April 2003 Archives by author

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Starting: Tue Apr 1 01:13:05 MST 2003
Ending: Wed Apr 30 23:12:56 MST 2003
Messages: 1553

• [Asterisk-Users] Up to 8 lines?   WipeOut .
• [Asterisk-Users] Setting up PABX   WipeOut .
• [Asterisk-Users] will this machine handle it   WipeOut .
• [Asterisk-Users] * on openmosix   WipeOut .
• [Asterisk-Users] Buying channel banks online?   WipeOut .
• [Asterisk-Users] Quesiton about SIP and MSN   WipeOut .
• [Asterisk-Users] Updating from CVS!   WipeOut .
• [Asterisk-Users] Is the CVS server working??   WipeOut .
• [Asterisk-Users] Comments on "transfer" feature request   WipeOut .
• [Asterisk-Users] Wiki for the * community.   WipeOut .
• [Asterisk-Users] Multiport FXO??   WipeOut .
• [Asterisk-Users] external trigger on caller id   WipeOut .
• [Asterisk-Users] Error compiling in RedHat 9   WipeOut .
• [Asterisk-Users] Error compiling in RedHat 9   WipeOut .
• [Asterisk-Users] Error compiling in RedHat 9   WipeOut .
• [Asterisk-Users] Asterisk-Redhat 9 install guide.   WipeOut .
• [Asterisk-Users] Where is zttool?   WipeOut .
• [Asterisk-Users] Where is zttool?   WipeOut .
• [Asterisk-Users] Modem connections   WipeOut .
• [Asterisk-Users] S100U hissing noise..   WipeOut .
• [Asterisk-Users] Asterisk and SNOM 200   WipeOut .
• [Asterisk-Users] Asterisk and SNOM 200   WipeOut .
• [Asterisk-Users] Trouble installing   WipeOut .
• [Asterisk-Users] SIP support status   WipeOut .
• [Asterisk-Users] S100U on RH9   WipeOut .
• [Asterisk-Users] S100U on RH9   WipeOut .
• [Asterisk-Users] S100U on RH9   WipeOut .
• [Asterisk-Users] SIP Proxy   WipeOut .
• [Asterisk-Users] SIP Proxy   WipeOut .
• [Asterisk-Users] S100U on RH9   WipeOut .
• [Asterisk-Users] SNOM 200 auth config   WipeOut .
• [Asterisk-Users] meetme config   WipeOut .
• [Asterisk-Users] Music-on-Hold radio input?   WipeOut .
• [Asterisk-Users] meetme config   WipeOut .
• [Asterisk-Users] meetme config   WipeOut .
• [Asterisk-Users] Redhat vs Mandrake.   WipeOut .
• [Asterisk-Users] Redhat vs Mandrake.   WipeOut .
• [Asterisk-Users] Taking a call for someone elses extension from my extension   WipeOut .
• [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extension from my extension)   WipeOut .
• [Asterisk-Users] netmeeting dial   WipeOut .
• [Asterisk-Users] Still can't get MeetMe working..   WipeOut .
• [Asterisk-Users] netmeeting dial   WipeOut .
• [Asterisk-Users] netmeeting dial   WipeOut .
• [Asterisk-Users] Best IP phone?   WipeOut .
• [Asterisk-Users] cdr_mysql table definition   WipeOut .
• [Asterisk-Users] testing asterisks   WipeOut .
• [Asterisk-Users] Call Monitoring   WipeOut .
• [Asterisk-Users] Zapata not required??   WipeOut .
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   WipeOut .
• [Asterisk-Users] SNOM 200 and MWI??   WipeOut .
• [Asterisk-Users] SNOM 200 and MWI??   WipeOut .
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   WipeOut .
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   WipeOut .
• [Asterisk-Users] Zapata not required??   WipeOut .
• [Asterisk-Users] Proxy / STUN??   WipeOut .
• [Asterisk-Users] MSN Messager and Asterisk   WipeOut .
• [Asterisk-Users] SNOM 200 and MWI??   WipeOut .
• [Asterisk-Users] Whats ENUM??   WipeOut .
• [Asterisk-Users] EICON Drivers and CAPI   ???????? ?????
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   Ask Bjørn Hansen
• [Asterisk-Users] false ringback   Petr Michálek
• [Asterisk-Users] CAPI and FAX errors   Petr Michálek
• [Asterisk-Users] RE: [Asterisk-Dev] Several patches, including recording and music -on-hold   Petr Michálek
• [Asterisk-Users] Rackmount PC hardware suggestions   Petr Michálek
• [Asterisk-Users] CVS build patch   Ragnar Kjørstad
• [Asterisk-Users] Eicon BRI-card   Jörg Bauer/Denic
• [Asterisk-Users] howto   Jörg Bauer/Denic
• [Asterisk-Users] howto (Also SIP VIA header)   Jörg Bauer/Denic
• [Asterisk-Users] howto (Also SIP VIA header)   Jörg Bauer/Denic
• [Asterisk-Users] Whats ENUM??   Jörg Bauer/Denic
• [Asterisk-Users] Recording Prompts   Omar Abhari
• [Asterisk-Users] Recording Prompts   Omar Abhari
• [Asterisk-Users] Recording Prompts   Omar Abhari
• [Asterisk-Users] * on openmosix   Andres Tello Abrego
• [Asterisk-Users] CE certification for Europe   Andres Tello Abrego
• [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?   T Aksoy
• [Asterisk-Users] CE certification for Europe   T Aksoy
• [Asterisk-Users] SIP Testing   T Aksoy
• [Asterisk-Users] ADSI Analogue phones   T Aksoy
• [Asterisk-Users] ADSI Analogue phones   T Aksoy
• [Asterisk-Users] adsi phones   T Aksoy
• [Asterisk-Users] adsi phones   T Aksoy
• [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?   Chris Albertson
• [Asterisk-Users] non-telephony use of T400P?   Chris Albertson
• [Asterisk-Users] Should List be Moderated?   Chris Albertson
• [Asterisk-Users] Music-on-Hold radio input?   Chris Albertson
• [Asterisk-Users] Redhat vs Mandrake.   Chris Albertson
• [Asterisk-Users] Rackmount PC hardware suggestions   Chris Albertson
• [Asterisk-Users] Re: Error with dot when making progdocs   Srini Amble
• [Asterisk-Users] music on hold   Joe Antkowiak
• [Asterisk-Users] adsi phones   Joe Antkowiak
• [Asterisk-Users] adsi phones   Joe Antkowiak
• [Asterisk-Users] web interface to vmail   Joe Antkowiak
• [Asterisk-Users] adsi phone codes?   Joe Antkowiak
• [Asterisk-Users] RE: [Asterisk-Dev] adsi phone codes?   Joe Antkowiak
• [Asterisk-Users] gastman   Joe Antkowiak
• [Asterisk-Users] gastman   Joe Antkowiak
• [Asterisk-Users] Multiple X100P cards   Jim Archer
• [Asterisk-Users] Music on hold distorted, whitenoise, maybe just too loud?   Jim Archer
• [Asterisk-Users] How to make an X100P answer only one distinctive ring cadence?   Jim Archer
• [Asterisk-Users] Multiple X100P cards   Jim Archer
• [Asterisk-Users] Multiple X100P cards   Jim Archer
• [Asterisk-Users] Multiple X100P cards   Jim Archer
• [Asterisk-Users] Multiple X100P cards   Jim Archer
• [Asterisk-Users] SIP and special functions - do they work?   Jim Archer
• [Asterisk-Users] Should List be Moderated?   Peter Armstrong
• [Asterisk-Users] Does anyone have the Cisco 7910 IP Phone working with Asterisk?   Peter Armstrong
• [Asterisk-Users] Grandstream BudgeTone 100   Domjan Attila
• [Asterisk-Users] Chan_capi errors compil   Rattana BIV
• [Asterisk-Users] ISDN4Linux problems   Rattana BIV
• [Asterisk-Users] Chan_capi errors compil   Rattana BIV
• [Asterisk-Users] ISDN4Linux problems   Rattana BIV
• [Asterisk-Users] Conferences   Rattana BIV
• [Asterisk-Users] Conferences   Rattana BIV
• [Asterisk-Users] Conferences   Rattana BIV
• [Asterisk-Users] Conferences without zaptel devices   Rattana BIV
• [Asterisk-Users] [Asterisk-Users]   Rattana BIV
• [Asterisk-Users] [Asterisk-Users]   Rattana BIV
• [Asterisk-Users] [Asterisk-Users]   Rattana BIV
• [Asterisk-Users] ztdummy hard to use   Rattana BIV
• [Asterisk-Users] Eicon driver?   Rattana BIV
• [Asterisk-Users] Channel Driver Architecture   Paul Bagyenda
• [Asterisk-Users] Voicetronix driver for V6PCI/V12PCI - first draft   Paul Bagyenda
• [Asterisk-Users] Voicetronix Openswitch driver - draft 2   Paul Bagyenda
• [Asterisk-Users] Redhat vs Mandrake.   Michael Baird
• [Asterisk-Users] PRI Setup   Michael Baird
• [Asterisk-Users] conf advices needed   Jean-Yves F. Barbier
• [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #239 - 13 msgs   Benjamin J. Bawkon
• [Asterisk-Users] NAT and 7960   Benjamin J. Bawkon
• [Asterisk-Users] AGI Script   Benjamin J. Bawkon
• [Asterisk-Users] Two Rings   Benjamin J. Bawkon
• [Asterisk-Users] Up to 8 lines?   George Bean
• [Asterisk-Users] Dlink DG-104S   George Bean
• [Asterisk-Users] Trouble installing   Derek Beaumont
• [Asterisk-Users] Asterisk Beginner   Derek Beaumont
• [Asterisk-Users] two computers set up ... now what???   Derek Beaumont
• [Asterisk-Users] Re: two computers set up ... now what?   Derek Beaumont
• [Asterisk-Users] Dialplan question   Derek Beaumont
• [Asterisk-Users] Dialing using X100P   Derek Beaumont
• [Asterisk-Users] Re: Dialing using X100P   Derek Beaumont
• [Asterisk-Users] Re: Dialing using X100P   Derek Beaumont
• [Asterisk-Users] Re: Dialing using X100P   Derek Beaumont
• [Asterisk-Users] Re: Dialing using X100P   Derek Beaumont
• [Asterisk-Users] Creating a phone channel   Derek Beaumont
• [Asterisk-Users] Re: Creating a phone channel   Derek Beaumont
• [Asterisk-Users] Creating a phone channel   Derek Beaumont
• [Asterisk-Users] Re: Creating a phone channel   Derek Beaumont
• [Asterisk-Users] TDM10B problem   Derek Beaumont
• [Asterisk-Users] Re: TDM10B problem   Derek Beaumont
• [Asterisk-Users] TDM10B problem   Derek Beaumont
• [Asterisk-Users] TDM10B problem   Derek Beaumont
• [Asterisk-Users] Nightsettings   Michiel Betel
• [Asterisk-Users] Patch to enable syslog logging   Michiel Betel
• [Asterisk-Users] Patch to enable syslog logging   Michiel Betel
• [Asterisk-Users] corrected syslog patch   Michiel Betel
• [Asterisk-Users] CE certification for Europe   Michiel Betel
• [Asterisk-Users] CE certification for Europe   Michiel Betel
• [Asterisk-Users] CE certification for Europe   Michiel Betel
• [Asterisk-Users] CE certification for Europe   Michiel Betel
• [Asterisk-Users] Fw: Predictive dialer   Michiel Betel
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Michiel Betel
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Michiel Betel
• [Asterisk-Users] Fw: Predictive dialer   Michiel Betel
• [Asterisk-Users] Running Asterisk on non-intel hardware   Michiel Betel
• [Asterisk-Users] Up to 8 lines?   Michael Bielicki
• [Asterisk-Users] will this machine handle it   Michael Bielicki
• [Asterisk-Users] MP3player problem   Michael Bielicki
• [Asterisk-Users] CE certification for Europe   Michael Bielicki
• [Asterisk-Users] false ringback   Michael Bielicki
• [Asterisk-Users] false ringback   Michael Bielicki
• [Asterisk-Users] TDM400 question   Michael Bielicki
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Michael Bielicki
• [Asterisk-Users] T100P Incoming Calls Drop   Michael Bielicki
• [Asterisk-Users] Whoah! My E400P system went AWOL   Michael Bielicki
• [Asterisk-Users] Recording Prompts   Michael Bielicki
• [Asterisk-Users] fix for typo in latest cvs in channels/chan_alsa.c   Michael Bielicki
• [Asterisk-Users] Account code on SIP   Michael Bielicki
• [Asterisk-Users] Account code on SIP   Michael Bielicki
• [Asterisk-Users] X100P   Michael Bielicki
• [Asterisk-Users] Re: TDM10B problem   Michael Bielicki
• [Asterisk-Users] will this machine handle it   Andre Bierwirth
• [Asterisk-Users] Asterisk and SNOM 200   Andre Bierwirth
• [Asterisk-Users] EICON Drivers and CAPI   Rattana Biv
• [Asterisk-Users] Chan_capi compil error   Rattana Biv
• [Asterisk-Users] PBX Console   Bluedemon
• [Asterisk-Users] Complie on RH8   Robert Boardman
• [Asterisk-Users] Re: Complie on RH8   Robert Boardman
• [Asterisk-Users] Conferences without zaptel devices   Leo Ann Boon
• [Asterisk-Users] Deependencies problem   Ahmed Boreau
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #235 - 5 msgs   Ahmed Boreau
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #237 - 11 msgs   Ahmed Boreau
• [Asterisk-Users] Don't be upset !!! Architecture is need !!!   Ahmed Boreau
• [Asterisk-Users] Extensions.conf   Ahmed Boreau
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #286 - 14 msgs   Ahmed Boreau
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #286 - 14 msgs   Ahmed Boreau
• [Asterisk-Users] Adjust volume for X100P   Patrick Bouchard
• [Asterisk-Users] Asterisk-Redhat 9 install guide.   Jason Boyd
• [Asterisk-Users] Zap flash bug?   Matteo Brancaleoni
• [Asterisk-Users] FAX over IAX   Matteo Brancaleoni
• [Asterisk-Users] SIP improvements   Matteo Brancaleoni
• [Asterisk-Users] IP hard phone devices (grandstream)   Matteo Brancaleoni
• [Asterisk-Users] Wiki for the * community.   Matteo Brancaleoni
• [Asterisk-Users] Wiki for the * community.   Matteo Brancaleoni
• [Asterisk-Users] Fw: Predictive dialer   Matteo Brancaleoni
• [Asterisk-Users] Strange Sip problem?   Matteo Brancaleoni
• [Asterisk-Users] Releasing Ast 0.4   Matteo Brancaleoni
• [Asterisk-Users] Asterisk and SNOM 200   Matteo Brancaleoni
• [Asterisk-Users] Account code on SIP   Matteo Brancaleoni
• R: [Asterisk-Users] Zhone re-provision   Matteo Brancaleoni
• [Asterisk-Users] Call Monitoring   Matteo Brancaleoni
• R: [Asterisk-Users] Zapata not required??   Matteo Brancaleoni
• [Asterisk-Users] TDM40B Heating ?   Matteo Brancaleoni
• [Asterisk-Users] cdr_mysql table definition   Gijsbert van den Brink
• [Asterisk-Users] cdr_mysql table definition   Gijsbert van den Brink
• [Asterisk-Users] TDM400P Pass Through Fax Support   Adrian Brown
• [Asterisk-Users] TDM400P Pass Through Fax Support   Adrian Brown
• [Asterisk-Users] Voicemail hangups/minimum message size   Peter Brown
• [Asterisk-Users] Music-on-Hold radio input?   Ryan Butler
• [Asterisk-Users] Music-on-Hold radio input?   Ryan Butler
• [Asterisk-Users] zaptel.conf and zapata.conf for Carrier Access   Kim C. Callis
• [Asterisk-Users] zaptel.conf and zapata.conf for Carrier Access   Kim C. Callis
• [Asterisk-Users] ATA186: "Call/Leg Transaction Doesn't Exist" on local call   Brian Capouch
• [Asterisk-Users] ATA186: "Call/Leg Transaction Doesn't Exist" on local call   Brian Capouch
• [Asterisk-Users] CE certification for Europe   Brian Capouch
• [Asterisk-Users] CE certification for Europe   Brian Capouch
• [Asterisk-Users] SIP Testing   Brian Capouch
• [Asterisk-Users] SIP Testing   Brian Capouch
• [Asterisk-Users] SIP Testing   Brian Capouch
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Brian Capouch
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Brian Capouch
• [Asterisk-Users] how to register * at FWD from behind NAT   Brian Capouch
• [Asterisk-Users] Parametric "sample.call?"   Brian Capouch
• [Asterisk-Users] Weirdness on "hookflash call pickup"   Brian Capouch
• [Asterisk-Users] Weirdness on "hookflash call pickup"   Brian Capouch
• [Asterisk-Users] Weirdness on "hookflash call pickup"   Brian Capouch
• [Asterisk-Users] * kernel panic   Brian Capouch
• [Asterisk-Users] iconnecthere bridging broken on recent CVS?   Brian Capouch
• [Asterisk-Users] iconnecthere bridging broken on recent CVS?   Brian Capouch
• [Asterisk-Users] What driver for the TDMX0B cards?   Brian Capouch
• [Asterisk-Users] Hmmm. RJ-45 on TDMx0B card?   Brian Capouch
• [Asterisk-Users] Anyone else lose iconnecthere service in recent CVS?   Brian Capouch
• [Asterisk-Users] Re: Dialing using X100P   Brian Capouch
• [Asterisk-Users] Sending CID to ATA186?   Brian Capouch
• [Asterisk-Users] Experiences with Zultys 4x4 SIP phone?   David Carr
• [Asterisk-Users] Error with dot when making progdocs   David Carr
• [Asterisk-Users] Modem connections   David Carr
• [Asterisk-Users] Hangup detection problems   David Carr
• [Asterisk-Users] Hangup detection problems   David Carr
• [Asterisk-Users] Voicemail silence/click/cadence/ring/hang-up detection   David Carr
• [Asterisk-Users] Grandstream BudgeTone 100   David Carr
• [Asterisk-Users] SIP transfer doesn't stop music on hold   David Carr
• [Asterisk-Users] I must be alone   Uriel Carrasquilla
• [Asterisk-Users] asterisk / channel bank   Uriel Carrasquilla
• [Asterisk-Users] asterisk /channel bank   Uriel Carrasquilla
• [Asterisk-Users] sounds /vm-*.gsm / apps_voicemail.c   Uriel Carrasquilla
• [Asterisk-Users] Firmware version in Adtran TA 750 Channel Bank   Uriel Carrasquilla
• [Asterisk-Users] Firmware in Adtran 750   Uriel Carrasquilla
• [Asterisk-Users] IAXCLIENT/Win32 - Source Compile   Uriel Carrasquilla
• [Asterisk-Users] Voice Mail Re-compile   Uriel Carrasquilla
• [Asterisk-Users] Xten / SIP Phones compared to GnoPhones   Uriel Carrasquilla
• [Asterisk-Users] Taking a call for someone elses extension from my extension   Uriel Carrasquilla
• [Asterisk-Users] SIP Phones Features   Uriel Carrasquilla
• [Asterisk-Users] SIP Phones / Asterisk Cook-Book   Uriel Carrasquilla
• [Asterisk-Users] Zapata not required??   Uriel Carrasquilla
• [Asterisk-Users] SIP call logging, called number not logged   Uriel Carrasquilla
• [Asterisk-Users] Dynamic IP Addrress Work Around   Uriel Carrasquilla
• [Asterisk-Users] vmail.agi   Uriel Carrasquilla
• [Asterisk-Users] vmail.agi   Uriel Carrasquilla
• [Asterisk-Users] DynExtenDB   Uriel Carrasquilla
• [Asterisk-Users] DynExtenDB   Uriel Carrasquilla
• [Asterisk-Users] Free Windows SIP Softclient S/W or component required!   Anil Chandra
• [Asterisk-Users] Vonage with Asterisk   Anil Chandra
• [Asterisk-Users] Asterisk-Redhat 9 install guide.   Vincent Choy
• [Asterisk-Users] No audio in Linejack   Lubomir Christov
• [Asterisk-Users] SIP support status   Lubomir Christov
• [Asterisk-Users] Anyone else lose iconnecthere service in recent CVS?   Lubomir Christov
• [Asterisk-Users] Anyone using Asterisks and a Quicknet Lineja ck in the UK?   Lubomir Christov
• [Asterisk-Users] Running Asterisk on non-intel hardware   Marco Cintolesi
• [Asterisk-Users] Running Asterisk on non-intel hardware - SOLVED   Marco Cintolesi
• [Asterisk-Users] ISDN and voice   Marco Cintolesi
• [Asterisk-Users] X100P problem   Jorge Cisneros
• [Asterisk-Users] Vonage with Asterisk   Ben Clark
• [Asterisk-Users] Running Asterisk on non-intel hardware   Jamin W. Collins
• [Asterisk-Users] SIP Proxy   Jamin W. Collins
• [Asterisk-Users] Best IP phone?   Jamin W. Collins
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   Jamin W. Collins
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   Jamin W. Collins
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   Jamin W. Collins
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   Jamin W. Collins
• [Asterisk-Users] Building own SIP CLient   Jamin W. Collins
• [Asterisk-Users] IP hard phone devices (grandstream)   John Congdon
• [Asterisk-Users] Setting up ADSI   John Congdon
• Fwd: [Asterisk-Users] Setting up ADSI   John Congdon
• [Asterisk-Users] Deependencies problem   Steven Critchfield
• [Asterisk-Users] Up to 8 lines?   Steven Critchfield
• [Asterisk-Users] low-cost * (newbie question)   Steven Critchfield
• [Asterisk-Users] SIP & Adtech 150   Steven Critchfield
• [Asterisk-Users] Setting up PABX   Steven Critchfield
• [Asterisk-Users] The best simple solution (newbee)   Steven Critchfield
• [Asterisk-Users] Setting up PABX   Steven Critchfield
• [Asterisk-Users] Line is stuck off hook...   Steven Critchfield
• [Asterisk-Users] Line is stuck off hook...   Steven Critchfield
• [Asterisk-Users] Line is stuck off hook...   Steven Critchfield
• [Asterisk-Users] low-cost * (newbie question)   Steven Critchfield
• [Asterisk-Users] low-cost * (newbie question)   Steven Critchfield
• [Asterisk-Users] Buying channel banks online?   Steven Critchfield
• [Asterisk-Users] Buying channel banks online?   Steven Critchfield
• [Asterisk-Users] "Broken Dial Tone" for voice mail?   Steven Critchfield
• [Asterisk-Users] CE certification for Europe   Steven Critchfield
• [Asterisk-Users] CE certification for Europe   Steven Critchfield
• [Asterisk-Users] MP3player problem   Steven Critchfield
• [Asterisk-Users] CE certification for Europe   Steven Critchfield
• [Asterisk-Users] "-p"   Steven Critchfield
• [Asterisk-Users] Priority usage: absolute sequential vs. sequential   Steven Critchfield
• [Asterisk-Users] Channel Driver Architecture   Steven Critchfield
• [Asterisk-Users] Chan_capi errors compil   Steven Critchfield
• [Asterisk-Users] Updating from CVS!   Steven Critchfield
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   Steven Critchfield
• [Asterisk-Users] Conferences   Steven Critchfield
• [Asterisk-Users] Error compiling in RedHat 9   Steven Critchfield
• [Asterisk-Users] Channel Banks   Steven Critchfield
• [Asterisk-Users] Channel Banks   Steven Critchfield
• [Asterisk-Users] Channel Banks   Steven Critchfield
• [Asterisk-Users] Channel Banks   Steven Critchfield
• [Asterisk-Users] Where is zttool?   Steven Critchfield
• [Asterisk-Users] external trigger on caller id   Steven Critchfield
• [Asterisk-Users] external trigger on caller id   Steven Critchfield
• [Asterisk-Users] Weird AGI/X100P behavior   Steven Critchfield
• [Asterisk-Users] Modem connections   Steven Critchfield
• [Asterisk-Users] dial plan etc via tftp (waas Voice-mail App)   Steven Critchfield
• [Asterisk-Users] Playback application   Steven Critchfield
• [Asterisk-Users] Parametric "sample.call?"   Steven Critchfield
• [Asterisk-Users] * kernel panic   Steven Critchfield
• [Asterisk-Users] conf advices needed   Steven Critchfield
• [Asterisk-Users] libpri functionality   Steven Critchfield
• [Asterisk-Users] libpri functionality   Steven Critchfield
• [Asterisk-Users] Trouble installing   Steven Critchfield
• [Asterisk-Users] dialplan problems cvs 04-15-03   Steven Critchfield
• [Asterisk-Users] X100P question   Steven Critchfield
• [Asterisk-Users] RE: iLBC   Steven Critchfield
• [Asterisk-Users] meetme config   Steven Critchfield
• [Asterisk-Users] Redhat vs Mandrake.   Steven Critchfield
• [Asterisk-Users] Redhat vs Mandrake.   Steven Critchfield
• [Asterisk-Users] Call screening   Steven Critchfield
• [Asterisk-Users] Zap channel to collect dialled number # terminated?   Steven Critchfield
• [Asterisk-Users] SS7   Steven Critchfield
• [Asterisk-Users] PBX Console   Steven Critchfield
• [Asterisk-Users] PBX Console   Steven Critchfield
• [Asterisk-Users] Call Queue Manager and DID Digits   Steven Critchfield
• [Asterisk-Users] Rackmount PC hardware suggestions   Steven Critchfield
• [Asterisk-Users] PBX Console   Steven Critchfield
• [Asterisk-Users] Anyone using Asterisks and a Quicknet Lineja ck in the UK?   Steven Critchfield
• [Asterisk-Users] Rackmount PC hardware suggestions   Steven Critchfield
• Wait doesn't read DTMF? Was Re: [Asterisk-Users] Collecting dialed digits   Steven Critchfield
• [Asterisk-Users] Meetme application en 0.4.0   Steven Critchfield
• [Asterisk-Users] No ring on extensions   Steven Critchfield
• [Asterisk-Users] new cisco VoIP phones   Steven Critchfield
• [Asterisk-Users] Re: Dialing using X100P   Steven Critchfield
• [Asterisk-Users] No ring on extensions   Steven Critchfield
• [Asterisk-Users] Re: Dialing using X100P   Steven Critchfield
• [Asterisk-Users] Re: Dialing using X100P   Steven Critchfield
• [Asterisk-Users] zaptel.conf and zapata.conf for Carrier Access   Steven Critchfield
• [Asterisk-Users] zaptel.conf and zapata.conf for Carrier Access   Steven Critchfield
• [Asterisk-Users] Hardware and E1   Steven Critchfield
• [Asterisk-Users] Mosix?   Steven Critchfield
• [Asterisk-Users] how many voicemail box asterisk can support   Steven Critchfield
• [Asterisk-Users] TDM10B problem   Steven Critchfield
• [Asterisk-Users] PRI Setup   Steven Critchfield
• [Asterisk-Users] iconnect and incoming DTMF   DA
• [Asterisk-Users] Unknown RTP codec 101 received   DA
• [Asterisk-Users] isdn config   Marian Danisek
• [Asterisk-Users] isdn config   Marian Danisek
• [Asterisk-Users] chan_h323 problems   Mark Darvill
• [Asterisk-Users] chan_h323 problems....   Mark Darvill
• [Asterisk-Users] chan_h323 problems....   Mark Darvill
• [Asterisk-Users] ATA186: "Call/Leg Transaction Doesn't Exist" on local call   Stephen Davies
• [Asterisk-Users] Bug? * not correctly honouring tag on To?   Stephen Davies
• [Asterisk-Users] SIP Testing   Stephen Davies
• [Asterisk-Users] SIP Testing   Stephen Davies
• [Asterisk-Users] SIP Testing   Stephen Davies
• [Asterisk-Users] SIP Testing   Stephen Davies
• [Asterisk-Users] SIP Testing   Stephen Davies
• [Asterisk-Users] SIP Testing   Stephen Davies
• [Asterisk-Users] Asterisk dies after 3-6 hours of operation. Help!   Stephen Davies
• [Asterisk-Users] Outgoing SIP Registration Fixed   Stephen Davies
• [Asterisk-Users] sip registration problems   Stephen Davies
• [Asterisk-Users] sip registration problems   Stephen Davies
• [Asterisk-Users] sip registration problems   Stephen Davies
• [Asterisk-Users] sip registration problems   Stephen Davies
• [Asterisk-Users] How to change login for iaxtel.com IAX?   Stephen Davies
• [Asterisk-Users] sip registration problems   Stephen Davies
• [Asterisk-Users] SIP and music on hold   Stephen Davies
• [Asterisk-Users] Any special kernel patches for good S100U performance?   Stephen Davies
• [Asterisk-Users] Zap channel to collect dialled number # terminated?   Stephen Davies
• [Asterisk-Users] Anyone else lose iconnecthere service in recent CVS?   Stephen Davies
• [Asterisk-Users] Anyone using Asterisks and a Quicknet Lineja ck in the UK?   Stephen Davies
• [Asterisk-Users] Packet8 New Area Codes and Rate Centers   Stephen Davies
• [Asterisk-Users] Registering with Pulver???   Stephen Davies
• [Asterisk-Users] Recording Prompts   David Davis
• [Asterisk-Users] Asterisk and SNOM 200   David Davis
• [Asterisk-Users] Cisco 7960 services and directory xml (CMXML, BTXML)   David Davis
• [Asterisk-Users] Redhat vs Mandrake.   Jean-Pierre Denis
• [Asterisk-Users] SIP support status   Francois Dessart
• [Asterisk-Users] SIP support status   Francois Dessart
• [Asterisk-Users] Xten / SIP Phones compared to GnoPhones   Francois Dessart
• [Asterisk-Users] Recording Prompts   Mike Diehl
• [Asterisk-Users] Voice Modem Support?   Mike Diehl
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   Kalin Dikov
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   Kalin Dikov
• [Asterisk-Users] asterisk online indicator   Shawn L. Djernes
• [Asterisk-Users] PBX Console   Shawn L. Djernes
• [Asterisk-Users] PBX Console   Shawn L. Djernes
• [Asterisk-Users] SIP to MGCP transcoding   Michael Van Donselaar
• [Asterisk-Users] SIP to MGCP transcoding   Michael Van Donselaar
• [Asterisk-Users] Xten - Free windows SIP client   Michael Van Donselaar
• [Asterisk-Users] No ring on extensions   Surfer Dude
• [Asterisk-Users] No ring on extensions   Surfer Dude
• [Asterisk-Users] No ring on extensions   Surfer Dude
• [Asterisk-Users] MGCP   Ray Dzek
• [Asterisk-Users] extension exited non-zero   Darrell Eldridge
• [Asterisk-Users] T100P Incoming Calls Drop   Matthew Farley
• [Asterisk-Users] T100P Incoming Calls Drop   Matthew Farley
• [Asterisk-Users] T100P Incoming Calls Drop   Matthew Farley
• [Asterisk-Users] T100P Incoming Calls Drop   Matthew Farley
• [Asterisk-Users] Rackmount PC hardware suggestions   Matthew Farley
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   Matthew Farley
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   Matthew Farley
• [Asterisk-Users] howto reduce number of rings ?   Dan Fernandez
• [Asterisk-Users] problem with X100P and wcfxo   Dan Fernandez
• [Asterisk-Users] Problems with apps (ie Voicemail, Meetme, etc) and SIP,g723   Dan Fernandez
• [Asterisk-Users] Grandstream BudgeTone 100   Dan Fernandez
• [Asterisk-Users] bandwith calculation   Dan Fernandez
• [Asterisk-Users] bandwith calculation   Dan Fernandez
• [Asterisk-Users] Does anyone have the Cisco 7910 IP Phone working with Asterisk?   Bruce Ferrell
• [Asterisk-Users] RE: [Asterisk-Dev] Several patches, includin g recording and music -on-hold   Fettahlioglu, Mahmut
• [Asterisk-Users] RE: [Asterisk-Dev] Several patches, includin g recording and music -on-hold   Fettahlioglu, Mahmut
• [Asterisk-Users] Problems Calling Toll-free number   Stefano Finetti
• [Asterisk-Users] Hardware requirements   Stefano Finetti
• [Asterisk-Users] isdn config   Stefano Finetti
• [Asterisk-Users] ISDN4Linux problems   Stefano Finetti
• [Asterisk-Users] I4l channel problems with latest CVS   Stefano Finetti
• [Asterisk-Users] I4l channel problems with latest CVS   Stefano Finetti
• [Asterisk-Users] I4l channel problems with latest CVS   Stefano Finetti
• [Asterisk-Users] Collecting dialed digits   Stefano Finetti
• [Asterisk-Users] Collecting dialed digits   Stefano Finetti
• [Asterisk-Users] Internet Dial-in security questions   Stefano Finetti
• [Asterisk-Users] Does anyone have the Cisco 7910 IP Phone working with Asterisk?   Stefano Finetti
• [Asterisk-Users] Does anyone have the Cisco 7910 IP Phone working with Asterisk?   Stefano Finetti
• [Asterisk-Users] Voice Modem Support?   Stefano Finetti
• [Asterisk-Users] Creating a phone channel   Stefano Finetti
• [Asterisk-Users] Creating a phone channel   Stefano Finetti
• [Asterisk-Users] Re: Building own SIP CLient   Ross Finlayson
• [Asterisk-Users] ActionTec with Asterisk   Jim Flagg
• [Asterisk-Users] Xten - Free windows SIP client   Jim Flagg
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   Jim Flagg
• [Asterisk-Users] X100P test help   Jorge Cisneros Flores
• [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?   Mathew Frank
• [Asterisk-Users] CE certification for Europe   Mathew Frank
• [Asterisk-Users] SIP Testing   Mathew Frank
• [Asterisk-Users] Should List be Moderated?   Mathew Frank
• [Asterisk-Users] Sip & Intercom   Christoph Frei
• [Asterisk-Users] Sip & Intercom   Christoph Frei
• [Asterisk-Users] TDM400 question   Ron Gage
• [Asterisk-Users] [OT] - Spectrum Technologies Channel bank   Ron Gage
• [Asterisk-Users] SIP Testing   Gary
• [Asterisk-Users] IP hard phone devices (grandstream)   Gary
• [Asterisk-Users] Globals ??   Gary
• [Asterisk-Users] Blocking off a line ?   Gary
• [Asterisk-Users] Busy out lines...   Gary
• [Asterisk-Users] Zhone and line length problems ??   Gary
• [Asterisk-Users] Zhone and line length problems ??   Gary
• [Asterisk-Users] Zhone and line length problems ??   Gary
• [Asterisk-Users] Callerid and tone zones ?   Gary
• [Asterisk-Users] PBX Console   Gary
• [Asterisk-Users] PBX Console   Gary
• [Asterisk-Users] Callerid and tone zones ?   Gary
• [Asterisk-Users] Music on hold ?   Gary
• [Asterisk-Users] Zhone and line length problems ??   Gary
• [Asterisk-Users] Various {was} ADSI Analogue phones   Gary
• [Asterisk-Users] Extened caller-id (to include called number)   Gary
• [Asterisk-Users] Three way calling   Gary
• [Asterisk-Users] Festival ?   Gary
• [Asterisk-Users] Radius app ?   Gary
• [Asterisk-Users] Dialing using X100P   Gary
• [Asterisk-Users] DynExtenDB Module for iax.conf ?   Gary
• [Asterisk-Users] Re: Dialing using X100P   Gary
• [Asterisk-Users] LineJack PSTN Out   Gary
• [Asterisk-Users] LineJack PSTN Out   Gary
• [Asterisk-Users] Whats ENUM??   Gary
• [Asterisk-Users] Creating a phone channel   Gary
• [Asterisk-Users] ISDN - Dialout MSN setting ??   Gary
• [Asterisk-Users] Creating a phone channel   Gary
• [Asterisk-Users] Creating a phone channel   Gary
• [Asterisk-Users] DynExtenDB   Gary
• FW: [Asterisk-Users] DynExtenDB   Gary
• [Asterisk-Users] unsubscribe   MOSCATO Giampiero
• [Asterisk-Users] RH9 won't finish Asterisk make   Jerry Gibson
• [Asterisk-Users] RH9 won't finish Asterisk make   Jerry Gibson
• [Asterisk-Users] Xten / SIP Phones compared to GnoPhones   Jerry Gibson
• [Asterisk-Users] Zapata not required??   Jerry Gibson
• [Asterisk-Users] Deependencies problem   James Golovich
• [Asterisk-Users] FAX over IAX   James Golovich
• [Asterisk-Users] MP3player problem   James Golovich
• [Asterisk-Users] T100P Incoming Calls Drop   James Golovich
• [Asterisk-Users] dialed number notify at invalid dial situation   James Golovich
• [Asterisk-Users] What driver for the TDMX0B cards?   James Golovich
• [Asterisk-Users] callerid send to shell script with system command   James Golovich
• [Asterisk-Users] change voicemail.conf, how to make it work   James Golovich
• [Asterisk-Users] OT - Hardware needed   Eduardo Goncalves
• [Asterisk-Users] E1 card with R2 signaling   Eduardo Goncalves
• [Asterisk-Users] E1 card with R2 signaling   Eduardo Goncalves
• [Asterisk-Users] Dialogic Card   Eduardo Goncalves
• [Asterisk-Users] Hardware and E1   Eduardo Goncalves
• [Asterisk-Users] Hardware and E1   Eduardo Goncalves
• [Asterisk-Users] Anyone using Asterisks and a Quicknet Linejack in the UK?   Colin Gorrie
• [Asterisk-Users] PPP by default in zapata   Adam Goryachev
• [Asterisk-Users] Music on Hold for SIP   Adam Goryachev
• [Asterisk-Users] SIP Testing   Adam Goryachev
• [Asterisk-Users] IP hard phone devices (grandstream)   Adam Goryachev
• [Asterisk-Users] I4l channel problems with latest CVS   Adam Goryachev
• [Asterisk-Users] exited non-zero   Adam Goryachev
• [Asterisk-Users] play message on bridged call   Adam Goryachev
• [Asterisk-Users] Asterisk Crashes   Adam Goryachev
• [Asterisk-Users] RE: [Asterisk-Dev] Several patches, including recording and music -on-hold   Adam Goryachev
• [Asterisk-Users] New 4port FXS   Adam Goryachev
• [Asterisk-Users] Running Asterisk on non-intel hardware   Adam Goryachev
• [Asterisk-Users] Running Asterisk on non-intel hardware   Adam Goryachev
• [Asterisk-Users] Taking a call for someone elses extension from my extension   Adam Goryachev
• [Asterisk-Users] Eicon BRI-card   Adam Goryachev
• [Asterisk-Users] Busy out lines...   Adam Goryachev
• [Asterisk-Users] Callerid and tone zones ?   Adam Goryachev
• [Asterisk-Users] PBX Console   Adam Goryachev
• [Asterisk-Users] ADSI Analogue phones   Adam Goryachev
• [Asterisk-Users] Revised example configurations available   Adam Goryachev
• [Asterisk-Users] enabling G.729 for SIP   Jim Gottlieb
• [Asterisk-Users] Re: SIP and ATA186   Jim Gottlieb
• [Asterisk-Users] Hangup detection problems   Jim Gottlieb
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Jim Gottlieb
• [Asterisk-Users] Two Rings   Jim Gottlieb
• [Asterisk-Users] Re: no audio after many transfers   Jim Gottlieb
• [Asterisk-Users] callerid send to shell script with system command   Michael Graff
• [Asterisk-Users] callerid send to shell script with system command   Michael Graff
• [Asterisk-Users] first few seconds of greeting cut-off   Michael Graff
• [Asterisk-Users] how many voicemail box asterisk can support   Michael Graff
• [Asterisk-Users] change voicemail.conf, how to make it work   Michael Graff
• [Asterisk-Users] FW: ipDialog Ethernet SIP Phone $199   Clay Graner
• [Asterisk-Users] Xten - Free windows SIP client   Clay Graner
• [Asterisk-Users] GnuGK -> Asterisk problem   Niclas Gustafsson
• [Asterisk-Users] Buying channel banks online?   Anton Hajducek
• [Asterisk-Users] Channel Banks   Anton Hajducek
• [Asterisk-Users] Multiport FXO?   Abdul Hakeem
• [Asterisk-Users] "Broken Dial Tone" for voice mail?   Drew Hamilton
• [Asterisk-Users] How many REN can the S100U drive?   Drew Hamilton
• [Asterisk-Users] how to register * at FWD from behind NAT   Ask Bjoern Hansen
• [Asterisk-Users] How to change login for iaxtel.com IAX?   Ask Bjoern Hansen
• [Asterisk-Users] dial plan etc via tftp (waas Voice-mail App)   Ask Bjoern Hansen
• [Asterisk-Users] dial plan etc via tftp (waas Voice-mail App)   Ask Bjoern Hansen
• [Asterisk-Users] Asterisk Crashes   Ask Bjoern Hansen
• [Asterisk-Users] DTMF tones not long enough   Ask Bjoern Hansen
• [Asterisk-Users] Disable callerid to pass incoming calls through faster - and hangup woes   Ask Bjoern Hansen
• [Asterisk-Users] Hidden voicemail feature - send message to pager   Ask Bjoern Hansen
• [Asterisk-Users] Hmmm. RJ-45 on TDMx0B card?   Ask Bjoern Hansen
• [Asterisk-Users] German voicemail prompts, anybody?   Ask Bjoern Hansen
• [Asterisk-Users] SS7   Tais M. Hansen
• [Asterisk-Users] Serious problem with z-plex 10   John Harragin
• [Asterisk-Users] Line is stuck off hook...   John Harragin
• [Asterisk-Users] Line is stuck off hook...   John Harragin
• [Asterisk-Users] FAX over IAX   John Harragin
• [Asterisk-Users] FAX over IAX   John Harragin
• [Asterisk-Users] FAX over IAX   John Harragin
• [Asterisk-Users] AT&T T1 Cable Needed!   John Harragin
• [Asterisk-Users] AT&T T1 Cable Needed!   John Harragin
• [Asterisk-Users] AT&T T1 Cable Needed!   John Harragin
• [Asterisk-Users] Priority usage: absolute sequential vs. sequential   John Harragin
• [Asterisk-Users] Priority usage: absolute sequential vs. sequential   John Harragin
• [Asterisk-Users] Priority usage: absolute sequential vs.sequential   John Harragin
• [Asterisk-Users] SIP improvements   John Harragin
• [Asterisk-Users] T1 vs PRI   John Harragin
• [Asterisk-Users] SIP support status   John Harragin
• [Asterisk-Users] How to pickup incoming calls immediately?   John Harragin
• [Asterisk-Users] Tor2 with em_w (or em) signalling pickup behavior?   John Harragin
• [Asterisk-Users] Tor2 with em_w (or em) signalling pickup behavior?   John Harragin
• [Asterisk-Users] German voicemail prompts, anybody?   John Harragin
• [Asterisk-Users] Anyone tried premisys channel bank?   John Harragin
• [Asterisk-Users] CE certification for Europe   Vlasis Hatzistavrou
• [Asterisk-Users] CE certification for Europe   Vlasis Hatzistavrou
• [Asterisk-Users] SS7   Vlasis Hatzistavrou
• [Asterisk-Users] sip registration problems   Juha Heinanen
• [Asterisk-Users] sip registration problems   Juha Heinanen
• [Asterisk-Users] sip registration problems   Juha Heinanen
• [Asterisk-Users] sip registration problems   Juha Heinanen
• [Asterisk-Users] Error with dot when making progdocs   Juha Heinanen
• [Asterisk-Users] sip registration problems   Juha Heinanen
• [Asterisk-Users] sip registration problems   Juha Heinanen
• [Asterisk-Users] NAT and 7960   Juha Heinanen
• [Asterisk-Users] sip registration problems   Juha Heinanen
• [Asterisk-Users] Asterisk and SNOM 200   Juha Heinanen
• [Asterisk-Users] iLBC   Juha Heinanen
• [Asterisk-Users] Asterisk and SNOM 200   Juha Heinanen
• [Asterisk-Users] voice problems on console   Juha Heinanen
• [Asterisk-Users] RE: iLBC   Juha Heinanen
• [Asterisk-Users] RE: iLBC   Juha Heinanen
• [Asterisk-Users] RE: iLBC   Juha Heinanen
• [Asterisk-Users] RE: iLBC   Juha Heinanen
• [Asterisk-Users] RE: iLBC   Juha Heinanen
• [Asterisk-Users] does * support digest authentication of sip clients   Juha Heinanen
• [Asterisk-Users] change voicemail.conf, how to make it work   Juha Heinanen
• [Asterisk-Users] change voicemail.conf, how to make it work   Juha Heinanen
• [Asterisk-Users] change voicemail.conf, how to make it work   Juha Heinanen
• [Asterisk-Users] Anyone using Asterisks and a Quicknet Lineja ck in the UK?   Juan Rodriguez Hervella
• [Asterisk-Users] segmentation fault at voicemail   NOHATA Hi
• [Asterisk-Users] Caller ID Issues   Matthew S. Hill
• Wait doesn't read DTMF? Was Re: [Asterisk-Users] Collecting dialed digits   James Hines
• Wait doesn't read DTMF? Was Re: [Asterisk-Users] Collecting dialed digits   James Hines
• [Asterisk-Users] Can you invoke an app before dialtone?   James Hines
• [Asterisk-Users] * process stops, after a few hours of uptime   Frank Hoonhout
• [Asterisk-Users] Asterisk dies after 3-6 hours of operation. Help!   Frank Hoonhout
• [Asterisk-Users] Sip & Intercom   Frank Hoonhout
• [Asterisk-Users] Outgoing SIP Registration Fixed   Frank Hoonhout
• [Asterisk-Users] FYI - New 79xx SIP image - 4.4.0 - Fixes premature CANCEL message with FWD through *   Frank Hoonhout
• [Asterisk-Users] zapata busy detect   Frank Hoonhout
• [Asterisk-Users] bandwith calculation   Frank Hoonhout
• [Asterisk-Users] Collecting dialed digits   Frank Hoonhout
• [Asterisk-Users] Collecting dialed digits   Frank Hoonhout
• [Asterisk-Users] Registering with Pulver???   Frank Hoonhout
• [Asterisk-Users] new cisco VoIP phones   Frank Hoonhout
• [Asterisk-Users] gastman   Frank Hoonhout
• [Asterisk-Users] gastman   Frank Hoonhout
• [Asterisk-Users] gastman   Frank Hoonhout
• [Asterisk-Users] gastman   Frank Hoonhout
• [Asterisk-Users] gastman   Frank Hoonhout
• [Asterisk-Users] regarding the new tdm?0b cards (another newbie question)   Luke Howard
• [Asterisk-Users] how many voicemail box asterisk can support   Yan Huang
• [Asterisk-Users] change voicemail.conf, how to make it work   Yan Huang
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   Klaus Hueske
• [Asterisk-Users] Using Asterisk on Call Center   Stephen Hui
• [Asterisk-Users] Sip & Intercom   James O. Sizemore III
• [Asterisk-Users] Sip & Intercom   James O. Sizemore III
• [Asterisk-Users] IP hard phone devices (grandstream)   Tomaz Izanc
• [Asterisk-Users] external trigger on caller id   Tomaz Izanc
• [Asterisk-Users] external trigger on caller id   Tomaz Izanc
• [Asterisk-Users] timeout music on hold or ring tone   Tomaz Izanc
• [Asterisk-Users] timeout music on hold or ring tone   Tomaz Izanc
• [Asterisk-Users] zapata busy detect   Tomaz Izanc
• [Asterisk-Users] zapata busy detect   Tomaz Izanc
• [Asterisk-Users] call queues   Tomaz Izanc
• [Asterisk-Users] call queues   Tomaz Izanc
• [Asterisk-Users] new cisco VoIP phones   Tomaz Izanc
• [Asterisk-Users] callerid send to shell script with system command   Tomaz Izanc
• [Asterisk-Users] dial even caller hungup   Tomaz Izanc
• [Asterisk-Users] Suppport for Asterisk, asterisk-h323 package and Voice Mail   MARTINEZ OGALDE RICARDO JAVIER
• [Asterisk-Users] X100P question   JKNUTSEN at UP.COM
• [Asterisk-Users] X100P question   JKNUTSEN at UP.COM
• [Asterisk-Users] Collecting dialed digits   JKNUTSEN at UP.COM
• [Asterisk-Users] Meetme conference capacity   JKNUTSEN at UP.COM
• [Asterisk-Users] Sound files   JKNUTSEN at UP.COM
• [Asterisk-Users] Asterisk-Redhat 9 install guide.   Brian Johnson
• [Asterisk-Users] FAX over IAX   Brian Jones
• [Asterisk-Users] CE certification for Europe   Klaus-Peter Junghanns
• [Asterisk-Users] * on openmosix   Klaus-Peter Junghanns
• [Asterisk-Users] CE certification for Europe   Klaus-Peter Junghanns
• [Asterisk-Users] FS: Cisco DT-24+, Dialogic D/240SC/T1, Natural Microsystem AG-T1   Klaus-Peter Junghanns
• [Asterisk-Users] isdn config   Klaus-Peter Junghanns
• [Asterisk-Users] Eicon driver?   Klaus-Peter Junghanns
• [Asterisk-Users] Re: Asterisk-Redhat 9 install guide.   Ajit M Kallingal
• [Asterisk-Users] OS X support?   Steve Kann
• [Asterisk-Users] OS X support?   Roy Sigurd Karlsbakk
• [Asterisk-Users] Is the CVS server working??   Roy Sigurd Karlsbakk
• [Asterisk-Users] can't use both controllers...   Roy Sigurd Karlsbakk
• [Asterisk-Users] * kernel panic   Roy Sigurd Karlsbakk
• [Asterisk-Users] * kernel panic   Roy Sigurd Karlsbakk
• [Asterisk-Users] core dump in capi somewhere   Roy Sigurd Karlsbakk
• [Asterisk-Users] new mgcp patch errors   Roy Sigurd Karlsbakk
• [Asterisk-Users] Modem connections   Jim Karutis
• [Asterisk-Users] Modem connections   Jim Karutis
• [Asterisk-Users] SIP and music on hold   Jim Karutis
• [Asterisk-Users] SIP and music on hold   Jim Karutis
• [Asterisk-Users] SIP and music on hold   Jim Karutis
• [Asterisk-Users] SIP and music on hold   Jim Karutis
• [Asterisk-Users] FAX over IAX   Ben Klang
• [Asterisk-Users] Rackmount PC hardware suggestions   Ben Klang
• [Asterisk-Users] unsubscribe   Gary Knapper
• [Asterisk-Users] Line is stuck off hook...   Gene Kochanowsky
• [Asterisk-Users] Line is stuck off hook...   Gene Kochanowsky
• [Asterisk-Users] Line is stuck off hook...   Gene Kochanowsky
• [Asterisk-Users] Line is stuck off hook...   Gene Kochanowsky
• [Asterisk-Users] CE certification for Europe   Gene Kochanowsky
• [Asterisk-Users] * with SIP behind NAT   Tjardick van der Kraan
• [Asterisk-Users] sip.conf & Pingtel   Tjardick van der Kraan
• [Asterisk-Users] DTMF tones not long enough   Tjardick van der Kraan
• [Asterisk-Users] howto (Also SIP VIA header)   Tjardick van der Kraan
• [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #239 - 13 msgs   Hemant Kumar
• [Asterisk-Users] Predictive dialer   Hemant Kumar
• [Asterisk-Users] Fw: Predictive dialer   Hemant Kumar
• [Asterisk-Users] Fw: Predictive dialer   Hemant Kumar
• [Asterisk-Users] Fw: Predictive dialer   Hemant Kumar
• [Asterisk-Users] Fw: Predictive dialer   Hemant Kumar
• [Asterisk-Users] Fw: Predictive dialer   Hemant Kumar
• [Asterisk-Users] Fw: Predictive dialer   Hemant Kumar
• [Asterisk-Users] Fw: Predictive dialer   Hemant Kumar
• [Asterisk-Users] Fw: Predictive dialer   Hemant Kumar
• [Asterisk-Users] Conferences   Hemant Kumar
• [Asterisk-Users] Conferences   Hemant Kumar
• [Asterisk-Users] play message on bridged call   Hemant Kumar
• [Asterisk-Users] Using Asterisk on Call Center   Hemant Kumar
• [Asterisk-Users] [Asterisk-Users]   Hemant Kumar
• [Asterisk-Users] [Asterisk-Users]   Hemant Kumar
• [Asterisk-Users] [Asterisk-Users]   Hemant Kumar
• [Asterisk-Users] Asterisk SRPM and RPMs   Hemant Kumar
• [Asterisk-Users] Whats ENUM??   Hemant Kumar
• [Asterisk-Users] Asterisk Crashes   Scott Lambert
• [Asterisk-Users] PBX Console   Scott Lambert
• [Asterisk-Users] German voicemail prompts, anybody?   Siggi Langauf
• [Asterisk-Users] Hidden voicemail feature - send message to pager   Robert Hajime Lanning
• [Asterisk-Users] LineJack PSTN Out   Robert Hajime Lanning
• [Asterisk-Users] Documentation   Randal Law
• [Asterisk-Users] CE certification for Europe   Gregg Lebovitz
• [Asterisk-Users] SIP Testing   Gregg Lebovitz
• [Asterisk-Users] SIP improvements   Gregg Lebovitz
• [Asterisk-Users] IP hard phone devices (grandstream)   Gregg Lebovitz
• [Asterisk-Users] IP hard phone devices (grandstream)   Gregg Lebovitz
• [Asterisk-Users] IP hard phone devices (grandstream)   Gregg Lebovitz
• [Asterisk-Users] Channel Banks   Gregg Lebovitz
• [Asterisk-Users] Channel Banks   Gregg Lebovitz
• [Asterisk-Users] Channel Banks   Gregg Lebovitz
• [Asterisk-Users] still looking for iconnect configuration help   Gregg Lebovitz
• [Asterisk-Users] Anyone else lose iconnecthere service in recent CVS?   Gregg Lebovitz
• [Asterisk-Users] Problems Calling Toll-free number   Tilghman Lesher
• [Asterisk-Users] Line is stuck off hook...   Tilghman Lesher
• [Asterisk-Users] FAX over IAX   Tilghman Lesher
• [Asterisk-Users] "Broken Dial Tone" for voice mail?   Tilghman Lesher
• [Asterisk-Users] howto reduce number of rings ?   Tilghman Lesher
• [Asterisk-Users] CE certification for Europe   Tilghman Lesher
• [Asterisk-Users] Call completion/error codes and extensions.conf call flow   Tilghman Lesher
• [Asterisk-Users] T100P Incoming Calls Drop   Tilghman Lesher
• [Asterisk-Users] How to make an X100P answer only one distinctive ring cadence?   Tilghman Lesher
• [Asterisk-Users] Recording Prompts   Tilghman Lesher
• [Asterisk-Users] Recording Prompts   Tilghman Lesher
• [Asterisk-Users] Running Asterisk on non-intel hardware   Tilghman Lesher
• [Asterisk-Users] Weirdness on "hookflash call pickup"   Tilghman Lesher
• [Asterisk-Users] Voice Mail Re-compile   Tilghman Lesher
• [Asterisk-Users] RH9 won't finish Asterisk make   Tilghman Lesher
• [Asterisk-Users] Globals ??   Tilghman Lesher
• [Asterisk-Users] Redhat vs Mandrake.   Tilghman Lesher
• [Asterisk-Users] module chan_sip.o failed to load   Tilghman Lesher
• [Asterisk-Users] Unable to call H323 phones via asterisk   Tilghman Lesher
• [Asterisk-Users] Unable to call H323 phones via asterisk   Tilghman Lesher
• [Asterisk-Users] Collecting dialed digits   Tilghman Lesher
• [Asterisk-Users] Collecting dialed digits   Tilghman Lesher
• [Asterisk-Users] Zhone + Digium T1 bug (?)   Tilghman Lesher
• [Asterisk-Users] Three way calling   Tilghman Lesher
• [Asterisk-Users] Dialing using X100P   Tilghman Lesher
• [Asterisk-Users] No ring on extensions   Tilghman Lesher
• [Asterisk-Users] Mosix?   Tilghman Lesher
• [Asterisk-Users] change voicemail.conf, how to make it work   Tilghman Lesher
• [Asterisk-Users] Channel Banks   Judson Lester
• [Asterisk-Users] external trigger on caller id   Judson Lester
• [Asterisk-Users] Channel Banks   Judson Lester
• [Asterisk-Users] external trigger on caller id   Judson Lester
• [Asterisk-Users] TDM400P w/ less than 4   Judson Lester
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Judson Lester
• [Asterisk-Users] MP3player problem   Tamas Levente
• [Asterisk-Users] MP3player problem   Tamas Levente
• [Asterisk-Users] MP3player problem   Tamas Levente
• [Asterisk-Users] PPP by default in zapata   Tamas Levente
• [Asterisk-Users] Zap asterisk prob   Tamas Levente
• [Asterisk-Users] * kernel panic   Tamas Levente
• [Asterisk-Users] * kernel panic   Tamas Levente
• [Asterisk-Users] * kernel panic   Tamas Levente
• [Asterisk-Users] Asterisk and SNOM 200   Tamas Levente
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   Tamas Levente
• [Asterisk-Users] Zapata not required??   Greg Lim
• [Asterisk-Users] Zapata not required??   Greg Lim
• [Asterisk-Users] Account code on SIP   George Lin
• [Asterisk-Users] preserve hostname for INVITE request-uri   Howard Lin
• [Asterisk-Users] preserve hostname for INVITE request-uri   Howard Lin
• [Asterisk-Users] ztdummy problems   Shawn Lipman
• [Asterisk-Users] Locked out of Cisco 7940   Pavel Litvinenko
• [Asterisk-Users] Cisco 7940 conversion software??   David Liu
• [Asterisk-Users] Cisco 7940 conversion software??   David Liu
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   John Lodden
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   John Lodden
• [Asterisk-Users] Lost Cisco ata to * cheat sheet.   Alex Lopez
• [Asterisk-Users] mpg123 hangs on close, but plays fine.   Alex Lopez
• [Asterisk-Users] mpg123 hangs on close, but plays fine.   Alex Lopez
• [Asterisk-Users] Two problems: Drops from conference and digital garbage in delay test   Leif Madsen
• [Asterisk-Users] Fw: Predictive dialer   C. Maj
• [Asterisk-Users] libpri functionality   C. Maj
• [Asterisk-Users] libpri functionality   C. Maj
• [Asterisk-Users] netmeeting dial   Paulo H. Mannheimer
• [Asterisk-Users] netmeeting dial   Paulo H. Mannheimer
• [Asterisk-Users] netmeeting dial   Paulo H. Mannheimer
• [Asterisk-Users] just static on netmeeting call to extension   Paulo H. Mannheimer
• [Asterisk-Users] still problems with oh323   Paulo H. Mannheimer
• [Asterisk-Users] still problems with oh323   Paulo H. Mannheimer
• [Asterisk-Users] H.323 support   Michael Manousos
• [Asterisk-Users] asterisk And OPen h323   Michael Manousos
• [Asterisk-Users] Suppport for Asterisk, asterisk-h323 package and Voice Mail   Michael Manousos
• Embedded/R-T/Small Linuxes [WAS Re: [Asterisk-Users] Redhat vs Mandrake.]   Michael Manousos
• [Asterisk-Users] X100P   Michael Manousos
• [Asterisk-Users] help can not install oh323 wrapper   Michael Manousos
• [Asterisk-Users] help please on oh323 wrapper   Michael Manousos
• [Asterisk-Users] Configuring Asterisk to Direct Dialling In   Marco.Morgato at dignatio.com.pt
• [Asterisk-Users] Configuring Asterisk to Direct Dialling In   Marco.Morgato at dignatio.com.pt
• [Asterisk-Users] Meetme application en 0.4.0   Marsico, Gustavo - (Arg)
• [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #339 - 11 msgs   Marsico, Gustavo - (Arg)
• [Asterisk-Users] Which VoIP, H.323, SIP reference books are recommended?   Gary Mart
• [Asterisk-Users] Redhat vs Mandrake.   Brancaleoni Matteo
• [Asterisk-Users] Account code on SIP   Brancaleoni Matteo
• [Asterisk-Users] Call screening   Brancaleoni Matteo
• [Asterisk-Users] netmeeting dial   Brancaleoni Matteo
• [Asterisk-Users] Zhone re-provision   Brancaleoni Matteo
• [Asterisk-Users] Zhone and line length problems ??   Brancaleoni Matteo
• [Asterisk-Users] call queues   Brancaleoni Matteo
• [Asterisk-Users] vmail.agi   Brancaleoni Matteo
• [Asterisk-Users] German voicemail prompts, anybody?   Brancaleoni Matteo
• [Asterisk-Users] will this machine handle it   Jeff McClure
• [Asterisk-Users] will this machine handle it   Jeff McClure
• [Asterisk-Users] will this machine handle it   Jeff McClure
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   Ken McKittrick
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   Ken McKittrick
• [Asterisk-Users] outbound calls not working   Ken McKittrick
• [Asterisk-Users] FAX over IAX   Jeremy McNamara
• [Asterisk-Users] chan_h323 problems....   Jeremy McNamara
• [Asterisk-Users] Channel driver architecture   Jeremy McNamara
• [Asterisk-Users] Experiences with Zultys 4x4 SIP phone?   Jeremy McNamara
• [Asterisk-Users] enabling G.729 for SIP   Jeremy McNamara
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Jeremy McNamara
• [Asterisk-Users] exited non-zero   Jeremy McNamara
• [Asterisk-Users] Where is zttool?   Jeremy McNamara
• [Asterisk-Users] Where is zttool?   Jeremy McNamara
• [Asterisk-Users] SIP to MGCP transcoding   Jeremy McNamara
• [Asterisk-Users] Taking a call for someone elses extension from my extension   Jeremy McNamara
• [Asterisk-Users] Taking a call for someone elses extension from my extension   Jeremy McNamara
• [Asterisk-Users] What driver for the TDMX0B cards?   Jeremy McNamara
• [Asterisk-Users] Unable to call H323 phones via asterisk   Jeremy McNamara
• [Asterisk-Users] Unable to call H323 phones via asterisk   Jeremy McNamara
• [Asterisk-Users] Unable to call H323 phones via asterisk   Jeremy McNamara
• [Asterisk-Users] still problems with oh323   Jeremy McNamara
• [Asterisk-Users] chan_h323 problems   Jeremy McNamara
• [Asterisk-Users] Building own SIP CLient   Jeremy McNamara
• [Asterisk-Users] TDM10B problem   Jeremy McNamara
• [Asterisk-Users] how to install chan_h323   Jeremy McNamara
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   Mercer, Jay
• [Asterisk-Users] * on openmosix   Liaan van der Merwe
• [Asterisk-Users] unsubscribe   Liaan van der Merwe
• [Asterisk-Users] [offtopic] phone system for sale/trade   Michael
• [Asterisk-Users] X100P odd state behavior on shared/split line.   Michael
• [SPAM] Re: [Asterisk-Users] modem.conf i4l issues   Pauline Middelink
• [Asterisk-Users] isdn config   Pauline Middelink
• [Asterisk-Users] connecting with a modem?   Mike
• [Asterisk-Users] CDR Format and Software   Benjamin Miller
• [Asterisk-Users] RE: [Asterisk-Dev] Several patches, including recording and music -on-hold   Benjamin Miller
• [Asterisk-Users] dialed number notify at invalid dial situation   Benjamin Miller
• [Asterisk-Users] RE: [Asterisk-Dev] Several patches, including recording and music -on-hold   Benjamin Miller
• [Asterisk-Users] ${EPOCH} and ${DATETIME} patch   Benjamin Miller
• [Asterisk-Users] ${EPOCH} and ${DATETIME} patch   Benjamin Miller
• [Asterisk-Users] Grandstream BudgeTone 100   Benjamin Miller
• Wait doesn't read DTMF? Was Re: [Asterisk-Users] Collecting dialed digits   Benjamin Miller
• [Asterisk-Users] struggling with Diva 4BRI and chan_capi   Louis-David Mitterrand
• [Asterisk-Users] Diva 4BRI, asterisk and capidrv support ?   Louis-David Mitterrand
• [Asterisk-Users] Eicon BRI-card   Jan Mvrn
• [Asterisk-Users] Eicon driver?   Jan Mvrn
• [Asterisk-Users] Eicon driver?   Jan Mvrn
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   Terje Myhre
• [Asterisk-Users] asterisk online indicator   Masakazu Nakano
• [Asterisk-Users] dialed number notify at invalid dial situation   Masakazu Nakano
• [Asterisk-Users] dialed number notify at invalid dial situation   Masakazu Nakano
• [Asterisk-Users] zaurus: tkcPhone is an interest   Masakazu Nakano
• [Asterisk-Users] asterisk online indicator   Masakazu Nakano
• [Asterisk-Users] mpg123 hangs on close, but plays fine.   Masakazu Nakano
• [Asterisk-Users] (Hardware (processor) requirements / Performance   Rahim Nathwani
• [Asterisk-Users] Quesiton about SIP and MSN   Thorsten Neumann
• [Asterisk-Users] voip quality   Ron North
• [Asterisk-Users] Cisco 7940 conversion software??   Ronald North
• [Asterisk-Users] Locked out of Cisco 7940   Ronald North
• [Asterisk-Users] re: locked out of 7940   Ronald North
• [Asterisk-Users] low-cost * (newbie question)   Grzegorz Nosek
• [Asterisk-Users] low-cost * (newbie question)   Grzegorz Nosek
• [Asterisk-Users] low-cost * (newbie question)   Grzegorz Nosek
• [Asterisk-Users] regarding the new tdm?0b cards (another newbie question)   Grzegorz Nosek
• [Asterisk-Users] struggling with Diva 4BRI and chan_capi   Jeff Noxon
• [Asterisk-Users] FW: ipDialog Ethernet SIP Phone $199   Alejandro Olchik
• [Asterisk-Users] SIP Testing   Alejandro Olchik
• [Asterisk-Users] Dialogic E1 card.   Alejandro Olchik
• [Asterisk-Users] Sample zaptel.conf and zapata.conf   Dan Erik Ollas
• [Asterisk-Users] howto reduce number of rings ?   Florian Overkamp
• [Asterisk-Users] IP hard phone devices (grandstream)   Florian Overkamp
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Florian Overkamp
• [Asterisk-Users] Whoah! My E400P system went AWOL   Florian Overkamp
• [Asterisk-Users] Whoah! My E400P system went AWOL   Florian Overkamp
• [Asterisk-Users] Does anyone have the Cisco 7910 IP Phone working with Asterisk?   Florian Overkamp
• [Asterisk-Users] new cisco VoIP phones   Florian Overkamp
• [Asterisk-Users] how many voicemail box asterisk can support   Florian Overkamp
• [Asterisk-Users] change voicemail.conf, how to make it work   Florian Overkamp
• [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan   Juan J. Sierralta P.
• [Asterisk-Users] * as a SIP/H323 gateway   Dave Packham
• [Asterisk-Users] Asterisk-Redhat 9 install guide.   Dave Packham
• [Asterisk-Users] Recording Prompts   Dave Packham
• [Asterisk-Users] call announce?   Dave Packham
• [Asterisk-Users] Recording Prompts   Dave Packham
• [Asterisk-Users] x100p line quality   Dave Packham
• [Asterisk-Users] x100p line quality   Dave Packham
• [Asterisk-Users] LineJACK Compatability   Dave Packham
• [Asterisk-Users] Can you invoke an app before dialtone?   Dave Packham
• [Asterisk-Users] ISDN phones & Asterisk, ASDI and then some   Patrick
• [Asterisk-Users] NT mode with hfc cards for emulation of internal isdn S0 bus   Patrick
• [Asterisk-Users] ISDN stuff (was pri pricing)   Patrick
• [Asterisk-Users] 32 E1 or 64 E1 Configuration ?   Patrick
• [Asterisk-Users] ADSI Analogue phones   Patrick
• [Asterisk-Users] Re: two computers set up ... now what?   Patrick
• [Asterisk-Users] PBX Console   Patrick
• [Asterisk-Users] Dialing using X100P   Patrick
• [Asterisk-Users] new cisco VoIP phones   Patrick
• [Asterisk-Users] Re: Complie on RH8   Patrick
• [Asterisk-Users] Started Zaptel SRPM & RPMs   Patrick
• [Asterisk-Users] LineJack PSTN Out   Patrick
• [Asterisk-Users] Asterisk SRPM and RPMs   Patrick
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   Michael Paull
• [Asterisk-Users] Recording Prompts   Francisco Perez-Landaeta
• [Asterisk-Users] Redhat vs Mandrake.   Francisco Perez-Landaeta
• [Asterisk-Users] Redhat vs Mandrake.   Francisco Perez-Landaeta
• [Asterisk-Users] Redhat vs Mandrake.   Francisco Perez-Landaeta
• [Asterisk-Users] ATA186: "Call/Leg Transaction Doesn't Exist" on local call   Pertti Pikkarainen
• [Asterisk-Users] non-telephony use of T400P?   Don Pobanz
• [Asterisk-Users] AT&T T1 Cable Needed!   Don Pobanz
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Don Pobanz
• [Asterisk-Users] No ring on extensions   Don Pobanz
• [Asterisk-Users] No ring on extensions   Don Pobanz
• [Asterisk-Users] IP hard phone devices (grandstream)   Lenny Post
• [Asterisk-Users] Codecs   Lenny Post
• [Asterisk-Users] ZHONE Fix !! (long)   Jon Pounder
• [Asterisk-Users] FAX over IAX   Jon Pounder
• [Asterisk-Users] Channel Banks   Jon Pounder
• [Asterisk-Users] Channel Banks   Jon Pounder
• [Asterisk-Users] external trigger on caller id   Jon Pounder
• [Asterisk-Users] external trigger on caller id   Jon Pounder
• [Asterisk-Users] Hmmm. RJ-45 on TDMx0B card?   Jon Pounder
• [Asterisk-Users] ADSI Analogue phones   Jon Pounder
• [Asterisk-Users] two computers set up ... now what???   Jon Pounder
• [Asterisk-Users] Re: two computers set up ... now what?   Jon Pounder
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Jon Pounder
• [Asterisk-Users] X100P odd state behavior on shared/split line.   Jon Pounder
• [Asterisk-Users] new cisco VoIP phones   Jon Pounder
• [Asterisk-Users] adsi phones   Jon Pounder
• [Asterisk-Users] LineJACK Compatability   Jon Pounder
• [Asterisk-Users] Two Rings   Jon Pounder
• [Asterisk-Users] Two Rings   Jon Pounder
• [Asterisk-Users] FAX over IAX   Karl Putland
• [Asterisk-Users] FAX over IAX   Karl Putland
• [Asterisk-Users] chan_local segfault   Karl Putland
• [Asterisk-Users] Fw: Predictive dialer   Karl Putland
• [Asterisk-Users] SIP to MGCP transcoding   Karl Putland
• [Asterisk-Users] ${EPOCH} and ${DATETIME} patch   Karl Putland
• [Asterisk-Users] transfering and parking experiment.   Karl Putland
• [Asterisk-Users] dlink updates   Karl Putland
• [Asterisk-Users] PBX Console   Karl Putland
• [Asterisk-Users] new mgcp patch errors   Karl Putland
• [Asterisk-Users] new mgcp patch errors   Karl Putland
• [Asterisk-Users] CallerID hosed   Karl Putland
• [Asterisk-Users] CallerID hosed   Karl Putland
• [Asterisk-Users] MeetMe over IAX2 Test   Karl Putland
• [Asterisk-Users] using asterisk as a mgcp <-> h.323 translator   Karl Putland
• [Asterisk-Users] callerid send to shell script with system command   Karl Putland
• [Asterisk-Users] OT - Hardware needed   Martin Pycko
• [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?   Martin Pycko
• [Asterisk-Users] Line is stuck off hook...   Martin Pycko
• [Asterisk-Users] segmentation fault   Martin Pycko
• [Asterisk-Users] Sip Transfer   Martin Pycko
• [Asterisk-Users] Multiple X100P cards   Martin Pycko
• [Asterisk-Users] MP3player problem   Martin Pycko
• [Asterisk-Users] MP3player problem   Martin Pycko
• [Asterisk-Users] MP3player problem   Martin Pycko
• [Asterisk-Users] FAX over IAX   Martin Pycko
• [Asterisk-Users] non-telephony use of T400P?   Martin Pycko
• [Asterisk-Users] Quesiton about SIP and MSN   Martin Pycko
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Martin Pycko
• [Asterisk-Users] T100P Incoming Calls Drop   Martin Pycko
• [Asterisk-Users] T100P Incoming Calls Drop   Martin Pycko
• [Asterisk-Users] asterisk online indicator   Martin Pycko
• [Asterisk-Users] external trigger on caller id   Martin Pycko
• [Asterisk-Users] Multiple X100P cards   Martin Pycko
• [Asterisk-Users] Multiple X100P cards   Martin Pycko
• [Asterisk-Users] sip registration problems   Martin Pycko
• [Asterisk-Users] Recording Prompts   Martin Pycko
• [Asterisk-Users] asterisk / channel bank   Martin Pycko
• [Asterisk-Users] * kernel panic   Martin Pycko
• [Asterisk-Users] Integrating cell phone into Asterisk Extension..   Martin Pycko
• [Asterisk-Users] RH9 won't finish Asterisk make   Martin Pycko
• [Asterisk-Users] New TDM400P no dialtone   Martin Pycko
• [Asterisk-Users] Globals ??   Martin Pycko
• [Asterisk-Users] timeout music on hold or ring tone   Martin Pycko
• [Asterisk-Users] X100P question   Martin Pycko
• [Asterisk-Users] RE: iLBC   Martin Pycko
• [Asterisk-Users] preserve hostname for INVITE request-uri   Martin Pycko
• [Asterisk-Users] meetme config   Martin Pycko
• [Asterisk-Users] New user questions   Martin Pycko
• [Asterisk-Users] Macros not working as expected with extension "h" in some circumstances   Martin Pycko
• [Asterisk-Users] Zap channel to collect dialled number # terminated?   Martin Pycko
• [Asterisk-Users] 32 E1 or 64 E1 Configuration ?   Martin Pycko
• [Asterisk-Users] SIP call logging, called number not logged   Martin Pycko
• [Asterisk-Users] testing asterisks   Martin Pycko
• [Asterisk-Users] Zhone and line length problems ??   Martin Pycko
• [Asterisk-Users] SS7   Martin Pycko
• [Asterisk-Users] SS7   Martin Pycko
• [Asterisk-Users] Tor2 with em_w (or em) signalling pickup behavior?   Martin Pycko
• [Asterisk-Users] Zapata not required??   Martin Pycko
• [Asterisk-Users] SIP call logging, called number not logged   Martin Pycko
• [Asterisk-Users] g729.so codec   Martin Pycko
• [Asterisk-Users] Zapata not required??   Martin Pycko
• [Asterisk-Users] two computers set up ... now what???   Martin Pycko
• [Asterisk-Users] Re: two computers set up ... now what?   Martin Pycko
• [Asterisk-Users] Re: two computers set up ... now what?   Martin Pycko
• [Asterisk-Users] Zapata not required??   Martin Pycko
• [Asterisk-Users] SIP call logging, called number not logged   Martin Pycko
• [Asterisk-Users] Internet Dial-in security questions   Martin Pycko
• [Asterisk-Users] SIP setup howto?   Martin Pycko
• [Asterisk-Users] Meetme application en 0.4.0   Martin Pycko
• [Asterisk-Users] Dialplan question   Martin Pycko
• [Asterisk-Users] Re: Dialing using X100P   Martin Pycko
• [Asterisk-Users] Re: Complie on RH8   Martin Pycko
• [Asterisk-Users] Documentation   Martin Pycko
• [Asterisk-Users] two X100P cards and one TDM400P   Martin Pycko
• [Asterisk-Users] H323   Martin Pycko
• [Asterisk-Users] compiling problems   Martin Pycko
• [Asterisk-Users] two X100P cards and one TDM400P   Martin Pycko
• [Asterisk-Users] Can you invoke an app before dialtone?   Martin Pycko
• [Asterisk-Users] Can you invoke an app before dialtone?   Martin Pycko
• [Asterisk-Users] two X100P cards and one TDM400P   Martin Pycko
• [Asterisk-Users] web interface to vmail   Martin Pycko
• [Asterisk-Users] Hardware and E1   Martin Pycko
• [Asterisk-Users] Hardware and E1   Martin Pycko
• [Asterisk-Users] first few seconds of greeting cut-off   Martin Pycko
• [Asterisk-Users] change voicemail.conf, how to make it work   Martin Pycko
• [Asterisk-Users] Turning off Bridging?   Martin Pycko
• [Asterisk-Users] TDM10B problem   Martin Pycko
• [Asterisk-Users] TDM10B problem   Martin Pycko
• [Asterisk-Users] gastman   Martin Pycko
• [Asterisk-Users] how to install chan_h323   Martin Pycko
• [Asterisk-Users] gastman   Martin Pycko
• [Asterisk-Users] gastman   Martin Pycko
• [Asterisk-Users] Dialogic E1 card.   Antonio Rabena
• [Asterisk-Users] Dialogic E1 card.   Antonio Rabena
• [Asterisk-Users] E1 Card   Antonio Rabena
• [Asterisk-Users] Integrating cell phone into Asterisk Extension..   Steve Radich
• [Asterisk-Users] Call screening   Steve Radich
• [Asterisk-Users] Zhone + Digium T1 bug (?)   Steve Radich
• [Asterisk-Users] MeetMe over IAX2 Test   Steve Radich
• [Asterisk-Users] Zhone + Digium T1 bug (?)   Steve Radich
• [Asterisk-Users] Two Rings   Steve Radich
• [Asterisk-Users] (no subject)   Mike Reiling
• [Asterisk-Users] OS X support?   Mike Reiling
• [Asterisk-Users] Voicemail hangups/minimum message size   Mike Reiling
• [Asterisk-Users] Sip & Intercom   Mike Reiling
• [Asterisk-Users] Sip & Intercom   Mike Reiling
• [Asterisk-Users] Sip & Intercom   Mike Reiling
• [Asterisk-Users] Caller press "0" in Voicemail   Mike Reiling
• [Asterisk-Users] Caller press "0" in Voicemail   Mike Reiling
• [Asterisk-Users] Sip & Intercom   Mike Reiling
• [Asterisk-Users] Recording Prompts   Mike Reiling
• [Asterisk-Users] Cisco 7960 services and directory xml (CMXML, BTXML)   Mike Reiling
• [Asterisk-Users] vmail.agi   Mike Reiling
• [Asterisk-Users] callerid send to shell script with system command   Mike Reiling
• [Asterisk-Users] Experiences with Zultys 4x4 SIP phone?   Greg Renouf
• [Asterisk-Users] I must be alone   Greg Renouf
• [Asterisk-Users] Grandstream BudgeTone 100   Greg Renouf
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   Greg Renouf
• [Asterisk-Users] Grandstream BudgeTone 100   Greg Renouf
• [Asterisk-Users] Zhone re-provision   Eris Riswanto
• [Asterisk-Users] Zhone re-provision   Eris Riswanto
• [Asterisk-Users] Zhone re-provision   Eris Riswanto
• [Asterisk-Users] Zhone re-provision   Eris Riswanto
• [Asterisk-Users] Unable to call H323 phones via asterisk   Michael Rose
• [Asterisk-Users] Unable to call H323 phones via asterisk   Michael Rose
• [Asterisk-Users] Unable to call H323 phones via asterisk   Michael Rose
• [Asterisk-Users] Softphones with IAX   Michael Rose
• [Asterisk-Users] first few seconds of greeting cut-off   Michael Rose
• [Asterisk-Users] first few seconds of greeting cut-off   Michael Rose
• [Asterisk-Users] SIP and music on hold   Roy
• [Asterisk-Users] SIP and music on hold   Roy
• [Asterisk-Users] Sip Transfer   Russ Beaupre, P.E.
• [Asterisk-Users] Re: [Asterisk] Kernel panic, ZapRAS & E400p   Thilo Salmon
• [Asterisk-Users] Whats ENUM??   Thilo Salmon
• [Asterisk-Users] Whats ENUM??   Thilo Salmon
• [Asterisk-Users] Eicon driver?   Thilo Salmon
• [Asterisk-Users] Does anyone have the Cisco 7910 IP Phone working with Asterisk?   Tycho Schenkeveld
• [Asterisk-Users] Cisco 7940 conversion software??   Tycho Schenkeveld
• [Asterisk-Users] Cisco 7940 conversion software??   Tycho Schenkeveld
• [Asterisk-Users] Does anyone have the Cisco 7910 IP Phone working with Asterisk?   Tycho Schenkeveld
• [Asterisk-Users] FAX over IAX   Brian J. Schrock
• [Asterisk-Users] New TDM400P no dialtone   Brian J. Schrock
• [Asterisk-Users] Recording Prompts   Richard Scobie
• [Asterisk-Users] zapata busy detect   Richard Scobie
• [Asterisk-Users] Call screening   Joel Scotkin
• [Asterisk-Users] Call screening   Joel Scotkin
• [Asterisk-Users] Three way calling   Joel Scotkin
• [Asterisk-Users] Three way calling   Joel Scotkin
• [Asterisk-Users] Drool? Cisco Wireless VOIP phone   Shado
• [Asterisk-Users] RE: Packet8 New Area Codes and Rate Centers   Shado
• [Asterisk-Users] Line is stuck off hook...   James Sharp
• [Asterisk-Users] FAX over IAX   James Sharp
• [Asterisk-Users] CE certification for Europe   James Sharp
• [Asterisk-Users] Weird AGI/X100P behavior   James Sharp
• [Asterisk-Users] * kernel panic   James Sharp
• [Asterisk-Users] Signals to AGI scripts   James Sharp
• [Asterisk-Users] Signals to AGI scripts   James Sharp
• [Asterisk-Users] SS7   James Sharp
• [Asterisk-Users] Asterisks features w/ LineJack   Vladimir Shved
• [Asterisk-Users] Recording Prompts   Nir Simionovich
• [Asterisk-Users] will this machine handle it   James Sizemore
• [Asterisk-Users] iconnect and incoming DTMF   James Sizemore
• [Asterisk-Users] "-p"   James Sizemore
• [Asterisk-Users] SIP hanging   Skuse, Phil
• [Asterisk-Users] SIP hanging   Skuse, Phil
• [Asterisk-Users] SIP Proxy   Skuse, Phil
• [Asterisk-Users] Grandstream BudgeTone 100   Skuse, Phil
• [Asterisk-Users] Grandstream BudgeTone 100   Skuse, Phil
• [Asterisk-Users] Anyone using Asterisks and a Quicknet Lineja ck in the UK?   Skuse, Phil
• [Asterisk-Users] Grandstream BudgeTone 100   Skuse, Phil
• [Asterisk-Users] Internet Dial-in security questions   Skuse, Phil
• [Asterisk-Users] Internet Dial-in security questions   Skuse, Phil
• [Asterisk-Users] Turning off Bridging?   Darren Smith
• [Asterisk-Users] Turning off Bridging?   Darren Smith
• [Asterisk-Users] RE: iLBC   Jared Smith
• [Asterisk-Users] No ring on extensions   Jared Smith
• [Asterisk-Users] Can you invoke an app before dialtone?   Jared Smith
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Randy Smith
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Randy Smith
• [Asterisk-Users] two X100P cards and one TDM400P   Randy Smith
• [Asterisk-Users] two X100P cards and one TDM400P   Randy Smith
• [Asterisk-Users] two X100P cards and one TDM400P   Randy Smith
• [Asterisk-Users] two X100P cards and one TDM400P   Randy Smith
• [Asterisk-Users] two X100P cards and one TDM400P   Randy Smith
• [Asterisk-Users] MSN Messager and Asterisk   Steven Smith
• [Asterisk-Users] MSN Messager and Asterisk   Steven Smith
• [Asterisk-Users] ATA186: "Call/Leg Transaction Doesn't Exist" on local call   Mark Spencer
• [Asterisk-Users] will this machine handle it   Mark Spencer
• [Asterisk-Users] ATA186: "Call/Leg Transaction Doesn't Exist" on local call   Mark Spencer
• [Asterisk-Users] segmentation fault   Mark Spencer
• [Asterisk-Users] segmentation fault   Mark Spencer
• [Asterisk-Users] "Broken Dial Tone" for voice mail?   Mark Spencer
• [Asterisk-Users] CE certification for Europe   Mark Spencer
• [Asterisk-Users] CE certification for Europe   Mark Spencer
• [Asterisk-Users] CE certification for Europe   Mark Spencer
• [Asterisk-Users] false ringback   Mark Spencer
• [Asterisk-Users] CE certification for Europe   Mark Spencer
• [Asterisk-Users] problem with X100P and wcfxo   Mark Spencer
• [Asterisk-Users] FS: Cisco DT-24+, Dialogic D/240SC/T1, Natural Microsystem AG-T1   Mark Spencer
• [Asterisk-Users] Bug? * not correctly honouring tag on To?   Mark Spencer
• [Asterisk-Users] SIP Testing   Mark Spencer
• [Asterisk-Users] SIP Testing   Mark Spencer
• [Asterisk-Users] SIP Testing   Mark Spencer
• [Asterisk-Users] SIP improvements   Mark Spencer
• [Asterisk-Users] Quesiton about SIP and MSN   Mark Spencer
• [Asterisk-Users] SIP improvements   Mark Spencer
• [Asterisk-Users] IP hard phone devices (grandstream)   Mark Spencer
• [Asterisk-Users] Should List be Moderated?   Mark Spencer
• [Asterisk-Users] IP hard phone devices (grandstream)   Mark Spencer
• [Asterisk-Users] SIP Testing   Mark Spencer
• [Asterisk-Users] enabling G.729 for SIP   Mark Spencer
• [Asterisk-Users] Should List be Moderated?   Mark Spencer
• [Asterisk-Users] chan_local segfault   Mark Spencer
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   Mark Spencer
• [Asterisk-Users] Voicemail hangups/minimum message size   Mark Spencer
• [Asterisk-Users] Fw: Predictive dialer   Mark Spencer
• [Asterisk-Users] Re: SIP and ATA186   Mark Spencer
• [Asterisk-Users] Fw: Predictive dialer   Mark Spencer
• [Asterisk-Users] Multiport FXO?   Mark Spencer
• [Asterisk-Users] Asterisk dies after 3-6 hours of operation. Help!   Mark Spencer
• [Asterisk-Users] Sip & Intercom   Mark Spencer
• [Asterisk-Users] Caller press "0" in Voicemail   Mark Spencer
• [Asterisk-Users] Multiport FXO?   Mark Spencer
• [Asterisk-Users] Outgoing SIP Registration Fixed   Mark Spencer
• [Asterisk-Users] Caller press "0" in Voicemail   Mark Spencer
• [Asterisk-Users] exited non-zero   Mark Spencer
• [Asterisk-Users] exited non-zero   Mark Spencer
• [Asterisk-Users] sip registration problems   Mark Spencer
• [Asterisk-Users] Where is zttool?   Mark Spencer
• [Asterisk-Users] sip registration problems   Mark Spencer
• [Asterisk-Users] extension exited non-zero   Mark Spencer
• [Asterisk-Users] sip registration problems   Mark Spencer
• [Asterisk-Users] Strange Sip problem?   Mark Spencer
• [Asterisk-Users] Outgoing SIP Registration Fixed   Mark Spencer
• [Asterisk-Users] sip registration problems   Mark Spencer
• [Asterisk-Users] SIP and special functions - do they work?   Mark Spencer
• [Asterisk-Users] sip registration problems   Mark Spencer
• Fwd: [Asterisk-Users] Setting up ADSI   Mark Spencer
• [Asterisk-Users] sip registration problems   Mark Spencer
• [Asterisk-Users] * kernel panic   Mark Spencer
• [Asterisk-Users] SIP hanging   Mark Spencer
• [Asterisk-Users] Asterisk and SNOM 200   Mark Spencer
• [Asterisk-Users] Asterisk and SNOM 200   Mark Spencer
• [Asterisk-Users] T1 vs PRI   Mark Spencer
• [Asterisk-Users] Dialogic E1 card.   Mark Spencer
• [Asterisk-Users] New 4port FXS   Mark Spencer
• [Asterisk-Users] S100U on RH9   Mark Spencer
• [Asterisk-Users] SIP and music on hold   Mark Spencer
• [Asterisk-Users] S100U on RH9   Mark Spencer
• [Asterisk-Users] SIP and music on hold   Mark Spencer
• [Asterisk-Users] SIP and music on hold   Mark Spencer
• [Asterisk-Users] iLBC   Mark Spencer
• [Asterisk-Users] Signals to AGI scripts   Mark Spencer
• [Asterisk-Users] Experimental MEC3   Mark Spencer
• [Asterisk-Users] SIP Proxy   Mark Spencer
• [Asterisk-Users] SIP and music on hold   Mark Spencer
• [Asterisk-Users] S100U on RH9   Mark Spencer
• [Asterisk-Users] RE: iLBC   Mark Spencer
• [Asterisk-Users] RE: iLBC   Mark Spencer
• [Asterisk-Users] Unknown RTP codec 101 received   Mark Spencer
• [Asterisk-Users] Account code on SIP   Mark Spencer
• [Asterisk-Users] RE: iLBC   Mark Spencer
• [Asterisk-Users] /proc info on X100P's, zapata.conf   Mark Spencer
• [Asterisk-Users] Still can't get MeetMe working..   Mark Spencer
• [Asterisk-Users] Hmmm. RJ-45 on TDMx0B card?   Mark Spencer
• [Asterisk-Users] E1 Card   Mark Spencer
• [Asterisk-Users] Zapata not required??   Mark Spencer
• [Asterisk-Users] SS7   Mark Spencer
• [Asterisk-Users] Tor2 with em_w (or em) signalling pickup behavior?   Mark Spencer
• [Asterisk-Users] Zapata not required??   Mark Spencer
• [Asterisk-Users] PBX Console   Mark Spencer
• [Asterisk-Users] g729.so codec   Mark Spencer
• [Asterisk-Users] PBX Console   Mark Spencer
• [Asterisk-Users] Outgoing SIP Call to unregistered Users   Mark Spencer
• [Asterisk-Users] Re: two computers set up ... now what?   Mark Spencer
• [Asterisk-Users] Internet Dial-in security questions   Mark Spencer
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Mark Spencer
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Mark Spencer
• [Asterisk-Users] SNOM 200 and MWI??   Mark Spencer
• [Asterisk-Users] Collecting dialed digits   Mark Spencer
• [Asterisk-Users] No ring on extensions   Mark Spencer
• [Asterisk-Users] PBX with modem   Borce Stevanoski
• [Asterisk-Users] ISDN phones & Asterisk, ASDI and then some   Iain Stevenson
• [Asterisk-Users] SIP call logging, called number not logged   Iain Stevenson
• [Asterisk-Users] SIP call logging, called number not logged   Iain Stevenson
• [Asterisk-Users] SIP call logging, called number not logged   Iain Stevenson
• [Asterisk-Users] SIP call logging, called number not logged   Iain Stevenson
• [Asterisk-Users] SIP call logging, called number not logged   Iain Stevenson
• [Asterisk-Users] ztdummy problems   Iain Stevenson
• [Asterisk-Users] SIP support status   Jon Stockill
• [Asterisk-Users] Softphones with IAX   Jon Stockill
• [Asterisk-Users] /proc info on X100P's, zapata.conf   Mark Street
• [Asterisk-Users] vmail.agi   Mark Street
• [Asterisk-Users] two X100P cards and one TDM400P   Mark Street
• [Asterisk-Users] CE certification for Europe   David Stubbs
• [Asterisk-Users] Call FWD & the new channel driver chan_local   TC
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   TC
• [Asterisk-Users] external trigger on caller id   TC
• [Asterisk-Users] How to make an X100P answer only one distinctivering cadence?   TC
• [Asterisk-Users] asterisk / channel bank   TC
• [Asterisk-Users] Firmware version in Adtran TA 750 Channel Bank   TC
• [Asterisk-Users] Experimental MEC3   TC
• [Asterisk-Users] IAX pauses   TC
• [Asterisk-Users] Another FXS to FXO conversion UNIT $45   TC
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   TC
• [Asterisk-Users] ADSI Analogue phones   TC
• [Asterisk-Users] RE: [Asterisk-Dev] adsi phone codes?   TC
• [Asterisk-Users] VoIP Gateway   Fabrice Tereszkiewicz
• [Asterisk-Users] VoIP gateway and Wilcard X100P   Fabrice Tereszkiewicz
• [Asterisk-Users] Sip & Intercom   James H. Thompson
• [Asterisk-Users] Channel Banks   Tim Thompson
• [Asterisk-Users] TP100   Tim Thompson
• [Asterisk-Users] module chan_sip.o failed to load   Tim Thompson
• [Asterisk-Users] module chan_sip.o failed to load   Tim Thompson
• [Asterisk-Users] module chan_sip.o failed to load   Tim Thompson
• [Asterisk-Users] module chan_sip.o failed to load   Tim Thompson
• [Asterisk-Users] Registering with Pulver???   Tim Thompson
• [Asterisk-Users] RE: [Asterisk] How to select server ardware?   Lars Boegild Thomsen
• [Asterisk-Users] new cisco VoIP phones   Lars Boegild Thomsen
• [Asterisk-Users] Real-time Asterisk hacking at VON   John Todd
• [Asterisk-Users] Experiences with Zultys 4x4 SIP phone?   John Todd
• [Asterisk-Users] Experiences with Zultys 4x4 SIP phone?   John Todd
• [Asterisk-Users] Call completion/error codes and extensions.conf call flow   John Todd
• [Asterisk-Users] Priority usage: absolute sequential vs. sequential   John Todd
• [Asterisk-Users] Which VoIP, H.323, SIP reference books are recommended?   John Todd
• [Asterisk-Users] Encryption at RTP level (was: Zultys SIP phone)   John Todd
• [Asterisk-Users] Priority usage: absolute sequential vs. sequential   John Todd
• [Asterisk-Users] Should List be Moderated?   John Todd
• [Asterisk-Users] Iconnect currently horribly broken?   John Todd
• [Asterisk-Users] IP hard phone devices (grandstream)   John Todd
• [Asterisk-Users] Comments on "transfer" feature request   John Todd
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   John Todd
• [Asterisk-Users] sip registration problems   John Todd
• [Asterisk-Users] Music-on-Hold radio input?   John Todd
• [Asterisk-Users] Unexpected behavior of X100P and * in no-dialtone situations   John Todd
• [Asterisk-Users] Macros not working as expected with extension "h" in some circumstances   John Todd
• [Asterisk-Users] ${EPOCH} and ${DATETIME} patch   John Todd
• [Asterisk-Users] Macros not working as expected with extension "h" in some circumstances   John Todd
• [Asterisk-Users] Rackmount PC hardware suggestions   John Todd
• [Asterisk-Users] PBX Console   John Todd
• [Asterisk-Users] g729.so codec   John Todd
• [Asterisk-Users] Revised example configurations available   John Todd
• [Asterisk-Users] Collecting dialed digits   John Todd
• [Asterisk-Users] Packet8 New Area Codes and Rate Centers   John Todd
• [Asterisk-Users] Cisco 7940 conversion software??   John Todd
• [Asterisk-Users] re: locked out of 7940   John Todd
• [Asterisk-Users] Can you invoke an app before dialtone?   John Todd
• [Asterisk-Users] Buzzword bingo: TLS and SRTP   John Todd
• [Asterisk-Users] H.323 support   Julio Tommasi
• [Asterisk-Users] IAXTEL Inbound, and Outbound Tollfree Changes   The Traveller
• [Asterisk-Users] still problems with oh323   The Traveller
• [Asterisk-Users] Dynamic IP Addrress Work Around   The Traveller
• [Asterisk-Users] (no subject)   Zara Trousk
• [Asterisk-Users] LineJack PSTN Out   Zara Trousk
• [Asterisk-Users] LineJack PSTN Out   Zara Trousk
• [Asterisk-Users] Creating a phone channel   Zara Trousk
• [Asterisk-Users] Creating a phone channel   Zara Trousk
• [Asterisk-Users] compiling problems   Zara Trousk
• [Asterisk-Users] Re: Creating a phone channel   Zara Trousk
• [Asterisk-Users] Configuring Asterisk to Direct Dialling In   Zara Trousk
• [Asterisk-Users] FAX over IAX   Steve Underwood
• [Asterisk-Users] CE certification for Europe   Steve Underwood
• [Asterisk-Users] CE certification for Europe   Steve Underwood
• [Asterisk-Users] CE certification for Europe   Steve Underwood
• [Asterisk-Users] CE certification for Europe   Steve Underwood
• [Asterisk-Users] torisa rev b   Steve Underwood
• [Asterisk-Users] torisa rev b   Steve Underwood
• [Asterisk-Users] Dialogic E1 card.   Steve Underwood
• [Asterisk-Users] oh323 failed to load   Makerere University
• [Asterisk-Users] help can not install oh323 wrapper   Makerere University
• [Asterisk-Users] help please on oh323 wrapper   Makerere University
• [Asterisk-Users] H323   Makerere University
• [Asterisk-Users] The Joys of ADSI   Jayson Vantuyl
• [Asterisk-Users] PBX Console   Jayson Vantuyl
• [Asterisk-Users] Call Queue Manager and DID Digits   Jayson Vantuyl
• [Asterisk-Users] Rackmount PC hardware suggestions   Jayson Vantuyl
• [Asterisk-Users] Rackmount PC hardware suggestions   Jayson Vantuyl
• [Asterisk-Users] ADSI Analogue phones   Jayson Vantuyl
• [Asterisk-Users] adsi phones   Jayson Vantuyl
• [Asterisk-Users] SIP & Adtech 150   Corrado Ventura
• [Asterisk-Users] New user questions   Vladislav
• [Asterisk-Users] New user questions   Vladislav
• [Asterisk-Users] explain me   Vladislav
• [Asterisk-Users] CE certification for Europe   John Vozza
• [Asterisk-Users] Multiple X100P cards   John Vozza
• [Asterisk-Users] Where is zttool?   John Vozza
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   DUSTIN WILDES
• [Asterisk-Users] Caller press "0" in Voicemail   DUSTIN WILDES
• [Asterisk-Users] Caller press "0" in Voicemail   DUSTIN WILDES
• [Asterisk-Users] Configuring for outbound calls with PRI on T100P   DUSTIN WILDES
• [Asterisk-Users] Error compiling in RedHat 9   DUSTIN WILDES
• [Asterisk-Users] Caller press "0" in Voicemail   DUSTIN WILDES
• [Asterisk-Users] Caller press "0" in Voicemail   DUSTIN WILDES
• [Asterisk-Users] Sip & Intercom   DUSTIN WILDES
• [Asterisk-Users] Sip & Intercom   DUSTIN WILDES
• [Asterisk-Users] Zap features using SIP   DUSTIN WILDES
• [Asterisk-Users] SIP support status   DUSTIN WILDES
• [Asterisk-Users] SIP support status   DUSTIN WILDES
• [Asterisk-Users] SIP and music on hold   DUSTIN WILDES
• [Asterisk-Users] SIP support status   DUSTIN WILDES
• [Asterisk-Users] cisco dial-peer help   DUSTIN WILDES
• [Asterisk-Users] Music-on-Hold radio input?   DUSTIN WILDES
• [Asterisk-Users] Music-on-Hold radio input?   DUSTIN WILDES
• [Asterisk-Users] Music-on-Hold radio input?   DUSTIN WILDES
• [Asterisk-Users] Recording Prompts   William Walsh
• [Asterisk-Users] Xten - Free windows SIP client   William Walsh
• [Asterisk-Users] Redhat vs Mandrake.   William Walsh
• [Asterisk-Users] Xten / SIP Phones compared to GnoPhones   William Walsh
• [Asterisk-Users] Redhat vs Mandrake.   William Walsh
• [Asterisk-Users] RE: Packet8 New Area Codes and Rate Centers   William Walsh
• [Asterisk-Users] Config??   Allan Wang
• [Asterisk-Users] Best IP phone?   Allan Wang
• [Asterisk-Users] using asterisk as a mgcp <-> h.323 translator   Sean Watkins
• [Asterisk-Users] Building own SIP CLient   God Knows Well
• [Asterisk-Users] How to write SIP Client   God Knows Well
• [Asterisk-Users] Building own SIP CLient   God Knows Well
• [Asterisk-Users] SIP Testing   Wade Weppler
• [Asterisk-Users] T100P Incoming Calls Drop   Wade Weppler
• [Asterisk-Users] How to make an X100P answer only one distinctive ring cadence?   Wade Weppler
• [Asterisk-Users] How to make an X100P answer only one distinctivering cadence?   Wade Weppler
• [Asterisk-Users] Channel Banks   Wade Weppler
• [Asterisk-Users] Modem connections   Wade Weppler
• [Asterisk-Users] iLBC   Wade Weppler
• [Asterisk-Users] * kernel panic   Wade Weppler
• [Asterisk-Users] Zhone re-provision   Wade Weppler
• [Asterisk-Users] Still can't get MeetMe working..   Wade Weppler
• [Asterisk-Users] Zhone re-provision   Wade Weppler
• [Asterisk-Users] Zhone re-provision   Wade Weppler
• [Asterisk-Users] Rackmount PC hardware suggestions   Wade Weppler
• [Asterisk-Users] two computers set up ... now what???   Wade Weppler
• [Asterisk-Users] Re: two computers set up ... now what?   Wade Weppler
• [Asterisk-Users] vmail.agi   Wade Weppler
• [Asterisk-Users] new cisco VoIP phones   Wade Weppler
• [Asterisk-Users] Re: Complie on RH8   Wade Weppler
• [Asterisk-Users] first few seconds of greeting cut-off   Wade Weppler
• [Asterisk-Users] Taking a call for someone elses extensionfrom my extension   Wade J. Weppler
• [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extensionfrom my extension)   Wade J. Weppler
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   Wade J. Weppler
• [Asterisk-Users] how to register * at FWD from behind NAT   Chris Wetemans
• [Asterisk-Users] SIP Testing   Eric Wieling
• [Asterisk-Users] Should List be Moderated?   Eric Wieling
• [Asterisk-Users] Problems compileing latest CVS   Eric Wieling
• [Asterisk-Users] Problems compileing latest CVS   Eric Wieling
• [Asterisk-Users] Dial Plan Problems (also IAX)   Eric Wieling
• [Asterisk-Users] Playback application   Eric Wieling
• [Asterisk-Users] Suppport for Asterisk, asterisk-h323 package and Voice Mail   Eric Wieling
• [Asterisk-Users] ATA-186 Dialplans   Eric Wieling
• [Asterisk-Users] DTMF Modes   Eric Wieling
• [Asterisk-Users] DTMF Modes   Eric Wieling
• [Asterisk-Users] Unable to call H323 phones via asterisk   Eric Wieling
• [Asterisk-Users] Unable to call H323 phones via asterisk   Eric Wieling
• [Asterisk-Users] CallerID hosed   Eric Wieling
• [Asterisk-Users] Packet8 New Area Codes and Rate Centers   Eric Wieling
• [Asterisk-Users] RE: Packet8 New Area Codes and Rate Centers   Eric Wieling
• [Asterisk-Users] Cisco 7940 conversion software??   Eric Wieling
• [Asterisk-Users] new cisco VoIP phones   Eric Wieling
• [Asterisk-Users] new cisco VoIP phones   Eric Wieling
• [Asterisk-Users] Cisco 7940 conversion software??   Eric Wieling
• [Asterisk-Users] Does anyone have the Cisco 7910 IP Phone working with Asterisk?   Eric Wieling
• [Asterisk-Users] Zaptel compile problems   Eric Wieling
• [Asterisk-Users] LineJACK Compatability   Eric Wieling
• [Asterisk-Users] callerid send to shell script with system command   Eric Wieling
• [Asterisk-Users] H323   Darran Wilson
• [Asterisk-Users] Mosix?   Darran Wilson
• [Asterisk-Users] H323   Darran Wilson
• [Asterisk-Users] Mosix?   Darran Wilson
• [Asterisk-Users] gastman   Darran Wilson
• [Asterisk-Users] gastman   Darran Wilson
• [Asterisk-Users] H323/GnuGK/Netmeeting   Darran Wilson
• [Asterisk-Users] gastman   Darran Wilson
• [Asterisk-Users] SIP Testing   Chris Witte
• [Asterisk-Users] SIP Testing   Dave Wolven
• [Asterisk-Users] DTMF Modes   Dave Wolven
• [Asterisk-Users] Xten - Free windows SIP client   Simon Woodhead
• [Asterisk-Users] ISDN4Linux problems   Stephen J Wright
• [Asterisk-Users] Setting up PABX   Only Yours
• [Asterisk-Users] segmentation fault   Alex Zarubin
• [Asterisk-Users] segmentation fault   Alex Zarubin
• [Asterisk-Users] dialplan problems cvs 04-15-03   Alex Zarubin
• [Asterisk-Users] Softphones with IAX   William Zhang
• [Asterisk-Users] Grandstream BudgeTone 100   William Zhang
• Embedded/R-T/Small Linuxes [WAS Re: [Asterisk-Users] Redhat vs Mandrake.]   Tom Zimnicki
• [Asterisk-Users] SIP Testing   Jodie Cunningham [LightningCrash{TSI}]
• [Asterisk-Users] Buying channel banks online?   alex at pilosoft.com
• [Asterisk-Users] new cisco VoIP phones   alex at pilosoft.com
• [Asterisk-Users] Redhat vs Mandrake.   ask at develooper.com
• [Asterisk-Users] will this machine handle it   asterisk at billheckel.com
• [Asterisk-Users] The Joys of ADSI   asterisk at billheckel.com
• [Asterisk-Users] Call Queue Manager and DID Digits   asterisk at billheckel.com
• [Asterisk-Users] ADSI Analogue phones   asterisk at billheckel.com
• [Asterisk-Users] SIP call logging, called number not logged   asterisk at billheckel.com
• Wait doesn't read DTMF? Was Re: [Asterisk-Users] Collecting dialed digits   asterisk at billheckel.com
• Wait doesn't read DTMF? Was Re: [Asterisk-Users] Collecting dialed digits   asterisk at billheckel.com
• [Asterisk-Users] No ring on extensions   asterisk at billheckel.com
• [Asterisk-Users] zaptel error while compiling   austino at skannet.com
• [Asterisk-Users] Channel driver architecture   bagyenda at imul.com
• [Asterisk-Users] Voicemail hangups/minimum message size   billp
• [Asterisk-Users] Unknown RTP 19, cisco vad   billp
• [Asterisk-Users] cisco dial-peer help   billp
• [Asterisk-Users] Two Rings   billp
• [Asterisk-Users] Error with EICON and CAPI   bnico at mailclub.net
• [Asterisk-Users] Channel Banks Quotation   chiardon at cable.net.co
• [Asterisk-Users] Asterisk-Redhat 9 install guide.   chiardon at cable.net.co
• [Asterisk-Users] SIP Testing   denon
• [Asterisk-Users] Voicemail hangups/minimum message size   denon
• [Asterisk-Users] Setting up ADSI   denon
• [Asterisk-Users] external trigger on caller id   denon
• [Asterisk-Users] Weirdness on "hookflash call pickup"   denon
• [Asterisk-Users] new cisco VoIP phones   denon
• [Asterisk-Users] new cisco VoIP phones   denon
• [Asterisk-Users] Building own SIP CLient   denon
• [Asterisk-Users] low-cost * (newbie question)   duncan
• [Asterisk-Users] ${EPOCH} and ${DATETIME} patch   duncan
• [Asterisk-Users] ISDN - Dialout MSN setting ??   duncan
• [Asterisk-Users] callerid send to shell script with system command   duncan
• [Asterisk-Users] callerid send to shell script with system command   duncan
• [Asterisk-Users] Asterisk-Redhat 9 install guide.   firedude at shorelinuxsolutions.com
• [Asterisk-Users] testing asterisks   firedude at shorelinuxsolutions.com
• [Asterisk-Users] Getting started   julian green
• [Asterisk-Users] RH9 won't finish Asterisk make   julian green
• [Asterisk-Users] new mgcp patch errors   julian green
• [Asterisk-Users] new mgcp patch errors   julian green
• [Asterisk-Users] SIP setup howto?   julian green
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] CE certification for Europe   d hinton
• [Asterisk-Users] unsubscribe   d hinton
• [Asterisk-Users] unsubscribe   d hinton
• [Asterisk-Users] torisa rev b   horape at highconnection.com.ar
• [Asterisk-Users] torisa: trying to understand the design   horape at highconnection.com.ar
• [Asterisk-Users] voicemail   mostafa ibrahim
• [Asterisk-Users] FS: Cisco DT-24+, Dialogic D/240SC/T1, Natural Microsystem AG-T1   info at aislecom.com
• [Asterisk-Users] Newbie problem?   isamar at isamarmaia.org
• [Asterisk-Users] Newbie problem?   isamar at isamarmaia.org
• [Asterisk-Users] No audio in Linejack   isamar at isamarmaia.org
• [Asterisk-Users] No audio in Linejack   isamar at isamarmaia.org
• [Asterisk-Users] SIP support status   isamar at isamarmaia.org
• [Asterisk-Users] LineJACK Compatability   isamar at isamarmaia.org
• [Asterisk-Users] Re: [Asterisk] IAXPHONE Win32 Alpha Released   it
• [Asterisk-Users] Re: [Asterisk] IAXPHONE Win32 Alpha Released   it
• [Asterisk-Users] A quesiton about MSN   it
• [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?   it
• [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?   it
• [Asterisk-Users] Buying channel banks online?   it
• [Asterisk-Users] Quesiton about SIP and MSN   it
• [Asterisk-Users] ISDN4Linux problems   it
• [Asterisk-Users] Quesiton about SIP and MSN   it
• [Asterisk-Users] ISDN4Linux problems   it
• [Asterisk-Users] Fw: Predictive dialer   it
• [Asterisk-Users] Fw: Predictive dialer   it
• [Asterisk-Users] Whoah! My E400P system went AWOL   it
• [Asterisk-Users] Whoah! My E400P system went AWOL   it
• [Asterisk-Users] torisa rev b   it
• [Asterisk-Users] torisa rev b   it
• [Asterisk-Users] Trouble installing   it
• [Asterisk-Users] Dialogic E1 card.   it
• [Asterisk-Users] Using Asterisk on Call Center   it
• [Asterisk-Users] Hangup detection problems   it
• [Asterisk-Users] Xten - Free windows SIP client   it
• [Asterisk-Users] Xten - Free windows SIP client   it
• [Asterisk-Users] Xten - Free windows SIP client   it
• [Asterisk-Users] MSN registered, but make phone call command is grayed   ping liu
• [Asterisk-Users] play message on bridged call   musashi at besite.be
• [Asterisk-Users] The best simple solution (newbee)   sjaak nabuurs
• [Asterisk-Users] Xten / SIP Phones compared to GnoPhones   stephane
• [Asterisk-Users] Best IP phone?   stephane
• [Asterisk-Users] Re: [Asterisk] H323 for Asterisk   steve
• [Asterisk-Users] exited non-zero   steve
• [Asterisk-Users] Should List be Moderated?   steve
• [Asterisk-Users] exited non-zero   steve
• [Asterisk-Users] exited non-zero   steve
• [Asterisk-Users] exited non-zero   steve
• [Asterisk-Users] PBX Console   steve
• [Asterisk-Users] PBX Console   steve
• [Asterisk-Users] PBX Console   steve
• [Asterisk-Users] PBX Console   steve
• [Asterisk-Users] PBX Console   steve
• [Asterisk-Users] Rackmount PC hardware suggestions   steve
• [Asterisk-Users] S100U hissing noise..   steve
• [Asterisk-Users] PBX Console   steve
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   steve
• [Asterisk-Users] Zhone + Digium T1 bug (?)   steve
• [Asterisk-Users] Sound files   steve
• [Asterisk-Users] [Asterisk-Users]   voip student
• [Asterisk-Users] how to install chan_h323   voip student
• [Asterisk-Users] Best IP phone?   tmassey at obscorp.com
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   tmassey at obscorp.com
• [Asterisk-Users] asterisk And OPen h323   victormedrano
• [Asterisk-Users] A quesiton about MSN   wasim at convergence.com.pk
• [Asterisk-Users] will this machine handle it   wasim at convergence.com.pk
• [Asterisk-Users] CE certification for Europe   wasim at convergence.com.pk
• [Asterisk-Users] Quesiton about SIP and MSN   wasim at convergence.com.pk
• [Asterisk-Users] SIP support status   wasim at convergence.com.pk
• [Asterisk-Users] OT: Multiple SIP phones behind NAT gateway?   wasim at convergence.com.pk
• [Asterisk-Users] voicemail   wasim at convergence.com.pk
• [Asterisk-Users] Proxy / STUN??   wasim at convergence.com.pk
• [Asterisk-Users] Wildcard TDM400P (TDM40B) connectors   wasim at convergence.com.pk

Last message date: Wed Apr 30 23:12:56 MST 2003
Archived on: Tue Sep 5 15:24:44 MST 2006

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