August 2006 Archives by author

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Starting: Tue Aug 1 00:02:23 MST 2006
Ending: Thu Aug 31 23:46:24 MST 2006
Messages: 3082

• [asterisk-users] Polycom 501 : How to make it ring when already ona call   Brian Vincent (C)
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Brian Vincent (C)
• [asterisk-users] Hotels...   Brian Vincent (C)
• [asterisk-users] Page()   Leonardo Kamache (Gmail)
• [asterisk-users] Where can i get a telephone number of Brasilia or Rio de Janeiro in Brazil   Leonardo Kamache (Gmail)
• [asterisk-users] Multiple Vulnerabilities in Asterisk 1.2.10 (Fixed in 1.2.11)   Matt Riddell (IT)
• [asterisk-users] can not get ${LEN(VAR)} and greater than ">" to work for me   Matt Riddell (IT)
• [asterisk-users] CDR Function - Asterisk-1.2.10   Matt Riddell (IT)
• [asterisk-users] Tracing audio problems   Matt Riddell (IT)
• [asterisk-users] Asterisk Development and Release Cycle   Matt Riddell (IT)
• [asterisk-users] Analyze core file prodeced after safe_asterisk crashh   Matt Riddell (IT)
• [asterisk-users] Detect if cell phone or users   Matt Riddell (IT)
• [asterisk-users] Analyze core file prodeced after safe_asterisk crashh   Matt Riddell (IT)
• [asterisk-users] Re: Asterisk Development and Release Cycle   Matt Riddell (IT)
• [asterisk-users] Asterisk 'Hosting'   Chris Mason (Lists)
• [asterisk-users] About Digium cards and HP DL servers   Matt Riddell (NZ)
• [asterisk-users] [Solution] Call Asterisk from GoogleTalk and have it tell you the status of your IAX2 links.   Matt Riddell (NZ)
• [asterisk-users] Fwd: * and GTalk testing   Matt Riddell (NZ)
• [asterisk-users] Fwd: * and GTalk testing   Matt Riddell (NZ)
• [asterisk-users] AGI doesn't execute PHP5 script   Matt Riddell (NZ)
• [asterisk-users] AGI doesn't execute PHP5 script   Matt Riddell (NZ)
• [asterisk-users] AGI doesn't execute PHP5 script [SOLVED]   Matt Riddell (NZ)
• [asterisk-users] Ignoring the # key on a call   Matt Riddell (NZ)
• [asterisk-users] Is anybody moderating this list?   Matt Riddell (NZ)
• [asterisk-users] Run As User Asterisk   Matt Riddell (NZ)
• [asterisk-users] Asterisk 'Hosting'   Matt Riddell (NZ)
• [asterisk-users] Asterisk 'Hosting'   Matt Riddell (NZ)
• [asterisk-users] Asterisk 'Hosting'   Matt Riddell (NZ)
• [asterisk-users] Asterisk 'Hosting'   Matt Riddell (NZ)
• [asterisk-users] Asterisk Load Balance   Andy Chung (Power-All)
• [asterisk-users] SER Dispatcher Load Balance How-To?   Andy Chung (Power-All)
• [asterisk-users] SER Dispatcher Load Balance How-To?   Andy Chung (Power-All)
• [asterisk-users] Prague PTT?   Jason Aarons (US)
• [asterisk-users] Does anyone use T.38?   Jason Aarons (US)
• [asterisk-users] Cisco 7960G SIP firmware 8.4   Jason Aarons (US)
• [asterisk-users] 911 versus 9.911   Jason Aarons (US)
• [asterisk-users] Running AGI in background   Bromont -
• [asterisk-users] Server Hardware   Woodoo People .pGa!
• [asterisk-users] Server Hardware   Woodoo People .pGa!
• [asterisk-users] Connecting an cellphone to asterisk   Woodoo People .pGa!
• [asterisk-users] Getting strange behavior on SIP channels after upgrade to 1.2.11   Álvaro Palma
• [asterisk-users] Re: transform bridged call into a conference   Álvaro Palma
• [asterisk-users] sip incoming stop working, what to look for in logs?   Adrià Vidal
• [asterisk-users] Cisco Buddies   Adrià Vidal
• [asterisk-users] Disable the flash hook hold capability on a SIP-to-SIP or SIP-to-ZAP call?   Antonio José dos Santos Brandão
• [asterisk-users] Re: [asterisk-dev] Recent additions to the Digium Asterisk development team   Denis Galvão - iSolve
• [asterisk-users] Realtime Extensions -- Comments?   Denis Galvão - iSolve
• [asterisk-users] 480 "Temporarily Unavailable" message   Diego Andrés Asenjo González
• [asterisk-users] Strange SIP response   Diego Andrés Asenjo González
• [asterisk-users] TRANSFER   Ing. Germán González B.
• [asterisk-users] TRANSFER   Ing. Germán González B.
• [asterisk-users] PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units   Josué Conti
• [asterisk-users] Providers that offer contract   Llorenç Suau
• [asterisk-users] Queue Management   Nicolás Gudiño
• [asterisk-users] Joining calls via manager.api or AGI   Nicolás Gudiño
• [asterisk-users] Phone status   Nicolás Gudiño
• [asterisk-users] Re: Manager API: matching an Originate to the Newchannel event   Miloš Kocbek
• [asterisk-users] Cisco Buddies   Adrià Vidal
• [asterisk-users] How to, astcc and virtuemart   Julio Cáceres A.
• [asterisk-users] 7970 'LoadID incorrect' problem   "Hans-Jürgen Brand"
• [asterisk-users] Dell Poweredge 1950 / 2950   Frédéric Marti
• SV: [asterisk-users] VOIP phone for Receptionist use   Jon Schøpzinsky
• [asterisk-users] IAX2 Trunking CPU usage   Jon Schøpzinsky
• SV: [asterisk-users] IAX trunk behing NAT with dynamic IP   Jon Schøpzinsky
• [asterisk-users] lbProxy   Jon Schøpzinsky
• SV: [asterisk-users] Hint extension issue - bug?   Jon Schøpzinsky
• SV: [asterisk-users] E61   Jon Schøpzinsky
• [asterisk-users] What I can use with ASTERISK to call clients to remind them about their appointments   Fabian Müller
• [asterisk-users] placing a call with the Manager interface   Fabian Müller
• [asterisk-users] New people in this world and his problem with ISDN   Dominik Kiełb
• [asterisk-users] Anybody using Eicon SoftIP with Asterisk   Dominik Kiełb
• [asterisk-users] Re: PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units   Tomislav Parčina
• [asterisk-users] Re: Asterisk Jobs Update   Tomislav Parčina
• [asterisk-users] Zaptel install - Fedora Core 5   Tomislav Parčina
• [asterisk-users] Re: Re: PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units   Tomislav Parčina
• [asterisk-users] Re: Zaptel install - Fedora Core 5   Tomislav Parčina
• [asterisk-users] GSM gateway and FXO ATA   Tomislav Parčina
• [asterisk-users] Asterisk forum - forum.globalvoicenet.com   Tomislav Parčina
• [asterisk-users] Re: Zaptel install - Fedora Core 5   Tomislav Parčina
• [asterisk-users] Re: Re: Zaptel install - Fedora Core 5   Tomislav Parčina
• [asterisk-users] Re: GSM gateway and FXO ATA   Tomislav Parčina
• [asterisk-users] Re: Re: GSM gateway and FXO ATA   Tomislav Parčina
• [asterisk-users] Re: Asterisk forum - forum.globalvoicenet.com   Tomislav Parčina
• [asterisk-users] H264   Tomislav Parčina
• [asterisk-users] Re: GSM gateway and FXO ATA   Tomislav Parčina
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Tomislav Parčina
• [asterisk-users] Re: DNS   Tomislav Parčina
• [asterisk-users] Re: H264   Tomislav Parčina
• [asterisk-users] Cisco 7970   Tomislav Parčina
• [asterisk-users] Asterisk Development and Release Cycle   Tomislav Parčina
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Tomislav Parčina
• [asterisk-users] Re: Asterisk Development and Release Cycle   Tomislav Parčina
• [asterisk-users] Re: Asterisk Development and Release Cycle   Tomislav Parčina
• [asterisk-users] Re: Asterisk Development and Release Cycle   Tomislav Parčina
• [asterisk-users] Re: Asterisk Development and Release Cycle   Tomislav Parčina
• [asterisk-users] Re: Cisco 7960G SIP firmware 8.4   Tomislav Parčina
• [asterisk-users] Re: Cisco 7960G SIP firmware 8.4   Tomislav Parčina
• [asterisk-users] Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] Re: GSM gateway and FXO ATA   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware   Tomislav Parčina
• [asterisk-users] CDR Variable   Abdul
• [asterisk-users] IAX2 Auto fallthrough   Abdul
• [asterisk-users] monmp3thread: Request to schedule in the past?!?!   Abdul
• [asterisk-users] quintum Calling Card   Abdul
• [asterisk-users] quintum Calling Card   Abdul
• [asterisk-users] Call Max Time   Abdul
• [asterisk-users] Call Max Time   Abdul
• [asterisk-users] Call Max Time   Abdul
• [asterisk-users] Dial C option   Master Abi
• [asterisk-users] Dial C option   Master Abi
• [asterisk-users] Dial C option   Master Abi
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   Juan Pablo Abuyeres
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   Juan Pablo Abuyeres
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   Juan Pablo Abuyeres
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   Juan Pablo Abuyeres
• [asterisk-users] Re: does OOH323 channel support Early Media?   Vlasis Hatzistavrou Mailing Lists Account
• [asterisk-users] Missing Fast AGI calling 'h' exten without hanging up   Rich Adamson
• [asterisk-users] Extend analog phone via SIP (OT)   Rich Adamson
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Rich Adamson
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Rich Adamson
• [asterisk-users] Detecting voicemail from CO on FXO port   Rich Adamson
• [asterisk-users] Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk   Rich Adamson
• [asterisk-users] Re: [asterisk-dev] Tuning Software Echo Cancellers   Rich Adamson
• [asterisk-users] codec_g729a.so coredump in SVN trunk   Rich Adamson
• [asterisk-users] Problems with Codecs in Asterisk   Rich Adamson
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Rich Adamson
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   Rich Adamson
• Autoreply: [asterisk-users] Snom MWI   Rich Adamson
• [asterisk-users] Mac Address Authentication Methods   Rich Adamson
• [asterisk-users] SIP trunks: order or type   Rich Adamson
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Rich Adamson
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Rich Adamson
• [asterisk-users] SIP header challenge   Rich Adamson
• [asterisk-users] 911 Testing   Rich Adamson
• [asterisk-users] SIP header challenge   Rich Adamson
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Rich Adamson
• [asterisk-users] SIP trunks: order or type   Rich Adamson
• [asterisk-users] CallerID is not displaying for my incoming calls   Rich Adamson
• [asterisk-users] 1.2.10 - g726 Issues   Rich Adamson
• [asterisk-users] 1.2.10 - g726 Issues   Rich Adamson
• [asterisk-users] Page()   Rich Adamson
• [asterisk-users] Digium TDM400P Vs Sangoma A200   Rich Adamson
• [asterisk-users] Digium TDM400P Vs Sangoma A200   Rich Adamson
• [asterisk-users] Dial statement problem   Rich Adamson
• [asterisk-users] Linksys SPA-3102   Rich Adamson
• [asterisk-users] Linksys SPA-3102   Rich Adamson
• [asterisk-users] Linksys SPA-941 Message Waiting Indicator   Rich Adamson
• [asterisk-users] No retry after DNS failure   Rich Adamson
• [asterisk-users] using asterisk + sangoma a102 to simulate telco PRI: is possible?   Rich Adamson
• [asterisk-users] Sipura 3000 dialplan strings.   Rich Adamson
• [asterisk-users] Re: Zaptel install - Fedora Core 5   Rich Adamson
• [asterisk-users] Re: Zaptel install - Fedora Core 5   Rich Adamson
• [asterisk-users] LOUD MP3 Hold Music   Rich Adamson
• [asterisk-users] Strange SIP response   Rich Adamson
• [asterisk-users] Calls over VPN   Rich Adamson
• [asterisk-users] VM - advanced options?   Rich Adamson
• [asterisk-users] Trunk with multiple IPs?   Rich Adamson
• [asterisk-users] Cisco Router QOS and IAX2   Rich Adamson
• [asterisk-users] Trunk with multiple IPs?   Rich Adamson
• [asterisk-users] Cisco Router QOS and IAX2   Rich Adamson
• [asterisk-users] Modems dialing over sangoma a104d   Rich Adamson
• [asterisk-users] Trunk with multiple IPs?   Rich Adamson
• [asterisk-users] DNS   Rich Adamson
• [asterisk-users] zap channel media volume   Rich Adamson
• [asterisk-users] DNS   Rich Adamson
• [asterisk-users] zap channel media volume   Rich Adamson
• [asterisk-users] Re: GSM gateway and FXO ATA   Rich Adamson
• [asterisk-users] Tracing audio problems   Rich Adamson
• [asterisk-users] lost packets when bridging zap and iax   Rich Adamson
• [asterisk-users] Missing number 2 in "advanced options" of VM   Rich Adamson
• [asterisk-users] Can anyone recommend a large button sip phone for the elderley.   Rich Adamson
• [asterisk-users] Problem with a TDM400P   Rich Adamson
• [asterisk-users] oddity with TDM400P / Asterisk setup   Rich Adamson
• [asterisk-users] HP ProLiant and Digium 24xxp   Rich Adamson
• [asterisk-users] iax vs. sip?   Rich Adamson
• [asterisk-users] High Availability with PRI failover   Vicente Aguilar
• [asterisk-users] Media direct from IAX Phone to IAX Phone   Kamran Ahmad
• [asterisk-users] Load balancing of IAX2   Kamran Ahmad
• [asterisk-users] autocreatepeer in iax   Kamran Ahmad
• [asterisk-users] Re: autocreatepeer in iax   Kamran Ahmad
• [asterisk-users] Re: autocreatepeer in iax   Kamran Ahmad
• [asterisk-users] Re: Load balancing of IAX2   Kamran Ahmad
• [asterisk-users] Re: Load balancing of IAX2   Kamran Ahmad
• [asterisk-users] Re: Load balancing of IAX2   Kamran Ahmad
• [asterisk-users] Polycom 601 Issues   Nathan Alberti
• [asterisk-users] Polycom 601 Issues   Nathan Alberti
• [asterisk-users] Polycom 601 Issues   Nathan Alberti
• [asterisk-users] Cisco 7960G SIP firmware 8.4   Nathan Alberti
• [asterisk-users] How to set externip in sip.conf automatically?   Larry Alkoff
• [asterisk-users] How to set externip in sip.conf automatically?   Larry Alkoff
• [asterisk-users] How to set externip in sip.conf automatically?   Larry Alkoff
• [asterisk-users] How to set externip in sip.conf automatically?   Larry Alkoff
• [asterisk-users] How to set externip in sip.conf automatically?   Larry Alkoff
• [asterisk-users] How to set externip in sip.conf automatically?   Larry Alkoff
• [asterisk-users] Attempt to setup paging and intercom   Larry Alkoff
• [asterisk-users] Re: Attempt to setup paging and intercom   Larry Alkoff
• [Asterisk-Users] Extension Ring on Multiple Phones   Larry Alkoff
• [Asterisk-Users] Extension Ring on Multiple Phones   Larry Alkoff
• [Asterisk-Users] Extension Ring on Multiple Phones   Larry Alkoff
• [Asterisk-Users] Extension Ring on Multiple Phones   Larry Alkoff
• [Asterisk-Users] Extension Ring on Multiple Phones   Larry Alkoff
• [Asterisk-Users] Extension Ring on Multiple Phones   Larry Alkoff
• [asterisk-users] How to use *411 using either last or first name?   Larry Alkoff
• [asterisk-users] Echo cancell   Pablo Allietti
• [asterisk-users] Anyone use TACI.pl for a click to call app? - Doesn't seem to want to work for me   Christopher Aloi
• [asterisk-users] Re: SIP Debug to file - Is it possible?   Christopher Aloi
• [asterisk-users] Re: SIP Debug to file - Is it possible?   Christopher Aloi
• [asterisk-users] Simple CDR parser to print to webpage   Christopher Aloi
• [asterisk-users] Help compiling asterisk-addons on Debian?   Christopher Aloi
• [asterisk-users] Help compiling asterisk-addons on Debian?   Christopher Aloi
• [asterisk-users] Asterisk Real Time Engine - Fails to Connect to MySQL   Christopher Aloi
• [asterisk-users] Re: Asterisk Real Time Engine - Fails to Connect to MySQL   Christopher Aloi
• [asterisk-users] Shared NFS or Shared MySQL for redundant secondary server?   Christopher Aloi
• [asterisk-users] app_conference   Aldo Alexander Leyva Alvarado
• [asterisk-users] app_conference   Aldo Alexander Leyva Alvarado
• [asterisk-users] Hint extension issue - bug?   Lucas Alvarez
• [asterisk-users] Hint extension issue - bug?   Lucas Alvarez
• [asterisk-users] Issue with IAX2 and Real Time configuration   Facundo Ameal
• [asterisk-users] Re: Issue with IAX2 and Real Time configuration   Facundo Ameal
• [asterisk-users] Re: How to configure NOKIA N70 with Asterisk?   Benny Amorsen
• [asterisk-users] Re: Snom MWI   Benny Amorsen
• [asterisk-users] Re: Asterisk 'Hosting'   Benny Amorsen
• [asterisk-users] Re: Asterisk 'Hosting'   Benny Amorsen
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Benny Amorsen
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Benny Amorsen
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Benny Amorsen
• [asterisk-users] sip phone networking question [possibly OT]   Colin Anderson
• [asterisk-users] Snom MWI   Colin Anderson
• [asterisk-users] Asterisk IAXmodem HylaFax?   Colin Anderson
• [asterisk-users] Asterisk IAXmodem HylaFax?   Colin Anderson
• [asterisk-users] Will Asterisk work with Exchange 2007 UM?   Colin Anderson
• [asterisk-users] OT: Any thoughts on the new Xserve?   Colin Anderson
• [asterisk-users] Sangoma A200 and Disconnected Cables   Andres
• [asterisk-users] Sangoma A200D and DTMF Detection   Andres
• [asterisk-users] Re: Sangoma A200D and DTMF Detection   Andres
• [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)   Andres
• [asterisk-users] Analog-to-VoIP: blade?   Cory Andrews
• [asterisk-users] Can anyone recommend a large button sip phone forthe elderley.   Cory Andrews
• [asterisk-users] Phone number lookup public database   Thameem Ansari
• [asterisk-users] MWI from Asterisk to Meridian   kritikus Araklidas
• [asterisk-users] MWI from Asterisk to Meridian   kritikus Araklidas
• [asterisk-users] RAM memory high Comsumption   kritikus Araklidas
• [asterisk-users] Asterisk suddenly die   kritikus Araklidas
• [asterisk-users] MWI from Asterisk to Meridian   kritikus Araklidas
• [asterisk-users] Asterisk Configuration   kritikus Araklidas
• [asterisk-users] voicemailmain   kritikus Araklidas
• [asterisk-users] voicemailmain   kritikus Araklidas
• [asterisk-users] Idiot questions   kritikus Araklidas
• [asterisk-users] SNOM 360   Julio Arruda
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] how to check the status of a channel   Thomas Artner
• [asterisk-users] how to check the status of a channel   Thomas Artner
• [asterisk-users] Attended Transfer call return with asterisk + sipura spa2002   Thomas Artner
• [asterisk-users] strange behaviour of a zaptel device   Thomas Artner
• [asterisk-users] strange behaviour of a zaptel device SOLVED   Thomas Artner
• [asterisk-users] Sending Email From A Dial Plan   Thomas Artner
• [asterisk-users] E61   Thomas Artner
• [asterisk-users] E61   Thomas Artner
• [asterisk-users] Idiot questions   Thomas Artner
• [asterisk-users] Asterisk 'Hosting'   Pablo L. Arturi
• [asterisk-users] calling in-out   Pablo L. Arturi
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Lacy Moore - Aspendora
• [asterisk-users] Set DID   Lacy Moore - Aspendora
• [asterisk-users] 7970 'LoadID incorrect' problem   Lacy Moore - Aspendora
• [asterisk-users] Connecting two asterisk servers   Lacy Moore - Aspendora
• [asterisk-users] Cisco 7960G SIP firmware 8.4   Lacy Moore - Aspendora
• [asterisk-users] visual indication of temp. closed mode   Lacy Moore - Aspendora
• [asterisk-users] Cisco 7970 8.0.4 SIP firmware   Lacy Moore - Aspendora
• [asterisk-users] RE: asterisk-users Digest, Vol 25, Issue 2   AstATN
• Subject: [asterisk-users] Slow dialing from PBX via E1   AstATN
• [asterisk-users] MWI from Asterisk to Meridian   AstATN
• [asterisk-users] Ringing all extensions   AstATN
• [asterisk-users] Asterisk and Siemens Legacy PBX   AstATN
• [asterisk-users] AddQueueMember and Local channel   Asterisk
• [asterisk-users] AddQueueMember and Local channel   Asterisk
• [asterisk-users] Is it possible to call System dialplan application via AMI?   Asterisk
• [asterisk-users] Asterisk - Comfort Noise   Asterisk at rpsuk.com
• [asterisk-users] Asterisk - Comfort Noise - Patch/Release   Asterisk at rpsuk.com
• [asterisk-users] does anyone offer truly unlimited voip in the US   Asterisk at rpsuk.com
• [asterisk-users] does anyone offer truly unlimited voip in the US   Asterisk at rpsuk.com
• [asterisk-users] Sipura SPA-841   Giedrius Augys
• [asterisk-users] MP-124   Giedrius Augys
• [asterisk-users] IAXMODEM   Giedrius Augys
• [asterisk-users] IAXMODEM   Giedrius Augys
• [asterisk-users] IAXMODEM   Giedrius Augys
• [asterisk-users] SIP NOTIFY   Giedrius Augys
• [asterisk-users] Problems with recording   Giedrius Augys
• [asterisk-users] Compilation   Dan Austin
• [asterisk-users] Compilation   Dan Austin
• [asterisk-users] 0005162: RTP Packetization : Few questions   Dan Austin
• [asterisk-users] zap channel media volume   JD Austin
• [asterisk-users] zap channel media volume   JD Austin
• [asterisk-users] determining meetme user number   Simon Austin
• [asterisk-users] detecting a users number using the dialplan or AGI   Simon Austin
• [asterisk-users] Permission for files generated by voicemail   Jean-Yves Avenard
• [asterisk-users] sms callback?   B
• [asterisk-users] Re: sms callback?   B
• [asterisk-users] DTMF + voipjet   B
• [asterisk-users] Queue Monitoring Broken?   J B
• [asterisk-users] postfix and asterisk   J B
• [asterisk-users] CTI   Fernando BERRETTA
• [asterisk-users] VoipNow 1.2.0 Beta   Michiel van Baak
• [asterisk-users] Dundi and Dial Arguments   Michiel van Baak
• [asterisk-users] Dundi and Dial Arguments   Michiel van Baak
• [asterisk-users] voicemail in mp3 format   Michiel van Baak
• [asterisk-users] IAX trunk behing NAT with dynamic IP   Michiel van Baak
• [asterisk-users] Iptables ,rtp   Michiel van Baak
• [asterisk-users] [RESOLVED] One way audion on Sangoma   Michiel van Baak
• [asterisk-users] [RESOLVED] One way audion on Sangoma   Michiel van Baak
• [asterisk-users] [RESOLVED] One way audion on Sangoma   Michiel van Baak
• [asterisk-users] Shared NFS or Shared MySQL for redundant secondary server?   Michiel van Baak
• [asterisk-users] How to set MWI   Michiel van Baak
• [asterisk-users] Detect if cell phone or users   Michiel van Baak
• [asterisk-users] dialplan help   Michiel van Baak
• [asterisk-users] MOH Silence   Wes Baehr
• [asterisk-users] HFC-S Cards in the UK   Chris Bagnall
• [asterisk-users] FXO module burn out !?   Rostislav Bagrov
• [asterisk-users] Presence SUBSCRIBE/NOTIFY behaviour   Shaun Bailey
• [asterisk-users] Rookie question, trying to learn   Time Bandit
• [asterisk-users] astbill white screen!!   Time Bandit
• [asterisk-users] Text to Speech   Time Bandit
• [asterisk-users] Text to Speech   Time Bandit
• [asterisk-users] About IVR and Oracle   Time Bandit
• [asterisk-users] Monitoring/Listening In   Time Bandit
• [asterisk-users] 7970 'LoadID incorrect' problem   Time Bandit
• [asterisk-users] detecting a users number using the dialplan or AGI   Time Bandit
• [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration   Time Bandit
• [asterisk-users] Asterisk t38passthrough   Edgar Barbosa
• [asterisk-users] No retry after DNS failure   Remco Barendse
• [asterisk-users] Unicall stack, right versions?   Barzilai
• [asterisk-users] Equivalent of channel switching?   Barzilai
• [asterisk-users] Recent additions to the Digium Asterisk development team   Barzilai
• [asterisk-users] Re: what is the real use of AEL?   Barzilai
• [asterisk-users] Recent additions to the Digium Asteriskdevelopment team   Barzilai
• [asterisk-users] Sipura 3000 and Asterisk   Barzilai
• [asterisk-users] Hola, en que andas?   Leonardo Bauchwitz
• [Asterisk-Users] Asterisk technician needed in Buenos Aires Argentina   Leonardo Bauchwitz
• [asterisk-users] does anyone offer truly unlimited voip in the US   Erv Bauman
• [asterisk-users] Run As User Asterisk   Forrest Beck
• [asterisk-users] Junk at beginning of frame   Forrest Beck
• [asterisk-users] Phone Newbie Questions   Brian Becker
• [asterisk-users] GROUP() and queues   Juraj Bednar
• [asterisk-users] asterisk presence (from manager API)   Juraj Bednar
• [asterisk-users] asterisk presence (from manager API)   Juraj Bednar
• [asterisk-users] weird sound with IAX   Juraj Bednar
• [asterisk-users] asterisk presence (from manager API)   Juraj Bednar
• [asterisk-users] queue announcements when using ringback   Nic Bellamy
• [asterisk-users] Manager API: matching an Originate to the Newchannel event   Nic Bellamy
• [asterisk-users] Arrays ???   Benchev
• [asterisk-users] Re: Arrays ???   Benchev
• [asterisk-users] Polycom IP600 HTTP Provisioning problem   Dovid Bender
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] Asterisk Manager Vb 6   Dovid Bender
• [asterisk-users] SNOM 360   Dovid Bender
• [asterisk-users] SNOM 360   Dovid Bender
• [asterisk-users] SIP and podcasts   Dovid Bender
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] How to forward a call to an outside line   Dovid Bender
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] g729 and trafic   Dovid Bender
• [asterisk-users] Using a DB for Configurations   Dovid Bender
• [asterisk-users] [update] SIP/Qualify   Dovid Bender
• [asterisk-users] SIP/Qualify   Dovid Bender
• [SPAM] Re: [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] Hotels...   Dovid Bender
• [asterisk-users] Ztdummy - No audio in BackGround function   Dovid Bender
• [asterisk-users] Sipura SPA-841   Dovid Bender
• [asterisk-users] Port Forwarding SIP rtp   Dovid Bender
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuckin a hole   Dovid Bender
• [asterisk-users] Ever donate Software to Digium? If you did your afool.   Dovid Bender
• [asterisk-users] 911 Testing   Dovid Bender
• [asterisk-users] Macro inside macro   Dovid Bender
• [asterisk-users] SIP Connection Problems   Dovid Bender
• [asterisk-users] SIP Connection Problems   Dovid Bender
• [asterisk-users] Problem with dtmf and voice mail   Dovid Bender
• [asterisk-users] multiple offices / hard phones / service provider   Dovid Bender
• [asterisk-users] Re: SIP Connection Problems   Dovid Bender
• [asterisk-users] Manager Interface API's   Dovid Bender
• [asterisk-users] SIP Qualify   Dovid Bender
• [asterisk-users] SIP Qualify   Dovid Bender
• [asterisk-users] Problem with dtmf and voice mail   Dovid Bender
• [asterisk-users] New Device   Dovid Bender
• [asterisk-users] New asterisk jukebox needs testing   Dovid Bender
• [asterisk-users] Polycom upgrade issue   Dovid Bender
• [asterisk-users] Asterisk 'Hosting'   Dovid Bender
• [asterisk-users] Asterisk 'Hosting'   Dovid Bender
• [asterisk-users] Changing CID   Dovid Bender
• [asterisk-users] how to add prefix 0 (zero) when coming from ISDNtrunk   Dovid Bender
• [asterisk-users] Sip suppression   Dovid Bender
• [asterisk-users] Realtime include   Dovid Bender
• SV: [Asterisk-Users] Nokia E61   Dovid Bender
• [asterisk-users] One way audion on Sangoma   Dovid Bender
• [asterisk-users] E61   Dovid Bender
• [asterisk-users] Realtime and hints   Dovid Bender
• [asterisk-users] Realtime Extensions -- Comments?   Dovid Bender
• [asterisk-users] Realtime and hints   Dovid Bender
• [asterisk-users] Realtime and hints   Dovid Bender
• [asterisk-users] [RESOLVED] One way audion on Sangoma   Dovid Bender
• [asterisk-users] Realtime and hints   Dovid Bender
• [asterisk-users] IP phone with 2 ethernet jacks   Dovid Bender
• [asterisk-users] Re: SV: E61   Dovid Bender
• [asterisk-users] Realtime and hints   Dovid Bender
• [asterisk-users] IP phone with 2 ethernet jacks   Dovid Bender
• [asterisk-users] Re: SV: E61   Dovid Bender
• [asterisk-users] [RESOLVED] One way audion on Sangoma   Dovid Bender
• [asterisk-users] Cannot dial out through SIP provider   Dovid Bender
• [asterisk-users] Prompts recording for Asterisk   Dovid Bender
• [asterisk-users] Asterisk with PABX   Dovid Bender
• [asterisk-users] Detect if cell phone or users   Dovid Bender
• [asterisk-users] cmd DIAL - Who picked up the call?   Vadim Berezniker
• [asterisk-users] cmd DIAL - Who picked up the call?   Vadim Berezniker
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Vadim Berezniker
• [asterisk-users] Re: asterisk-users Digest, Vol 25, Issue 119   Aldo Bergamini
• [asterisk-users] nat and qualify questions   BerkHolz, Steven
• [asterisk-users] Iaxy and SendDTMF??   BerkHolz, Steven
• [asterisk-users] iax vs. sip?   BerkHolz, Steven
• [asterisk-users] Queue Monitoring Broken?   Jeremy Betts
• [asterisk-users] Will Asterisk work with Exchange 2007 UM?   Matt Birmingham
• [asterisk-users] IP phone with 2 ethernet jacks   Matt Birmingham
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Gareth Blades
• [asterisk-users] Is there a smarter way to ban expensive calls in dial plan?   Chris Blunt
• [asterisk-users] Mix Monitor call quality   robert Boardman
• [asterisk-users] VoipNow 1.2.0 Beta   Greg Boehnlein
• [asterisk-users] Polycom 1.6.7 Firmware Messages Button   Greg Boehnlein
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Greg Boehnlein
• [asterisk-users] VoipNow 1.2.0 Beta   Greg Boehnlein
• [asterisk-users] RE VoipNow 1.2.0 Beta   Greg Boehnlein
• [asterisk-users] High Availability with PRI failover   Greg Boehnlein
• [asterisk-users] Asterisk Performance without RTP?   Greg Boehnlein
• [asterisk-users] Asterisk Performance without RTP?   Greg Boehnlein
• [asterisk-users] Asterisk Performance without RTP?   Greg Boehnlein
• [asterisk-users] DTMF between cisco and sipura going through asterisk   Greg Boehnlein
• [asterisk-users] Call Routing based on Caller-Id   Leo Ann Boon
• [asterisk-users] MWI from Asterisk to Meridian   Leo Ann Boon
• [asterisk-users] PRI Dropouts (Solved)   Leo Ann Boon
• [asterisk-users] VoiceMail and Fax on same extension   Leo Ann Boon
• [asterisk-users] Accessing SIP URI (not ${SIPURI})   Leo Ann Boon
• [asterisk-users] Changing CID   Leo Ann Boon
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Leo Ann Boon
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Leo Ann Boon
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Leo Ann Boon
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Leo Ann Boon
• [asterisk-users] SIP Encryption in China   Leo Ann Boon
• [asterisk-users] Re: SIP Debug to file - Is it possible?   Leo Ann Boon
• [asterisk-users] How to modify incoming DNIS?   Leo Ann Boon
• [asterisk-users] No outbound with A2Billing   Leo Ann Boon
• [asterisk-users] Asterisk 'Hosting'   Casey Boone
• [asterisk-users] Digium TDM400P Vs Sangoma A200   Jonathan Borden
• [asterisk-users] Detecting voicemail from CO on FXO port and passing to H.323 phone. Possible?   Bob Bosiljevac
• [asterisk-users] Detecting voicemail from CO on FXO port   Bob Bosiljevac
• [asterisk-users] RE VoipNow 1.2.0 Beta   Peter Bowyer
• [asterisk-users] Port Forwarding SIP rtp   Peter Bowyer
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Peter Bowyer
• [asterisk-users] Asterisk Jobs Update   Peter Bowyer
• [asterisk-users] cannot received calls in pstn line   Crazy Boy
• [asterisk-users] How to connect Snom softphone from my home?   Crazy Boy
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Crazy Boy
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Crazy Boy
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Crazy Boy
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Crazy Boy
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Crazy Boy
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Crazy Boy
• [asterisk-users] CallerID is not displaying for my incoming calls   Crazy Boy
• [asterisk-users] Problems with incoming authentication   Crazy Boy
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Crazy Boy
• [asterisk-users] CallerID is not displaying for my incoming calls   Crazy Boy
• [asterisk-users] CallerID is not displaying for my incoming calls   Crazy Boy
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Crazy Boy
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Crazy Boy
• [asterisk-users] CallerID is not displaying for my incoming calls   Crazy Boy
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Crazy Boy
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Crazy Boy
• [asterisk-users] CallerID is not displaying for my incoming calls   Crazy Boy
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Crazy Boy
• [asterisk-users] How can I implement Music on Call Transfer?   Crazy Boy
• [asterisk-users] How can I implement Music on Call Transfer?   Crazy Boy
• [asterisk-users] Nobody is responding. Why? (Implement music on transfer)   Crazy Boy
• [asterisk-users] Connecting two asterisk servers   Crazy Boy
• [asterisk-users] does anyone offer truly unlimited voip in the US   Crazy Boy
• [asterisk-users] does anyone offer truly unlimited voip in the US   Crazy Boy
• [asterisk-users] Linksys SPA-941 Message Waiting Indicator   Cliff Brake
• [asterisk-users] how to enable REACHABLE/UNREACHABLE messages in logs   Cliff Brake
• [asterisk-users] BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN   Matteo Brancaleoni
• [asterisk-users] Auto Congestion   David Brazier
• [asterisk-users] Asterisk GUI tool needed   Tijl Van den Broeck
• [asterisk-users] Quick One - PHP Script to restart Asterisk   Tijl Van den Broeck
• [asterisk-users] Re: How to forward a call to an outside line   Greg Broiles
• [asterisk-users] ARI   Ryder Brook
• [asterisk-users] Problem with dtmf and voice mail   Paul A Brown
• [asterisk-users] Problem with dtmf and voice mail   Paul A Brown
• [asterisk-users] Problem with dtmf and voice mail   Paul A Brown
• [asterisk-users] 7970 SIP image   Paul A Brown
• [asterisk-users] 7970 SIP image   Paul A Brown
• [asterisk-users] Is there an Asterisk@Home specific list?   Paul A Brown
• [asterisk-users] 7970 'LoadID incorrect' problem   Paul A Brown
• [asterisk-users] 7970 'LoadID incorrect' problem   Paul A Brown
• [asterisk-users] 7970 'LoadID incorrect' problem   Paul A Brown
• [asterisk-users] 7970 'LoadID incorrect' problem   Paul A Brown
• [asterisk-users] ExtensionState always returns 1   Paco Brufal
• [asterisk-users] By week extension dialing   Dan Brummer
• [asterisk-users] RE: By week extension dialing   Dan Brummer
• [asterisk-users] By week extension dialing   Dan Brummer
• [asterisk-users] Re: By week extension dialing   Dan Brummer
• [asterisk-users] By week extension dialing   Dan Brummer
• [asterisk-users] MySQL undefined symbol: __pure_virtual   Dan Brummer
• [asterisk-users] codec conversion   Russell Bryant
• [asterisk-users] Asterisk 1.0.1 in SuSE 10.0   Russell Bryant
• [asterisk-users] Using Flite in a call file.   Russell Bryant
• [asterisk-users] Simple config question   Russell Bryant
• [asterisk-users] autocreatepeer in iax   Russell Bryant
• [asterisk-users] Help with perl AGI script   Russell Bryant
• [asterisk-users] Help with perl AGI script   Russell Bryant
• [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?   Russell Bryant
• [asterisk-users] SLA Doc   Russell Bryant
• [asterisk-users] Can anyone recommend a large button sip phone for the elderley.   Chuck Bunn
• [asterisk-users] Can anyone recommend a large button sip phone for the elderly.   Chuck Bunn
• [asterisk-users] Is there a Blue tooth wireless headset that will work with asterisk?   Chuck Bunn
• [asterisk-users] Can anyone recommend a large button sip phone for the elderly.   Chuck Bunn
• [asterisk-users] How to play music on hold from within PHP AGI scripts?   Leo Burd
• [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?   Leo Burd
• [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?   Leo Burd
• [asterisk-users] DTMF intermittent on menu.   Shane Burrell
• [asterisk-users] VOIP phone for Receptionist use   Jeff Busch
• [asterisk-users] Asterisk 1.2.4 I hear other party's voice only when I speack need help   Tommaso Calosi
• [asterisk-users] Asterisk 1.2.4 I hear other party's voice only when I speack need help - SOLVED -   Tommaso Calosi
• [asterisk-users] Inbound problems, no audio   Pablo Salvador Capo
• [asterisk-users] Garbled initial voicemail prompt   Brian Capouch
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Brian Capouch
• [asterisk-users] Manager Interface API's   Brian Capouch
• [asterisk-users] Manager Interface API's   Brian Capouch
• [asterisk-users] Realtime and labels   Brian Capouch
• [asterisk-users] Realtime Extensions -- Comments?   Brian Capouch
• [asterisk-users] how to check the status of a channel   Marcus Carlson
• [asterisk-users] Asterisk 1.0.1 in SuSE 10.0   Oscar Carrillo
• [asterisk-users] SER + Asterisk PSTN calls don't hung up   Ricardo Carvalho
• [asterisk-users] SER + Asterisk PSTN calls don't hung up   Ricardo Carvalho
• [asterisk-users] Asterisk Real Time and sip.conf file used at the same time   Ricardo Carvalho
• [asterisk-users] Portuguese sound files available?   Ricardo Carvalho
• [asterisk-users] Asterisk t38passthrough   Ricardo Carvalho
• [asterisk-users] Asterisk t38passthrough   Ricardo Carvalho
• [asterisk-users] Does anyone use T.38?   Ricardo Carvalho
• [asterisk-users] Asterisk t38passthrough   Ricardo Carvalho
• [asterisk-users] Handytone 286 T.38 SDP parameters   Ricardo Carvalho
• [asterisk-users] Handytone 286 T.38 SDP parameters   Ricardo Carvalho
• [Fwd: Re: [asterisk-users] Asterisk t38passthrough]   Ricardo Carvalho
• [asterisk-users] How to forward a call to an outside line   Dan Casey
• [asterisk-users] Re: How to forward a call to an outside line   Dan Casey
• [asterisk-users] Re: How to forward a call to an outside line   Dan Casey
• [asterisk-users] Re: How to forward a call to an outside line   Dan Casey
• [asterisk-users] Re: Re: How to forward a call to an outside line   Dan Casey
• [asterisk-users] macro-dialout without specifying trunk   Dan Casey
• [asterisk-users] SIP asterisk over Linksys VPN   Dan Casey
• [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch   Santiago del Castillo
• [asterisk-users] IAX Trunking Only ONE-WAY   Lic. Miguel Cavazos
• [asterisk-users] MoH native volume   Carlos Chavez
• [asterisk-users] Asterisk H323 and Alcatel 4400   Carlos Chavez
• [asterisk-users] Video Conferencing over Asterisk   Carlos Chavez
• [asterisk-users] Simple incomming fax solution ...   Carlos Chavez
• [asterisk-users] E1 for Voice and Data with MFC/R2   Carlos Chavez
• [asterisk-users] E1 for Voice and Data with MFC/R2   Carlos Chavez
• [asterisk-users] E1 for Voice and Data with MFC/R2   Carlos Chavez
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Carlos Chavez
• [asterisk-users] CallerID is not displaying for my incoming calls   Carlos Chavez
• [asterisk-users] Asterisk Real Time and sip.conf file used at the same time   Carlos Chavez
• [asterisk-users] Asterisk Real Time and sip.conf file used at   Carlos Chavez
• [asterisk-users] Voicemail and languages other than english doesn't seem to work well   Carlos Chavez
• [asterisk-users] 3COM NBX and Digium Cards...   Carlos Chavez
• [asterisk-users] H264   Carlos Chavez
• [asterisk-users] Just a test   Edgar Alonso Lopez Chavez
• [asterisk-users] anothet tes   Edgar Alonso Lopez Chavez
• [asterisk-users] SRTP help   Khaled Chehab
• [asterisk-users] G729   Khaled Chehab
• [asterisk-users] Security   Khaled Chehab
• [asterisk-users] Security   Khaled Chehab
• [asterisk-users] Compilation   Khaled Chehab
• [asterisk-users] Compilation   Khaled Chehab
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Neil Cherry
• [asterisk-users] does anyone offer truly unlimited voip in the US   Bob Chiodini
• [asterisk-users] Softphone for Windows Mobile 5?   Christian
• [asterisk-users] Softphone for Windows Mobile 5?   Christian
• [asterisk-users] VOIP phone for Receptionist use   Dr. Michael J. Chudobiak
• [asterisk-users] Metermaid - Parking Slot   Dr. Michael J. Chudobiak
• [asterisk-users] [RESOLVED] One way audion on Sangoma   Dr. Michael J. Chudobiak
• [asterisk-users] E1 for Voice and Data with MFC/R2   Jorge Cisneros
• [asterisk-users] lost packets when bridging zap and iax   Simone Cittadini
• [asterisk-users] Page()   Dennis P. Clark
• [asterisk-users] Page()   Dennis P. Clark
• [asterisk-users] Page()   Dennis P. Clark
• [asterisk-users] Page()   Dennis P. Clark
• [asterisk-users] Page()   Dennis P. Clark
• [asterisk-users] Zaptel install - Fedora Core 5   Dennis P. Clark
• [asterisk-users] LOUD MP3 Hold Music   Dennis P. Clark
• [asterisk-users] LOUD MP3 Hold Music   Dennis P. Clark
• [asterisk-users] Re: what is the real use of AEL?   Mike Clark
• [asterisk-users] Calls over VPN   Henry J. Cobb
• [asterisk-users] Standard for transfer via IAX provider?   Henry J. Cobb
• [asterisk-users] DNS   Henry J. Cobb
• [asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!   Henry J. Cobb
• [asterisk-users] Re: question about asterisk DB   Tzafrir Cohen
• [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?   Tzafrir Cohen
• [asterisk-users] RAM memory high Comsumption   Tzafrir Cohen
• [asterisk-users] need dialout help in python script   Tzafrir Cohen
• [asterisk-users] Two card NT-TE mode   Tzafrir Cohen
• [asterisk-users] prob with star input agi-bin   Tzafrir Cohen
• [asterisk-users] Dapper Drake, Asterisk, and Faxing   Tzafrir Cohen
• [asterisk-users] Run As User Asterisk   Tzafrir Cohen
• [asterisk-users] Issues compiling addons on Fedora Core 3   Tzafrir Cohen
• [asterisk-users] Asterisk & Gizmo?   Tzafrir Cohen
• [asterisk-users] modprobe wctdm fails in /etc/rc.local on FC5   Tzafrir Cohen
• [asterisk-users] modprobe wctdm fails in /etc/rc.local on FC5   Tzafrir Cohen
• [asterisk-users] postfix and asterisk   Tzafrir Cohen
• [asterisk-users] Manager Interface API's   Tzafrir Cohen
• [asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5   Tzafrir Cohen
• [asterisk-users] Security   Tzafrir Cohen
• [asterisk-users] DMTF issues on voicemail on Zap   Tzafrir Cohen
• [asterisk-users] valgrind + Asterisk   Tzafrir Cohen
• [asterisk-users] Server Hardware   Tzafrir Cohen
• [asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5   Tzafrir Cohen
• [asterisk-users] Sending Email From A Dial Plan   Tzafrir Cohen
• [asterisk-users] Apache for FastAGI   Tzafrir Cohen
• [asterisk-users] Server Hardware   Tzafrir Cohen
• [asterisk-users] loading the prompt files in memory on Asterisk startup   Tzafrir Cohen
• [asterisk-users] sox &gsm   Tzafrir Cohen
• [asterisk-users] Sending signals to asterisk   Tzafrir Cohen
• [asterisk-users] Configure mailserver to deliver voicemail   Tzafrir Cohen
• [asterisk-users] Re: Zaptel install - Fedora Core 5   Tzafrir Cohen
• [asterisk-users] Re: Zaptel install - Fedora Core 5   Tzafrir Cohen
• [asterisk-users] Asterisk and spandsp   Tzafrir Cohen
• [asterisk-users] Re: Asterisk IAXmodem HylaFax?   Tzafrir Cohen
• [asterisk-users] Re: Asterisk IAXmodem HylaFax?   Tzafrir Cohen
• [asterisk-users] Annoying Bristuff   Tzafrir Cohen
• [asterisk-users] SSH connection hangs on logout?   Tzafrir Cohen
• [asterisk-users] Re: [Users] Mysql problem   Tzafrir Cohen
• [asterisk-users] voicemailmain   Tzafrir Cohen
• [asterisk-users] Re: GSM gateway and FXO ATA   Tzafrir Cohen
• [asterisk-users] Debian and Asterisk IAX2 channel driver   Tzafrir Cohen
• [asterisk-users] Asterisk with PABX   Tzafrir Cohen
• [asterisk-users] manual mods with GUI in place   Tzafrir Cohen
• [asterisk-users] does misdn-mqueue work if compiled with gcc 4?   Tzafrir Cohen
• [asterisk-users] Asterisk speaks Russian!   Tzafrir Cohen
• [asterisk-users] How to run a batch file on the asterisk CLI   Tzafrir Cohen
• [asterisk-users] Re: voicemail as email and attachment   Tzafrir Cohen
• [asterisk-users] NVFaxDetect and 1.2.10   John D. Coleman
• [asterisk-users] SendText Queue Notification   John D. Coleman
• [asterisk-users] Not Working   Adam Collard
• [asterisk-users] Idiot questions   Adam Collard
• [asterisk-users] Question about context for incoming calls   Adam Collard
• [asterisk-users] Polycom 1.6.7 firmware?   Dean Collins
• [asterisk-users] polycom headset question   Dean Collins
• [asterisk-users] polycom headset question   Dean Collins
• [asterisk-users] Polycom 1.6.7 firmware?   Dean Collins
• [asterisk-users] Polycom 1.6.7 firmware?   Dean Collins
• [asterisk-users] polycom config script   Dean Collins
• [asterisk-users] Ever donate Software to Digium? If you did your afool.   Dean Collins
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   Dean Collins
• [asterisk-users] Set DID?   Dean Collins
• [asterisk-users] Set DID?   Dean Collins
• [asterisk-users] Set DID?   Dean Collins
• [asterisk-users] Set DID?   Dean Collins
• [asterisk-users] Problem with dtmf and voice mail   Dean Collins
• [asterisk-users] Set DID   Dean Collins
• [asterisk-users] Set DID   Dean Collins
• [asterisk-users] Set DID   Dean Collins
• [asterisk-users] Always On Top 100 private companies   Dean Collins
• [asterisk-users] Is there an Asterisk@Home specific list?   Dean Collins
• [asterisk-users] Speech Recognition Apps   Dean Collins
• [asterisk-users] RE: [asterisk-dev] Phone status   Dean Collins
• [asterisk-users] Singapore   Dean Collins
• [asterisk-users] New Asterisk Voice Changer 0.4   Dean Collins
• [asterisk-users] Asterisk with PABX   Dean Collins
• [asterisk-users] Asterisk with PABX   Dean Collins
• [asterisk-users] Voicemail/Email Integration   Dean Collins
• [asterisk-users] Voicemail/Email Integration   Dean Collins
• [asterisk-users] Asterisk with PABX   Dean Collins
• [asterisk-users] does anyone offer truly unlimited voip in the US   Dean Collins
• [asterisk-users] does anyone offer truly unlimited voip in the US   Dean Collins
• [asterisk-users] does anyone offer truly unlimited voip in the US   Dean Collins
• [asterisk-users] Polycom HD Voice   Dean Collins
• [asterisk-users] Manager Interface API's   Michael Collins
• [asterisk-users] Manager Interface API's   Michael Collins
• [asterisk-users] Manager Interface API's (OSS doc discussion)   Michael Collins
• [asterisk-users] Recent additions to the Digium Asteriskdevelopment team   Michael Collins
• [asterisk-users] Asterisk with PABX   Michael Collins
• [asterisk-users] manual mods with GUI in place   Michael Collins
• [asterisk-users] IAX and Accountcode   Joshua Colp
• [asterisk-users] IAX and Accountcode   Joshua Colp
• [asterisk-users] canreinvite=yes and RTP dropping in and out   Joshua Colp
• [asterisk-users] Limitations of IAX   Joshua Colp
• [asterisk-users] canreinvite=yes and RTP dropping in and out   Joshua Colp
• [asterisk-users] Limitations of IAX   Joshua Colp
• [asterisk-users] SIP_HEADER() read-only   Joshua Colp
• [asterisk-users] MoH native volume   Joshua Colp
• [asterisk-users] wip 300 opensource code - changes to support SIPMESSAGE   Joshua Colp
• [asterisk-users] Ringing all extensions   Joshua Colp
• [asterisk-users] RAM memory high Comsumption   Joshua Colp
• [asterisk-users] Garbled initial voicemail prompt   Joshua Colp
• [asterisk-users] IAX Variables   Joshua Colp
• [asterisk-users] asterisk dosenot compile   Joshua Colp
• [asterisk-users] IAXy behind NAT, unable to transfer?   Joshua Colp
• [asterisk-users] IAXy behind NAT, unable to transfer?   Joshua Colp
• [asterisk-users] Problems with Hangup   Joshua Colp
• [asterisk-users] Problems with Hangup   Joshua Colp
• [asterisk-users] Sending INVITE to an unavailable phone - Bug?   Joshua Colp
• [asterisk-users] SIP Qualify   Joshua Colp
• [asterisk-users] Question about context for incoming calls   Joshua Colp
• [asterisk-users] Max number of SIP devices registered to anextension   Joshua Colp
• [asterisk-users] REGISTER attempt   Joshua Colp
• [asterisk-users] Max number of SIP devices registered to anextension   Joshua Colp
• [asterisk-users] Max number of SIP devices registered to anextension   Joshua Colp
• [asterisk-users] Got error when compiling asterisk 1.2.11   Joshua Colp
• [asterisk-users] isdn30 uk setup problem   Nicholas Colyer
• [asterisk-users] Does anyone use T.38?   Juan Jose Comellas
• [asterisk-users] MixMonitor and g729 licenses   Juan Jose Comellas
• [asterisk-users] Need help   Tech Con
• [asterisk-users] Anyone use TACI.pl for a click to call app? - Doesn't seem to want to work for me   Mitchel Constantin
• [asterisk-users] IAX trunk behing NAT with dynamic IP   Andre Courchesne - Consultant
• V: [asterisk-users] IAX trunk behing NAT with dynamic IP   Andre Courchesne - Consultant
• [asterisk-users] Asterisk time not the same as unit time ?   Andre Courchesne - Consultant
• [asterisk-users] Asterisk time not the same as unit time ?   Andre Courchesne - Consultant
• [asterisk-users] Reason to hit failed extension   Andre Courchesne - Consultant
• [asterisk-users] Double dial dtmf sounds   Andre Courchesne - Consultant
• [asterisk-users] IP interface "box" for Meridian type digital phone   Andre Courchesne - Consultant
• [asterisk-users] Re: Sipura SPA-3000 vs Sangoma A200   David Cook
• [asterisk-users] Dialplan "or" matching   David Cook
• [asterisk-users] Dialplan "or" matching   David Cook
• [asterisk-users] hints causing hang in reload   Sean Cook
• [asterisk-users] e&m wink, TE110P, * answers too soon   Sean Cook
• [asterisk-users] Modems dialing over sangoma a104d   Sean Cook
• [asterisk-users] Connecting an cellphone to asterisk   Alvaro Cornejo
• [asterisk-users] Asterisk 'Hosting'   Dave Cotton
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   David Coulson
• [asterisk-users] Hotels...   Jonathan k. Creasy
• [asterisk-users] Quick One - PHP Script to restart Asterisk   Jonathan k. Creasy
• [asterisk-users] failed calls   Jonathan k. Creasy
• [asterisk-users] quintum Calling Card   Jonathan k. Creasy
• [asterisk-users] quintum Calling Card   Jonathan k. Creasy
• [asterisk-users] Polycom 501 config questions   Jonathan k. Creasy
• [asterisk-users] Copying a recording to a voice mail box   Nate Criss
• [asterisk-users] Copying a recording to a voice mail box   Nate Criss
• [asterisk-users] Call Routing based on Caller-Id   Matthew Crocker
• [asterisk-users] Dialplan routing based on CallerID   Matthew Crocker
• [asterisk-users] Analog-to-VoIP: blade?   Matthew Crocker
• [asterisk-users] Realtime Extensions -- Comments?   Matthew Crocker
• [asterisk-users] Realtime Extensions -- Comments?   Matthew Crocker
• [asterisk-users] MySQL CDR   Diego Quintana Cruz
• [asterisk-users] Timeout Registration IAX2   Diego Quintana Cruz
• [asterisk-users] Slightly off-topic: Opinions of Comcast and Bellsouth?   Alistair Cunningham
• [asterisk-users] Hotels...   Ken D'Ambrosio
• [asterisk-users] No "zap" command?   Ken D'Ambrosio
• [asterisk-users] Analog-to-VoIP: blade?   Ken D'Ambrosio
• [asterisk-users] Sipura 3000 dialplan strings.   Ken D'Ambrosio
• [asterisk-users] Strange SIP response   Sergio R. D'Ippolito
• [asterisk-users] problems with wevbmail   Sergio R. D'Ippolito
• [asterisk-users] Re: problems with wevbmail   Sergio R. D'Ippolito
• [asterisk-users] NAT problems   Sergio R. D'Ippolito
• [asterisk-users] Problem with Tycho Voicemail   Sergio R. D'Ippolito
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   M D
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   M D
• [asterisk-users] problem with mfcr2 protocol   JOSE MANUEL CORTES DAVID
• [asterisk-users] problem with unicall   JOSE MANUEL CORTES DAVID
• [asterisk-users] problem with asterisk billing time...   JOSE MANUEL CORTES DAVID
• [asterisk-users] VOIP phone for Receptionist use   DM
• [asterisk-users] IAX trunk behing NAT with dynamic IP   DM
• [asterisk-users] Polycom IP430 won't finish boot   DM
• [asterisk-users] Polycom IP430 won't finish boot   DM
• [asterisk-users] Trixbox – Called party can't hangup   Allan Dalton
• [asterisk-users] DUNDi with SIP   Aaron Daniel
• [asterisk-users] DUNDi with SIP   Aaron Daniel
• [asterisk-users] Re: DUNDi with SIP   Aaron Daniel
• [asterisk-users] Re: DUNDi with SIP   Aaron Daniel
• [asterisk-users] Re: DUNDi with SIP   Aaron Daniel
• [asterisk-users] Macro inside macro   Aaron Daniel
• [asterisk-users] Asterisk Real Time and sip.conf file used at the same time   Aaron Daniel
• [asterisk-users] How To NOT Generate A CDR For A Call?   Aaron Daniel
• [asterisk-users] Realtime Extensions -- Comments?   Aaron Daniel
• [asterisk-users] Realtime Extensions -- Comments?   Aaron Daniel
• [asterisk-users] Direct to Voicemail   Aaron Daniel
• [asterisk-users] Realtime and hints   Aaron Daniel
• [asterisk-users] voicemailmain   Aaron Daniel
• [asterisk-users] Realtime and hints   Aaron Daniel
• [asterisk-users] SER Dispatcher Load Balance How-To?   Aaron Daniel
• [asterisk-users] SER Dispatcher Load Balance How-To?   Aaron Daniel
• [Asterisk-Users] Using HINT with Cisco 7940/SIP   Aaron Daniel
• [asterisk-users] 911 versus 9.911   Aaron Daniel
• [asterisk-users] Cisco 7970 8.0.4 SIP firmware   Aaron Daniel
• [asterisk-users] Cisco 7970 8.0.4 SIP firmware   Aaron Daniel
• [asterisk-users] Two card NT-TE mode   Klaus Darilion
• [asterisk-users] Re: PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units   Klaus Darilion
• [asterisk-users] SIP ActiveX?   Klaus Darilion
• [asterisk-users] transform bridged call into a conference   Klaus Darilion
• [asterisk-users] transform bridged call into a conference   Klaus Darilion
• [asterisk-users] Sangoma Problems - A104d not detected   Klaus Darilion
• [asterisk-users] Sangoma Problems - A104d not detected - solved   Klaus Darilion
• [asterisk-users] Voicemail and languages other than english doesn't seem to work well   Dominique Dartois
• [asterisk-users] Voicemail and languages other than englishdoesn't seem to work well   Dominique Dartois
• [asterisk-users] DISA + Voicemail + DTMF   Dave
• [asterisk-users] Detecting voicemail from CO on FXO port   Paul Davidson
• [asterisk-users] Detecting voicemail from CO on FXO   Paul Davidson
• [asterisk-users] SNOM 360   Steve Davies
• [asterisk-users] SNOM 360   Steve Davies
• [asterisk-users] Odd Busy tone on Aastra phones   Steve Davies
• [asterisk-users] Possible to To Have Different Outgoing VM Messages, but One Mailbox?   Robert DeVries
• [asterisk-users] US Toll-Free DID Providers with Caller ID NAME?   Robert DeVries
• [asterisk-users] US Toll-Free DID Providers with Caller ID NAME?   Robert DeVries
• [asterisk-users] US Toll-Free DID Providers with Caller ID NAME?   Robert DeVries
• [asterisk-users] Snom phones locking up   Peter J Dean
• [asterisk-users] Zap difficulties   Rusty Dekema
• [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch   Delca
• [asterisk-users] Asterisk, two eth and two providers   Delca
• [asterisk-users] Missing Agent Function   Delca
• [asterisk-users] Missing Agent Function   Delca
• [asterisk-users] Caller ID problem on TDM400 FXO   Greg Delgado
• [asterisk-users] DTMF-CallerID on POTS   Greg Delgado
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Greg Delgado
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   Greg Delgado
• [asterisk-users] How to set externip in sip.conf automatically?   Greg Delgado
• [asterisk-users] [Linksys 3102] Couple of issues   Vincent Delporte
• [asterisk-users] Working Sipura 3000 or Linksys 3102 configuration?   Vincent Delporte
• [asterisk-users] Re: Working Sipura 3000 or Linksys 3102 configuration?   Vincent Delporte
• [asterisk-users] Page()   Joe Dennick
• [asterisk-users] Server Hardware   Joe Dennick
• [asterisk-users] Realtime Extensions -- Comments?   Joe Dennick
• [asterisk-users] Simple CDR parser to print to webpage   Joe Dennick
• [asterisk-users] Strange SIP response   Joe Dennick
• [asterisk-users] Missing Agent Function   Joe Dennick
• [asterisk-users] Missing Agent Function   Joe Dennick
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Mike Dent
• Autoreply: [asterisk-users] Snom MWI   Austin Denyer
• [asterisk-users] Abstraction for a newbie   Austin Denyer
• [asterisk-users] Asterisk with AVM B1 and HFC   Marco Dieckhoff
• [asterisk-users] Equivalent of channel switching?   Christopher Dobbs
• [asterisk-users] ISDN incoming call - inband info and announcements BEFORE ANSWER   Michal Dole ž el
• [asterisk-users] BRI<->PRI switching and synchronization (data/fax calls)   Michal Dole ž el
• [asterisk-users] Tri-Link Technologies?   Don
• Autoreply: [asterisk-users] Tri-Link Technologies?   Don
• Autoreply: [asterisk-users] Tri-Link Technologies?   Don
• [asterisk-users] Re: Ever donate Software to Digium? If you didyour afool.   Don
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Don
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Don
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Don
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Don
• [asterisk-users] Manager Interface API's   Don
• [asterisk-users] Manager Interface API's   Don
• [asterisk-users] Manager Interface API's   Don
• [asterisk-users] Manager Interface API's   Don
• [asterisk-users] Cepstral and Asterisk   Don
• [asterisk-users] Cepstral and Asterisk   Don
• [asterisk-users] Cepstral and Asterisk   Don
• [asterisk-users] PRI problems - no D channel   Don
• [asterisk-users] Cepstral and Asterisk again...   Don
• [asterisk-users] Resolved: PRI problems - no D channel   Don
• [asterisk-users] Static in Monitor recordings   Don
• [asterisk-users] Text to Speech   Don
• [asterisk-users] Text to Speech   Don
• [asterisk-users] Realtime Extensions -- Comments?   Don
• [asterisk-users] Config quesiton: all inbound on PRI   Dave Donovan
• [asterisk-users] rx_fax problem   Paradise Dove
• [asterisk-users] Asterisk 'Hosting'   Dualcall.com
• [asterisk-users] Idiot questions   Dualcall.com
• [asterisk-users] Singapore   Dualcall.com
• [asterisk-users] Quick One - PHP Script to restart Asterisk   Corporate IT Solutions - Michael Dunne
• [asterisk-users] Re: Sangoma A200D and DTMF Detection   Rana Dutt
• [asterisk-users] Trunk with multiple IPs?   Lists at EMS
• [asterisk-users] ESCAUX releases net.PBX Free Edition   VOIP ESCAUX
• [asterisk-users] IAX call drops, recent instability   Chris Earle
• [asterisk-users] Playback() does not work   Camilo Echeverry
• [Asterisk-Users] Duration for billing   Steve Edwards
• [asterisk-users] What to use beyond T1's?   Steve Edwards
• [asterisk-users] What to use beyond T1's?   Steve Edwards
• [asterisk-users] Meetme bug or feature?   Steve Edwards
• [asterisk-users] Non-zaptel hardware based timing sources   Steve Edwards
• [asterisk-users] SSH connection hangs on logout?   Steve Edwards
• [asterisk-users] Can I increase DTMF sensitivity?   Steve Edwards
• [asterisk-users] Analyze core file prodeced after safe_asterisk crashh   Steve Edwards
• [asterisk-users] Analyze core file prodeced after safe_asterisk crashh   Steve Edwards
• [asterisk-users] Static vs dynamic meetme rooms   Steve Edwards
• [asterisk-users] How to run a batch file on the asterisk CLI   Steve Edwards
• [asterisk-users] Asterisk Sending Data to a Web Page   Steve Edwards
• [asterisk-users] multiple offices / hard phones / service provider   Joseph Ellis
• [asterisk-users] MOH help needed with fresh install   Nick Ellson
• Solved: <sigh> Re: [asterisk-users] MOH help needed with fresh install   Nick Ellson
• [asterisk-users] placing a call with the Manager interface   Eric
• [asterisk-users] placing a call with the Manager interface - solved   Eric
• [asterisk-users] Dialling from extension to extension with Manager   Eric
• [asterisk-users] Force peer source ip   Robin Ericsson
• [asterisk-users] Tracing audio problems   Erik
• [asterisk-users] Asterisk IAXmodem HylaFax?   Damon Estep
• [asterisk-users] Polycom just disconnects   Damon Estep
• [asterisk-users] hint status not updating on inbound   Damon Estep
• [asterisk-users] CDR inaccuracies   Dumpolid Exeplish
• [asterisk-users] Toll-Free numbers   Dumpolid Exeplish
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] MWI from Asterisk to Meridian   C F
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] Detecting voicemail from CO on FXO port   C F
• [asterisk-users] Detecting voicemail from CO on FXO port andpassing to H.323 phone. Possible?   C F
• [asterisk-users] Polycom 501 : How to make it ring whenalreadyona call   C F
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] Detecting voicemail from CO on FXO   C F
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] Dialplan routing based on CallerID   C F
• [asterisk-users] Variables sip redirects and call forward   C F
• [asterisk-users] Variables sip redirects and call forward   C F
• [asterisk-users] Variables sip redirects and call forward   C F
• [asterisk-users] Hotels...   C F
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   C F
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   C F
• [asterisk-users] Hotels...   C F
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   C F
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   C F
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   C F
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   C F
• [asterisk-users] How many digits are collected   C F
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   C F
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   C F
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   C F
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   C F
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   C F
• [asterisk-users] MailboxExists not branching to n+101   C F
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   C F
• [asterisk-users] Fast busy signals... Satisfying my curiousity   C F
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   C F
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   C F
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   C F
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   C F
• [asterisk-users] SPA-942 TFTP Provisioning   C F
• [asterisk-users] Config quesiton: all inbound on PRI   C F
• [asterisk-users] Page Groups   C F
• [asterisk-users] SIP asterisk over Linksys VPN   C F
• [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)   C F
• [asterisk-users] Page()   C F
• [asterisk-users] Extension for Incoming Call through Zap Channel   C F
• [asterisk-users] Restricting Incoming SIP Calls Without "call-limit"   C F
• [asterisk-users] SIP asterisk over Linksys VPN   C F
• [asterisk-users] SIP asterisk over Linksys VPN   C F
• [asterisk-users] Page()   C F
• [asterisk-users] SIP asterisk over Linksys VPN   C F
• [asterisk-users] Set DID   C F
• [asterisk-users] Not Working   C F
• [asterisk-users] Set DID   C F
• [asterisk-users] TRANSFER   C F
• [asterisk-users] Set DID   C F
• [asterisk-users] PRI problems - no D channel   C F
• [asterisk-users] PRI problems - no D channel   C F
• [asterisk-users] PRI problems - no D channel   C F
• [asterisk-users] Equivalent of channel switching?   C F
• [asterisk-users] Ignoring PRI call?   C F
• [asterisk-users] Ignoring PRI call?   C F
• [asterisk-users] Sending signals to asterisk   C F
• [asterisk-users] Ignoring PRI call?   C F
• [asterisk-users] How to modify incoming DNIS?   C F
• [asterisk-users] Missing Extension   C F
• [asterisk-users] RE: [asterisk-dev] Phone status   C F
• [asterisk-users] Re: Intercom mode on Polycom and/or SPA9xx   C F
• [asterisk-users] Will Asterisk work with Exchange 2007 UM?   C F
• [asterisk-users] Nobody is responding. Why? (Implement music on transfer)   C F
• [asterisk-users] Can anyone recommend a large button sip phone for the elderly.   C F
• [asterisk-users] Re: [asterisk-biz] Asterisk Tools   C F
• [asterisk-users] Off Topic: Hardware Required   C F
• [asterisk-users] Re: How to configure NOKIA N70 with Asterisk?   FaberK
• [asterisk-users] cisco 2600   FaberK
• [asterisk-users] Asterisk with PABX   Fabio
• [asterisk-users] Ringing after answered on zaptel   Don Fanning
• [asterisk-users] Asterfax and Gentoo   Don Fanning
• [asterisk-users] Asterfax and Gentoo   Don Fanning
• [asterisk-users] Asterisk 'Hosting'   Don Fanning
• [asterisk-users] Manager Interface API's   Don Fanning
• [asterisk-users] Connecting an cellphone to asterisk   Don Fanning
• [asterisk-users] load average with MOH   Jon Farmer
• [asterisk-users] Check call duration on active call in CLI?   Jon Farmer
• [asterisk-users] Asterisk and Siemens Legacy PBX   Jon Farmer
• [asterisk-users] Asterisk and Siemens Legacy PBX   Jon Farmer
• [asterisk-users] Using a DB for Configurations   Barry Fawthrop
• [asterisk-users] Using a DB for Configurations   Barry Fawthrop
• [asterisk-users] SIP Connection Problems   Barry Fawthrop
• [asterisk-users] SIP Connection Problems   Barry Fawthrop
• [asterisk-users] Cisco 7960 password reset   Barry Fawthrop
• [asterisk-users] In CDR record not what I want   Matthias Fechner
• [asterisk-users] In CDR record not what I want   Matthias Fechner
• [asterisk-users] GXP-2000 auf Betafirmware updaten?   Matthias Fechner
• [asterisk-users] GXP-2000 update to betafirmware?   Matthias Fechner
• [asterisk-users] Fax with asterisk?