Guenther
• [asterisk-users] Dialplan flow on device state change   Andrea Spadaccini
• [asterisk-users] Question about queues and the definition of agents   Mark Michelson
• [asterisk-users] Intercom & Paging with Polycoms   George Pajari
• [asterisk-users] MRCP Asterisk Integration   Cavalera Claudio Luigi
• [asterisk-users] Deadlock of asterisk on app_system   Atis Lezdins
• [asterisk-users] Authenticate problems (Extensions)   Joris Cras
• [asterisk-users] Question about queues and the definition of agents   PGentilini at intellicare.com
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   Hans Witvliet
• [asterisk-users] HFC-S zap channels always busy   Tzafrir Cohen
• [asterisk-users] Fwd: Trying to build MFC/R2   Brian Hutchinson
• [asterisk-users] Asterisk 1.4 and ISDN-BRI support   IT-Connect
• [asterisk-users] MONITOR CMD()   Bruno de Assumpção Loureiro
• [asterisk-users] Question about queues and the definition ofagents   Don Pobanz
• [asterisk-users] HFC-S zap channels always busy   Jaap Winius
• [asterisk-users] Soundcard necessary on an asterisk server toget output of playback()?? -> Next step   Tzafrir Cohen
• [asterisk-users] Asterisk Realtime unixODBC timeout?   Tilghman Lesher
• [asterisk-users] HPEC   clive.chan(Atn)
• [asterisk-users] Bandwidth and Colocation (was: Re: Is it possible to use spandsp and patton to do fax2mail ?)   Philipp Kempgen
• [asterisk-users] Congestion/Forbidden issue with new carrier   Kevin S
• [asterisk-users] Congestion/Forbidden issue with new carrier   Kevin S
• [asterisk-users] Developing Help   Anthony Francis
• [asterisk-users] zaptel digit problem   Bhrugu Mehta
• [asterisk-users] MFC/R2 Signaling configuration   Brian Hutchinson
• [asterisk-users] Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!   Johansson Olle E
• [asterisk-users] My latest MFC/R2 update with asterisk-1.4.17   Brian Hutchinson
• [asterisk-users] MFC/R2 Signaling configuration   Tzafrir Cohen
• [asterisk-users] Perl-AGI process   Abdul
• [asterisk-users] ISDN channels not properly released after call   Antonios Tsakiridis
• [asterisk-users] Perl-AGI process   Trevor Peirce
• [asterisk-users] [FreeBSD 6.2] Error compiling Zaptel from Ports?   Vincent
• [asterisk-users] Perl-AGI process   Abdul
• [asterisk-users] My latest MFC/R2 update with asterisk-1.4.17   Brian Hutchinson
• [asterisk-users] MFC/R2 Signaling configuration   Brian Hutchinson
• [asterisk-users] My latest MFC/R2 update with asterisk-1.4.17   Moises Silva
• [asterisk-users] Fwd: Trying to build MFC/R2   Moises Silva
• [asterisk-users] MFC/R2 Signaling configuration   Brian Hutchinson
• [asterisk-users] Fwd: Trying to build MFC/R2   Brian Hutchinson
• [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF conversion erros.   Mayur
• [asterisk-users] PRI Down but zaptel lets calls through   Michael J. Liberatore
• [asterisk-users] Packet2Packet bridging occurring when not wanted   Cameron Jenkins
• [asterisk-users] Asterisk ports and CentOS firewall   Ed Nunez
• [asterisk-users] PRI Down but zaptel lets calls through   Andrew Joakimsen
• [asterisk-users] Packet2Packet bridging occurring when not wanted   Andrew Joakimsen
• [asterisk-users] Packet2Packet bridging occurring when not wanted   Andrew Joakimsen
• [asterisk-users] PRI Down but zaptel lets calls through   Michael J. Liberatore
• [asterisk-users] Asterisk ports and CentOS firewall   dave cantera
• [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF conversion erros.   dave cantera
• [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()?? -> Next step   Stefan Guenther
• [asterisk-users] Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!   Matthew Rubenstein
• [asterisk-users] MRCP Asterisk Integration   Olivier
• [asterisk-users] interconnecting an asterisk server with an old alcatel PBX through a Digium B410P   Olivier
• [asterisk-users] Adtran 750 and E&M Wink   Ron McCarthy
• [asterisk-users] HPEC   Thomas Kenyon
• [asterisk-users] Question about queues and the definition and agents   Stefan Guenther
• [asterisk-users] problems with zaptel and Udev   robert boardman
• [asterisk-users] Perl-AGI process   Trevor Peirce
• [asterisk-users] problems with zaptel and Udev   Tzafrir Cohen
• [asterisk-users] Asterisk ports and CentOS firewall   MatsK
• [asterisk-users] problems with zaptel and Udev   robert boardman
• [asterisk-users] ProxyPal for AMI Proxy Development   Lee Jenkins
• [asterisk-users] ProxyPal for AMI Proxy Development   Julian Lyndon-Smith
• [asterisk-users] problems with zaptel and Udev   Ed Nunez
• [asterisk-users] ProxyPal for AMI Proxy Development   Lee Jenkins
• [asterisk-users] Question about queues and the definition and agents   Paul Hales
• [asterisk-users] Aastra Venture   Michael Graves
• [asterisk-users] Call parking   Paul Hales
• [asterisk-users] ProxyPal for AMI Proxy Development   Julian Lyndon-Smith
• [asterisk-users] [SOLVED + EXPLANATION]: Strange ISDN-problem with incoming calls out of the same city   Stefan Guenther
• [asterisk-users] Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!   Philippe Sultan
• [asterisk-users] State of the application chan_spy   Jan-Hendrik Palic
• [asterisk-users] MRCP Asterisk Integration   Cavalera Claudio Luigi
• [asterisk-users] State of the application chan_spy   Atis Lezdins
• [asterisk-users] MRCP Asterisk Integration   Olivier
• [asterisk-users] AGISTATUS is SUCCESS even though my PHP script returned -1   Brian Hutchinson
• [asterisk-users] g729 codec - simultaneous calls   Naveen Palani
• [asterisk-users] g729 codec - simultaneous calls   Thomas Kenyon
• [asterisk-users] State of the application chan_spy   Jan-Hendrik Palic
• [asterisk-users] g729 codec - simultaneous calls   Steve Totaro
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   bilal ghayyad
• [asterisk-users] Video Call and Asterisk   bilal ghayyad
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Steve Totaro
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Chris Bagnall
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Tzafrir Cohen
• [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?   Olivier
• [asterisk-users] Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!   Johansson Olle E
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Gordon Henderson
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Steve Kennedy
• [asterisk-users] Different ringing tones ...   Gordon Henderson
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Tzafrir Cohen
• [asterisk-users] ProxyPal for AMI Proxy Development   Lee Jenkins
• [asterisk-users] OT: reverse DNS error for lists.digium.com   Armin Schindler
• [asterisk-users] ISDN channels not properly released after call   Armin Schindler
• [asterisk-users] Help needed for Fax2Email with Welltech FXO 3804   Ronald Wiplinger
• [asterisk-users] Meetme Record Format   Rob Schall
• [asterisk-users] problems with zaptel and Udev   robert boardman
• [asterisk-users] Temporary Service - Dominican Republic DID   Ryan M. Colbert
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Steve Totaro
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Steve Totaro
• [asterisk-users] Asterisk 1.4 Call Recording   Mike Hammett
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Tzafrir Cohen
• [asterisk-users] OT: reverse DNS error for lists.digium.com   Trevor Peirce
• [asterisk-users] OT: reverse DNS error for lists.digium.com   Gaëtan Minet
• [asterisk-users] [SOLVED + EXPLANATION]: Strange ISDN-problem with incoming calls out of the same city   Philipp Kempgen
• [asterisk-users] OT: reverse DNS error for lists.digium.com   Armin Schindler
• [asterisk-users] Asterisk 1.4 Call Recording   Steve Johnson
• [asterisk-users] Verficar VoiceMail   Gilberto Nunes
• [asterisk-users] [asterisk-dev] Unstable releases lately   Jared Smith
• [asterisk-users] Voicemail check   Gilberto Nunes
• [asterisk-users] VoiceMail Check   Gilberto Nunes
• [asterisk-users] Verficar VoiceMail   Shane D
• [asterisk-users] Verficar VoiceMail   Gilberto Nunes
• [asterisk-users] Voicemail check   Steve Johnson
• [asterisk-users] OT: reverse DNS error for lists.digium.com   Kevin P. Fleming
• [asterisk-users] Voicemail check   Gilberto Nunes
• [asterisk-users] Asterisk 1.4 Call Recording   Mike Hammett
• [asterisk-users] OT: reverse DNS error for lists.digium.com   Armin Schindler
• [asterisk-users] OT: reverse DNS error for lists.digium.com   Gaëtan Minet
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Gordon Henderson
• [asterisk-users] Verficar VoiceMail   Kristian Kielhofner
• [asterisk-users] Asterisk 1.4 Call Recording   Steve Johnson
• [asterisk-users] Transfer/Speed-Dial   Ken Williams
• [asterisk-users] Different ringing tones ...   Ken Williams
• [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor   Sam Tam
• [asterisk-users] Question about queues and the definition and agents   Stefan Guenther
• [asterisk-users] app_voicemail for spanish   Anton Krall
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Alex Balashov
• [asterisk-users] CID blocking ...   J. Oquendo
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Steve Totaro
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Alex Balashov
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Steve Totaro
• [asterisk-users] Asterisk 1.4.17 crashing more   Abdul
• [asterisk-users] CID blocking ...   Paul Hales
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Andrew Joakimsen
• [asterisk-users] app_voicemail for spanish   Andrew Joakimsen
• [asterisk-users] SVN servers down for maintenance   Russell Bryant
• [asterisk-users] Asterisk 1.4.17 crashing more   Steve Totaro
• [asterisk-users] Zaptel 1.2.23 and 1.4.8 released   The Asterisk Development Team
• [asterisk-users] AGISTATUS is SUCCESS even though my PHP script returned -1   Matt Riddell
• [asterisk-users] Video Call and Asterisk   Matt Riddell
• [asterisk-users] app_voicemail for spanish   Anton Krall
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Steve Totaro
• [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF conversion erros.   Mayur
• [asterisk-users] AGISTATUS is SUCCESS even though my PHP script returned -1   Steve Edwards
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Tzafrir Cohen
• [asterisk-users] Park() help, extension not heard   Rob
• [asterisk-users] AGISTATUS is SUCCESS even though my PHP script returned -1   Brian Hutchinson
• [asterisk-users] Asterisk 1.4.17 crashing more   Brian Hutchinson
• [asterisk-users] app_voicemail for spanish   Andrew Joakimsen
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Andrew Joakimsen
• [asterisk-users] Park() help, extension not heard   Rob
• [asterisk-users] pickupchan without bristuffed version?   Stefan Guenther
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Bruce McAlister
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Thomas Kenyon
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Andrew Joakimsen
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Thomas Kenyon
• [asterisk-users] Meetme recording   Lees, James (UK)
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Bruce McAlister
• [asterisk-users] Asterisk 1.4.17 crashing more   Abdul
• [asterisk-users] Meetme recording   Gordon Henderson
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Tzafrir Cohen
• [asterisk-users] Meetme recording   Tony Mountifield
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Thomas Kenyon
• [asterisk-users] Console app   Gilberto Nunes
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Tzafrir Cohen
• [asterisk-users] CID blocking ...   Philipp Kempgen
• [asterisk-users] Meetme recording   John covici
• [asterisk-users] Fax machine detect   Naveen Palani
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Steve Totaro
• [asterisk-users] SIP Reason   Carles Pina i Estany
• [asterisk-users] Playing DTMF tones down a channel   Lees, James (UK)
• [asterisk-users] cisco ip phne 7911G with asterisk   Christian Pinedo
• [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.   Thomas Kenyon
• [asterisk-users] SIP Reason   Johansson Olle E
• [asterisk-users] busy/congestion random   Sasa
• [asterisk-users] SIP Reason   Carles Pina i Estany
• [asterisk-users] SIP Reason   Steve Langstaff
• [asterisk-users] sip channel error - extension pattern matching problem   Tomasz Zieleniewski
• [asterisk-users] SIP Reason   Johansson Olle E
• [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions   Johansson Olle E
• [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions   Patrick
• [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions   Atis Lezdins
• [asterisk-users] Playing DTMF tones down a channel   Tony Mountifield
• [asterisk-users] SIP Reason   Steve Langstaff
• [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions   Philipp Kempgen
• [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions   CSB
• [asterisk-users] Asterisk 1.4 Call Recording   Mike Hammett
• [asterisk-users] SIP Reason   Carles Pina i Estany
• [asterisk-users] app_voicemail for spanish   Anton Krall
• [asterisk-users] Fax machine detect   Naveen Palani
• [asterisk-users] Park() help, extension not heard   Jared Smith
• [asterisk-users] Asterisk 1.4 Call Recording   Patrick
• [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions   Tilghman Lesher
• [asterisk-users] Interrupt the swift text   Naveen Palani
• [asterisk-users] Asterisk 1.4 Call Recording   Mike Hammett
• [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions   Atis Lezdins
• [asterisk-users] Asterisk 1.4 Call Recording   Mike Hammett
• [asterisk-users] Interrupt the swift text   Steve Totaro
• [asterisk-users] Interrupt the swift text   Ron Joffe
• [asterisk-users] Heartbeat   Jeremy Mann
• [asterisk-users] Record calls then send them to users voicemail   Anciso, Roy
• [asterisk-users] asterisk 1.4 context   Jerry Geis
• [asterisk-users] Interrupt the swift text