September 2008 Archives by date

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Starting: Mon Sep 1 01:49:22 CDT 2008
Ending: Tue Sep 30 23:31:28 CDT 2008
Messages: 1665

• [asterisk-users] Asterisk Queue's   Tobias Ahlander
• [asterisk-users] Gateway errors   voip crazy
• [asterisk-users] Gateway errors   Igor Hernandez
• [asterisk-users] Gateway errors   voip crazy
• [asterisk-users] Asterisk Queue's   Philipp Kempgen
• [asterisk-users] Gateway errors   Andrea Spadaccini
• [asterisk-users] Gateway errors   Igor Hernandez
• [asterisk-users] Gateway errors   hatem moiz
• [asterisk-users] Gateway errors   voip crazy
• [asterisk-users] Penalties for agents   Tobias Ahlander
• [asterisk-users] not able to make call to landline no...to mobile works fine   bikrish at aim.com
• [asterisk-users] Problematic Trunk SIP: Got SIP response 405 "Method not allowed"   daniele visaggio
• [asterisk-users] PRI Splitter   Christian Victor
• [asterisk-users] Gateway errors   voip crazy
• [asterisk-users] Problematic Trunk SIP: Got SIP response 405 "Method not allowed"   Steven Howes
• [asterisk-users] Problematic Trunk SIP: Got SIP response 405 "Method not allowed"   daniele visaggio
• [asterisk-users] lists.digium.com monthly reminders   Tony Mountifield
• [asterisk-users] lists.digium.com monthly reminders   Michiel van Baak
• [asterisk-users] lists.digium.com monthly reminders   randulo
• [asterisk-users] lists.digium.com monthly reminders   Tzafrir Cohen
• [asterisk-users] lists.digium.com monthly reminders   Tony Mountifield
• [asterisk-users] lists.digium.com monthly reminders   Kevin P. Fleming
• [asterisk-users] lists.digium.com monthly reminders   Tony Mountifield
• [asterisk-users] Asterisk 1.6 beta   VoIP Cyprus
• [asterisk-users] PRI Splitter   Karl Fife
• [asterisk-users] PRI Splitter   Michael Graves
• [asterisk-users] Get call status and hangup   Loic Didelot
• [asterisk-users] PRI Splitter   Karl Fife
• [asterisk-users] PRI Splitter   Karl Fife
• [asterisk-users] Asterisk 1.6 beta   Rob Hillis
• [asterisk-users] Asterisk 1.6 beta   Tim Panton
• [asterisk-users] Documentation of users.conf   Nestor A. Diaz
• [asterisk-users] PRI Splitter   Olivier
• [asterisk-users] Documentation of users.conf   Robert Lister
• [asterisk-users] lists.digium.com monthly reminders   Ira
• [asterisk-users] PRI Splitter   Karl Fife
• [asterisk-users] lists.digium.com monthly reminders   Tony Mountifield
• [asterisk-users] Asterisk 1.6 beta   Benny Amorsen
• [asterisk-users] lists.digium.com monthly reminders   Benny Amorsen
• [asterisk-users] PRI Splitter   Karl Fife
• [asterisk-users] PRI Splitter   Christian Victor
• [asterisk-users] Still badly in need for mISDN help!   Julien Claassen
• [asterisk-users] PRI Splitter   Karl Fife
• [asterisk-users] Redundant PSTN PRI Gateways using Asterisk   Michael Melia Jr.
• [asterisk-users] Dialplan terminates when the caller hangs up   Cong Van Nguyen
• [asterisk-users] Asterisk Queue's   Paul Crane
• [asterisk-users] Gateway errors   hatem moiz
• [asterisk-users] Redundant PSTN PRI Gateways using Asterisk   Alex Balashov
• [asterisk-users] play remote file   Pezhman Lali
• [asterisk-users] Problem with Call Forward   Dpto. Datos Television Costa Blanca
• [asterisk-users] SetCallerPres   Steven Howes
• [asterisk-users] SetCallerPres   Steven Howes
• [asterisk-users] Problem with Call Forward   Philipp Kempgen
• [asterisk-users] zaptel 1.2.27 ?   Leonardo Gomes Figueira
• [asterisk-users] Redundant PSTN PRI Gateways using Asterisk   Karl Fife
• [asterisk-users] Asterisk Trunk and normal   bilal ghayyad
• [asterisk-users] Asterisk Trunk and normal   Doug Lytle
• [asterisk-users] Asterisk Trunk and normal   Dan Julius
• [asterisk-users] Pri to sip interfaces   Christian Victor
• [asterisk-users] lists.digium.com monthly reminders   Jay R. Ashworth
• [asterisk-users] Asterisk Trunk and normal   Philipp Kempgen
• [asterisk-users] Redundant PSTN PRI Gateways using Asterisk   Steve Totaro
• [asterisk-users] AgentCallbackLogin AddQueueMember   Krzysztof Zimnicki
• [asterisk-users] Asterisk Trunk and normal   bilal ghayyad
• [asterisk-users] PRI Splitter   FailSafe Inc.
• [asterisk-users] Asterisk Trunk and normal   Erik Anderson
• [asterisk-users] Asterisk Trunk and normal   bilal ghayyad
• [asterisk-users] Asterisk Trunk and normal   Erik Anderson
• [asterisk-users] AgentCallbackLogin AddQueueMember   Atis Lezdins
• [asterisk-users] play remote file   Atis Lezdins
• [asterisk-users] Faxing through Zap cards   James Sneeringer
• [asterisk-users] SALE 71% OFF on Pfizer   asterisk-users at lists.digium.com
• [asterisk-users] Dial timeout to cell phones   Steve Repo
• [asterisk-users] Problematic Trunk SIP: Got SIP response 405 "Method not allowed"   Kristian Kielhofner
• [asterisk-users] Dial timeout to cell phones   John Novack
• [asterisk-users] Dial timeout to cell phones   Mark G. Thomas
• [asterisk-users] Problem with Call Forward   Nhadie
• [asterisk-users] Selectively disable echo cancellation?   Ken D'Ambrosio
• [asterisk-users] Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte & Randal Schwartz   Michael Graves
• [asterisk-users] Asterisk CDR Problem   Steve Murphy
• [asterisk-users] beta9: how to set callerid on incoming iax?   sean darcy
• [asterisk-users] AgentCallbackLogin AddQueueMember   Lee, John (Sydney)
• [asterisk-users] DUNDI Help   ronald ramos
• [asterisk-users] Selectively disable echo cancellation?   Octavio Ruiz
• [asterisk-users] Congestion in Outgoing call through PRI   Octavio Ruiz
• [asterisk-users] PRI Splitter   Olivier
• [asterisk-users] Offering FIFO service to receptionist with LIFO hardphone ...   Olivier
• [asterisk-users] Asterisk 1.6 beta   Olivier
• [asterisk-users] Asterisk Queue's   Tobias Ahlander
• [asterisk-users] Newbie Polycom: ACD AgentLogin display on phone   Lee, John (Sydney)
• [asterisk-users] Dial timeout to cell phones   Steve Repo
• [asterisk-users] multiple passwords for one meetme!   fateme fatah
• [asterisk-users] Newbie Polycom: ACD AgentLogin display on phone   Paul Hales
• [asterisk-users] AgentCallbackLogin AddQueueMember   Krzysztof Zimnicki
• [asterisk-users] AgentCallbackLogin AddQueueMember   Lee, John (Sydney)
• [asterisk-users] Asterisk Queue's   Alejandro Kauffmann
• [asterisk-users] multiple passwords for one meetme!   Philipp Kempgen
• [asterisk-users] MixMonitor-Saving Recorded file with AgentId.   Syed Nasruddin
• [asterisk-users] Inefficient Codec Translation   Jim Boykin
• [asterisk-users] Asterisk Queue's   José Carlos Messias
• [asterisk-users] Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte & Randal Schwartz   randulo
• [asterisk-users] AgentCallbackLogin AddQueueMember   Atis Lezdins
• [asterisk-users] Asterisk Queue's   Atis Lezdins
• [asterisk-users] Live operator as a service?   Michael Graves
• [asterisk-users] Live operator as a service?   Dean Collins
• [asterisk-users] Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte & Randal Schwartz   Eric Moniz
• [asterisk-users] Selectively disable echo cancellation?   Jerry Jones
• [asterisk-users] Asterisk Crash   Josiah Bryan
• [asterisk-users] res_cepstral.so   Norman Franke
• [asterisk-users] Asterisk Crash   Andrew Latham
• [asterisk-users] Congestion in Outgoing call through PRI   Richard Lyman
• [asterisk-users] Asterisk Crash   Josiah Bryan
• [asterisk-users] Problem with Call Forward   Dpto. Datos Television Costa Blanca
• [asterisk-users] Asterisk Crash   Andrew Latham
• [asterisk-users] Inefficient Codec Translation   Steve Davies
• [asterisk-users] Live operator as a service?   Michael Graves
• [asterisk-users] SIP TLS / Nokia E51   Stefan Gofferje
• [asterisk-users] Asterisk Crash   Josiah Bryan
• [asterisk-users] Problem with Call Forward   Nhadie
• [asterisk-users] Problem with Call Forward   Philipp Kempgen
• [asterisk-users] Congestion in Outgoing call through PRI   Octavio Ruiz
• [asterisk-users] sip to sip unplanned conference! help!!   RoLaNd RoLaNd
• [asterisk-users] Inefficient Codec Translation   Kevin P. Fleming
• [asterisk-users] G722 and Asterisk 1.6   Steve Repo
• [asterisk-users] Faxing through Zap cards   Karl Fife
• [asterisk-users] G722 and Asterisk 1.6   Michael Graves
• [asterisk-users] PRI Splitter   Karl Fife
• [asterisk-users] Asterisk voicemail message order   Robert Lister
• [asterisk-users] PRI Splitter   Igor Hernandez
• [asterisk-users] Asterisk Crash   Josiah Bryan
• [asterisk-users] Congestion in Outgoing call through PRI   Richard Lyman
• [asterisk-users] Faxing through Zap cards   Luis Morales
• [asterisk-users] Ringing on Console after a page   Josiah Bryan
• [asterisk-users] G722 and Asterisk 1.6   Thomas Kenyon
• [asterisk-users] DID number   michel freiha
• [asterisk-users] Asterisk Crash   Andrew Latham
• [asterisk-users] G722 and Asterisk 1.6   Michael Graves
• [asterisk-users] Ringing on Console after a page   Barry L. Kline
• [asterisk-users] DID number   Igor Hernandez
• [asterisk-users] Asterisk with E1 interface vs IP PBX   Alejandro Facultad
• [asterisk-users] Asterisk Crash   Josiah Bryan
• [asterisk-users] Ringing on Console after a page   Josiah Bryan
• [asterisk-users] DID number   michel freiha
• [asterisk-users] DID number   michel freiha
• [asterisk-users] DID number   Igor Hernandez
• [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI   asteriskteam at digium.com
• [asterisk-users] Faxing through Zap cards   C F
• [asterisk-users] Combine sip audio and video from different sources   Artem Makhutov
• [asterisk-users] G722 and Asterisk 1.6   Thomas Kenyon
• [asterisk-users] Asterisk with E1 interface vs IP PBX   Luis Morales
• [asterisk-users] All calls want to go out only on interface ZAP/g0   Mark Best
• [asterisk-users] All calls want to go out only on interface ZAP/g0   Paul Hales
• [asterisk-users] All calls want to go out only on interface ZAP/g0   Paul Hales
• [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI   sean darcy
• [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI   Russell Bryant
• [asterisk-users] G722 and Asterisk 1.6   Russell Bryant
• [asterisk-users] Voicemail "from an unknown caller"   Andrew Joakimsen
• [asterisk-users] ASTERISK supported Video phone   Tharanga
• [asterisk-users] Z-Wave or Zigbee for Office or Home automation using XML Browser enabled Screen Phone   Karl Fife
• [asterisk-users] PRI Splitter   Craig Guy
• [asterisk-users] DID number   Jaswinder Singh
• [asterisk-users] Congestion in Outgoing call through PRI   Shariq Khan
• [asterisk-users] MixMonitor-Saving Recorded file with AgentId.   Syed Nasruddin
• [asterisk-users] Dial L( x [: y ][: z ]) option truncates colon (:) using AGI /_   selmak se
• [asterisk-users] ASTERISK supported Video phone   Gordon Henderson
• [asterisk-users] Asterisk Queue's   Tobias Ahlander
• [asterisk-users] Reliable wireless SIP phones   Thomas Kenyon
• [asterisk-users] Stability problems in Asterisk 1.4.18 (and other 1.4.xx versions)   z_gringo
• [asterisk-users] #include changes in 1.4   Chris Bagnall
• [asterisk-users] Faxing through Zap cards   z_gringo
• [asterisk-users] Installing ValetParking?   Tobias Ahlander
• [asterisk-users] How to check mailbox exists (Received SIP subscribe for peer without mailbox)   Olivier
• [asterisk-users] How to check mailbox exists (Received SIP subscribe for peer without mailbox)   Rob Hillis
• [asterisk-users] iLBC codec   Nhadie
• [asterisk-users] PRI Splitter   Igor Hernandez
• [asterisk-users] dial out via fxo gateway   ACL
• [asterisk-users] G722 and Asterisk 1.6   Steve Repo
• [asterisk-users] DID number   michel freiha
• [asterisk-users] dial out via fxo gateway   Steve Repo
• [asterisk-users] MixMonitor + Originate   Geraint Lee
• [asterisk-users] How to check mailbox exists (Received SIP subscribe for peer without mailbox)   Olivier
• [asterisk-users] New Install using DAHDI   Joseph L. Casale
• [asterisk-users] Z-Wave or Zigbee for Office or Home automation using XML Browser enabled Screen Phone   Michael Graves
• [asterisk-users] Logs: messages, events, queue   bilal ghayyad
• [asterisk-users] G722 and Asterisk 1.6   Russell Bryant
• [asterisk-users] Stability problems in Asterisk 1.4.18 (and other 1.4.xx versions)   Russell Bryant
• [asterisk-users] conf files for dahdi   sean darcy
• [asterisk-users] Dial L( x [: y ][: z ]) option truncates colon (:) using AGI /_   Tilghman Lesher
• [asterisk-users] #include changes in 1.4   Tilghman Lesher
• [asterisk-users] iLBC codec   Tilghman Lesher
• [asterisk-users] G722 and Asterisk 1.6   Peder at NetworkOblivion
• [asterisk-users] strange transfer problem   Rizwan Hisham
• [asterisk-users] How to check mailbox exists (Received SIP subscribe for peer without mailbox)   Rob Hillis
• [asterisk-users] ringback when the channel is answered   eng. Anatoli Marinov
• [asterisk-users] ringback when the channel is answered   Eric "ManxPower" Wieling
• [asterisk-users] G722 and Asterisk 1.6   Steve Repo
• [asterisk-users] ASTERISK supported Video phone   Steve Repo
• [asterisk-users] G722 and Asterisk 1.6   Michael Graves
• [asterisk-users] ringback when the channel is answered   eng. Anatoli Marinov
• [asterisk-users] ASTERISK supported Video phone   eng. Anatoli Marinov
• [asterisk-users] PRI Splitter   FailSafe Inc.
• [asterisk-users] extensions.conf programming?   Ken D'Ambrosio
• [asterisk-users] ringback when the channel is answered   Eric "ManxPower" Wieling
• [asterisk-users] G722 and Asterisk 1.6   Thomas Kenyon
• [asterisk-users] 1.6rc4 chan_iax2 messages   MFH
• [asterisk-users] extensions.conf programming?   Mark Michelson
• [asterisk-users] ASTERISK supported Video phone   Gordon Henderson
• [asterisk-users] extensions.conf programming?   Darren Sessions
• [asterisk-users] G722 and Asterisk 1.6   Michael Graves
• [asterisk-users] G722 and Asterisk 1.6   Gordon Henderson
• [asterisk-users] 1.6rc4 chan_iax2 messages   Tilghman Lesher
• [asterisk-users] 1.6rc4 chan_iax2 messages   MFH
• [asterisk-users] ringback when the channel is answered   eng. Anatoli Marinov
• [asterisk-users] DAHDI FAQ not up. Anyplace else?   sean darcy
• [asterisk-users] ringback when the channel is answered   Steve Totaro
• [asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination   Shaun Wingrin
• [asterisk-users] ringback when the channel is answered   eng. Anatoli Marinov
• [asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination   Anthony Francis
• [asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination   Terry Wilson
• [asterisk-users] ASTERISK supported Video phone   bilal ghayyad
• [asterisk-users] Asterisk supported Video phone   Philipp Kempgen
• [asterisk-users] DAHDI FAQ not up. Anyplace else?   Kevin P. Fleming
• [asterisk-users] Polycom BLF - multiple buddies   Robert McNaught
• [asterisk-users] DID number   Marcelo Freitas
• [asterisk-users] G722 and Asterisk 1.6   Steve Underwood
• [asterisk-users] DID number   Philipp Kempgen
• [asterisk-users] extensions.conf programming?   Lee, John (Sydney)
• [asterisk-users] Polycom BLF - multiple buddies   Lee, John (Sydney)
• [asterisk-users] iLBC codec   ronald
• [asterisk-users] The question about the M(X)option of Dial   larry
• [asterisk-users] libpri 1.4.5 priindication   Jorge Mendoza
• [asterisk-users] Asterisk supported Video phone   bilal ghayyad
• [asterisk-users] Linksys 3102 - Call Waiting   Joseph
• [asterisk-users] G722 and Asterisk 1.6   Darrick Hartman
• [asterisk-users] Polycom BLF - multiple buddies   James Sneeringer
• [asterisk-users] Polycom BLF - multiple buddies   Lee, John (Sydney)
• [asterisk-users] G722 and Asterisk 1.6   Michael Graves
• [asterisk-users] dahdi & tdm400p: no luck   sean darcy
• [asterisk-users] Polycom BLF - multiple buddies   James Sneeringer
• [asterisk-users] Polycom BLF - multiple buddies   James Sneeringer
• [asterisk-users] Polycom BLF - multiple buddies   Lee, John (Sydney)
• [asterisk-users] ASTERISK supported Video phone   eng. Anatoli Marinov
• [asterisk-users] dahdi & tdm400p: no luck   Tzafrir Cohen
• [asterisk-users] ringback when the channel is answered   eng. Anatoli Marinov
• [asterisk-users] The question about the M(X)option of Dial   Alex Balashov
• [asterisk-users] Gateway errors   voip crazy
• [asterisk-users] Bridge 2 incoming calls   Tim Panton
• [asterisk-users] Grandstream Video Phones & Asterisk..   Gordon Henderson
• [asterisk-users] Call-leg stays on MusicOnHold forever   Andreas Brodmann
• [asterisk-users] FW: Vivox SLim   Dean Collins
• [asterisk-users] dahdi & tdm400p: no luck   sean darcy
• [asterisk-users] G722 and Asterisk 1.6   Olivier
• [asterisk-users] Bridge 2 incoming calls   Andreas Brodmann
• [asterisk-users] dahdi & tdm400p: no luck   Tzafrir Cohen
• [asterisk-users] svn branches for dhadi and its tools   John covici
• [asterisk-users] Dear asterisk-users at lists.digium.com 79% OFF on Pfizer   VIAGRA ®
• [asterisk-users] Asterisk Crash   Tony Mountifield
• [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI   Tony Mountifield
• [asterisk-users] Bridge 2 incoming calls   Steve Murphy
• [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI   Steve Repo
• [asterisk-users] Dear asterisk-users at lists.digium.com 79% OFF on Pfizer   Joe Pukepail
• [asterisk-users] Bridge 2 incoming calls   Tim Panton
• [asterisk-users] Bridge 2 incoming calls   Tim Panton
• [asterisk-users] Polycom BLF - multiple buddies   Robert McNaught
• [asterisk-users] Transfers on AgentLogin()   Mark Hamilton
• [asterisk-users] Call-leg stays on MusicOnHold forever   Jorge Mendoza
• [asterisk-users] PRI Splitter   Craig Guy
• [asterisk-users] FAX over T1 Question   Amaru Netapshaak
• [asterisk-users] FAX over T1 Question   Bob Pierce
• [asterisk-users] FAX over T1 Question   Tim Nelson
• [asterisk-users] FAX over T1 Question   Steve Totaro
• [asterisk-users] Call monitor/barge/train   David Backeberg
• [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI   Kevin P. Fleming
• [asterisk-users] Call monitor/barge/train   Steve Totaro
• [asterisk-users] FAX over T1 Question   Eric "ManxPower" Wieling
• [asterisk-users] FAX over T1 Question   Steve Totaro
• [asterisk-users] FAX over T1 Question   Alex Balashov
• [asterisk-users] FAX over T1 Question   Vinícius Fontes
• [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI   Tzafrir Cohen
• [asterisk-users] svn branches for dhadi and its tools   Tzafrir Cohen
• [asterisk-users] (no subject)   Bill Andersen
• [asterisk-users] FAX over T1 Question   Amaru Netapshaak
• [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI   Tony Mountifield
• [asterisk-users] soft hangup (was: Re: (no subject))   Philipp Kempgen
• [asterisk-users] svn branches for dhadi and its tools   John covici
• [asterisk-users] FAX over T1 Question   Eric "ManxPower" Wieling
• [asterisk-users] (no subject)   Shariq Khan
• [asterisk-users] dahdi & tdm400p: no luck   sean darcy
• [asterisk-users] dahdi & tdm400p: no luck   Tzafrir Cohen
• [asterisk-users] dahdi & tdm400p: no luck   sean darcy
• [asterisk-users] dahdi & tdm400p: no luck   sean darcy
• [asterisk-users] Polycom BLF - multiple buddies   James Sneeringer
• [asterisk-users] Transfers on AgentLogin()   James Sneeringer
• [asterisk-users] Polycom BLF - multiple buddies   Matt Riddell
• [asterisk-users] Transfers on AgentLogin()   Mark Hamilton
• [asterisk-users] Transfers on AgentLogin()   Matt Riddell
• [asterisk-users] call-limit problem   Nhadie
• [asterisk-users] dahdi & tdm400p: no luck   Matt Gibson
• [asterisk-users] dahdi & tdm400p: no luck   sean darcy
• [asterisk-users] dahdi & tdm400p: no luck   Tzafrir Cohen
• [asterisk-users] dahdi & tdm400p: no luck   Matt Gibson
• [asterisk-users] [FreeBSD 6.3] Right-way to recover Zaptel?   Vincent
• [asterisk-users] dahdi & tdm400p: no luck   sean darcy
• [asterisk-users] realtime queue reload   Thomas Winter
• [asterisk-users] [FreeBSD 6.3] Right-way to recover Zaptel?   Anthony Francis
• [asterisk-users] realtime queue reload   Brian
• [asterisk-users] [FreeBSD 6.3] Right-way to recover Zaptel?   Vincent
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   hugolivude
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   Steve Totaro
• [asterisk-users] Transfers on AgentLogin()   Mark Hamilton
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   Gordon Henderson
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   Tim Panton
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   Steve Totaro
• [asterisk-users] Transfers on AgentLogin()   Doug Lytle
• [asterisk-users] error while trying to compile dahdi-tools-trunk   John covici
• [asterisk-users] realtime queue reload   Thomas Winter
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   Michiel van Baak
• [asterisk-users] Occassional garbled voicemail   Bruce Komito
• [asterisk-users] Occassional garbled voicemail   Steven Howes
• [asterisk-users] error while trying to compile dahdi-tools-trunk   Tzafrir Cohen
• [asterisk-users] error while trying to compile dahdi-tools-trunk   John covici
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   Artem Makhutov
• [asterisk-users] realtime queue reload   Atis Lezdins
• [asterisk-users] IAX vs SIP   Edgar Guadamuz
• [asterisk-users] iLBC and G729 codecs   Edgar Guadamuz
• [asterisk-users] lost DTMF   András DORN
• [asterisk-users] iLBC and G729 codecs   Christopher Dobbs
• [asterisk-users] iLBC and G729 codecs   Stelios Koroneos
• [asterisk-users] Streaming live music into a conference room   Leotis buchanan
• [asterisk-users] realtime queue reload   Thomas Winter
• [asterisk-users] CPU Usage 100% when Voicemail Notification is sent   Sip Support
• [asterisk-users] how to disallow the native bridge between the two channel   bala krishnan
• [asterisk-users] IAX vs SIP   Tim Panton
• [asterisk-users] Which kernel mISDN to choose   Julien Claassen
• [asterisk-users] Which kernel mISDN to choose   Julien Claassen
• [asterisk-users] mISDN or BRIstuff ...   Gordon Henderson
• [asterisk-users] mISDN or BRIstuff ...   Stefan Gofferje
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   Steve Totaro
• [asterisk-users] realtime queue reload   Atis Lezdins
• [asterisk-users] how to disallow the native bridge between the two channel   Atis Lezdins
• [asterisk-users] Help about the Rxfax on asterisk   Max Alex
• [asterisk-users] mISDN or BRIstuff ...   Olivier
• [asterisk-users] IAX2 was Re: Problems with 2 Asterisk servers on same LAN   Tim Panton
• [asterisk-users] Asterisk realtime MySQL clients from same IP problem   Patrick Maartense
• [asterisk-users] Video on Hold?   Gordon Henderson
• [asterisk-users] Help about the Rxfax on asterisk   Steve Davies
• [asterisk-users] Asterisk realtime MySQL clients from same IP problem   Philipp Kempgen
• [asterisk-users] Asterisk realtime MySQL clients from same IP problem   Jay R. Ashworth
• [asterisk-users] Asterisk realtime MySQL clients from same IP problem   Patrick Maartense
• [asterisk-users] Help about the Rxfax on asterisk   Philipp Kempgen
• [asterisk-users] FAX over T1 Question   Amaru Netapshaak
• [asterisk-users] FAX over T1 Question   Steve Totaro
• [asterisk-users] Help Astmanproxy - AutoFilter   Rodrigo Pinto
• [asterisk-users] [OT] Re: Asterisk realtime MySQL clients from same IP problem   Philipp Kempgen
• [asterisk-users] Originate   Krzysiek
• [asterisk-users] Asterisk realtime MySQL clients from same IP problem   Philipp Kempgen
• [asterisk-users] Help about the Rxfax on asterisk   Tzafrir Cohen
• [asterisk-users] FAX over T1 Question   Eric "ManxPower" Wieling
• [asterisk-users] How to read DTFMs from MEETME_AGI_BACKGROUND without blocking?   Cosmin Prund
• [asterisk-users] FAX over T1 Question   Jay R. Ashworth
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   Jay R. Ashworth
• [asterisk-users] Help Astmanproxy - AutoFilter   Steve Davies
• [asterisk-users] FAX over T1 Question   Jeremy Mann
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   Matthew Fredrickson
• [asterisk-users] [OT] Re: Asterisk realtime MySQL clients from same IP problem   Patrick Maartense
• [asterisk-users] fax detection without answer   JD
• [asterisk-users] Newbie questions: seting up extension for miSDN   Julien Claassen
• [asterisk-users] [OT] email netiquette (was: Re: Re: Asterisk realtime MySQL clients from same IP problem)   Philipp Kempgen
• [asterisk-users] fax detection without answer   Vinícius Fontes
• [asterisk-users] fax detection without answer   Steven Howes
• [asterisk-users] Pointers to replace astdb   Nguyen
• [asterisk-users] FAX over T1 Question   Amaru Netapshaak
• [asterisk-users] Multihomed Server Issues   Joseph L. Casale
• [asterisk-users] FAX over T1 Question   Amaru Netapshaak
• [asterisk-users] Pointers to replace astdb   Gordon Henderson
• [asterisk-users] fax detection without answer   C F
• [asterisk-users] Asterisk T38 and Dialogic DMG 2000   JR Richardson
• [asterisk-users] fax detection without answer   JD
• [asterisk-users] SIP Extension Config Issue   Joseph L. Casale
• [asterisk-users] SIP Extension Config Issue   Eric "ManxPower" Wieling
• [asterisk-users] Multihomed Server Issues   Tilghman Lesher
• [asterisk-users] SIP Extension Config Issue   Joseph L. Casale
• [asterisk-users] Auto Attendant help   Ron Hertz
• [asterisk-users] Asterisk realtime MySQL clients from same IP problem   Patrick Maartense
• [asterisk-users] Multihomed Server Issues   Joseph L. Casale
• [asterisk-users] Asterisk T38 and Dialogic DMG 2000   Raj Jain
• [asterisk-users] SIP Extension Config Issue   Doug Lytle
• [asterisk-users] SIP Extension Config Issue   Joseph L. Casale
• [asterisk-users] New Versions of Asterisk and DAHDI   Asterisk Development Team
• [asterisk-users] OT: ARI   Mark Hamilton
• [asterisk-users] Digium Hardware Echo Cancellation   Klaverstyn, David C
• [asterisk-users] OT: ARI   David Backeberg
• [asterisk-users] Pointers to replace astdb   Tilghman Lesher
• [asterisk-users] OT: ARI   Mark Hamilton
• [asterisk-users] Video on Hold?   Russell Bryant
• [asterisk-users] dahdi & tdm400p: no luck - filed BUG   sean darcy
• [asterisk-users] Video on Hold?   Russell Bryant
• [asterisk-users] OT: ARI   David Backeberg
• [asterisk-users] Bridge 2 incoming calls   Russell Bryant
• [asterisk-users] OT: ARI   Mark Hamilton
• [asterisk-users] OT: ARI   Paul Hales
• [asterisk-users] Pointers to replace astdb   Steve Edwards
• [asterisk-users] Aastra Phone Parts   mark at njycamps.org
• [asterisk-users] Help needed creating gateway   Rudolf Ladyzhenskii
• [asterisk-users] Help needed creating gateway   Paul Hales
• [asterisk-users] Pointers to replace astdb   Tilghman Lesher
• [asterisk-users] Dundi Help   technocrat voip
• [asterisk-users] Pointers to replace astdb   Nguyen
• [asterisk-users] Pointers to replace astdb   Nguyen
• [asterisk-users] cli-originate and features   Julien Claassen
• [asterisk-users] Session Progress   michel freiha
• [asterisk-users] OT: ARI   Mark Hamilton
• [asterisk-users] realtime queue reload   Thomas Winter
• [asterisk-users] Dear asterisk-users at lists.digium.com , Viagra Official Site 2008!   Doctor Jerrod Duffy
• [asterisk-users] realtime queue reload   Atis Lezdins
• [asterisk-users] Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)   Hiren Mistry
• [asterisk-users] Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)   Hiren Mistry
• [asterisk-users] Call-Limit on Asterisk Cluster   Nhadie
• [asterisk-users] Manager API -> call to agi   Chriss
• [asterisk-users] OT: ARI   Steve Totaro
• [asterisk-users] Asterisk and Network Monitoring   Jacobus van Niekerk
• [asterisk-users] how to disallow the native bridge between the two channel   Atis Lezdins
• [asterisk-users] Digium Hardware Echo Cancellation   Remi Quezada
• [asterisk-users] Asterisk and Network Monitoring   Michiel van Baak
• [asterisk-users] Does X-Lite 'remember' Congestion state? (halfway OT)   Paul Schewietzek
• [asterisk-users] Asterisk and Network Monitoring   Gareth Llewellyn
• [asterisk-users] Video on Hold?   k4rjj at bellsouth.net
• [asterisk-users] Asterisk and Network Monitoring   Dean Collins
• [asterisk-users] Asterisk and Network Monitoring   Martin Smith
• [asterisk-users] Asterisk and Network Monitoring   Darrick Hartman (lists)
• [asterisk-users] Asterisk and Network Monitoring   EdPimentl
• [asterisk-users] Does X-Lite 'remember' Congestion state? (halfway OT)   Kristian Kielhofner
• [asterisk-users] Asterisk and Network Monitoring   Jay R. Ashworth
• [asterisk-users] PRI auto-configure - continued from DEV list   Bill Michaelson
• [asterisk-users] AstriCon 2008 - Two Weeks To Go - Register Today   Steven Sokol
• [asterisk-users] OT: ARI   Mark Hamilton
• [asterisk-users] Asterisk - Operator switch billing   Sriram
• [asterisk-users] Asterisk - Operator switch billing   Josiah Bryan
• [asterisk-users] CLI and AGI question   Julien Claassen
• [asterisk-users] Asterisk - Operator switch billing   Sriram
• [asterisk-users] PRI auto-configure - continued from DEV list   Tzafrir Cohen
• [asterisk-users] Asterisk - Operator switch billing   Jay R. Ashworth
• [asterisk-users] Choppy Audio in One Direction   Kevin Ragsdale
• [asterisk-users] Asterisk and Network Monitoring   Tzafrir Cohen
• [asterisk-users] DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!   Ming Yong
• [asterisk-users] Asterisk and Network Monitoring   Lyle Giese
• [asterisk-users] iLBC and G729 codecs   Edgar Guadamuz
• [asterisk-users] SIP to IAX?   Mattias Andersson
• [asterisk-users] SIP to IAX?   Darren Sessions
• [asterisk-users] iLBC and G729 codecs   Guillermo V. Salas
• [asterisk-users] DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!   Tilghman Lesher
• [asterisk-users] Asterisk 1.4.22-rc5, 1.6.0-rc6 and Zaptel 1.4.12.1 Released   Asterisk Development Team
• [asterisk-users] DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!   Dean Collins
• [asterisk-users] Asterisk REFER   Al lists
• [asterisk-users] DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!   Tilghman Lesher
• [asterisk-users] Pressing 0 to get an external line   Christian Victor
• [asterisk-users] DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!   Charles Alvis
• [asterisk-users] Asterisk phone conferencing performance   George Williams
• [asterisk-users] Asterisk phone conferencing performance   Darren Sessions
• [asterisk-users] DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!   k4rjj at bellsouth.net
• [asterisk-users] nagios check_sip plugin   Robert McNaught
• [asterisk-users] Asterisk and Network Monitoring   Paul Hales
• [asterisk-users] Asterisk and Network Monitoring   Paul Hales
• [asterisk-users] Pressing 0 to get an external line   Lee, John (Sydney)
• [asterisk-users] Pressing 0 to get an external line   Marco Mouta
• [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)   Steve Finkelstein
• [asterisk-users] Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)   Max Alex
• [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)   Alex Balashov
• [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)   David Quinton
• [asterisk-users] how to disallow the native bridge between the two channel   bala krishnan
• [asterisk-users] Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)   Patrick
• [asterisk-users] Newbie AEL2: Syntax for Hint   Lee, John (Sydney)
• [asterisk-users] Dundi Help   Giorgio Incantalupo
• [asterisk-users] Dundi Help   technocrat voip
• [asterisk-users] How to make outgoing call from Pri ?   Hiren Mistry
• [asterisk-users] Resilience using DNS or phone feature ?   Olivier
• [asterisk-users] Dundi Help   Giorgio Incantalupo
• [asterisk-users] how to disallow the native bridge between the two channel   Atis Lezdins
• [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)   henry at henrythebig.com
• [asterisk-users] Asterisk and Network Monitoring   Faraz Khan
• [asterisk-users] realtime queue reload   Thomas Winter
• [asterisk-users] Newbie AEL2: Syntax for Hint   Steve Murphy
• [asterisk-users] Bell Canada (Nortel DMS100) PRI Outbound CNAM issue   Iain McBride
• [asterisk-users] iLBC and G729 codecs   Thomas Kenyon
• [asterisk-users] Asterisk phone conferencing performance   George Williams
• [asterisk-users] Bell Canada (Nortel DMS100) PRI Outbound CNAM issue   Jonn R Taylor
• [asterisk-users] Bell Canada (Nortel DMS100) PRI Outbound CNAM issue   Jay R. Ashworth
• [asterisk-users] Write Asterisk CDR MySQL records to multiple servers   Ricardo Melendez
• [asterisk-users] Write Asterisk CDR MySQL records to multiple servers   Vahan Yerkanian
• [asterisk-users] Write Asterisk CDR MySQL records to multiple servers   Tilghman Lesher
• [asterisk-users] Write Asterisk CDR MySQL records to multipleservers   Watkins, Bradley
• [asterisk-users] How to make outgoing call from Pri ?   C F
• [asterisk-users] Is there a way to get the Call-ID into the CDR?   Eric Chamberlain
• [asterisk-users] Resilience using DNS or phone feature ?   CunningPike
• [asterisk-users] Zero time (nearly) call transfer.   Karl Fife
• [asterisk-users] Newbie AEL2: Syntax for Hint   Lee, John (Sydney)
• [asterisk-users] Write Asterisk CDR MySQL records to multiple servers   Eric Chamberlain
• [asterisk-users] distinguish trunk from same host   Nhadie
• [asterisk-users] Write Asterisk CDR MySQL records to multiple servers   Tilghman Lesher
• [asterisk-users] Write Asterisk CDR MySQL records to multiple servers   Andres
• [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)   Chris Brentano
• [asterisk-users] Newbie AEL2: Syntax for Hint   Lee, John (Sydney)
• [asterisk-users] Language for app_queue, chan_local, chan_agent or whatever?   Benjamin Meyer
• [asterisk-users] SHELL function strangeness   Julien Claassen
• [asterisk-users] Newbie AEL2: Syntax for Hint   Eric Dantie
• [asterisk-users] Resilience using DNS or phone feature ?   Olivier
• [asterisk-users] ASTERISK supported Video phone   Artem Makhutov
• [asterisk-users] meetme without zaptel   Pezhman Lali
• [asterisk-users] Newbie AEL2: Syntax for Hint   Lee, John (Sydney)
• [asterisk-users] BLF call pickup on Linksys SPA932   Chris Bagnall
• [asterisk-users] meetme without zaptel   Quan Nguyen, (NCS)
• [asterisk-users] SIP to IAX?   Chris Bagnall
• [asterisk-users] SIP to IAX?   John Millican
• [asterisk-users] meetme without zaptel   Pezhman Lali
• [asterisk-users] meetme without zaptel   Tzafrir Cohen
• [asterisk-users] BLF call pickup on Linksys SPA932   Stefan Schmidt
• [asterisk-users] Newbie AEL2: Syntax for Hint   Dave Fullerton
• [asterisk-users] redirection of called   ameur brahim
• [asterisk-users] Sarfaraz has invited you to join iDeezire - Keeps Connected!   Sarfaraz
• [asterisk-users] Outside SIP Caller accessing voivemail   Joseph L. Casale
• [asterisk-users] Pressing 0 to get an external line   Grygoriy Dobrovolskyy
• [asterisk-users] IVR response of the pound key   sipxuser sipx
• [asterisk-users] Outside SIP Caller accessing voivemail   Steven Howes
• [asterisk-users] SHELL function strangeness   Tilghman Lesher
• [asterisk-users] Video on Hold?   Russell Bryant
• [asterisk-users] asterisk 1.6.0rc6 make menuselect failed.   Thomas Kenyon
• [asterisk-users] [Re: Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)]   Hiren Mistry
• [asterisk-users] asterisk 1.6.0rc6 make menuselect failed.   Alex Balashov
• [asterisk-users] BLF call pickup on Linksys SPA932   Steve Davies
• [asterisk-users] g729 passthrough   Yacine Boukaba
• [asterisk-users] dahdi vs zap with latest version of asterisk -- having some problems   John covici
• [asterisk-users] SHELL function strangeness   Julien Claassen
• [asterisk-users] Asterisk calleri id resolution   Todd Reese
• [asterisk-users] SIP to IAX?   Tim Panton
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   Jay R. Ashworth
• [asterisk-users] dahdi vs zap with latest version of asterisk -- having some problems   Matt Gibson
• [asterisk-users] SHELL function strangeness   Steve Edwards
• [asterisk-users] BLF call pickup on Linksys SPA932   Stefan Schmidt
• [asterisk-users] dahdi vs zap with latest version of asterisk -- having some problems   John covici
• [asterisk-users] iLBC and G729 codecs   Edgar Guadamuz
• [asterisk-users] iLBC and G729 codecs   Edgar Guadamuz
• [asterisk-users] Video on Hold?   Gordon Henderson
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   Jay R. Ashworth
• [asterisk-users] dahdi vs zap with latest version of asterisk -- having some problems   Matt Gibson
• [asterisk-users] SHELL function strangeness   Julien Claassen
• [asterisk-users] about application Jack and its runtime   Julien Claassen
• [asterisk-users] dahdi vs zap with latest version of asterisk -- having some problems   Tzafrir Cohen
• [asterisk-users] Probably very simple... call a number and play a sound?   Mike Johnson
• [asterisk-users] Asterisk and CallerID Resolution   Todd Reese
• [asterisk-users] Unable to run make menuselect for asterisk-addons   Jonn R Taylor
• [asterisk-users] Unable to run make menuselect for asterisk-addons   Jonn R Taylor
• [asterisk-users] Unable to run make menuselect for asterisk-addons   Sean Dennis
• [asterisk-users] distinguish trunk from same host   bkruse
• [asterisk-users] Write Asterisk CDR MySQL records to multiple servers   Eric Chamberlain
• [asterisk-users] SHELL function strangeness   Steve Edwards
• [asterisk-users] Write Asterisk CDR MySQL records to multiple servers   Tilghman Lesher
• [asterisk-users] Unable to run make menuselect for asterisk-addons   Tzafrir Cohen
• [asterisk-users] Probably very simple... call a number and play a sound?   C. Chad Wallace
• [asterisk-users] Unable to run make menuselect for asterisk-addons   Sean Bright
• [asterisk-users] Unable to run make menuselect for asterisk-addons   Sean Bright
• [asterisk-users] Possible Packet loss but need an opinion   Andrew Matthews
• [asterisk-users] SIP to IAX?   C. Chad Wallace
• [asterisk-users] about application Jack and its runtime   Russell Bryant
• [asterisk-users] BLF call pickup on Linksys SPA932   Paul Hales
• [asterisk-users] distinguish trunk from same host   Nhadie
• [asterisk-users] asterisk 1.6.0rc6 make menuselect failed.   Sean Bright
• [asterisk-users] Outside SIP Caller accessing voivemail   Joseph L. Casale
• [asterisk-users] Asterisk and Fedora 9   Pascal Bruno
• [asterisk-users] Asterisk and Fedora 9   Anthony Messina
• [asterisk-users] BLF call pickup on Linksys SPA932   Klaverstyn, David C
• [asterisk-users] Asterisk and Fedora 9   Pascal Bruno
• [asterisk-users] Asterisk and Fedora 9   Pascal Bruno
• [asterisk-users] Asterisk and Fedora 9   Anthony Messina
• [asterisk-users] how to pass a variable in extensions.conf to AGI file   Zen Kato
• [asterisk-users] SIP to IAX?   Kristian Kielhofner
• [asterisk-users] SIP to IAX?   Kristian Kielhofner
• [asterisk-users] VoIP Users Conference today at 12 Noon EDT   randulo
• [asterisk-users] about application Jack and its runtime   Julien Claassen
• [asterisk-users] about application Jack and its runtime   Julien Claassen
• [asterisk-users] about application Jack and its runtime   Julien Claassen
• [asterisk-users] Amazing "show uptime"   Stephen Davies
• [asterisk-users] Amazing "show uptime"   Michiel van Baak
• [asterisk-users] Amazing "show uptime"   Tim Panton
• [asterisk-users] BLF call pickup on Linksys SPA932   Steve Davies
• [asterisk-users] Outside SIP Caller accessing voivemail   Doug Lytle
• [asterisk-users] Amazing "show uptime"   Tzafrir Cohen
• [asterisk-users] Video on Hold?   Atis Lezdins
• [asterisk-users] Amazing "show uptime"   Steven Howes
• [asterisk-users] Amazing "show uptime"   randulo
• [asterisk-users] Executing dialplan after the call normaly ended   Gergo Csibra
• [asterisk-users] Executing dialplan after the call normaly ended   Atis Lezdins
• [asterisk-users] Executing dialplan after the call normaly ended   Steve Totaro
• [asterisk-users] show g729 seems to no longer work in latest 1.4 version. What do I use please?   Shaun Wingrin
• [asterisk-users] asterisk 1.6.0rc6 make menuselect failed.   Thomas Kenyon
• [asterisk-users] Dial function, and no telephone line fixed in the fxo port   bilal ghayyad
• [asterisk-users] Amazing "show uptime"   Stephen Davies
• [asterisk-users] Asterisk and Fedora 9   MFH
• [asterisk-users] Amazing "show uptime"   Doug Lytle
• [asterisk-users] Amazing "show uptime"   Christopher Hoff
• [asterisk-users] Amazing "show uptime"   randulo
• [asterisk-users] Amazing "show uptime"   Steve Totaro
• [asterisk-users] about application Jack and its runtime   Jay R. Ashworth
• [asterisk-users] about application Jack and its runtime   Russell Bryant
• [asterisk-users] PRI auto-configure - continued from DEV list   Bill Michaelson
• [asterisk-users] about application Jack and its runtime   Russell Bryant
• [asterisk-users] Asterisk and Fedora 9   Pascal Bruno
• [asterisk-users] about application Jack and its runtime   Jay R. Ashworth
• [asterisk-users] Extension not found   michel freiha
• [asterisk-users] Read one or X DTMF   Ruddy Gbaguidi
• [asterisk-users] asterisk 16 and zapata   hh174
• [asterisk-users] PRI auto-configure - continued from DEV list   Tzafrir Cohen
• [asterisk-users] asterisk 16 and zapata   Steve Totaro
• [asterisk-users] asterisk 16 and zapata   Steve Totaro
• [asterisk-users] Amazing "show uptime"   Stephen Davies
• [asterisk-users] about application Jack and its runtime   Julien Claassen
• [asterisk-users] SCCP port numbers used for audio stram?   OCG Technical Support
• [asterisk-users] Amazing "show uptime"   Doug Lytle
• [asterisk-users] Which internet phone protocol best to choose   Julien Claassen
• [asterisk-users] asterisk 1.6.0rc6 make menuselect failed.   Sean Bright
• [asterisk-users] SCCP port numbers used for audio stram?   Kristian Kielhofner
• [asterisk-users] SCCP port numbers used for audio stram?   Eric "ManxPower" Wieling
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   Matthew Fredrickson
• [asterisk-users] Read one or X DTMF   Karsten Wemheuer
• [asterisk-users] Read one or X DTMF   Tony Mountifield
• [asterisk-users] SCCP - max lines per phone limit   OCG Technical Support
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   Jay R. Ashworth
• [asterisk-users] Encrypted IP phone compatible with Asterisk   Alejandro Cabrera Obed
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   Matthew Fredrickson
• [asterisk-users] Setup speed dials on Cisco 7921   OCG Technical Support
• [asterisk-users] Read one or X DTMF   Ruddy Gbaguidi
• [asterisk-users] Read one or X DTMF   Ruddy Gbaguidi
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   Jay R. Ashworth
• [asterisk-users] SCCP - max lines per phone limit   Michiel van Baak
• [asterisk-users] Extension not found   Karsten Wemheuer
• [asterisk-users] Setup speed dials on Cisco 7921   Michiel van Baak
• [asterisk-users] echo cancellation problem with dahdi   John covici
• [asterisk-users] Amazing "show uptime"   Justin Coffi
• [asterisk-users] Read one or X DTMF   Karsten Wemheuer
• [asterisk-users] Setup speed dials on Cisco 7921   OCG Technical Support
• [asterisk-users] Transfer via AMI   Nicholas Blasgen
• [asterisk-users] Setup speed dials on Cisco 7921   Michiel van Baak
• [asterisk-users] SIp Signalling   Il Neofita
• [asterisk-users] Asterisk and Fedora 9   Jonn R Taylor
• [asterisk-users] asterisk 16 and zapata   Jonn R Taylor
• [asterisk-users] Asterisk and Fedora 9   Benny Amorsen
• [asterisk-users] asterisk 16 and zapata   Jonn R Taylor
• [asterisk-users] Asterisk and Fedora 9   Nicholas Blasgen
• [asterisk-users] Asterisk and Fedora 9   Pascal Bruno
• [asterisk-users] Asterisk and Fedora 9   MFH
• [asterisk-users] OpenStage20 Problem   Stefan Tichy
• [asterisk-users] SIp Signalling   Alex Balashov
• [asterisk-users] Read one or X DTMF   Tony Mountifield
• [asterisk-users] [FreeBSD 6.3/Ports] Make does nothing   Vincent
• [asterisk-users] Which internet phone protocol best to choose   Stefan Gofferje
• [asterisk-users] Which internet phone protocol best to choose   Alex Balashov
• [asterisk-users] Which internet phone protocol best to choose   OCG Technical Support
• [asterisk-users] Which internet phone protocol best to choose   Julien Claassen
• [asterisk-users] Which internet phone protocol best to choose   Tilghman Lesher
• [asterisk-users] Which internet phone protocol best to choose   Stefan Gofferje
• [asterisk-users] Which internet phone protocol best to choose   Stefan Gofferje
• [asterisk-users] Sip Info events   robb
• [asterisk-users] cdr_adaptive_odbc writing CDR before h extension is processed   Eric Chamberlain
• [asterisk-users] Which internet phone protocol best to choose   Julien Claassen
• [asterisk-users] Which internet phone protocol best to choose   Alex Balashov
• [asterisk-users] Which internet phone protocol best to choose   Steve Totaro
• [asterisk-users] Which internet phone protocol best to choose   SIP
• [asterisk-users] Read one or X DTMF   Ruddy Gbaguidi
• [asterisk-users] Append String to CIDNAME   Sip Support
• [asterisk-users] Read one or X DTMF   Ruddy Gbaguidi
• [asterisk-users] Is there a way to get the remote User-Agent info from an outbound leg?   Eric Chamberlain
• [asterisk-users] Can the outbound SIP leg Call-ID be set to match the inbound SIP leg Call-ID?   Eric Chamberlain
• [asterisk-users] cdr_adaptive_odbc writing CDR before h extension is processed   Tilghman Lesher
• [asterisk-users] Amazing "show uptime"   John Faubion
• [asterisk-users] IVR response of the pound key   John Faubion
• [asterisk-users] Read one or X DTMF   Karsten Wemheuer
• [asterisk-users] Which internet phone protocol best to choose   Tzafrir Cohen
• [asterisk-users] Which internet phone protocol best to choose   Stefan Gofferje
• [asterisk-users] Which internet phone protocol best to choose   Julien Claassen
• [asterisk-users] meetme without zaptel   Philipp Kempgen
• [asterisk-users] Asterisk and Fedora 9   Philipp Kempgen
• [asterisk-users] app_conference   Pezhman Lali
• [asterisk-users] Which internet phone protocol best to, choose   Bill Michaelson
• [asterisk-users] Append String to CIDNAME   Sean Bright
• [asterisk-users] about application Jack and its runtime   Russell Bryant
• [asterisk-users] Newber   Dean Collins
• [asterisk-users] how to monitor traffic against specific Calling ID   Masood Ahmed
• [asterisk-users] dahdi & tdm400p: no luck - filed BUG   sean darcy
• [asterisk-users] Append String to CIDNAME   Sip Support
• [asterisk-users] rc6: can't get a dialtone   sean darcy
• [asterisk-users] Which internet phone protocol best to choose   Alex Balashov
• [asterisk-users] how to pass a variable in extensions.conf to AGI file   Steve Edwards
• [asterisk-users] Help...Failed to initialize G.729 copy protection!   Shaun Wingrin
• [asterisk-users] how to pass a variable in extensions.conf to AGI file   Steve Totaro
• [asterisk-users] Outside SIP Caller accessing voivemail   Joseph L. Casale
• [asterisk-users] about application Jack and its runtime   Julien Claassen
• [asterisk-users] Getting realtime ASR and ACD from Asterisk   Shaun Wingrin
• [asterisk-users] Queue Calls getting stuck in there   Tariq ..
• [asterisk-users] What if some phone picks up   Julien Claassen
• [asterisk-users] cdr_adaptive_odbc writing CDR before h extension is processed   Eric Chamberlain
• [asterisk-users] rc6: can't get a dialtone   Quan Nguyen, (NCS)
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Eric Chamberlain
• [asterisk-users] What if some phone picks up   Alex Balashov
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Alex Balashov
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Eric Chamberlain
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Igor Hernandez
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Tilghman Lesher
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Alex Balashov
• [asterisk-users] rc6: can't get a dialtone   sean darcy
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Tilghman Lesher
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Alex Balashov
• [asterisk-users] how to pass a variable in extensions.conf to AGI file   Zen Kato
• [asterisk-users] Streaming MoH on 1.4   Mark Hamilton
• [asterisk-users] Queue Calls getting stuck in there   Mark Hamilton
• [asterisk-users] rc6: can't get a dialtone   Tzafrir Cohen
• [asterisk-users] asterisk 16 and zapata   hatem moiz
• [asterisk-users] What if some phone picks up   Phil Reynolds
• [asterisk-users] Read dublicate dtmf   Il Neofita
• [asterisk-users] Read dublicate dtmf   Steve Totaro
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Eric Chamberlain
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Tilghman Lesher
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Tilghman Lesher
• [asterisk-users] cdr_adaptive_odbc writing CDR before h extension is processed   Eric Chamberlain
• [asterisk-users] Read dublicate dtmf   Il Neofita
• [asterisk-users] rc6: can't get a dialtone   sean darcy
• [asterisk-users] [FreeBSD 6.3/Ports] Make does nothing   Vincent
• [asterisk-users] MoH with an Aastra 9112i   Trevor Peirce
• [asterisk-users] MoH with an Aastra 9112i   Alex Balashov
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Alex Balashov
• [asterisk-users] IAX vs SIP   Edgar Guadamuz
• [asterisk-users] Is there a way to get the remote User-Agent info from an outbound leg?   Eric Chamberlain
• [asterisk-users] Newber   Eric Chamberlain
• [asterisk-users] Is there a way to get the remote User-Agent info from an outbound leg?   Alex Balashov
• [asterisk-users] Queue Calls getting stuck in there   Tariq ..
• [asterisk-users] MGCP Configuration <=> ADIT 600 <=> T1 Port   Barton Fisher
• [asterisk-users] Newber   Dean Collins
• [asterisk-users] Newber   Alex Balashov
• [asterisk-users] Streaming MoH on 1.4   Mark Hamilton
• [asterisk-users] Streaming MoH on 1.4   Klaverstyn, David C
• [asterisk-users] Degum Hardware Echo Cancellation   Klaverstyn, David C
• [asterisk-users] Streaming MoH on 1.4   Klaverstyn, David C
• [asterisk-users] Streaming MoH on 1.4   Klaverstyn, David C
• [asterisk-users] asterisk-users Digest, Vol 50, Issue 37   gianrico.fichera
• [asterisk-users] Streaming MoH on 1.4   Mark Hamilton
• [asterisk-users] Streaming MoH on 1.4   Mark Hamilton
• [asterisk-users] Streaming MoH on 1.4   Klaverstyn, David C
• [asterisk-users] Asterisk and Fedora 9   Pascal Bruno
• [asterisk-users] CallerID Resolution   Todd Reese
• [asterisk-users] Streaming MoH on 1.4   Mark Hamilton
• [asterisk-users] Streaming MoH on 1.4   Paul Hales
• [asterisk-users] asterisk-users Digest, Vol 50, Issue 38   gianrico.fichera
• [asterisk-users] Streaming MoH on 1.4   Mark Hamilton
• [asterisk-users] Asterisk and Fedora 9   Steve Totaro
• [asterisk-users] Asterisk and Fedora 9   Alex Balashov
• [asterisk-users] Asterisk and Fedora 9   Pascal Bruno
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   logan
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   Nhadie
• [asterisk-users] CallerID Resolution   OCG Technical Support
• [asterisk-users] Newber   Ming Yong
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   logan
• [asterisk-users] Amazing "show uptime"   Stephen Davies
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   Nhadie
• [asterisk-users] Asterisk REFER   Patrick Maartense
• [asterisk-users] [OT] email netiquette (was: Re: Re: Asterisk realtime MySQL clients from same IP problem)   Patrick Maartense
• [asterisk-users] call files hacking...   Gergo Csibra
• [asterisk-users] Streaming MoH on 1.4   Mark Hamilton
• [asterisk-users] Asterisk   michel freiha
• [asterisk-users] UK call initiating party hangup control on analog home lines   roberts at latnet.lv
• [asterisk-users] Asterisk   Quan Nguyen, (NCS)
• [asterisk-users] PBX appliances   Femi
• [asterisk-users] Asterisk   michel freiha
• [asterisk-users] UK call initiating party hangup control on analoghome lines   Don Kelly
• [asterisk-users] Callcenter monitoring tool   voip crazy
• [asterisk-users] cdr_adaptive_odbc writing CDR before h extension is processed   Steve Murphy
• [asterisk-users] Callcenter monitoring tool   Steve Totaro
• [asterisk-users] PBX appliances   Dean Collins
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Bill Andersen
• [asterisk-users] Is there a way to get the remote User-Agent info from an outbound leg?   Tilghman Lesher
• [asterisk-users] call files hacking...   Steve Edwards
• [asterisk-users] Callcenter monitoring tool   Jay R. Ashworth
• [asterisk-users] [FreeBSD] Right way to upgrade Zaptel from ports?   Vincent
• [asterisk-users] FW: open source PBX survey   Dean Collins
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   Alex Balashov
• [asterisk-users] cdr_adaptive_odbc writing CDR before h extension is processed   Eric Chamberlain
• [asterisk-users] Is there a way to get the remote User-Agent info from an outbound leg?   Eric Chamberlain
• [asterisk-users] Can someone give a plain english explanation of the HASH function?   MAGUIRE MATTHEW
• [asterisk-users] Need help with PHP script to authenticate user from database   Zeeshan Zakaria
• [asterisk-users] What characters can be present in SIP dial string passwords?   Eric Chamberlain
• [asterisk-users] Need help with PHP script to authenticate user from database   Steve Edwards
• [asterisk-users] Looking for an Asterisk consultant recommendation - LONDON, ON, CANADA   Alvin Austin
• [asterisk-users] Is there a way to get the remote User-Agent info from an outbound leg?   Tilghman Lesher
• [asterisk-users] PBX appliances   Stelios Koroneos
• [asterisk-users] RTCP-XR   Mike Hammett
• [asterisk-users] [FreeBSD] Right way to upgrade Zaptel from ports?   Vincent
• [asterisk-users] rc6: Dunno what to do with STUN message 0101 ??   sean darcy
• [asterisk-users] PBX appliances   Paul Hales
• [asterisk-users] PBX appliances   Shidan
• [asterisk-users] RTCP-XR   John Todd
• [asterisk-users] Context always defaults to PSTN   Mark Best
• [asterisk-users] dundi   hugolivude
• [asterisk-users] dundi   hugolivude
• [asterisk-users] Problems with 2 Asterisk servers on same LAN   hugolivude
• [asterisk-users] Context always defaults to PSTN   Paul Hales
• [asterisk-users] Callcenter monitoring tool   Al Baker
• [asterisk-users] dundi   Michiel van Baak
• [asterisk-users] Callcenter monitoring tool   Alex Balashov
• [asterisk-users] PBX appliances   Femi
• [asterisk-users] call files hacking...   Ian FREISLICH
• [asterisk-users] Callcenter monitoring tool   Tony Mountifield
• [asterisk-users] Redirecting SIP RTP with one Asterisk behind a NAT. The other is on Public IP   Shaun Wingrin
• [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle   Shaun Wingrin
• [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle   Julien Claassen
• [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle   Gordon Henderson
• [asterisk-users] how to force Asterisk 1.4 to use soxmix   Giorgio Incantalupo
• [asterisk-users] Voicemail: Thunderbird extension to play wav file in attachment?   Klaus Darilion
• [asterisk-users] One Week Until AstriCon 2008   Steven Sokol
• [asterisk-users] Snom phones and P-asserted   Loic Didelot
• [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle   Al Baker
• [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle   Eric "ManxPower" Wieling
• [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle   Al Baker
• [asterisk-users] how to force Asterisk 1.4 to use soxmix   Julien Claassen
• [asterisk-users] What is worng with that include in contrast to the example   Julien Claassen
• [asterisk-users] Callcenter monitoring tool   Senad Jordanovic
• [asterisk-users] What is worng with that include in contrast to the example   Doug Lytle
• [asterisk-users] Callcenter monitoring tool   Matt Florell
• [asterisk-users] What is worng with that include in contrast to the example   Julien Claassen
• [asterisk-users] RTCP-XR   Olivier
• [asterisk-users] Streaming MoH on 1.4   Olivier
• [asterisk-users] Streaming MoH on 1.4   SIP
• [asterisk-users] addons will not load when compiling Asterisk with DEBUG_THREADS   Andres
• [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle   SIP
• [asterisk-users] addons will not load when compiling Asterisk with DEBUG_THREADS   Mark Michelson
• [asterisk-users] Cisco + Asterisk   Guilherme Loch Waltrick Góes
• [asterisk-users] Streaming MoH on 1.4   Patrick Maartense
• [asterisk-users] addons will not load when compiling Asterisk with DEBUG_THREADS   Andres
• [asterisk-users] Parked Calls   Jeremy Mann
• [asterisk-users] iax reject using domain name   Nhadie
• [asterisk-users] Parked Calls   Doug Lytle
• [asterisk-users] Cisco + Asterisk   Yehavi Bourvine
• [asterisk-users] Parked Calls   Jeremy Mann
• [asterisk-users] Cisco + Asterisk   Darren Sessions
• [asterisk-users] UK call initiating party hangup control on analoghome lines   Roberts Klotins
• [asterisk-users] Parked Calls   Doug Lytle
• [asterisk-users] Cisco + Asterisk   David Backeberg
• [asterisk-users] Cisco + Asterisk   Guilherme Loch Waltrick Góes
• [asterisk-users] Parked Calls   Jeremy Mann
• [asterisk-users] Snom phones and P-asserted   Christian Stredicke
• [asterisk-users] Parked Calls   Doug Lytle
• [asterisk-users] UK call initiating party hangup control on analoghome lines   Gordon Henderson
• [asterisk-users] UK call initiating party hangup control on analoghome lines   Don Kelly
• [asterisk-users] UK call initiating party hangup control on analoghome lines   Roberts Klotins
• [asterisk-users] UK call initiating party hangup control on analoghome lines   Roberts Klotins
• [asterisk-users] Cellroute setup with asterisk   Roberts Klotins
• [asterisk-users] Voicemail: Thunderbird extension to play wav file in attachment?   Roderick A. Anderson
• [asterisk-users] PBX appliances   Paul Hales
• [asterisk-users] Cisco + Asterisk   Paul Hales
• [asterisk-users] Cisco + Asterisk   Alex Balashov
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] UK call initiating party hangup control on analoghome lines   Gordon Henderson
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   logan
• [asterisk-users] PBX appliances   Femi
• [asterisk-users] Cellroute setup with asterisk   Roberts Klotins
• [asterisk-users] how to force Asterisk 1.4 to use soxmix   Giorgio Incantalupo
• [asterisk-users] realtime queue asterisk 1.6.0-beta5   Ralf Träskman
• [asterisk-users] Asterisk 1.6.0-beta5 voicemail problem   Ralf Träskman
• [asterisk-users] realtime queue asterisk 1.6.0-beta5   Atis Lezdins
• [asterisk-users] SIP URI Forwarding   Rizwan Hisham
• [asterisk-users] dtmf passthru   Rizwan Hisham
• [asterisk-users] dtmf passthru   Benjamin Jacob
• [asterisk-users] dtmf passthru   Benjamin Jacob
• [asterisk-users] SIP URI Forwarding   Alex Balashov
• [asterisk-users] dtmf passthru   Alex Balashov
• [asterisk-users] codec of channels   Pezhman Lali
• [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use   hugolivude
• [asterisk-users] codec of channels-solved   Pezhman Lali
• [asterisk-users] Help with MFC/R2   Moises Silva
• [asterisk-users] chan_iax2.c: No more space   Philipp Kempgen
• [asterisk-users] Help with MFC/R2   Luis Morales
• [asterisk-users] dtmf passthru   Rizwan Hisham
• [asterisk-users] SIP URI Forwarding   Rizwan Hisham
• [asterisk-users] How to remove dialtone from DISA?   Zeeshan Zakaria
• [asterisk-users] DTMF detection problem on DISA   Zeeshan Zakaria
• [asterisk-users] DTMF detection problem on DISA   Tilghman Lesher
• [asterisk-users] DTMF detection problem on DISA   Zeeshan Zakaria
• [asterisk-users] how to force Asterisk 1.4 to use soxmix   Jorge Nunes
• [asterisk-users] Part of some calls does not get recorded   Jorge Nunes
• [asterisk-users] Format ulaw|h ?   R. Raja
• [asterisk-users] Restrict SIP registration to one ip address only?   Remco Barendse
• [asterisk-users] DTMF detection problem on DISA   Tilghman Lesher
• [asterisk-users] Digium training course   Pascal Bruno
• [asterisk-users] Restrict SIP registration to one ip address only?   Mr Shunz
• [asterisk-users] codec of channels   Yavuz YILDIRIM
• [asterisk-users] Restrict SIP registration to one ip address only?   Jared Smith
• [asterisk-users] Parked Calls   Jeremy Mann
• [asterisk-users] Parked Calls   Jeremy Mann
• [asterisk-users] app_confrence with loud voices   Pezhman Lali
• [asterisk-users] app_confrence with loud voices   Anthony Francis
• [asterisk-users] app_confrence with loud voices   Pezhman Lali
• [asterisk-users] chan_iax2.c: No more space   Tim Panton
• [asterisk-users] app_confrence with loud voices   Alex Balashov
• [asterisk-users] Restrict SIP registration to one ip address only?   JD
• [asterisk-users] Understanding of SIP Info Messages   robb
• [asterisk-users] chan_iax2.c: No more space   Philipp Kempgen
• [asterisk-users] chan_iax2.c: No more space   Philipp Kempgen
• [asterisk-users] strategy for measuring conference audio delay   George Williams
• [asterisk-users] strategy for measuring conference audio delay   Steve Edwards
• [asterisk-users] Speech recognition on simultaneous SIP / PSTN calls   Allann Jones
• [asterisk-users] PBX appliances   Paul Hales
• [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use   hugolivude
• [asterisk-users] Digium training course   Steve Totaro
• [asterisk-users] Digium training course   Pascal Bruno
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] Help with MFC/R2   Moises Silva
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] how to detect pickup...   Gergo Csibra
• [asterisk-users] how to detect pickup...   Chris Maciejewski
• [asterisk-users] Digium training course   Steve Totaro
• [asterisk-users] 482 Loop Detected   remi.druilhe at orange-ftgroup.com
• [asterisk-users] Sip Info events   Grey Man
• [asterisk-users] Get rid of "Really destroying SIP dialog"   Olivier
• [asterisk-users] BRI or PRI callerid   Loic Didelot
• [asterisk-users] Verbosity best practice   Olivier
• [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use   Anthony Messina
• [asterisk-users] BRI or PRI callerid   Jorge Nunes
• [asterisk-users] BRI or PRI callerid   Igor Zamocky
• [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use   Tzafrir Cohen
• [asterisk-users] BRI or PRI callerid   Loic Didelot
• [asterisk-users] rxfax and txfax   Rizwan Hisham
• [asterisk-users] Verbosity best practice   Duncan Turnbull
• [asterisk-users] How to make a Outgoing Call from Asterisk ?   Hiren Mistry
• [asterisk-users] Pre-paid Billing   Jim Boykin
• [asterisk-users] rxfax and txfax   Thomas Stein
• [asterisk-users] Sip Info events   Jared Smith
• [asterisk-users] Verbosity best practice   Olivier
• [asterisk-users] Restrict SIP registration to one ip address only?   Remco Barendse
• [asterisk-users] RTCP-XR   Olivier
• [asterisk-users] CHANNEL((rtpqos|audio|local_ssrc)) (was: Re: RTCP-XR)   Philipp Kempgen
• [asterisk-users] Help with MFC/R2   Luis Morales
• [asterisk-users] Help with MFC/R2   Moises Silva
• [asterisk-users] Pre-paid Billing   Alex Balashov
• [asterisk-users] Pre-paid Billing   Igor Zamocky
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] CHANNEL((rtpqos|audio|local_ssrc)) (was: Re: RTCP-XR)   Olivier
• [asterisk-users] Pre-paid Billing   Alex Balashov
• [asterisk-users] Custom Voicemail emails   Steve Anness
• [asterisk-users] Restrict SIP registration to one ip address only?   Stefan Gofferje
• [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?   Olivier
• [asterisk-users] Help with MFC/R2   Luis Morales
• [asterisk-users] Pre-paid Billing   Jim Boykin
• [asterisk-users] server farm workarounds? (was Re: Restrict SIP registration to one ip address only?)   JD
• [asterisk-users] Pre-paid Billing   Alex Balashov
• [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?   Alex Balashov
• [asterisk-users] Digium training course   Tilghman Lesher
• [asterisk-users] Custom Voicemail emails   Tilghman Lesher
• [asterisk-users] Get rid of "Really destroying SIP dialog"   Tilghman Lesher
• [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?   Gordon Henderson
• [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?   Barton Fisher
• [asterisk-users] Digium training course   Eric "ManxPower" Wieling
• [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?   Cory Andrews
• [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?   Philipp Kempgen
• [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?   Olivier
• [asterisk-users] Pre-paid Billing   Jai Rangi
• [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?   Olivier
• [asterisk-users] device probe order question   Jason T. Nelson
• [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?   Stefan Gofferje
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] device probe order question   Tzafrir Cohen
• [asterisk-users] Help with MFC/R2   Luis Morales
• [asterisk-users] device probe order question   Jason T. Nelson
• [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?   Gordon Henderson
• [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?   Kristian Kielhofner
• [asterisk-users] Digium training course   Steve Totaro
• [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?   Andres
• [asterisk-users] device probe order question   Tzafrir Cohen
• [asterisk-users] Digium training course   Jon Pounder
• [asterisk-users] Help with MFC/R2   Moises Silva
• [asterisk-users] Old voicemail bounces users   David A. Bandel
• [asterisk-users] Digium training course   Jared Smith
• [asterisk-users] Old voicemail bounces users   Tilghman Lesher
• [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?   Stefan Gofferje
• [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?   Kristian Kielhofner
• [asterisk-users] Polycom phones and DNS SRV   CunningPike
• [asterisk-users] device probe order question   Ioan Indreias
• [asterisk-users] Get rid of "Really destroying SIP dialog"   Olivier
• [asterisk-users] Digium training course   Steve Totaro
• [asterisk-users] Custom Voicemail emails   mitcheloc
• [asterisk-users] device probe order question   Steve Totaro
• [asterisk-users] Custom Voicemail emails   Matt Gibson
• [asterisk-users] Streaming MoH on 1.4   Jay R. Ashworth
• [asterisk-users] device probe order question   Philipp Kempgen
• [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?   peder at networkoblivion.com
• [asterisk-users] device probe order question   Andres
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] Help with MFC/R2   Humberto Figuera
• [asterisk-users] Digium training course   Craig Guy
• [asterisk-users] Follow Me app question   Mark Phillips
• [asterisk-users] Old voicemail bounces users   David A. Bandel
• [asterisk-users] what codec is sip using?   sean darcy
• [asterisk-users] what codec is sip using?   David Gibbons
• [asterisk-users] Asterisk REFER   Al lists
• [asterisk-users] what codec is sip using?   sean darcy
• [asterisk-users] what codec is sip using?   Alex Balashov
• [asterisk-users] T100P detection.   Alex Balashov
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   ram
• [asterisk-users] PRI E1 Inbound calls hangup with busy after a few seconds   Daniel Johnson
• [asterisk-users] How to make a Outgoing Call from Asterisk ?   Hiren Mistry
• [asterisk-users] PRI E1 Inbound calls hangup with busy after a few seconds   Daniel Johnson
• [asterisk-users] PRI E1 Inbound calls hangup with busy after a few seconds   Paul Hales
• [asterisk-users] T38 FAX over a Broadsoft   eng. Anatoli Marinov
• [asterisk-users] Dialing a 60anything number issue!   Brad
• [asterisk-users] Dialing a 60anything number issue!   Steven Howes
• [asterisk-users] SIP request send me 482 error   remi.druilhe at orange-ftgroup.com
• [asterisk-users] SIP request send me 482 error   Stefan Gofferje
• [asterisk-users] SIP request send me 482 error   remi.druilhe at orange-ftgroup.com
• [asterisk-users] How to make a Outgoing Call from Asterisk ?   David Backeberg
• [asterisk-users] Preventing a call forward   Julian Lyndon-Smith
• [asterisk-users] Preventing a call forward   Julian Lyndon-Smith
• [asterisk-users] Help with MFC/R2   Luis Morales
• [asterisk-users] Follow Me app question   BJ Weschke
• [asterisk-users] Cisco + Asterisk   David Backeberg
• [asterisk-users] dundi and zap devices   Giorgio Incantalupo
• [asterisk-users] chan_iax2.c: No more space   Tim Panton
• [asterisk-users] what codec is sip using?   sean darcy
• [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?   Benny Amorsen
• [asterisk-users] VoIP Users Friday Conference @12 Noon EDT: Astricon run up meeting and more   randulo
• [asterisk-users] TE110P or TE120P   Gordon Henderson
• [asterisk-users] Streaming MoH on 1.4   Tilghman Lesher
• [asterisk-users] TE110P or TE120P   Cory Andrews
• [asterisk-users] TE110P or TE120P   Gordon Henderson
• [asterisk-users] PRI E1 Inbound calls hangup with busy after a few seconds   Matthew Fredrickson
• [asterisk-users] SIP request send me 482 error   Stefan Gofferje
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   logan
• [asterisk-users] Weird "permissions" issue when permissions check out...   Jason Lixfeld
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] Help with MFC/R2   Luis Morales
• [asterisk-users] Follow Me app question   David Van Ginneken
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   Jai Rangi
• [asterisk-users] Help with MFC/R2   Dae Yeung Um
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   logan
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   Jai Rangi
• [asterisk-users] Dropping Phone Calls   Todd Reese
• [asterisk-users] getting results messages from CLI commands via -rx   George Williams
• [asterisk-users] Last 2 days for early bird tickets to DruidCON 2008, 1-2 Oct in Atlanta GA   Ming Yong
• [asterisk-users] Dropping Phone Calls   Matt Gibson
• [asterisk-users] Dropping Phone Calls   Todd Reese
• [asterisk-users] getting results messages from CLI commands via -rx   Tilghman Lesher
• [asterisk-users] Loud noise on Zap port...   Carlos Chavez
• [asterisk-users] SVN 1.6.0 / current does not compile   Stefan Gofferje
• [asterisk-users] Specific SIP answers on incoming calls?   Stefan Gofferje
• [asterisk-users] Help with MFC/R2   Moises Silva
• [asterisk-users] SVN 1.6.0 / current does not compile   Sean Bright
• [asterisk-users] [CID] Unknown IE 18/21?   Vincent
• [asterisk-users] Specific SIP answers on incoming calls?   Alex Balashov
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   Nhadie
• [asterisk-users] Anyone flying with BA on Monday for Astricon ?   Stelios Koroneos
• [asterisk-users] 1.6.0-rc6 - SIP hold logic broken?   Stefan Gofferje
• [asterisk-users] 1.6.0-rc6 - SIP hold logic broken?   Stefan Gofferje
• [asterisk-users] Specific SIP answers on incoming calls?   Philipp von Klitzing
• [asterisk-users] getting results messages from CLI commands via -rx   Tzafrir Cohen
• [asterisk-users] SIP request send me 482 error   Raj Jain
• [asterisk-users] Cisco acquires Jabber   Dean Collins
• [asterisk-users] Cisco acquires Jabber   David Backeberg
• [asterisk-users] callwaiting callerid   robb
• [asterisk-users] Cisco acquires Jabber   Dean Collins
• [asterisk-users] Cisco acquires Jabber   Steve Kennedy
• [asterisk-users] Help with MFC/R2   Luis Morales
• [asterisk-users] Cisco acquires Jabber   mitcheloc
• [asterisk-users] broadcast ability   DesRae Mason
• [asterisk-users] broadcast ability   Doug Lytle
• [asterisk-users] Help with MFC/R2   Moises Silva
• [asterisk-users] Anyone flying with BA on Monday for Astricon ?   Stelios Koroneos
• [asterisk-users] how to add extensions and sip registrations dynamically   George Williams
• [asterisk-users] Astricon: Throw the dice, give a talk   John Todd
• [asterisk-users] chan_iax2.c: No more space   Philipp Kempgen
• [asterisk-users] Asterisk weird behavior after upgrading   Andrew Nowrot
• [asterisk-users] DNS Query Overload   Adam Lovegrove
• [asterisk-users] Asterisk weird behavior after upgrading   Philipp Kempgen
• [asterisk-users] DNS Query Overload   David Boyd
• [asterisk-users] DNS Query Overload   Philipp Kempgen
• [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use   hugolivude
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   logan
• [asterisk-users] Digium training course   Tilghman Lesher
• [asterisk-users] how to add extensions and sip registrations dynamically   Tilghman Lesher
• [asterisk-users] Digium training course   Chris Brentano
• [asterisk-users] Digium training course   Chris Brentano
• [asterisk-users] How to notify an event to every user   Olivier
• [asterisk-users] How to notify an event to every user   Doug Lytle
• [asterisk-users] How to notify an event to every user   Gordon Henderson
• [asterisk-users] How to notify an event to every user   alex.robar at gmail.com
• [asterisk-users] PRI E1 Inbound calls hangup with busy after a few seconds   Daniel Johnson
• [asterisk-users] How to notify an event to every user   Steve Totaro
• [asterisk-users] Digium training course   Jared Smith
• [asterisk-users] DNS Query Overload   Adam Lovegrove
• [asterisk-users] How to notify an event to every user   Paul Hales
• [asterisk-users] Digium training course   Steve Totaro
• [asterisk-users] DNS Query Overload   Anthony Francis
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   ram
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   Nhadie
• [asterisk-users] Seemingly easy question: NPA/NXX   Karl Fife
• [asterisk-users] Seemingly easy question: NPA/NXX   Shane Young
• [asterisk-users] Seemingly easy question: NPA/NXX   c.savinovich at itntelecom.com
• [asterisk-users] DNS Query Overload   Grey Man
• [asterisk-users] SIP request send me 482 error   remi.druilhe at orange-ftgroup.com
• [asterisk-users] SIP request send me 482 error   Stefan Gofferje
• [asterisk-users] Astricon news online?   randulo
• [asterisk-users] Astricon news online?   Ming Yong
• [asterisk-users] Astricon news online?   Tzafrir Cohen
• [asterisk-users] SIP request send me 482 error   remi.druilhe at orange-ftgroup.com
• [asterisk-users] setvar for outgoing SIP channels?   Klaus Darilion
• [asterisk-users] Problem using AJAM on asterisk 1.4.17   Jason Martin
• [asterisk-users] Problem using AJAM on asterisk 1.4.17   Sylvain Boily
• [asterisk-users] wad happen if there is nothing wrong in conf but still can't make calls?   Cindy Tan
• [asterisk-users] DNS Query Overload   Mik Cheez
• [asterisk-users] SIP request send me 482 error   Stefan Gofferje
• [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line   logan
• [asterisk-users] Seemingly easy question: NPA/NXX   Luki
• [asterisk-users] Seemingly easy question: NPA/NXX   Ira
• [asterisk-users] GotoIfTime and timezone specification   Klaus Darilion
• [asterisk-users] Seemingly easy question: NPA/NXX   Philipp Kempgen
• [asterisk-users] Seemingly easy question: NPA/NXX   Philipp Kempgen
• [asterisk-users] SIP request send me 482 error   Philipp von Klitzing
• [asterisk-users] RNB (was: Re: Seemingly easy question: NPA/NXX)   Philipp Kempgen
• [asterisk-users] I can't call my remote users?   Steve Anness
• [asterisk-users] I can't call my remote users?   Steve Anness
• [asterisk-users] Custom Voicemail emails   Steve Edwards
• [asterisk-users] Custom Voicemail emails   Steve Edwards
• [asterisk-users] Problem using AJAM on asterisk 1.4.17   Jason Martin
• [asterisk-users] E&M wink/no audio   Bill Michaelson
• [asterisk-users] Problem using AJAM on asterisk 1.4.17   Sylvain Boily
• [asterisk-users] ast_func_write: Function not registered   Anis Maatoug
• [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone   Zeeshan Zakaria
• [asterisk-users] Send us your suggestions on exhibits & tutorials to cover (video) at Voiceroute   Ming Yong
• [asterisk-users] PSTN Simulator   mark morreny
• [asterisk-users] PSTN Simulator   Anthony Francis
• [asterisk-users] How to notify an event to every user   Olivier
• [asterisk-users] How to notify an event to every user   Olivier
• [asterisk-users] How to notify an event to every user   Olivier
• [asterisk-users] How to notify an event to every user   Paul Hales
• [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone   Tony Mountifield
• [asterisk-users] PSTN Simulator   Alex Balashov
• [asterisk-users] [1.4.21.2] Checking that already off-hook?   Vincent
• [asterisk-users] Transcoding G.729 files   Thomas Kenyon
• [asterisk-users] Transcoding G.729 files   Alex Balashov
• [asterisk-users] Fax with asterisk   Erik Haider Forsen
• [asterisk-users] t38modem on OpenSuse   Marco Signorini
• [asterisk-users] Registration by IP address   michel freiha
• [asterisk-users] AGI and prepaid billing   bilal ghayyad
• [asterisk-users] chan_misdn troubles   Thanos Koukoulis
• [asterisk-users] chan_misdn troubles   Gergo Csibra
• [asterisk-users] chan_misdn troubles   Julien Claassen
• [asterisk-users] [1.4.21.2] Checking that already off-hook?   Julien Claassen
• [asterisk-users] [1.4.21.2] Checking that already off-hook?   Vincent
• [asterisk-users] AGI and prepaid billing   Benjamin Jacob
• [asterisk-users] Fax with asterisk   Giorgio Incantalupo
• [asterisk-users] chan_misdn troubles   Thanos Koukoulis
• [asterisk-users] chan_misdn troubles   Thanos Koukoulis
• [asterisk-users] Digium training course   Craig Guy
• [asterisk-users] Transcoding G.729 files   Thomas Kenyon
• [asterisk-users] How to notify an event to every user   Olivier
• [asterisk-users] Fax with asterisk   Olivier
• [asterisk-users] Fax with asterisk   Luis Morales
• [asterisk-users] Fax with asterisk   Giorgio Incantalupo
• [asterisk-users] Extension registration   michel freiha
• [asterisk-users] Extension registration   Vinícius Fontes
• [asterisk-users] How to notify an event to every user   Administrator TOOTAI
• [asterisk-users] Extension registration   Administrator TOOTAI
• [asterisk-users] Extension registration   Doug Lytle
• [asterisk-users] Extension registration   michel freiha
• [asterisk-users] Extension registration   Fred Posner
• [asterisk-users] Extension registration   michel freiha
• [asterisk-users] Extension registration   michel freiha
• [asterisk-users] Extension registration   John covici
• [asterisk-users] Fax with asterisk   Matthew Marion
• [asterisk-users] Fax with asterisk   Erik Haider Forsen
• [asterisk-users] Fax with asterisk   Erik Haider Forsen
• [asterisk-users] 1.6b9 Audio Issue   MFH
• [asterisk-users] Extension registration   Fred Posner
• [asterisk-users] Extension registration   Tariq ..
• [asterisk-users] Fax with asterisk   Steve Totaro
• [asterisk-users] Extension registration   michel freiha
• [asterisk-users] Fax with asterisk   Matthew Marion
• [asterisk-users] Fax with asterisk   Matthew Marion
• [asterisk-users] PRI incoming call forward / call redirect   Andrew Kohlsmith (lists)
• [asterisk-users] Extension registration   Fred Posner
• [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive   Steve Totaro
• [asterisk-users] Linksys 3102 with rfc2833 - NOT WORKING   Joseph
• [asterisk-users] Transcoding G.729 files   Kristian Kielhofner
• [asterisk-users] Seemingly easy question: NPA/NXX   Jay R. Ashworth
• [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive   Gordon Henderson
• [asterisk-users] Seemingly easy question: NPA/NXX   Jay R. Ashworth
• [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive   Alan Lord
• [asterisk-users] Fax with asterisk   Giorgio Incantalupo
• [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive   Tzafrir Cohen
• [asterisk-users] Connecting TE212p to NEC XenMaster   Mikel Lindsaar
• [asterisk-users] Fax with asterisk   Andrew Joakimsen
• [asterisk-users] extension definition   michel freiha
• [asterisk-users] Short question: CPU hardware requirements for Asterisk   Alejandro Cabrera Obed
• [asterisk-users] [1.4.21.2] Checking that already off-hook?   Vincent
• [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive   Michael Graves
• [asterisk-users] extension definition   Steve Edwards
• [asterisk-users] Short question: CPU hardware requirements for Asterisk   Steve Edwards
• [asterisk-users] extension definition   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk 1.4 or 1.6   Joseph
• [asterisk-users] Asterisk 1.4 or 1.6   Gordon Henderson
• [asterisk-users] AGI and prepaid billing + Radius   bilal ghayyad
• [asterisk-users] AGI and prepaid billing + Radius   Alex Balashov
• [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive   Michael Graves
• [asterisk-users] AGI and prepaid billing + Radius   Philippe Sultan
• [asterisk-users] "No route to destination" error   Martin Seebach
• [asterisk-users] AGI and prepaid billing + Radius   Vlasis Hatzistavrou (KTI)
• [asterisk-users] AGI and prepaid billing + Radius   bilal ghayyad
• [asterisk-users] A2Billing Callback Hangup after/about 20 sec!   Muhammad Zulqarnain
• [asterisk-users] "No route to destination" error   Philipp Kempgen
• [asterisk-users] extension definition   michel freiha
• [asterisk-users] Cellroute setup with asterisk   Roberts Klotins
• [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone   Zeeshan Zakaria
• [asterisk-users] How to send indicating call privacy using P-Asserted-Identity?   Zeeshan Zakaria
• [asterisk-users] Asterisk 1.4 or 1.6   Rob Hillis
• [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone   Steve Totaro
• [asterisk-users] How to notify an event to every user   Paul Hales
• [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone   Zeeshan Zakaria
• [asterisk-users] Short question: CPU hardware requirements for Asterisk   Senad Jordanovic
• [asterisk-users] Asterisk mysql CDR   Nhadie
• [asterisk-users] setvar for outgoing SIP channels?   Klaus Darilion
• [asterisk-users] asterisk console: "quit" is twice in history   Klaus Darilion
• [asterisk-users] SIP request send me 482 error   remi.druilhe at orange-ftgroup.com
• [asterisk-users] "No route to destination" error   Martin Seebach
• [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP   Nathan Dennis
• [asterisk-users] AGI and prepaid billing + Radius   Philippe Sultan
• [asterisk-users] Asterisk mysql CDR   Rizwan Hisham
• [asterisk-users] Fax with asterisk   Rizwan Hisham
• [asterisk-users] extension definition   Rizwan Hisham
• [asterisk-users] "No route to destination" error   Steve Totaro
• [asterisk-users] AGI and prepaid billing   Rizwan Hisham
• [asterisk-users] wad happen if there is nothing wrong in conf but still can't make calls?   Rizwan Hisham
• [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP   Tony Mountifield
• [asterisk-users] "No route to destination" error   Martin Seebach
• [asterisk-users] Voicemail cutting out after about 30 seconds   Chris Bagnall
• [asterisk-users] Asterisk 1.4 or 1.6   Artem Makhutov
• [asterisk-users] chan_misdn troubles   Thanos Koukoulis
• [asterisk-users] Voicemail cutting out after about 30 seconds   Steven Howes
• [asterisk-users] Asterisk 1.4 or 1.6   Chris Bagnall
• [asterisk-users] Debug dropped calls   Mark Engelhardt
• [asterisk-users] Asterisk is covering the peers IP address in SIP and SDP messages   Arno Scholz
• [asterisk-users] DID mode   michel freiha
• [asterisk-users] dundi and regcontext   ronald ramos
• [asterisk-users] asterisk console: "quit" is twice in history   Gordon Henderson
• [asterisk-users] Voicemail cutting out after about 30 seconds   Gordon Henderson
• [asterisk-users] Asterisk is covering the peers IP address in SIP and SDP messages   Vinícius Fontes
• [asterisk-users] Timeout question   Vadim Lebedev
• [asterisk-users] Fax with asterisk   C F
• [asterisk-users] Astricon 08 Videos & interviews & Voiceroute twitters on astricon   Ming Yong
• [asterisk-users] Asterisk mysql CDR   Nhadie
• [asterisk-users] Asterisk 1.4 or 1.6   Joseph
• [asterisk-users] Zaptel/DAHDI ztdummy only   Roderick A. Anderson
• [asterisk-users] "No route to destination" error   Andres
• [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP   Nathan Dennis
• [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6   Joseph
• [asterisk-users] Asterisk 1.4 or 1.6   Steve Totaro
• [asterisk-users] Asterisk 1.4 or 1.6   Paul Hales
• [asterisk-users] Asterisk on VMware Workstation 6   Michael J. Liberatore
• [asterisk-users] Asterisk on VMware Workstation 6   Matt Gibson
• [asterisk-users] Asterisk 1.4 is asking me for Mailbox #   Joseph
• [asterisk-users] g729 capacity   Robert McNaught
• [asterisk-users] Asterisk on VMware Workstation 6   Dean Collins
• [asterisk-users] g729 capacity   Igor H
• [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6   Jonn R Taylor
• [asterisk-users] What happened to the register= setting in sip.conf?   David Kerr
• [asterisk-users] g729 capacity   Steve Totaro
• [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6   Tzafrir Cohen
• [asterisk-users] Zaptel/DAHDI ztdummy only   Tzafrir Cohen
• [asterisk-users] g729 capacity   Igor Hernandez
• [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6   Joseph
• [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6   Joseph
• [asterisk-users] dundi and regcontext   technocrat voip
• [asterisk-users] g729 capacity   Robert McNaught
• [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6   Tzafrir Cohen
• [asterisk-users] Problem making international calls   f.beijer
• [asterisk-users] Recommend quality wholesale termination - Singapore and Sydney, Aus   broadband Voice
• [asterisk-users] Require Billing solution for Calling Cards retail...   broadband Voice
• [asterisk-users] Terrible Experience Net2phone A-Z termination   broadband Voice
• [asterisk-users] Terrible Experience Net2phone A-Z termination   Steve Totaro
• [asterisk-users] Terrible Experience Net2phone A-Z termination   Bruno Castelo Branco
• [asterisk-users] Current available allarms in the Asterisk   bilal ghayyad
• [asterisk-users] appconference low quality g729   Pezhman Lali
• [asterisk-users] IMAP voicemail import   Andreas Anderson
• [asterisk-users] Asterisk on VMware Workstation 6   Steven Howes
• [asterisk-users] Terrible Experience Net2phone A-Z termination   Igor Hernandez
• [asterisk-users] Fax with asterisk   Rizwan Hisham
• [asterisk-users] Terrible Experience Net2phone A-Z termination   broadband Voice
• [asterisk-users] SIP TLS   Rafael Puga
• [asterisk-users] Terrible Experience Net2phone A-Z termination   Tom Moore
• [asterisk-users] Terrible Experience Net2phone A-Z termination   Igor H
• [asterisk-users] Ringing after console dsp hangup   Jerry Geis
• [asterisk-users] OT: Do You Know What the Problem With CDMA is?   Dean Collins
• [asterisk-users] Monitoring trunk   Rodrigo Pinto
• [asterisk-users] Terrible Experience Net2phone A-Z termination   Sherwood McGowan
• [asterisk-users] OT: Do You Know What the Problem With CDMA is?   Andrew Kohlsmith (lists)
• [asterisk-users] Terrible Experience Net2phone A-Z termination   Rafael Canchola
• [asterisk-users] Dial Plan Issues   Tariq ..
• [asterisk-users] Dial Plan Issues   Steve Murphy
• [asterisk-users] OT: Do You Know What the Problem With CDMA is?   Drew Gibson
• [asterisk-users] OT: Do You Know What the Problem With CDMA is?   Steve Totaro
• [asterisk-users] OT: Do You Know What the Problem With CDMA is?   Andrew Kohlsmith (lists)
• [asterisk-users] Astricon people please post the announcement   randulo
• [asterisk-users] OT: Do You Know What the Problem With CDMA is?   Jason Aarons (US)
• [asterisk-users] Asterisk 1.4 is asking me for Mailbox #   Mark Michelson
• [asterisk-users] Astricon people please post the announcement   randulo
• [asterisk-users] Astricon people please post the announcement   randulo
• [asterisk-users] Astricon people please post the announcement   Steve Anness
• [asterisk-users] Astricon people please post the announcement   randulo
• [asterisk-users] Skype-asterisk connection announced (was Astricon people please post the announcement)   randulo
• [asterisk-users] Astricon people please post the announcement   Brian J. Murrell
• [asterisk-users] Astricon people please post the announcement   Tim Panton
• [asterisk-users] Astricon people please post the announcement   Tim Panton
• [asterisk-users] PRI TE110P Configuration   Shyju K
• [asterisk-users] OT: Do You Know What the Problem With CDMA is?   Drew Gibson
• [asterisk-users] Astricon people please post the announcement   Ming Yong
• [asterisk-users] Astricon people please post the announcement   Tzafrir Cohen
• [asterisk-users] Astricon people please post the announcement   Brian J. Murrell
• [asterisk-users] Asterisk 1.4 is asking me for Mailbox #   Joseph
• [asterisk-users] PRI TE110P Configuration   Steven Howes
• [asterisk-users] Astricon people please post the announcement   Steve Anness
• [asterisk-users] sip forking needed for ekiga 3.0   Brian J. Murrell
• [asterisk-users] asterisk-app_nv_faxdetect - Gentoo ebuild for *-1.4 was: NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6   Joseph
• [asterisk-users] sip forking needed for ekiga 3.0   SIP
• [asterisk-users] sip forking needed for ekiga 3.0   Brian J. Murrell
• [asterisk-users] Server Dimensioning   Jon Weisman
• [asterisk-users] sip forking needed for ekiga 3.0   SIP
• [asterisk-users] Server Dimensioning   Sam Tam
• [asterisk-users] sip forking needed for ekiga 3.0   Brian J. Murrell
• [asterisk-users] Create virtual extension   Manolet Gmail
• [asterisk-users] Server Dimensioning   Philipp Kempgen
• [asterisk-users] Server Dimensioning   Philipp Kempgen
• [asterisk-users] Skype + Asterisk Interview at Astricon   Ming Yong
• [asterisk-users] Create virtual extension   Brent Davidson
• [asterisk-users] Mysql Command and number rows returned   David Murphy
• [asterisk-users] users.conf behavior   Dave Poirier
• [asterisk-users] sip forking needed for ekiga 3.0   Alex Balashov
• [asterisk-users] Server Dimensioning   Alex Balashov
• [asterisk-users] Server Dimensioning   Jai Rangi
• [asterisk-users] Server Dimensioning   Igor Hernandez
• [asterisk-users] sip forking needed for ekiga 3.0   SIP
• [asterisk-users] Astricon people please post the announcement   Eric Fort
• [asterisk-users] Astricon people please post the announcement   Fred Posner
• [asterisk-users] Ringing after console dsp hangup   Doug Lytle
• [asterisk-users] How to send indicating call privacy using P-Asserted-Identity?   Zeeshan Zakaria
• [asterisk-users] Queue Calls getting stuck in there   Andrew Prowant
• [asterisk-users] Music on hold for sub tenants   carl Lougher
• [asterisk-users] Sip reload casuing issues   carl Lougher
• [asterisk-users] Monitoring simul calls   carl Lougher
• [asterisk-users] ZAP not answering call   Daniel Johnson
• [asterisk-users] Music on hold for sub tenants   Nhadie
• [asterisk-users] Astricon people please post the announcement   Brian J. Murrell
• [asterisk-users] Music on hold for sub tenants   carl Lougher
• [asterisk-users] Astricon people please post the announcement   Dean Collins
• [asterisk-users] Music on hold for sub tenants   Darrick Hartman
• [asterisk-users] Astricon people please post the announcement   Darrick Hartman
• [asterisk-users] Monitoring simul calls   Matt Gibson
• [asterisk-users] ZAP not answering call   Paul Hales
• [asterisk-users] Skype channel beta   randulo
• [asterisk-users] ZAP not answering call   Daniel Johnson
• [asterisk-users] Server Dimensioning   Gordon Henderson
• [asterisk-users] Asterisk on VMware Workstation 6   Michael J. Liberatore
• [asterisk-users] Asterisk on VMware Workstation 6   Michael J. Liberatore
• [asterisk-users] sip forking needed for ekiga 3.0   Grey Man
• [asterisk-users] Astricon people please post the announcement   Fred Posner
• [asterisk-users] Asterisk on VMware Workstation 6   Adam Goryachev
• [asterisk-users] Astricon people please post the announcement   Dinesh Nair
• [asterisk-users] Astricon people please post the announcement   Tzafrir Cohen
• [asterisk-users] OT: Do You Know What the Problem With CDMA is?   Dinesh Nair
• [asterisk-users] FAX t.38 on Asterisk 1.6?   Adam Goryachev
• [asterisk-users] T38 fax gateway announcement   Daniel Ferenci
• [asterisk-users] Astricon people please post the announcement   Grygoriy Dobrovolskyy
• [asterisk-users] FAX t.38 on Asterisk 1.6?   Daniel Ferenci
• [asterisk-users] Push presence from one asterisk to another   Loic Didelot
• [asterisk-users] setting DNID   Giorgio Incantalupo
• [asterisk-users] Get Call Length of Calls   Ali Jawad
• [asterisk-users] Astricon people please post the announcement   Tim Panton
• [asterisk-users] Incoming URL handling Problem (Asterisk problem ?)   Fabio Mosti
• [asterisk-users] Dell (was: Re: Server Dimensioning)   Philipp Kempgen
• [asterisk-users] Astricon people please post the announcement   Tim Panton
• [asterisk-users] Push presence from one asterisk to another   Philipp Kempgen
• [asterisk-users] sip forking needed for ekiga 3.0   Brian J. Murrell
• [asterisk-users] Get Call Length of Calls   Julien Claassen
• [asterisk-users] Friday 2008-09-26 12:00:00 Asterisk + Skype on your box   randulo
• [asterisk-users] setting DNID   Philipp Kempgen
• [asterisk-users] Astricon people please post the announcement   randulo
• [asterisk-users] Queue Calls getting stuck in there   Tariq ..
• [asterisk-users] Get Call Length of Calls   Ali Jawad
• [asterisk-users] sip forking needed for ekiga 3.0   Grey Man
• [asterisk-users] Astricon people please post the announcement   Tzafrir Cohen
• [asterisk-users] sip forking needed for ekiga 3.0   Brian J. Murrell
• [asterisk-users] sip forking needed for ekiga 3.0   SIP
• [asterisk-users] sip forking needed for ekiga 3.0   Watkins, Bradley
• [asterisk-users] Astricon people please post the announcement   Grygoriy Dobrovolskyy
• [asterisk-users] Sip reload casuing issues   Andres
• [asterisk-users] PRI TE110P Configuration (Solved)   Shyju K
• [asterisk-users] sip forking needed for ekiga 3.0   Brian J. Murrell
• [asterisk-users] sip forking needed for ekiga 3.0   SIP
• [asterisk-users] sip forking needed for ekiga 3.0   Brian J. Murrell
• [asterisk-users] Astricon people please post the announcement   Tzafrir Cohen
• [asterisk-users] Asterisk on VMware Workstation 6   Tilghman Lesher
• [asterisk-users] Dial Plan Issues   Brent Davidson
• [asterisk-users] Astricon people please post the announcement   Kevin P. Fleming
• [asterisk-users] server and 2 uniden phones no ringing   Jerry Geis
• [asterisk-users] Astricon people please post the announcement   Jay R. Ashworth
• [asterisk-users] Dial issue   equis software
• [asterisk-users] Voicemail retention   Asterisk User List
• [asterisk-users] Bizarre international call problem.   Ken D'Ambrosio
• [asterisk-users] server and 2 uniden phones no ringing   Mark Michelson
• [asterisk-users] sip forking needed for ekiga 3.0   Mark Michelson
• [asterisk-users] Voicemail retention   Gordon Henderson
• [asterisk-users] server and 2 uniden phones no ringing   Jerry Geis
• [asterisk-users] Voicemail retention   Stefan Schmidt
• [asterisk-users] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document   randulo
• [asterisk-users] Asterisk on VMware Workstation 6   Michael J. Liberatore
• [asterisk-users] Astricon people please post the announcement   Kevin P. Fleming
• [asterisk-users] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document   Philipp Kempgen
• [asterisk-users] Astricon people please post the announcement   Grygoriy Dobrovolskyy
• [asterisk-users] Fax with asterisk   Andrew Joakimsen
• [asterisk-users] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document   Paul Hales
• [asterisk-users] Audio Files   Abel Monzon
• [asterisk-users] Audio Files   Julien Claassen
• [asterisk-users] Astricon people please post the announcement   Kevin P. Fleming
• [asterisk-users] Push presence from one asterisk to another   Kevin P. Fleming
• [asterisk-users] users.conf behavior   Kevin P. Fleming
• [asterisk-users] Bizarre international call problem.   David Backeberg
• [asterisk-users] Audio Files   Abel Monzon
• [asterisk-users] Asterisk on VMware Workstation 6   David Backeberg
• [asterisk-users] iPhone Sip App   Forrest Beck
• [asterisk-users] Split incoming call volume across queues on several asterisk servers   Haider Raza
• [asterisk-users] Split incoming call volume across queues on several asterisk servers   Alex Balashov
• [asterisk-users] iPhone Sip App   Guillermo V. Salas
• [asterisk-users] iPhone Sip App   Eric Monoz
• [asterisk-users] Split incoming call volume across queues on several asterisk servers   Alex Balashov
• [asterisk-users] Split incoming call volume across queues on several asterisk servers   Haider Raza
• [asterisk-users] Split incoming call volume across queues on several asterisk servers   Haider Raza
• [asterisk-users] Split incoming call volume across queues on several asterisk servers   Alex Balashov
• [asterisk-users] Split incoming call volume across queues on several asterisk servers   Haider Raza
• [asterisk-users] Push presence from one asterisk to another   Loic Didelot
• [asterisk-users] Dial issue   Max Alex
• [asterisk-users] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document   randulo
• [asterisk-users] test call generator   Sam Tam
• [asterisk-users] test call generator   Jai Rangi
• [asterisk-users] test call generator   Igor Hernandez
• [asterisk-users] Problem with pickup extension *8 from features.conf using IAX   coco
• [asterisk-users] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document   Steve Totaro
• [asterisk-users] test call generator   Steve Totaro
• [asterisk-users] running out of disk space   Mr surfit
• [asterisk-users] Set A-Number in Sip Header   Michael Litzel
• [asterisk-users] test call generator   Alex Balashov
• [asterisk-users] Dial issue   equis software
• [asterisk-users] running out of disk space   Alexander Lopez
• [asterisk-users] Philippines   Dean Collins
• [asterisk-users] Asterisk and VoIP educational resources   randulo
• [asterisk-users] test call generator   Sam Tam
• [asterisk-users] test call generator   Sam Tam
• [asterisk-users] Problem with pickup extension *8 from features.conf using IAX   Shazaum
• [asterisk-users] Set A-Number in Sip Header   Shazaum
• [asterisk-users] Set A-Number in Sip Header   Michael Litzel
• [asterisk-users] Set A-Number in Sip Header   Shazaum
• [asterisk-users] test call generator   Steve Totaro
• [asterisk-users] running out of disk space   Patrick Maartense
• [asterisk-users] iPhone Sip App   Patrick Maartense
• [asterisk-users] FW: Google Alert - "dean collins"   Dean Collins
• [asterisk-users] FW: Google Alert - "dean collins"   Brian J. Murrell
• [asterisk-users] Asterisk 1.4 or 1.6   Chris Bagnall
• [asterisk-users] FW: Google Alert - "dean collins"   Eric Chamberlain
• [asterisk-users] FW: Google Alert - "dean collins"   Tzafrir Cohen
• [asterisk-users] rtpkeepalive problem ?   tic tac
• [asterisk-users] FW: Google Alert - "dean collins"   Steve Totaro
• [asterisk-users] Bizarre international call problem.   Ken D'Ambrosio
• [asterisk-users] Problem with pickup extension *8 from features.conf using IAX   Eric "ManxPower" Wieling
• [asterisk-users] FW: Google Alert - "dean collins"   BJ Weschke
• [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?   Philip Prindeville
• [asterisk-users] FW: Google Alert - "dean collins"   Steve Totaro
• [asterisk-users] New User with Calling Card Question   Babcock, Michael Alex
• [asterisk-users] Keeps Ringing After Answer   broadband Voice
• [asterisk-users] New User with Calling Card Question   Igor Hernandez
• [asterisk-users] New User with Calling Card Question   broadband Voice
• [asterisk-users] New User with Calling Card Question   Babcock, Michael Alex
• [asterisk-users] credit card processing   Ruddy Gbaguidi
• [asterisk-users] credit card processing   Babcock, Michael Alex
• [asterisk-users] credit card processing   Chris Bagnall
• [asterisk-users] credit card processing   Babcock, Michael Alex
• [asterisk-users] credit card processing   Ruddy Gbaguidi
• [asterisk-users] test call generator   Grey Man
• [asterisk-users] G.722 between Eyebeam and a Polycom IP650   Michael Graves
• [asterisk-users] credit card processing   Babcock, Michael Alex
• [asterisk-users] Eye P Media Soft Phone?   Michael Graves
• [asterisk-users] Vividial issue   Brad
• [asterisk-users] G.722 between Eyebeam and a Polycom IP650   Steve Underwood
• [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?   Sergio
• [asterisk-users] Vividial issue   ram
• [asterisk-users] New User with Calling Card Question   ram
• [asterisk-users] test call generator   Gnu Devel
• [asterisk-users] New User with Calling Card Question   Babcock, Michael Alex
• [asterisk-users] New User with Calling Card Question   Babcock, Michael Alex
• [asterisk-users] Conferencing Hardware   Jim Boykin
• [asterisk-users] G.722 between Eyebeam and a Polycom IP650   Thomas Kenyon
• [asterisk-users] Problem with pickup extension *8 from features.conf using IAX   coco
• [asterisk-users] Dial Plan Issues   Tariq ..
• [asterisk-users] G.722 between Eyebeam and a Polycom IP650   Michael Graves
• [asterisk-users] Dial Plan Issues   Steve Totaro
• [asterisk-users] New User with Calling Card Question   Steve Totaro
• [asterisk-users] Conferencing Hardware   Gordon Henderson
• [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?   Kristian Kielhofner
• [asterisk-users] Need help with Cisco 7960   Christian
• [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?   Philip Prindeville
• [asterisk-users] Dial Plan Issues   Tariq ..
• [asterisk-users] Dial Plan Issues   Tzafrir Cohen
• [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?   Andres
• [asterisk-users] Problem with my softphone   Abel Monzon
• [asterisk-users] Conferencing Hardware   Mike Trest
• [asterisk-users] Conferencing Hardware   Jim Boykin
• [asterisk-users] uk tole-free dids?   Babcock, Michael Alex
• [asterisk-users] uk tole-free dids?   Geraint Lee
• [asterisk-users] Conferencing Hardware   Gordon Henderson
• [asterisk-users] AGI defunct processes + GSM Playback - HELP!   Hakan C
• [asterisk-users] uk tole-free dids?   Gordon Henderson
• [asterisk-users] Creating Asterisk Binary Package   Jim Boykin
• [asterisk-users] Creating Asterisk Binary Package   Tzafrir Cohen
• [asterisk-users] Creating Asterisk Binary Package   Steven Howes
• [asterisk-users] ATA for large networks   Vieri
• [asterisk-users] identify/find a channel to pick it up   Stefan Schmidt
• [asterisk-users] Creating Asterisk Binary Package   Alan Lord
• [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?   Philip Prindeville
• [asterisk-users] ATA for large networks   Sam Tam
• [asterisk-users] ATA for large networks   Vieri
• [asterisk-users] Knowing incoming call technology and channel   Olivier
• [asterisk-users] Knowing incoming call technology and channel   Alex Balashov
• [asterisk-users] Creating Asterisk Binary Package   Tzafrir Cohen
• [asterisk-users] Creating Asterisk Binary Package   Jim Boykin
• [asterisk-users] Creating Asterisk Binary Package   Tzafrir Cohen
• [asterisk-users] Knowing incoming call technology and channel [SOLVED]   Olivier
• [asterisk-users] Creating Asterisk Binary Package   Jim Boykin
• [asterisk-users] Disable CDR?   Vincent
• [asterisk-users] Creating Asterisk Binary Package   Tzafrir Cohen
• [asterisk-users] Creating Asterisk Binary Package   Jim Boykin
• [asterisk-users] ATA for large networks   Thanos Koukoulis
• [asterisk-users] Knowing incoming call technology and channel   Philipp Kempgen
• [asterisk-users] OT - Avantages of ISDN PtP and PtmP   Olivier
• [asterisk-users] ATA for large networks   Olivier
• [asterisk-users] SIP/IAX Interworking ip CANCEL behavior   Andreas M.
• [asterisk-users] OT - Avantages of ISDN PtP and PtmP   Tzafrir Cohen
• [asterisk-users] OT - Avantages of ISDN PtP and PtmP   Philipp Kempgen
• [asterisk-users] Disable CDR?   Atis Lezdins
• [asterisk-users] OT - Avantages of ISDN PtP and PtmP   Steve Davies
• [asterisk-users] Bizarre international call problem.   Dinesh Nair
• [asterisk-users] OT - Avantages of ISDN PtP and PtmP   Olivier
• [asterisk-users] ATA for large networks   Darrick Hartman
• [asterisk-users] CLI and verbosity level   Olivier
• [asterisk-users] CLI and verbosity level   Darrick Hartman
• [asterisk-users] Problem with pickup extension *8 from features.conf using IAX   Serghei Gutanu
• [asterisk-users] OT - Avantages of ISDN PtP and PtmP   Steve Davies
• [asterisk-users] ATA for large networks   Andres
• [asterisk-users] ATA for large networks   Thanos Koukoulis
• [asterisk-users] ATA for large networks   Gordon Henderson
• [asterisk-users] CLI and verbosity level [SOLVED]   Olivier
• [asterisk-users] ATA for large networks   Tom Moore
• [asterisk-users] ATA for large networks   Eric "ManxPower" Wieling
• [asterisk-users] Bizarre international call problem.   Ken D'Ambrosio
• [asterisk-users] Knowing incoming call technology and channel [SOLVED]   Eric "ManxPower" Wieling
• [asterisk-users] ATA for large networks   Steve Totaro
• [asterisk-users] ATA for large networks   Andres
• [asterisk-users] Dial Plan Issues   Tariq ..
• [asterisk-users] ATA for large networks   C F
• [asterisk-users] ATA for large networks   Yehavi Bourvine
• [asterisk-users] Knowing incoming call technology and channel [SOLVED]   SIP
• [asterisk-users] Zaptel Lines - How many are in use..   Singer Wang
• [asterisk-users] Knowing incoming call technology and channel [SOLVED]   Josiah Bryan
• [asterisk-users] Zaptel Lines - How many are in use..   Josiah Bryan
• [asterisk-users] ATA for large networks   Steve Totaro
• [asterisk-users] Zaptel Lines - How many are in use..   Luis Morales
• [asterisk-users] Source of SIP "Remote host can't match request NOTIFY"   Olivier
• [asterisk-users] Knowing incoming call technology and channel [SOLVED]   Mark Hamilton
• [asterisk-users] Knowing incoming call technology and channel [SOLVED]   Eric "ManxPower" Wieling
• [asterisk-users] uk tole-free dids?   Babcock, Michael Alex
• [asterisk-users] ATA for large networks   Carlos Chavez
• [asterisk-users] uk tole-free dids?   Babcock, Michael Alex
• [asterisk-users] Zaptel Lines - How many are in use..   Singer Wang
• [asterisk-users] uk tole-free dids?   Gordon Henderson
• [asterisk-users] How can Block a pri channel   Alireza Shokoienia
• [asterisk-users] uk tole-free dids?   Babcock, Michael Alex
• [asterisk-users] uk tole-free dids?   Steve Kennedy
• [asterisk-users] Dial Plan Issues   Steve Murphy
• [asterisk-users] Cheap FXO Card?   Andrew Joakimsen
• [asterisk-users] Zaptel/DAHDI ztdummy only   Roderick A. Anderson
• [asterisk-users] uk tole-free dids?   asterisk
• [asterisk-users] ATA for large networks   C F
• [asterisk-users] ATA for large networks   C F
• [asterisk-users] uk tole-free dids?   Geraint Lee
• [asterisk-users] ATA for large networks   Jerry Jones
• [asterisk-users] ATA for large networks   Brian Webster
• [asterisk-users] Channel variables "materializing" ...   Julian Lyndon-Smith
• [asterisk-users] Source of SIP "Remote host can't match request NOTIFY"   Olivier
• [asterisk-users] Channel variables "materializing" ...   Brent Davidson
• [asterisk-users] Asterisk Documentation on voip-info.org (was: Re: Knowing incoming call technology and channel [SOLVED])   Josiah Bryan
• [asterisk-users] ATA for large networks   broadband Voice
• [asterisk-users] credit card processing   Andrew Joakimsen
• [asterisk-users] ATA for large networks   Vieri
• [asterisk-users] ATA for large networks   Vinícius Fontes
• [asterisk-users] Maybe OT - routing calls in PSTN   Bill Michaelson
• [asterisk-users] Maybe OT - routing calls in PSTN   Alex Balashov
• [asterisk-users] Maybe OT - routing calls in PSTN   Alex Balashov
• [asterisk-users] problem with my softphone   Abel Monzon
• [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting   Brian J. Murrell
• [asterisk-users] ATA for large networks   Lafras Henning
• [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?   Philip Prindeville
• [asterisk-users] test call generator   Steve Totaro
• [asterisk-users] test call generator   Sam Tam
• [asterisk-users] problem with my softphone   David
• [asterisk-users] ATA for large networks   Johann Steinwendtner
• [asterisk-users] Cisco 7911g   Sean Lowry
• [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting   Lyle Giese
• [asterisk-users] test call generator   Steve Totaro
• [asterisk-users] ATA for large networks   Jerry Jones
• [asterisk-users] ATA for large networks   Loic Didelot
• [asterisk-users] Question about Asterisk and Java   Santiago Panchi
• [asterisk-users] asterisk app store   Dean Collins
• [asterisk-users] Question about Asterisk and Java   Martin Smith
• [asterisk-users] asterisk app store   Steve Totaro
• [asterisk-users] Maybe OT - routing calls in PSTN   Bill Michaelson
• [asterisk-users] asterisk app store   Dean Collins
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Josiah Bryan
• [asterisk-users] Maybe OT - routing calls in PSTN   Steve Totaro
• [asterisk-users] Maybe OT - routing calls in PSTN   Igor Hernandez
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Steve Totaro
• [asterisk-users] asterisk-users Digest, Vol 50, Issue 89   Bill Michaelson
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Tzafrir Cohen
• [asterisk-users] Maybe OT - routing calls in PSTN   Alex Balashov
• [asterisk-users] Channel variables "materializing" ...   Julian Lyndon-Smith
• [asterisk-users] Problem with pickup extension *8 from features.conf using IAX   coco
• [asterisk-users] asterisk app store   zac wolfe
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Josiah Bryan
• [asterisk-users] test call generator   zac wolfe
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Mark Hamilton
• [asterisk-users] Using AMI to View ZAP Channels   David Budny
• [asterisk-users] asterisk app store   Dean Collins
• [asterisk-users] credit card processing   Gerald Begumisa
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Tzafrir Cohen
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Josiah Bryan
• [asterisk-users] How to tell the underlying carrier for your ITSP.   Alex Balashov
• [asterisk-users] OT- NIU Framing   Cory Andrews
• [asterisk-users] OT- NIU Framing   Alex Balashov
• [asterisk-users] OT- NIU Framing   Cory Andrews
• [asterisk-users] Question about Asterisk and Java   Santiago Panchi
• [asterisk-users] Transfer a call without announce : no sound   Nicolas Ross
• [asterisk-users] Asterisk Documentation now on voip-info.org Wiki   Steve Totaro
• [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting   Brian J. Murrell
• [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting   Lyle Giese
• [asterisk-users] Asterisk in VM.   Alex Balashov
• [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting   Brian J. Murrell
• [asterisk-users] zap destroy   Jeff LaCoursiere
• [asterisk-users] is DNS SRV enough for failover?   Nhadie
• [asterisk-users] is DNS SRV enough for failover?   Alex Balashov
• [asterisk-users] is DNS SRV enough for failover?   Andres
• [asterisk-users] Cisco Dropping SIP support?   Michael Graves
• [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting   Andrew Joakimsen
• [asterisk-users] No reply to our critical packet   Andrew Joakimsen
• [asterisk-users] Polycom 3.1.0RevB   Andrew Joakimsen
• [asterisk-users] G723 on asterisk 1.4.1   Andrew Joakimsen
• [asterisk-users] ATA for large networks   Col Ferguson

Last message date: Tue Sep 30 23:31:28 CDT 2008
Archived on: Tue Sep 30 23:31:52 CDT 2008

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