September 2006 Archives by date

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Starting: Fri Sep 1 00:03:26 MST 2006
Ending: Sat Sep 30 20:54:35 MST 2006
Messages: 3621

• [asterisk-users] asterisk presence (from manager API)   harrygaillac-sip at yahoo.fr
• [asterisk-users] quadbri & TDM400P on same pbx ?   Giorgio Incantalupo
• [asterisk-users] Any way to go from factory reset 7970 to SIP without Call Manager?   Jason Lixfeld
• [asterisk-users] Re: Any way to go from factory reset 7970 to SIP without Call Manager?   Tomislav Parčina
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] Outgoing Call group ?   Noc Phibee
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Andrea Spadaccini
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Andrea Spadaccini
• [asterisk-users] help me!!Problem on incoming calls   Marco Mouta
• [asterisk-users] Probelm with incoming calls to my DID-Please help me   Crazy Boy
• [asterisk-users] voicemail as email and attachment   Tim St. Pierre
• [asterisk-users] Any way to go from factory reset 7970 to SIP without Call Manager?   Matthew Crocker
• [asterisk-users] 911 versus 9.911   end1r
• [asterisk-users] looking for GXV-3000 users   marek cervenka
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] 911 versus 9.911   Matthew Crocker
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Dave Fullerton
• [asterisk-users] Sipura SPA3000   Rich Adamson
• [asterisk-users] Asterisk as a SER client   Andrea Spadaccini
• [asterisk-users] help me!!Problem on incoming calls   Marco Mouta
• [asterisk-users] balance anouncement   ram
• [asterisk-users] Different MOH in parked calls??   equis software
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Andrea Spadaccini
• [asterisk-users] Asterisk as a SER client   David Hindmarsh
• [asterisk-users] 911 versus 9.911   Strom Carlson
• [asterisk-users] balance anouncement   John Millican
• [asterisk-users] phpagi syntax and SendDTMF   Frank Church
• [asterisk-users] Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Adit 3104 randomly reboot   Jerry Jones
• [asterisk-users] [Slightly OT] Grandstream configurator tool   Dave Fullerton
• [asterisk-users] Sipura SPA3000   Steve Kennedy
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Kevin Smith
• [asterisk-users] Missing Agent Function   William Piper
• [asterisk-users] question of CLI   William Piper
• [asterisk-users] Outgoing Call group ?   William Piper
• [asterisk-users] balance anouncement   ram
• [asterisk-users] Asterisk Core dump   Anthony Musaluke
• [asterisk-users] Re: 911 versus 9.911   Dave Fullerton
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] Sipura SPA3000   Kevin Collins
• [asterisk-users] balance anouncement   John Millican
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Henrik Woffinden
• [asterisk-users] 911 versus 9.911   Jay Milk
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Dave Fullerton
• [asterisk-users] Sipura SPA3000   Rich Adamson
• [asterisk-users] Toll-Free numbers   Jay Milk
• [asterisk-users] Re: 911 versus 9.911   Doug Lytle
• [asterisk-users] Asterisk as a SER client   Arnd Vehling
• [asterisk-users] incompatible hardware?   Roy Sigurd Karlsbakk
• [asterisk-users] US Toll-Free DID Providers with Caller ID NAME?   Jay Milk
• [asterisk-users] Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Sipura SPA3000   Bob Chiodini
• [asterisk-users] help me!!Problem on incoming calls   Marco Mouta
• [asterisk-users] Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Sipura 3000 and Asterisk   Francisco Seratti
• [asterisk-users] Re: Any way to go from factory reset 7970 to SIP without Call Manager?   Jason Lixfeld
• [asterisk-users] Probelm with incoming calls to my DID-Please help me   Marco Mouta
• [asterisk-users] balance anouncement   ram
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Jonn R Taylor
• [asterisk-users] Re: Re: 911 versus 9.911   Jim Rice
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Steven Ringwald
• [asterisk-users] Outgoing Call group ?   Ira
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Marco Mouta
• [asterisk-users] Re: Re: 911 versus 9.911   Derek Whitten
• [asterisk-users] balance anouncement   Kevin Savoy
• [asterisk-users] Sipura SPA3000   Rich Adamson
• [asterisk-users] quadbri & TDM400P on same pbx ?   mike
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] Re: Re: 911 versus 9.911   Doug Lytle
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] SPA-942 Sound Quality   Andres
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bob Chiodini
• [asterisk-users] Re: Re: 911 versus 9.911   ahester at galacticltd.com
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] balance anouncement   ram
• [asterisk-users] Re: Re: 911 versus 9.911   Matt
• [asterisk-users] Re: Re: 911 versus 9.911   Mojo with Horan & Company, LLC
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   J. Oquendo
• [asterisk-users] Re: Any way to go from factory reset 7970 to SIP without Call Manager?   Lacy Moore - Aspendora
• [asterisk-users] Re: Re: Re: 911 versus 9.911   Steven
• [asterisk-users] Re: Re: Re: 911 versus 9.911   Steven
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Steven Ringwald
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Steven Ringwald
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Crazy Boy
• [asterisk-users] Asterisk speaks Italian!   Stuart
• [asterisk-users] Callback + dtmf problem   Insider KT
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bob Chiodini
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Tim Sharp
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bruce Ferrell
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bruce Ferrell
• [asterisk-users] Probelm with incoming calls to my DID-Please help me   Crazy Boy
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Steven Ringwald
• [asterisk-users] Asterisk speaks Italian!   Michael Collins
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Richard Scobie
• [asterisk-users] Asterisk speaks Italian!   Dean Collins
• [asterisk-users] Cisco 7960 won't download dialplan.xml   Peter Pauly
• [asterisk-users] Cisco 7960 won't download dialplan.xml   Aaron Daniel
• [asterisk-users] Sipura SPA3000   Rich Adamson
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Richard Klingler
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Mr. Jones
• [asterisk-users] Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF?   Gary G. Hendershot
• [asterisk-users] Callback + dtmf problem   James
• [asterisk-users] Fax with asterisk?   Matthias Fechner
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Kevin P. Fleming
• [asterisk-users] What does 'trunk' mean in outgoing and incoming?   Larry Alkoff
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Elpidio Ramos
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Lacy Moore - Aspendora
• [asterisk-users] Operator Console(s)/Shared Call Appearances   Tim St. Pierre
• [asterisk-users] Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF?   Tim St. Pierre
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Tim St. Pierre
• [asterisk-users] What does 'trunk' mean in outgoing and incoming?   Tim St. Pierre
• [asterisk-users] Asterisk speaks Italian!   Tzafrir Cohen
• [asterisk-users] Help with blind transfer   George A. Roberts IV
• [asterisk-users] Asterisk speaks Italian!   Tzafrir Cohen
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Tzafrir Cohen
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Cory Andrews
• [asterisk-users] Asterisk speaks Italian!   Dean Collins
• [asterisk-users] Unable to open pseudo channel for timing... Sound may be choppy.   Dominik Weber
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   Anthony Rodgers
• [asterisk-users] "best" BRI card ?   Tzafrir Cohen
• [asterisk-users] Re: Any way to go from factory reset 7970 to SIP without Call Manager?   Jason Lixfeld
• [asterisk-users] Asterisk mysql cdr   Abdul
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Help compiling asterisk-addons on Debian?   Rushowr
• [asterisk-users] Keys pressed not registering ...   Rushowr
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Asterisk mysql cdr   Matt Riddell (IT)
• [asterisk-users] Keys pressed not registering ...   John covici
• [asterisk-users] Asterisk speaks Italian!   Tzafrir Cohen
• [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] Re: Cisco 7960G SIP firmware 8.4   Nathan Alberti
• [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaperthan existing one.   Kannaiyan Natesan
• [asterisk-users] Help compiling asterisk-addons on Debian?   Tzafrir Cohen
• [asterisk-users] Asterisk server crashes after two years   Tzafrir Cohen
• [asterisk-users] Blind transfer 3/4 digits   David Gagnon
• [asterisk-users] Problems compil 1.2.11   Tzafrir Cohen
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Jerry Jones
• [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060   Bob Chiodini
• [asterisk-users] Toll-Free numbers   Tim Panton
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Nokia N80   Dean Collins
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Problems compil 1.2.11   Lenny
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Problems compil 1.2.11   Andrea Spadaccini
• [asterisk-users] Keys pressed not registering ...   Ronald Wiplinger
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Ronald Wiplinger
• [asterisk-users] Keys pressed not registering ...   John covici
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Bob Chiodini
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Hermann Wecke
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Tim St. Pierre
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Re: [asterisk-biz] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] Sipura SPA3000   sdcharly at gmail.com
• [asterisk-users] Re: [asterisk-biz] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] [asterisk-dev] Re: [asterisk-biz] G729 Replacement Codec - FREE ormay ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF?   Gary G. Hendershot
• [asterisk-users] Problems compil 1.2.11   Tzafrir Cohen
• [asterisk-users] Blind transfer 3/4 digits   Kevin Smith
• [asterisk-users] Asterisk server crashes after two years   Nir Simionovich
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Jonn R Taylor
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] res_osp.c not compiled   neil
• [asterisk-users] Blind transfer 3/4 digits   Nick Ellson
• [asterisk-users] Caller ID has extra digits to strip   Bart Fisher
• [asterisk-users] How to send correct Caller ID on PRI   Zeeshan Zakaria
• [asterisk-users] SIPP problem   Diego Quintana Cruz
• [asterisk-users] Roundrobin not working on PRI   Zeeshan Zakaria
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Zeeshan Zakaria
• [asterisk-users] Queue timeout problems   Mr. Jones
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Bob Chiodini
• [asterisk-users] Caller ID has extra digits to strip   Ira
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Nic Bellamy
• [asterisk-users] Blind transfer 3/4 digits   Tim St. Pierre
• [asterisk-users] How to send correct Caller ID on PRI   Tim St. Pierre
• [asterisk-users] Queue timeout problems   Guido Hecken
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Roundrobin not working on PRI   Tim St. Pierre
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Tim St. Pierre
• [asterisk-users] Caller ID has extra digits to strip   Tim St. Pierre
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Tim St. Pierre
• [asterisk-users] Roundrobin not working on PRI   Andres
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Larry Alkoff
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Hardware ? Analog DID trunks (ILT)   Jonn R Taylor
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Nick Ellson
• [asterisk-users] SIPP problem   Greg Boehnlein
• [asterisk-users] Queue timeout problems   Mr. Jones
• [asterisk-users] What I always get asked in SME * deployments   Eric Bishop
• [asterisk-users] Help with blind transfer   George A. Roberts IV
• [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaper than existing one.   Remco Barendse
• [asterisk-users] SER+Asterisk integration   Siqhamo Sifo
• [asterisk-users] File structure question   Tzafrir Cohen
• [asterisk-users] Queue timeout problems   Guido Hecken
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Using Thunderbird (mail client) to call Contacts from Address Book   Matt Riddell (IT)
• [asterisk-users] Asterisk not sending RTP   Matt Riddell (IT)
• [asterisk-users] UK (BT) Problem with TDM 400P   Mark Muffett
• [asterisk-users] UK (BT) Problem with TDM 400P   Matt Riddell (IT)
• [asterisk-users] UK (BT) Problem with TDM 400P   Tzafrir Cohen
• [asterisk-users] Asterisk+ser+docs   Siqhamo Sifo
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Unable to open pseudo channel for timing... Sound may be choppy.   Tzafrir Cohen
• [asterisk-users] UK (BT) Problem with TDM 400P   Mark Muffett
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Asterisk not sending RTP   Jeremy McNamara
• [asterisk-users] Query about Call Detail Record in Asterisk   Chan Kwang Mien
• [asterisk-users] Asterisk not sending RTP   Matt Riddell (IT)
• [asterisk-users] UK (BT) Problem with TDM 400P   Matt Riddell (IT)
• [asterisk-users] UK (BT) Problem with TDM 400P   Rich Adamson
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Asterisk not sending RTP   Jeremy McNamara
• [asterisk-users] UK (BT) Problem with TDM 400P   Mark Muffett
• [asterisk-users] SIPP problem   Diego Quintana Cruz
• [asterisk-users] UK (BT) Problem with TDM 400P   Nick Chalk
• [asterisk-users] Asterisk not sending RTP   Nir Simionovich
• [asterisk-users] Asterisk+ser+docs   Victor Toofic
• [asterisk-users] UK (BT) Problem with TDM 400P   Mark Muffett
• [asterisk-users] Asterisk+ser+docs   ram
• [asterisk-users] SER+Asterisk integration   Rob Lith
• [asterisk-users] SIPP problem   Tzafrir Cohen
• [asterisk-users] AgentCallBackLogin and cdrupdate   Garth van Sittert
• [asterisk-users] Queue timeout problems   Mr. Jones
• [asterisk-users] SER+Asterisk integration   Arnd Vehling
• [asterisk-users] SIPP problem   Diego Quintana Cruz
• [asterisk-users] What I always get asked in SME * deployments   Dovid Bender
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Kevin Smith
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Rich Adamson
• [asterisk-users] PBX -> VoIP migration   Richard Klingler
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Avi Miller
• [asterisk-users] Roundrobin not working on PRI   Zeeshan Zakaria
• [asterisk-users] Blind transfer 3/4 digits   Fabio
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Avi Miller
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Avi Miller
• [asterisk-users] Please help route incoming PSTN calls to Asterisk   Larry Alkoff
• [asterisk-users] PBX -> VoIP migration   Paul Hales
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Paul Hales
• [asterisk-users] Help with blind transfer   William Piper
• [asterisk-users] Help with blind transfer   George A. Roberts IV
• [asterisk-users] Help with blind transfer   George A. Roberts IV
• [asterisk-users] Asterisk calling through FWD?   Nick Ellson
• [asterisk-users] Blind transfer 3/4 digits   Koopmann, Jan-Peter
• [asterisk-users] Re:sip giving problems, please help.   Ma Zhiyong
• [asterisk-users] Fax with asterisk?   DRi at b-w-computer.de
• [asterisk-users] No more linux/compiler.h in Fedora Core 6.   William F. Acker WB2FLW +1-303-722-7209
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Zoa
• [asterisk-users] "Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com>,   Vidura Senadeera
• [asterisk-users] Zaptel-1.2.8 compile problem   Vidura Senadeera
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] Architecture:MainOffice(AstServer)-SmallOffices(ATA.-LegacyPBX)   Marco Mouta
• [asterisk-users] Zaptel-1.2.8 compile problem   Rushowr
• [asterisk-users] Zaptel-1.2.8 compile problem   Tzafrir Cohen
• [asterisk-users] Zaptel-1.2.8 compile problem   yusuf
• [asterisk-users] Any Hardphone with VPNClient embedded?   Marco Mouta
• [asterisk-users] External calls from Asterisk over a Siemens(legacy) RDSI PBX   Llorenç Suau
• [asterisk-users] What I always get asked in SME * deployments   Colin MacMillan
• [Asterisk-Users] External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX   Llorenç Suau
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] usereqphone=yes seems to don't work   Alexandre VERNIOL
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Victor Toofic
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] vim syntax highlighting( for Asterisk.conf files)   Marco Mouta
• [asterisk-users] FAX handling   Jose Limeres
• [asterisk-users] Any Hardphone with VPNClient embedded?   Leo Ann Boon
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Any Hardphone with VPNClient embedded?   Cory Andrews
• [asterisk-users] PAP2-NA + Asterisk   joy panlilio
• [asterisk-users] Handling Disconnection Causes   Rafael J. Risco G.V.
• [asterisk-users] Blind transfer 3/4 digits   David Gagnon
• [asterisk-users] Asterisk 1.2.11 and # key   David Gagnon
• [asterisk-users] Submenus   Mir
• [asterisk-users] Dropping extra frame of G.729 ?   Noc Phibee
• [asterisk-users] FAX handling   phil.dawson at marnock.com
• [asterisk-users] FAX handling   Marco Mouta
• [asterisk-users] Dropping extra frame of G.729 ?   Hermann Wecke
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Elpidio Ramos
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Justin Tunney
• [asterisk-users] File structure question   Jay Moore
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Carlos Chavez
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Rich Adamson
• [asterisk-users] missing pri connect (wwomera to pri)   Rosario Pingaro
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Elpidio Ramos
• [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK   Elpidio Ramos
• [asterisk-users] File structure question   Peter Bowyer
• [asterisk-users] File structure question   Marco Mouta
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Jean-Louis curty
• [asterisk-users] Roundrobin not working on PRI   Andres
• [asterisk-users] Astbill DIALSTRING doesn't work   Sebastian Milioto
• [asterisk-users] playback some digits to the caller from the callee (involves DTMF) prob   umar tarar
• [asterisk-users] Grandstream and H.264 !   Sergio (Red)
• [asterisk-users] Grandstream and H.264 !   Carlos Chavez
• [asterisk-users] File structure question   Jay Moore
• [asterisk-users] Looks like Nufone is changing around...   Justin Newman
• [asterisk-users] Any Hardphone with VPNClient embedded?   Francesco Peeters (Asterisk)
• [asterisk-users] File structure question   Peter Bowyer
• [asterisk-users] app_conference not working for me   Steve Edwards
• [asterisk-users] Looks like Nufone is changing around...   Jeremy McNamara
• [asterisk-users] includes in realtime ??   Rushowr
• [asterisk-users] Call center reports   Technical Support
• [asterisk-users] FAX handling   Tzafrir Cohen
• [asterisk-users] missing pri connect (wwomera to pri)   Tzafrir Cohen
• [asterisk-users] Call center reports   Hermann Wecke
• [asterisk-users] FAX handling   Technical Support
• [asterisk-users] FAX handling   Tzafrir Cohen
• [asterisk-users] app_conference not working for me   Matt Riddell (IT)
• [asterisk-users] Re: Nufone making changes   Justin Newman
• [asterisk-users] Asterisk calling through FWD?   Nick Ellson
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Leo Ann Boon
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Blind transfer 3/4 digits   wendell hamilton
• [asterisk-users] Digum g729 and g723   Joe shmoe
• [asterisk-users] SNMP with 1.2.11 stable   Azfhasterisk
• H.264 basic backport (was Re: [asterisk-users] Grandstream and H.264 !)   Nic Bellamy
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] Digum g729 and g723   Kannaiyan Natesan
• [asterisk-users] Asterisk calling through FWD?   Nick Ellson
• [asterisk-users] Asterisk calling through FWD?   Derek Whitten
• [asterisk-users] Blind transfer 3/4 digits   Ronald Wiplinger
• [asterisk-users] Digum g729 and g723   brandon kruz
• [asterisk-users] Asterisk calling through FWD?   Nick Ellson
• [asterisk-users] File structure question   Jay Moore
• [asterisk-users] Codec Thread   Joe shmoe
• [asterisk-users] Re: FAX handling   Justin Newman
• [asterisk-users] Warning about using PAP2-NA ATA recent firmware 3.1.12 LS   joy panlilio
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] Blind transfer 3/4 digits   David Gagnon
• [asterisk-users] HITBSecConf2006 Final Call !   Praburaajan
• [asterisk-users] Asterisk 1.2.11 and # key   David Gagnon
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] Blind transfer 3/4 digits   Koopmann, Jan-Peter
• [asterisk-users] Reading the raw E1 channels ?   Azher Amin
• [asterisk-users] Blind transfer 3/4 digits   Koopmann, Jan-Peter
• [asterisk-users] Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one.   Kannaiyan Natesan
• [asterisk-users] includes in realtime ??   RR
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] End of call   Michael Strelnikov
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] File structure question   Peter Bowyer
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Richard Klingler
• [asterisk-users] connect with two servers multiple time   Arjan Kroon
• [asterisk-users] Re: FAX handling   Jose Limeres
• [asterisk-dev] Re: [asterisk-users] Digum g729 and g723   Kannaiyan Natesan
• [asterisk-users] Re: How to use Grandstream GX-2000 phones for paging   Tomislav Parčina
• [asterisk-dev] Re: [asterisk-users] Digum g729 and g723   Kannaiyan Natesan
• [asterisk-users] includes in realtime ??   RR
• [asterisk-users] A couple more interviews with Digium staff   Matt Riddell (IT)
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] Codec Thread   Matt Riddell (IT)
• [asterisk-users] Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one.   Matt Riddell (IT)
• [asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...   Marco Mouta
• [asterisk-users] includes in realtime ??   RR
• [Asterisk-Users] External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX   Wolfgang Zweimueller
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...   Marco Mouta
• [asterisk-users] Can not hear the telco System Announcement   stoffell
• [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...   Tony Mountifield
• [asterisk-users] Matra 6501   Richard Klingler
• [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...   Marco Mouta
• [asterisk-users] How to manipulate a plus in a phone number   Tim Panton
• [asterisk-users] Any Hardphone with VPNClient embedded?   Philipp von Klitzing
• [asterisk-users] How to manipulate a plus in a phone number   Doug Lytle
• [asterisk-users] latest CentOS-asterisk-freepbx installation procedure   Roland
• [asterisk-users] How to manipulate a plus in a phone number   Tim Panton
• [asterisk-users] latest CentOS-asterisk-freepbx installation procedure   Marco Mouta
• [asterisk-users] Experience Patton BRI gateways and Asterisk?   Koopmann, Jan-Peter
• [asterisk-users] latest CentOS-asterisk-freepbx installation procedure   Avi Miller
• [asterisk-users] Re: How to manipulate a plus in a phone number   Stefan Tichy
• [asterisk-users] Re: How to manipulate a plus in a phone number   Tim Panton
• [asterisk-users] Codec Thread   Jean-Michel Hiver
• [asterisk-users] telco error message on PRI and BRI   stoffell
• [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010   Jopi at fastwebnet.it
• [asterisk-users] ISDN config EWSD   Virmones Pereira Tavares de Miranda
• [asterisk-users] Can not hear the telco System Announcement   Jean-Michel Hiver
• [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010   Mike Lynchfield
• [asterisk-users] Zero length queue   Artifex Maximus
• R: Re: [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010   Jopi at fastwebnet.it
• [asterisk-users] ISDN config EWSD   Roger Schreiter
• [asterisk-users] Can not hear the telco System Announcement   stoffell
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] Keys pressed not registering ...   Lenny
• [asterisk-users] includes in realtime ??   Douglas Garstang
• [asterisk-users] Zero length queue   Wes Baehr
• [asterisk-users] Unable to make calls from CallManager to Asterisk   Anantha Padmanabha.M.L
• [asterisk-users] File structure question   Marco Mouta
• [asterisk-users] Different MOH in waiting calls and parked calls   equis software
• [asterisk-users] Experience Patton BRI gateways and Asterisk?   Guido Hecken
• [asterisk-users] Wrong CallerID passed to SIP phone   Richard Klingler
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server   Marco Mouta
• [asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server   Rich Adamson
• [Asterisk-Users] T1 echo canceller   Michael Araba
• [asterisk-users] How to manipulate a plus in a phone number   Ira
• [asterisk-users] includes in realtime ??   RR
• [asterisk-users] Find-Me/Follow-ME   Roger Workman
• [asterisk-users] Asterisk vicidial question   Gustavo Alejandro Gonzalez
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Timothy R. McKee
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] Asterisk vicidial question   ggonzalez at telviso.com.ar
• [asterisk-users] Find-Me/Follow-ME   Joel Vandal
• [Asterisk-Users] T1 echo canceller   Matthew Crocker
• [asterisk-users] Faxing ..   Lenny
• [asterisk-users] Find-Me/Follow-ME   Marnus van Niekerk
• [asterisk-users] Find-Me/Follow-ME   Roger Workman
• [Asterisk-Users] T1 echo canceller   Doug Lytle
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Dave Fullerton
• [asterisk-users] Asterisk vicidial question   Matt Florell
• [asterisk-users] Faxing ..   Steve Totaro
• [asterisk-users] Different MOH between waiting calls and transfer calls   equis software
• [asterisk-users] IAX and rsa   andrutto
• [asterisk-users] Different MOH between waiting calls and transfer   Doug Lytle
• [asterisk-users] blf aastra 9133i working but can't pickup calls   shadowym
• [asterisk-users] Meet-me recording formats   Michael Lively
• [asterisk-users] Asterisk & Linksys PAP2 ATA   Tim St. Pierre
• [asterisk-users] config include issues   Curt Shaffer
• [asterisk-users] asterisk t.38 fax failed   Kokfoo Soo
• [asterisk-users] Catch an event   Olivier Saulnier
• [asterisk-users] Articulation Palm client and Asterisk   Jorge Alayon
• [asterisk-users] config include issues   Doug Lytle
• [asterisk-users] config include issues   Curt Shaffer
• [asterisk-users] asterisk t.38 fax failed   Ricardo Carvalho
• [asterisk-users] Asterisk and REFER authentication   KEN KANGAN
• [asterisk-users] Linking Asterisk with PBX through E1   Marlon Dutra
• [asterisk-users] Adding custom fields (more than one) to CDR DB   Mike
• [asterisk-users] Is this a warning or not...MYSQL Fetch   Mike
• [asterisk-users] Adding custom fields (more than one) to CDR DB   Steven Ringwald
• [asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...   Eric "ManxPower" Wieling
• [Asterisk-Users] Asterisk Cygwin Port.   joshnet at nbnet.nb.ca
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• [asterisk-users] How to notify an ACD agent before he/she picks up   Manrique Feoli
• [asterisk-users] Adding custom fields (more than one) to CDR DB   Matt Riddell (IT)
• [asterisk-users] Deadlock   Michael Welter
• [asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...   Jason Parker
• [asterisk-users] config include issues   Doug Lytle
• [asterisk-users] Merlin Legend - Working Now!   Sterling Moses
• [asterisk-users] Caller ID has extra digits to strip   Bart Fisher
• [asterisk-users] Caller ID has extra digits to strip   Steve Totaro
• [asterisk-users] Caller ID has extra digits to strip   Bart Fisher
• [asterisk-users] Merlin Legend - Working Now!   Steve Totaro
• [asterisk-users] Native Chinese speaker needed   John Williams
• [asterisk-users] Need somebody for video phone testing   Ronald Wiplinger
• [asterisk-users] Has anyone tried to install both digital card and analog card in one machine   Xue Liangliang
• [asterisk-users] End of call   Michael Strelnikov
• [asterisk-users] Call center reports   David Gagnon
• [asterisk-users] Asterisk 1.2.11 and # key   David Gagnon
• [asterisk-users] Really bad phone line.. possible causes?   Jeff Turner
• [asterisk-users] Merlin Legend - Working Now!   Sterling Moses
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] Has anyone tried to install both digital card and analog card in one machine   Raphael Jacquot
• [asterisk-users] macros in Realtime   Benjamin Jacob
• [Asterisk-Users] T1 echo canceller   Michael Araba
• [asterisk-users] Asterisk + Samsung OffServ 500   Eugeniy Khvastunov
• [asterisk-users] Catch an event   Tzafrir Cohen
• [asterisk-users] Codec Thread   Erik
• [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry   Matthew Thompson
• [asterisk-users] core dumps   Anthony Musaluke
• [asterisk-users] Asterisk AGI and Firebird   Steve Rawlings
• [asterisk-users] Wrong CallerID passed to SIP phone   Richard Klingler
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] macros in Realtime   Simon Woodhead
• [asterisk-users] How to test TE405P T1   Andy Chung (Power-All)
• [asterisk-users] Budgetones - multiple phones losing IP address during day   Garth van Sittert
• [asterisk-users] Budgetones - multiple phones losing IP address during day   Brandon Galbraith
• [asterisk-users] Merlin Legend - Working Now!   Steve Totaro
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] sangoma A104d echo canceller and fax   Klaus Darilion
• [asterisk-users] flag 'g' in Dail() is'nt working with agentcallbacklogin()   umar tarar
• [asterisk-users] core dumps   Marco Mouta
• [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...   Tony Mountifield
• [asterisk-users] IAX and rsa   picciuX
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] mobile refusing call   René Enskat [Teamware GmbH]
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] Budgetones - multiple phones losing IP address during day   Rob Lith
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] how to setup poxy sip server   Ranjeet Kumar
• [asterisk-users] Answer Machine detection   Mark Ackroyd
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Marco Mouta
• [asterisk-users] how to setup poxy sip server   brian at neotiq.com
• [asterisk-users] Answer Machine detection   Matt Florell
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] cmd SET time value   Benjamin Jacob
• [asterisk-users] macros in Realtime   Benjamin Jacob
• [asterisk-users] includes in realtime ??   Benjamin Jacob
• [asterisk-users] Asterisk + Samsung OffServ 500   Garth van Sittert
• [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages   Giorgio Incantalupo
• [asterisk-users] How to test TE405P T1   Garth van Sittert
• [asterisk-users] Unable to make calls from CallManager to Asterisk   Gary Richardson
• [asterisk-users] Deadlock   BJ Weschke
• [asterisk-users] SIP client with video???   Joao Pereira
• [asterisk-users] Asterisk video support   Joao Pereira
• [asterisk-users] asterisk t.38 fax failed   Kokfoo Soo
• [asterisk-users] app_rxfax Only Receives One Page   Steve Totaro
• [asterisk-users] Native Chinese speaker needed   Steve Underwood
• [asterisk-users] asterisk t.38 fax failed   Ricardo Carvalho
• [asterisk-users] How to notify an ACD agent before he/she picks up   MF
• [asterisk-users] Has anyone tried to install both digital card and analog card in one machine   MF
• [asterisk-users] app_rxfax Only Receives One Page   Doug Lytle
• [Asterisk-Users] Which SIP hardphone with embedded VPNClient ?   Olivier
• [asterisk-users] Budgetones - multiple phones losing IP address during day   Jessee J Holmes
• [asterisk-users] SIP client with video???   Blake Krone
• [asterisk-users] Cisco PIX firewall and nat=yes   Bill Gibbs
• [asterisk-users] Budgetones - multiple phones losing IP addressduring day   Harden, Bob
• [asterisk-users] Cisco PIX firewall and nat=yes   Peder at NetworkOblivion
• [asterisk-users] asterisk t.38 fax failed   Kokfoo Soo
• [asterisk-users] Cisco MWI   Steve Kennedy
• [asterisk-users] Cisco PIX firewall and nat=yes   Bill Gibbs
• [asterisk-users] Budgetones - multiple phones losing IP addressduring day   Jessee J Holmes
• [asterisk-users] Submenus   Mojo with Horan & Company, LLC
• [asterisk-users] app_rxfax Only Receives One Page   Steve Totaro
• [asterisk-users] app_rxfax Only Receives One Page   Marco Mouta
• [asterisk-users] app_rxfax Only Receives One Page   Doug Lytle
• [asterisk-users] Cisco MWI   Doug Lytle
• [asterisk-users] Conditional IF based on IP address?   Steve Hsieh
• [asterisk-users] app_rxfax Only Receives One Page   Steve Totaro
• [asterisk-users] app_rxfax Only Receives One Page   Steve Underwood
• [asterisk-users] Is asterisk's mgcp support(NAS) Network access server package   Ibrar Ahmed
• [asterisk-users] Is asterisk's mgcp support(NAS) Network access server package   Davor Grgicevic
• [asterisk-users] Asterisk 1.2.11 and # key   Matt
• [asterisk-users] Submenus   Mojo with Horan & Company, LLC
• [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry   Matt Birmingham
• [asterisk-users] Native Chinese speaker needed   John Williams
• [asterisk-users] Call parking and RTP traffic   Dave Fullerton
• [asterisk-users] Conditional IF based on IP address?   Rushowr
• [asterisk-users] Really bad phone line.. possible causes?   Mojo with Horan & Company, LLC
• [asterisk-users] Call parking and ringbacks   J. Oquendo
• [asterisk-users] Cisco MWI   Doug Lytle
• [asterisk-users] Conditional IF based on IP address?   Rushowr
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Dan Serban
• [asterisk-users] Native Chinese speaker needed   Steve Hsieh
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Alberto Sagredo
• [asterisk-users] Conditional IF based on IP address?   Steve Hsieh
• [asterisk-users] Conditional IF based on IP address?   Rushowr
• [asterisk-users] Different MOH between waiting calls and transfer   equis software
• [asterisk-users] Different MOH between waiting calls and transfer   Doug Lytle
• [asterisk-users] app_rxfax Only Receives One Page   Marco Mouta
• [asterisk-users] Volume events causing talk off on Asterisk with Digium 411P   Servetas, Andrew
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Dan Serban
• [asterisk-users] Cisco MWI   Steve Kennedy
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Alberto Sagredo
• [asterisk-users] Cisco MWI   Doug Lytle
• [asterisk-users] Cisco MWI   Michiel van Baak
• [asterisk-users] Cisco MWI   Aaron Daniel
• [asterisk-users] faktortel   Dean Collins
• [asterisk-users] Cisco MWI   Doug Lytle
• [asterisk-users] Digium G.729 codec binaries updated   Kevin P. Fleming
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Mojo with Horan & Company, LLC
• [asterisk-users] Digium's response to posting of G.729 and G.723 source code   Kevin P. Fleming
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Leo Ann Boon
• [asterisk-users] Garbled (quality probs) IAX2 & SIP calls Asterisk-to-Asterisk   lists.digium.com at tgice.com
• [asterisk-users] Re: Really bad phone line.. possible causes?   M.Hockings
• [asterisk-users] Native Chinese speaker needed   Steve Underwood
• [asterisk-users] app_rxfax Only Receives One Page   Steve Underwood
• [asterisk-users] using SIP to connect remote other VoIP server   tengulre
• [asterisk-users] How to test TE405P T1   Andy Chung (Power-All)
• [asterisk-users] Digium G.729 codec binaries updated   Kristian Kielhofner
• [asterisk-users] Asterisk 1.2.11 and # key   Michael Strelnikov
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Polycom new firmware and bootrom   Chris Dos
• [asterisk-users] using SIP to connect remote other VoIP server   Tim St. Pierre
• [asterisk-users] How to check which rtp ports my firewall let through?   Ronald Wiplinger
• [asterisk-users] the sounds quality of IAX2 channels are not good as SIP channels?   Ma Zhiyong
• [asterisk-users] cmd SET time value   Benjamin Jacob
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Rich Adamson
• [asterisk-users] cmd SET time value   Tim St. Pierre
• [asterisk-users] Query on Call Forward Feature codes for SIP users..   A C Sathish-a22713
• [asterisk-users] Native Chinese speaker needed   Steve Hsieh
• [asterisk-users] Re: [asterisk-dev] UUI in calls   John Todd
• [asterisk-users] cmd SET time value   Benjamin Jacob
• [asterisk-users] Volume events causing talk off on Asterisk with Digium 411P   Zoa
• [asterisk-users] Response to KP Flemming...   Joe Shmoe
• [asterisk-users] ast_parse_allow_disallow: Cannot allow unknown format 'h264'   Ronald Wiplinger
• [asterisk-users] Response to KP Flemming...   Joe Shmoe
• [asterisk-users] Response to KP Flemming...   Joe Shmoe
• [asterisk-users] How to send and receiving fax with asterisk?   Andrea infoteam
• [asterisk-users] Configuring new IAX2 Jitter Buffer for IVR application.   John Melody
• [asterisk-users] New polycom firmware / presence   harrygaillac-sip at yahoo.fr
• [asterisk-users] netmask   Dean Collins
• [asterisk-users] WG: mobile refusing call   René Enskat [Teamware GmbH]
• [asterisk-users] netmask   Richard Klingler
• [asterisk-users] bristuff compile problems with kernel 2.6.17.11   Arik Raffael Funke
• [asterisk-users] netmask   Dean Collins
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Cisco 7970 directories and services xml   Tomislav Parčina
• [asterisk-users] Cisco 7970 directories and services xml   Richard Klingler
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Bob Chiodini
• [asterisk-users] Response to KP Flemming...   Rushowr
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Incoming call problem-calling part is busy(IPKall)   Crazy Boy
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Bruce Reeves
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   Rushowr
• [asterisk-users] Incoming call problem-calling part is busy(IPKall)   Doug Lytle
• [asterisk-users] Response to KP Flemming...   Matt Riddell (IT)
• [asterisk-users] Asterisk "Clusters"   Mitch Thompson
• [asterisk-users] Incoming call problem-calling part is busy(I PKall)   Guido Hecken
• [asterisk-users] RE: Volume events causing talk off on Asterisk with Digium 411P   Servetas, Andrew
• [asterisk-users] Response to KP Flemming...   Andrew Kohlsmith
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Re: Cisco 7970 directories and services xml   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 directories and services xml   Richard Klingler
• [asterisk-users] Re: Volume events causing talk off on Asterisk withDigium 411P   Steven
• [asterisk-users] netmask   Kokfoo Soo
• [asterisk-users] Response to KP Flemming...   Aaron Daniel
• [asterisk-users] Re: Cisco 7970 directories and services xml   Tomislav Parčina
• [asterisk-users] Capacity for transcode G711 to G729   Kokfoo Soo
• [asterisk-users] using SIP to connect remote other VoIP server   Elpidio Ramos
• [asterisk-users] New polycom firmware / presence   Douglas Garstang
• [asterisk-users] Response to KP Flemming...   Eric "ManxPower" Wieling
• [asterisk-users] Capacity for transcode G711 to G729   Matt Riddell (IT)
• [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls   Dan Serban
• [asterisk-users] svn trunk or branches ???   Ronald Wiplinger
• [asterisk-users] g729 failover when out of licenses   Tod Detre (CampusEAI Consortium)
• [asterisk-users] Voicemail Delete Bug?   Douglas Garstang
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Gareth Owen
• [asterisk-users] Asterisk and NAT ?   Noc Phibee
• [asterisk-users] Asterisk hangs up after 10-15 minutes when SIP Phone is on mute   Mike
• [asterisk-users] bristuff compile problems with kernel 2.6.17.11   Tzafrir Cohen
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Dave Cotton
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] Asterisk and NAT ?   yusuf
• [asterisk-users] Re: Really bad phone line.. possible causes?   Mojo with Horan & Company, LLC
• [asterisk-users] uConnect Voip device   Frank Church
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIP Phone is on mute   Tony Mountifield
• [asterisk-users] Sound (or lack of it) problems   Jordan Kirby
• [asterisk-users] Capacity for transcode G711 to G729   RR
• [asterisk-users] Polycom new firmware and bootrom   Nathan Alberti
• [asterisk-users] 0005162: RTP Packetization : Few questions   yusuf
• [asterisk-users] Polycom new firmware and bootrom   Douglas Garstang
• [asterisk-users] How to Install H323   Wasif
• [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)   Crazy Boy
• [asterisk-users] Polycom new firmware and bootrom   Bruce Reeves
• [asterisk-users] 0005162: RTP Packetization : Few questions   Dan Austin
• [asterisk-users] Polycom new firmware and bootrom   Jessee J Holmes
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIPPhone is on mute   Mike
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIPPhone is on mute   Dr. Michael J. Chudobiak
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Bruce Reeves
• [asterisk-users] Asterisk and NAT ?   Noc Phibee
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Brandon Galbraith
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute   Mike
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] Asterisk Outgoing Spool Failed   Arun Kumar
• [asterisk-users] Polycom new firmware and bootrom   Douglas Garstang
• [asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P   Servetas, Andrew
• [asterisk-users] Re: Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute   Rich Adamson
• [asterisk-users] Polycom new firmware and bootrom   Brandon Galbraith
• [asterisk-users] How to Install H323   Alberto Sagredo
• [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)   Rich Adamson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Polycom new firmware and bootrom   Jessee J Holmes
• [asterisk-users] Polycom new firmware and bootrom   harrygaillac-sip at yahoo.fr
• [asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P   Zoa
• [asterisk-users] Polycom new firmware and bootrom   Jessee J Holmes
• [asterisk-users] Re: bristuff compile problems with kernel 2.6.17.11   Arik Raffael Funke
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)   Elpidio Ramos
• [asterisk-users] 0005162: RTP Packetization : Few questions   Dan Austin
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Bruce Reeves
• [asterisk-users] Polycom new firmware and bootrom   Chris Dos
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Blake Krone
• [asterisk-users] Polycom new firmware and bootrom   Douglas Garstang
• [asterisk-users] Capacity for transcode G711 to G729   Matt Riddell (IT)
• [asterisk-users] Asterisk Outgoing Spool Failed   Matt Riddell (IT)
• [asterisk-users] svn trunk or branches ???   Matt Riddell (IT)
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Ferguson, Michael
• [asterisk-users] TDM400 and T100 config on same asterisk   Rich
• [asterisk-users] Open source G.729 and G.723.1 release for 1.2 and 1.4   Daniel Pocock
• [asterisk-users] Speex Codex - Eyebean to Asterisk   Kokfoo Soo
• [asterisk-users] g729 failover when out of licenses   Mark Phillips
• [asterisk-users] Capacity for transcode G711 to G729   Mark Phillips
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] g729 failover when out of licenses   Henry J. Cobb
• [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 39   Servetas, Andrew
• [asterisk-users] RE: Volume events causing talk off on Asterisk with Digium 411P   Servetas, Andrew
• [asterisk-users] Re: Really bad phone line.. possible causes?   Jeff Turner
• [asterisk-users] te110p and te205p behavioural differences   Mark Edwards
• [asterisk-users] Capacity for transcode G711 to G729   Matt Riddell (IT)
• [asterisk-users] Re: bristuff compile problems with kernel 2.6.17.11   Tzafrir Cohen
• [asterisk-users] Re: Really bad phone line.. possible causes?   Mojo with Horan & Company, LLC
• [asterisk-users] TDM400 and T100 config on same asterisk   Paul Hales
• [asterisk-users] Re: bristuff compile problems with kernel 2.6.17.11   Arik Raffael Funke
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   Leo Ann Boon
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   Tzafrir Cohen
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   Bob Chiodini
• [asterisk-users] Re: uConnect Voip device   Frank Church
• [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute   David Gagnon
• [asterisk-users] Polycom new firmware and bootrom   David Gagnon
• [asterisk-users] Re: uConnect Voip device   Alex Robar
• [asterisk-users] Re: uConnect Voip device   Frank Church
• [asterisk-users] Prompts playback changing tempo w/ SMP kernel   RR
• [asterisk-users] Capacity for transcode G711 to G729   RR
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   RR
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Nick Ellson
• [asterisk-users] Re: Softphones IAX vs. SIP, remote connectivity.   Martin Joseph
• [asterisk-users] Re: the sounds quality of IAX2 channels are not good as SIP channels?   Martin Joseph
• [asterisk-users] Query on Call Forward Feature codes for SIP users..   William Piper
• [asterisk-users] Asterisk and NAT ?   William Piper
• [asterisk-users] Call Forwarding in SIP.conf   Tim St. Pierre
• [asterisk-users] Softphones IAX vs. SIP, remote connectivity.   Cristian Draghici
• [asterisk-users] Intel 945G and Digium TE110P compatibility issue   Xue Liangliang
• [asterisk-users] cmd SET time value   Benjamin Jacob
• [asterisk-users] Asterisk 1.2 and SATA drives   Tharanga
• [asterisk-users] app_amd and voicemail   David Koski
• [asterisk-users] cmd SET time value   Tim St. Pierre
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] Asterisk "Clusters"   Koen Van Impe
• [asterisk-users] Re: the sounds quality of IAX2 channels are notgood as SIP channels?   Ma Zhiyong
• [asterisk-users] Asterisk 1.2 and SATA drives   Gabriel Afana
• [asterisk-users] Asterisk "Clusters"   Rob Lith
• [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?   Rushowr
• [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute   Daniel Pocock
• [asterisk-users] dialplan applications   Robert Bielik
• [asterisk-users] Asterisk 1.2 and SATA drives   Guido Hecken
• [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990   harrygaillac-sip at yahoo.fr
• [asterisk-users] Asterisk 1.2 and SATA drives   Raphael Jacquot
• [asterisk-users] Asterisk "Clusters"   Raphael Jacquot
• [asterisk-users] 0005162: RTP Packetization : Few questions   yusuf
• [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990 [SOLVED]   harrygaillac-sip at yahoo.fr
• [asterisk-users] blf aastra 9133i working but can't pickup calls   Jean-Louis curty
• [asterisk-users] 0005162: RTP Packetization : Few questions   yusuf
• [asterisk-users] New polycom firmware / presence   harrygaillac-sip at yahoo.fr
• [asterisk-users] sip peer question   Dijkstra, Roelof
• [asterisk-users] Problems with app_directed_pickup   Jon Schøpzinsky
• [asterisk-users] Digits are played in english in french voicemail   harrygaillac-sip at yahoo.fr
• [asterisk-users] Trouble with rxfax multi-page printing with cups   Artifex Maximus
• [asterisk-users] Digits are played in english in french voicemail   Dijkstra, Roelof
• [asterisk-users] Digits are played in english in french voicemail   Giorgio Incantalupo
• [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990   Tzafrir Cohen
• [asterisk-users] Trouble with rxfax multi-page printing with cups   Doug Lytle
• [asterisk-users] Caller ID display on 7970G   Richard Klingler
• [asterisk-users] Digits are played in english in french voicemail   Tzafrir Cohen
• [asterisk-users] Trouble with rxfax multi-page printing with cups   Steve Hanselman
• [asterisk-users] Asterisk and "Maximum retries exceeded"   Noc Phibee
• [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990   harrygaillac-sip at yahoo.fr
• [asterisk-users] Asterisk and NAT ?   Bob Chiodini
• [asterisk-users] distinguishing users by their domain   Ricardo Carvalho
• [asterisk-users] Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute   Dean Collins
• [asterisk-users] distinguishing users by their domain   Benjamin Jacob
• [asterisk-users] IAX and rsa   Andrew Nowrot
• [asterisk-users] Asterisk hangs up after 10-15 minuteswhenSIPPhone is on mute   Mike
• [asterisk-users] sip peer question   Rushowr
• [asterisk-users] Problems with KG1000 voip gateway and DTMF   Terence Haddock
• [asterisk-users] Trouble with rxfax multi-page printing with cups   Artifex Maximus
• [asterisk-users] Grandstream GX-2000, doesn't send calls to free lines   Zeeshan Zakaria
• [asterisk-users] How can I set CDR data in dialplan? Set(CDR(src)=foo)   Mike
• [asterisk-users] Call Forwarding in SIP.conf   William Piper
• [asterisk-users] Grandstream, how to use the configuration tool   Zeeshan Zakaria
• [asterisk-users] Grandstream GX-2000, doesn't send calls to free lines   Daniel Salama
• [asterisk-users] Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] Grandstream, how to use the configuration tool   William Piper
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC is arrested at JFK!!   Dean Collins
• [asterisk-users] Reload question   Mailing List
• [asterisk-users] Reload question   Patrick
• [asterisk-users] Re: Tracking the source of a disconnect?   Tony Mountifield
• [asterisk-users] Asterisk and SIP Redirect message   Michel Zenone
• [asterisk-users] Asterisk 1.2 and SATA drives   shadowym
• [asterisk-users] Transcode Speex to G711-ulaw   Kokfoo Soo
• [asterisk-users] Asterisk 1.2 and SATA drives   Brandon Galbraith
• [asterisk-users] sip peer question   Tim St. Pierre
• [asterisk-users] Re: Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC is arrested at JFK!!   Dean Collins
• [asterisk-users] Asterisk and SIP Redirect message   Johansson Olle E
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC is arrested at JFK!!   Alex Robar
• [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC isarrested at JFK!!   Dean Collins
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] 0005162: RTP Packetization : Few questions   Dan Austin
• [asterisk-users] Re: FW: Peter Dicks Chairman of Sportingbet PLCisarrested at JFK!!   Steven
• {Fraud?} RE: [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC isarrested at JFK!!   Jay Milk
• [asterisk-users] Re: FW: Peter Dicks Chairman of SportingbetPLCisarrested at JFK!!   Dean Collins
• [asterisk-users] Asterisk and "Maximum retries exceeded"   Noc Phibee
• [asterisk-users] FW: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!!   Dean Collins
• [asterisk-users] Use PauseQueueMember   gc
• [asterisk-users] Use PauseQueueMember   Julian Lyndon-Smith
• [asterisk-users] Call Forwarding in SIP.conf   Tim St. Pierre
• [asterisk-users] FW: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!!   Alex Robar
• [asterisk-users] distinguishing users by their domain   Ricardo Carvalho
• [asterisk-users] Want to support a better SIP stack in Asterisk?   Olle E Johansson
• [asterisk-users] Use PauseQueueMember   BJ Weschke
• [asterisk-users] Use PauseQueueMember   gc
• [asterisk-users] ISDN HFC card cannot 'detect remote answer'   Edoardo Serra
• [asterisk-users] Call Forwarding in SIP.conf   broadbandvoice at comcast.net
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] Asterisk Outgoing Spool Failed   Arun Kumar
• [asterisk-users] Asterisk and "Maximum retries exceeded"   Anthony Rodgers
• [asterisk-users] No dialtone, just directly busy   Henrik Woffinden
• [asterisk-users] Call Forwarding in SIP.conf   Tim St. Pierre
• [asterisk-users] What don't I get about SIP?   Tim St. Pierre
• [asterisk-users] distinguishing users by their domain   Tim St. Pierre
• [asterisk-users] Re: Tracking the source of a disconnect?   Jamin W. Collins
• [asterisk-users] No dialtone, just directly busy   Tim St. Pierre
• [asterisk-users] What don't I get about SIP?   Mike
• [asterisk-users] What don't I get about SIP?   Dave Fullerton
• [asterisk-users] How to use Grandstream GX-2000 phones for paging   Barry D. Hassler
• [asterisk-users] What don't I get about SIP?   Tim St. Pierre
• [asterisk-users] RE: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!!   Dean Collins
• [asterisk-users] Re: FW: Peter Dicks Chairman ofSportingbet PLC isarrested at JFK!!   Steven
• [asterisk-users] Re: FW: Peter Dicks Chairman ofSpor