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?   Matthias Fechner
• [asterisk-users] Fax with asterisk?   Matthias Fechner
• [asterisk-users] How is GXP2000 with latest firmware   Matthias Fechner
• [asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer   Manrique Feoli
• [asterisk-users] Is the manager good for high traffic?? but only with one connection to it   Manrique Feoli
• [asterisk-users] Meetme chat room with many users, and only 4 can talk, is there a max amount of users?   Manrique Feoli
• [asterisk-users] Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users?   Manrique Feoli
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   Manrique Feoli
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   Manrique Feoli
• [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)   Manrique Feoli
• [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN) In a Hurry   Manrique Feoli
• [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)   Manrique Feoli
• [asterisk-users] Steve Totaro I am trying to reach you.   Ferguson, Michael
• [asterisk-users] Cisco 7960 password reset   Ferguson, Michael
• [asterisk-users] Cisco 7960 password reset   Ferguson, Michael
• [asterisk-users] Cisco 7960 password reset   Ferguson, Michael
• [asterisk-users] Cisco 7960 password reset   Ferguson, Michael
• [asterisk-users] Cisco 7960 password reset   Ferguson, Michael
• [asterisk-users] Registration Error   Ferguson, Michael
• [asterisk-users] Frustration cubed   Ferguson, Michael
• [asterisk-users] Registration Error   Ferguson, Michael
• [asterisk-users] Configure mailserver to deliver voicemail   Ferguson, Michael
• [asterisk-users] What are my logs telling me here?   Ferguson, Michael
• [asterisk-users] What are my logs telling me here?   Ferguson, Michael
• [asterisk-users] modprobe wctdm fails in /etc/rc.local on FC5   Robert La Ferla
• [asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5   Robert La Ferla
• [asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5   Robert La Ferla
• [asterisk-users] How many digits are collected   Bruce Ferrell
• [asterisk-users] Asterisk with PABX   Bruce Ferrell
• [asterisk-users] Snom 360 Function Keys   Alessandro De Filippo
• [asterisk-users] [ANN] - Coder Needed for Patch   Bart Fisher
• [asterisk-users] Run a script at certain CLI writes   Bart Fisher
• [asterisk-users] Run a script at certain CLI writes   Bart Fisher
• [asterisk-users] Hotels...   Bart Fisher
• [asterisk-users] Strange CLI Output   Bart Fisher
• [asterisk-users] e&m wink, TE110P, * answers too soon   Bart Fisher
• [asterisk-users] Clocking Multiple T1 Cards   Kevin P. Fleming
• [asterisk-users] create custom cdr's   Kevin P. Fleming
• [asterisk-users] Binary/unreadable configuration files?   Kevin P. Fleming
• [asterisk-users] Caller ID on Transfers   Kevin P. Fleming
• [asterisk-users] Recent additions to the Digium Asterisk development team   Kevin P. Fleming
• [asterisk-users] SLA.conf   Kevin P. Fleming
• [asterisk-users] Can the codec/format for name/greeting in voicemail be changed?   Kevin P. Fleming
• [asterisk-users] Asterisk Development and Release Cycle   Kevin P. Fleming
• [asterisk-users] Problem with distortion of initial voicemail prompt   Kevin P. Fleming
• [asterisk-users] HP ProLiant and Digium 24xxp   Kevin P. Fleming
• [asterisk-users] Is the manager good for high traffic?? but only with one connection to it   Matt Florell
• [asterisk-users] Polycom 1.6.7 firmware?   Matt Florell
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Matt Florell
• [asterisk-users] Manager Interface API's   Matt Florell
• [asterisk-users] What to use beyond T1's?   Matt Florell
• [asterisk-users] TAMPA BAY Asterisk Users Meeting Monday   Matt Florell
• [asterisk-users] Joining calls via manager.api or AGI   Matt Florell
• [asterisk-users] Re: Manager API: matching an Originate to the Newchannel event   Matt Florell
• [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)   El Flynn
• [asterisk-users] Nokia E60/61/70 and SIP   El Flynn
• [asterisk-users] Ateus Easy gate call progress   Jan Fousek
• [asterisk-users] asterisk dosenot compile   Jan Fousek
• [asterisk-users] Jitterbuffer on SIP   Jan Fousek
• [asterisk-users] Re: SIP Debug to file - Is it possible?   Jan Fousek
• [asterisk-users] Asterisk and spandsp   Jan Fousek
• [asterisk-users] Asterisk and spandsp   Jan Fousek
• [asterisk-users] ISDN incoming call - inband info and announcements BEFORE ANSWER   Matthew Fredrickson
• [asterisk-users] ANNOUNCE: libss7   Matthew Fredrickson
• [asterisk-users] Fwd: * and GTalk testing   David Freeman
• [asterisk-users] Fwd: * and GTalk testing   David Freeman
• [asterisk-users] Probelm with IAX peers   David Freeman
• Autoreply: [asterisk-users] Tri-Link Technologies?   David Freeman
• [asterisk-users] safe_asterisk to start latest version from SVN - trying asterisk with googletalk   David Freeman
• [asterisk-users] Problems with incoming authentication   David Freeman
• [asterisk-users] Problems with incoming authentication   David Freeman
• [asterisk-users] Asterisk 'Hosting'   David Freeman
• [asterisk-users] Asterisk 'Hosting'   David Freeman
• [asterisk-users] LOUD MP3 Hold Music   David Freeman
• [asterisk-users] Asterisk with PABX   David Freeman
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Jim Freeze
• [asterisk-users] Cisco Router QOS and IAX2   Dave Fullerton
• [asterisk-users] Sipura 3000 and Asterisk   Dave Fullerton
• [asterisk-users] Speex Problemz   Dave Fullerton
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Stephen G
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Stephen G
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Stephen G
• [asterisk-users] Strange SIP response   Diego Andres Asenjo G.
• [asterisk-users] does OOH323 channel support Early Media?   Rafael J. Risco G.V.
• [asterisk-users] Re: does OOH323 channel support Early Media?   Rafael J. Risco G.V.
• [asterisk-users] Re: Cisco 7970 MWI not working (Was: Problem with Cisco7970 SIP load / call transfer)   Michael J. Tubby G8TIC
• [asterisk-users] Re: Cisco 7970 MWI not working (Was: Problem withCisco7970 SIP load / call transfer)   Michael J. Tubby G8TIC
• [asterisk-users] g.711 Codec Question   Michael J. Tubby G8TIC
• [asterisk-users] Sending SIP 183 Session Progressing   Michael J. Tubby B.Sc (Hons) G8TIC
• [asterisk-users] VoiceMail being cutoff when leaving message   BILL GITONGA
• [asterisk-users] Restricting Incoming SIP Calls Without "call-limit"   Damien Gabrielson
• [asterisk-users] Restricting Incoming SIP Calls Without "call-limit"   Damien Gabrielson
• [asterisk-users] Sending Email From A Dial Plan   Damien Gabrielson
• [asterisk-users] Sending Email From A Dial Plan   Damien Gabrielson
• [asterisk-users] Per DID Codec Negotiation   Damien Gabrielson
• [asterisk-users] Config quesiton: all inbound on PRI   Ron Gage
• [asterisk-users] PRI problems - no D channel   Ron Gage
• [asterisk-users] PRI problems - no D channel   Ron Gage
• [asterisk-users] PRI problems - no D channel   Ron Gage
• [asterisk-users] PRI problems - no D channel   Ron Gage
• [asterisk-users] PRI problems - no D channel   Ron Gage
• [asterisk-users] Resolved: PRI problems - no D channel   Ron Gage
• [asterisk-users] Advice needed - asterisk & Mitel 200SX   Ron Gage
• [asterisk-users] SLA Doc   David Gagnon
• [asterisk-users] loading the prompt files in memory on Asteriskstartup   David Gagnon
• [asterisk-users] SLA Doc   David Gagnon
• [asterisk-users] recommended hardware specs   David Gagnon
• [asterisk-users] Metermaid - Parking Slot   David Gagnon
• [asterisk-users] Metermaid - Parking Slot   David Gagnon
• [asterisk-users] LOUD MP3 Hold Music   David Gagnon
• [asterisk-users] Polycom 501 vs 601 provisioning   David Gagnon
• [asterisk-users] Hint extension issue - bug?   David Gagnon
• [asterisk-users] Hint extension issue - bug?   David Gagnon
• [asterisk-users] Hint extension issue - bug?   David Gagnon
• [asterisk-users] Hint extension issue - bug?   David Gagnon
• [asterisk-users] Registering IP Phone To Asterisk   David Gagnon
• [asterisk-users] Call Parking Ring Back (Snoms)   David Gagnon
• [asterisk-users] Can I increase DTMF sensitivity?   David Gagnon
• [asterisk-users] IP interface "box" for Meridian type digital phone   David Gagnon
• [asterisk-users] Call Routing based on Caller-Id   Brandon Galbraith
• [asterisk-users] Asterisk 'Hosting'   Brandon Galbraith
• [asterisk-users] Asterisk 'Hosting'   Brandon Galbraith
• [asterisk-users] Asterisk 'Hosting'   Brandon Galbraith
• [asterisk-users] Re: SIP Debug to file - Is it possible?   Brandon Galbraith
• [asterisk-users] Calls over VPN   Brandon Galbraith
• [asterisk-users] Idiot questions   Brandon Galbraith
• [asterisk-users] Max number of SIP devices registered to an extension   Brandon Galbraith
• [asterisk-users] Max number of SIP devices registered to anextension   Brandon Galbraith
• [asterisk-users] Detect if cell phone or users   Brandon Galbraith
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   George Gardiner
• [asterisk-users] IAX and Accountcode   Douglas Garstang
• [asterisk-users] IAX and Accountcode   Douglas Garstang
• [asterisk-users] IAX and Accountcode   Douglas Garstang
• [asterisk-users] Dundi and Dial Arguments   Douglas Garstang
• [asterisk-users] Dundi and Dial Arguments   Douglas Garstang
• [asterisk-users] Limitations of IAX   Douglas Garstang
• [asterisk-users] Limitations of IAX   Douglas Garstang
• [asterisk-users] IAX Trunking   Douglas Garstang
• [asterisk-users] Limitations of IAX   Douglas Garstang
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] SIP_HEADER() read-only   Douglas Garstang
• [asterisk-users] IAX Trunking   Douglas Garstang
• [asterisk-users] IAX Variables   Douglas Garstang
• [asterisk-users] IAX Variables   Douglas Garstang
• [asterisk-users] SIP_HEADER() read-only   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] MOH Silence   Douglas Garstang
• [asterisk-users] SIP musicclass   Douglas Garstang
• [asterisk-users] Music On Hold Class Not Makin' Sense   Douglas Garstang
• [asterisk-users] Re: Polycom 1.6.7 firmware?   Douglas Garstang
• [asterisk-users] Polycom 1.6.7 firmware?   Douglas Garstang
• [asterisk-users] Agent Transfer Locking up Queue() Application   Douglas Garstang
• [asterisk-users] Digit timeout on Asterisk Assisted Transfers   Douglas Garstang
• [asterisk-users] AgentcallbackLogin()   Douglas Garstang
• [asterisk-users] Re: AgentcallbackLogin()   Douglas Garstang
• [asterisk-users] pyAst   Douglas Garstang
• [asterisk-users] Sending INVITE to an unavailable phone - Bug?   Douglas Garstang
• [asterisk-users] SIP Qualify   Douglas Garstang
• [asterisk-users] SIP Qualify   Douglas Garstang
• [asterisk-users] SIP Qualify   Douglas Garstang
• [asterisk-users] SIP Qualify   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [Asterisk-Users] OPENSER / SER and Asterisk   Douglas Garstang
• [Asterisk-Users] OPENSER / SER and Asterisk   Douglas Garstang
• [asterisk-users] Polycom upgrade issue   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Manager Interface API's   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Realtime include   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Realtime Extension Lookups   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Return data from Fast AGI   Douglas Garstang
• [asterisk-users] Return data from Fast AGI   Douglas Garstang
• [asterisk-users] Sending Email From A Dial Plan   Douglas Garstang
• [asterisk-users] Return data from Fast AGI   Douglas Garstang
• [asterisk-users] Return data from Fast AGI   Douglas Garstang
• [asterisk-users] Recent additions to the Digium Asteriskdevelopment team   Douglas Garstang
• [asterisk-users] Realtime Peers Disappearing   Douglas Garstang
• [asterisk-users] Apache for FastAGI   Douglas Garstang
• [asterisk-users] Realtime Extensions and 'include =>'   Douglas Garstang
• [asterisk-users] Apache for FastAGI   Douglas Garstang
• [asterisk-users] Asterisk Jobs Update   Douglas Garstang
• [asterisk-users] Apache for FastAGI   Douglas Garstang
• [asterisk-users] Re: Asterisk 'Hosting'   Douglas Garstang
• [asterisk-users] Announce caller-id   Douglas Garstang
• [asterisk-users] Realtime and labels   Douglas Garstang
• [asterisk-users] Realtime and hints   Douglas Garstang
• [asterisk-users] Apache for FastAGI   Douglas Garstang
• [asterisk-users] Re: SIP Debug to file - Is it possible?   Douglas Garstang
• [asterisk-users] Realtime Extensions -- Comments?   Douglas Garstang
• [asterisk-users] Apache for FastAGI   Douglas Garstang
• [asterisk-users] Apache for FastAGI   Douglas Garstang
• [asterisk-users] Polycom 501 vs 601 provisioning   Douglas Garstang
• [asterisk-users] Realtime Extensions -- Comments?   Douglas Garstang
• [asterisk-users] Realtime Extensions -- Comments?   Douglas Garstang
• [asterisk-users] Realtime Extensions -- Comments?   Douglas Garstang
• [asterisk-users] Realtime Extensions -- Comments?   Douglas Garstang
• [asterisk-users] Realtime Extensions -- Comments?   Douglas Garstang
• [asterisk-users] Realtime Extensions -- Comments?   Douglas Garstang
• [asterisk-users] Realtime and hints   Douglas Garstang
• [asterisk-users] Realtime and hints   Douglas Garstang
• [asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS   Douglas Garstang
• [asterisk-users] Realtime and hints   Douglas Garstang
• [asterisk-users] Realtime and hints   Douglas Garstang
• [asterisk-users] SER Dispatcher Load Balance How-To?   Douglas Garstang
• [asterisk-users] SER Dispatcher Load Balance How-To?   Douglas Garstang
• [asterisk-users] SER Dispatcher Load Balance How-To?   Douglas Garstang
• [asterisk-users] SER Dispatcher Load Balance How-To?   Douglas Garstang
• [asterisk-users] SER Dispatcher Load Balance How-To?   Douglas Garstang
• [asterisk-users] Hola, en que andas?   Benjamin Garzon
• [asterisk-users] PrivacyManager   Jeremy G. Gault
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] AGI doesn't execute PHP5 script   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] AGI doesn't execute PHP5 script   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] AGI doesn't execute PHP5 script   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] AGI doesn't execute PHP5 script [SOLVED]   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Re: Asterisk load testing   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Missing number 2 in "advanced options" of VM   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Missing number 2 in "advanced options" of VM   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] wip 300 opensource code - changes to support SIP MESSAGE   Jerry Geis
• [asterisk-users] sendtext() to another machine   Jerry Geis
• [asterisk-users] can i detect a voice with asterisk ?   Jerry Geis
• [asterisk-users] VOIP phone for Receptionist use   Bill Gibbs
• [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects   Bill Gibbs
• [asterisk-users] polycom headset question   Bill Gibbs
• [asterisk-users] Ever donate Software to Digium? If you didyoura fool.   Bill Gibbs
• [asterisk-users] Intel D945G chipset   Bill Gibbs
• [asterisk-users] Page Groups   Bill Gibbs
• [asterisk-users] PRI problems - no D channel   Bill Gibbs
• [asterisk-users] PRI problems - no D channel   Bill Gibbs
• [asterisk-users] Cisco PIX firewall and nat=yes   Bill Gibbs
• [asterisk-users] Cisco PIX firewall and nat=yes   Bill Gibbs
• [asterisk-users] DNS   Bill Gibbs
• [asterisk-users] Call parking with Polycom's - works but MOH stops in one scenario   Bill Gibbs
• [asterisk-users] RE: Call parking with Polycom's - works but MOH stops in one scenario   Bill Gibbs
• [asterisk-users] Asterisk Jobs Update   Matt Gibson
• [asterisk-users] Asterisk Jobs Update   Matt Gibson
• [asterisk-users] Text to Speech   Jean-Denis Girard
• [asterisk-users] Callback feature in voicemail broke?   Steve Gladden
• [asterisk-users] Callback in within voicemail broken   Steve Gladden
• [asterisk-users] Callback in within voicemail broken   Steve Gladden
• [asterisk-users] SEXY WOMAN wants to know about =>Callback in within voicemail broken   Steve Gladden
• [asterisk-users] ISDN BRI, and Trixbox   Nigel Godfrey
• [asterisk-users] BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN   Stefan Gofferje
• [asterisk-users] Active Directory Listing Feauture   Mohamed A. Gombolaty
• [asterisk-users] Make Asterisk server initiate a Call   Mohamed A. Gombolaty
• [asterisk-users] About Digium cards and HP DL servers   Angel Gomez
• [asterisk-users] Snom360 with 6.2.2 firmware   Giordano Grandis
• R: [asterisk-users] Text to Speech   Giordano Grandis
• R: [asterisk-users] Snom360 with 6.2.2 firmware   Giordano Grandis
• R: [asterisk-users] Snom360 with 6.2.2 firmware   Giordano Grandis
• [asterisk-users] g729 and trafic   Michael Graves
• [asterisk-users] g729 and trafic   Michael Graves
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Michael Graves
• [asterisk-users] Astribank   Ed Greenberg
• [asterisk-users] Macro inside macro   Attilla De Groot
• [asterisk-users] Macro inside macro   Attilla De Groot
• [asterisk-users] Macro inside macro   Attilla De Groot
• [asterisk-users] Macro inside macro   Attilla De Groot
• [asterisk-users] Macro inside macro   Attilla De Groot
• [asterisk-users] Macro inside macro   Attilla De Groot
• [asterisk-users] Macro inside macro   Attilla De Groot
• [asterisk-users] Asterisk@Home, call reporting and performance   Esteban Guana-Jarrin
• [asterisk-users] Asterisk load testing   Nitin Gupta
• [asterisk-users] loading the prompt files in memory on Asterisk startup   Nitin Gupta
• [asterisk-users] loading the prompt files in memory on Asteriskstartup   Nitin Gupta
• [asterisk-users] Prompts recording for Asterisk   Nitin Gupta
• [asterisk-users] Prompts recording for Asterisk   Nitin Gupta
• [asterisk-users] Off Topic: Hardware Required   Sahil Gupta
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Randall H.
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Paul Hales
• [asterisk-users] Quick One - PHP Script to restart Asterisk   Paul Hales
• [asterisk-users] Quick One - PHP Script to restart Asterisk   Paul Hales
• [asterisk-users] IpSwitchboard   Paul Hales
• [asterisk-users] Polycom 601 Issues   Paul Hales
• [asterisk-users] Polycom 601 Issues   Paul Hales
• [asterisk-users] GSM gateway and FXO ATA   Paul Hales
• [asterisk-users] Calls over VPN   Paul Hales
• [asterisk-users] Calls over VPN   Paul Hales
• [asterisk-users] hint status not updating on inbound   Paul Hales
• [asterisk-users] PorTech GSM gateway   Paul Hales
• [asterisk-users] polycom headset question   Evan P. Hall
• [asterisk-users] polycom headset question   Evan P. Hall
• [asterisk-users] Slow dialing from PBX via E1   Gavin Hamill
• Subject: [asterisk-users] Slow dialing from PBX via E1   Gavin Hamill
• [asterisk-users] rx_fax problem   Steve Hanselman
• [asterisk-users] Unable to match on CallerID in an include block   Steve Hanselman
• [asterisk-users] Unable to match on CallerID in an include block   Steve Hanselman
• [asterisk-users] Unable to match on CallerID in an include block   Steve Hanselman
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Dana Harding
• [asterisk-users] No retry after DNS failure   James Harper
• [asterisk-users] No retry after DNS failure   James Harper
• [asterisk-users] MixMonitor stops after attended call transfer   Haspers
• [asterisk-users] Nokia E60/61/70 and SIP   Haspers
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Haspers
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Haspers
• [asterisk-users] Linksys SPA-3102   Barry D. Hassler
• [asterisk-users] Linksys SPA-3102   Barry D. Hassler
• [asterisk-users] Conflict between S & L option in Dial?   Vlasis Hatzistavrou
• [asterisk-users] Asterisk 1.2.11 and ${SIPDOMAIN} variable   Gary Hawkins
• [asterisk-users] Host 207.174.202.2 failed to authenticate as teliax   Ryan Hayward
• [asterisk-users] MailboxExists not branching to n+101   Ryan Hayward
• [asterisk-users] How to connect Snom softphone from my home?   Guido Hecken
• [asterisk-users] AgentCallBackLogin+Queue   Guido Hecken
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Guido Hecken
• [asterisk-users] IP phone with 2 ethernet jacks   Guido Hecken
• [asterisk-users] Changes in handling anonymous calls entering ast erisk   Guido Hecken
• [asterisk-users] Changes in handling anonymous calls entering asterisk   Guido Hecken
• [asterisk-users] Snom Function keys   Guido Hecken
• [asterisk-users] Ring Groups   Chris Hembrow
• [asterisk-users] Ring Groups   Chris Hembrow
• [asterisk-users] getting SIP to listen on multiple ports   Gary G. Hendershot
• [asterisk-users] AgentCallBackLogin+Queue   Gleidson Antonio Henriques
• [asterisk-users] RemoveQueueMember isn't working.   Keith Herrington
• [asterisk-users] unsubscribe   Keith Herrington
• [asterisk-users] Queue bug: When 2 callers call in, only one is processed until the first is answered   Keith Herrington
• [asterisk-users] Cisco 7961 SIP & Presence / BLF   Scott Higginbotham
• [asterisk-users] Call file do 2 outbound call   Daniel Hikel
• [asterisk-users] Call file do 2 outbound call   Daniel Hikel
• [asterisk-users] Asterisk GUI tool needed   Rizwan Hisham
• [asterisk-users] TrixBox install   Rizwan Hisham
• [asterisk-users] High Availability with PRI failover   Jean-Michel Hiver
• [asterisk-users] Recent additions to the Digium Asterisk development team   Jean-Michel Hiver
• [asterisk-users] NuFone chan_h323   Jean-Michel Hiver
• [asterisk-users] Apache for FastAGI   Jean-Michel Hiver
• [asterisk-users] Re: what is the real use of AEL?   Jean-Michel Hiver
• [asterisk-users] Call file do 2 outbound call   Jean-Michel Hiver
• [asterisk-users] Apache for FastAGI   Jean-Michel Hiver
• [asterisk-users] H264   Jean-Michel Hiver
• [asterisk-users] Asterisk codec strangeness   Jean-Michel Hiver
• [asterisk-users] Connecting two asterisk servers   Jean-Michel Hiver
• [asterisk-users] Asterisk codec strangeness   Jean-Michel Hiver
• [asterisk-users] Asterisk codec strangeness   Jean-Michel Hiver
• [asterisk-users] Asterisk + DTMF + g729   Chris Hodgetts
• [asterisk-users] softhangup() problem   Shaun Hofer
• [asterisk-users] SIP trunks: order or type   Shaun Hofer
• [asterisk-users] SIP trunks: order or type   Shaun Hofer
• [asterisk-users] SIP trunks: order or type   Shaun Hofer
• [asterisk-users] SIP trunks: order or type   Shaun Hofer
• [asterisk-users] sip host and registering   Shaun Hofer
• [asterisk-users] column width in CLI   Shaun Hofer
• [asterisk-users] Run a script at certain CLI writes   Nick Hoffman
• [asterisk-users] Asterisk Performance without RTP?   Nick Hoffman
• [asterisk-users] polycom soundstation 501 crash   Jessee J Holmes
• [asterisk-users] polycom soundstation 501 crash   Jessee J Holmes
• [asterisk-users] Polycom 1.6.7 firmware?   Jessee J Holmes
• [asterisk-users] polycom headset question   Jessee J Holmes
• [asterisk-users] Polycom 1.6.7 firmware?   Jessee J Holmes
• [asterisk-users] Can budgetone 101 display name part of cid?   Jessee J Holmes
• [asterisk-users] Asterisk in Xen 3.0   Tomer Horn
• [asterisk-users] Cancelling outbound call: is Asterisk behaving correctly   Wolfgang Hottgenroth
• [asterisk-users] Can budgetone 101 display name part of cid?   Guus Houtzager
• [asterisk-users] Can budgetone 101 display name part of cid?   Guus Houtzager
• [asterisk-users] Can budgetone 101 display name part of cid?   Guus Houtzager
• [asterisk-users] Asterisk IAXmodem HylaFax?   Lee Howard
• [asterisk-users] Asterisk IAXmodem HylaFax?   Lee Howard
• [asterisk-users] Asterisk IAXmodem HylaFax?   Lee Howard
• [asterisk-users] IAXMODEM   Lee Howard
• [asterisk-users] Recent additions to the Digium Asteriskdevelopment team   Lee Howard
• [asterisk-users] Set DID?   Mark Hulber
• [asterisk-users] DTMF-CallerID on POTS   Mark Hulber
• [asterisk-users] Where can i get a telephone number of Brasilia or Rio de Janeiro in Brazil   Han van Hulst
• [asterisk-users] Can i use the FXO of a addpack in Asterisk   Han van Hulst
• [asterisk-users] Is there a Blue tooth wireless headset that willwork with asterisk?   Hunt, Bill
• [asterisk-users] Re: Zaptel install - Fedora Core 5   Bruno Wolff III
• [asterisk-users] SoftHangup with Polycom_acd_functions release of asterisk   Dean at INKnBITs
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Dean at INKnBITs
• [asterisk-users] G729, IAX, polycom - trying to using 2 codecs   Dean at INKnBITs
• [asterisk-users] Problems with Codecs in Asterisk   Dean at INKnBITs
• [asterisk-users] extensions.ael - calling an exten from a macro   Dean at INKnBITs
• [asterisk-users] Sip suppression   Dean at INKnBITs
• [asterisk-users] polycom_acd_functions branch and outboundproxy   Dean at INKnBITs
• [asterisk-users] IAXy behind NAT, unable to transfer?   Jeff Iddings
• [asterisk-users] IAXy behind NAT, unable to transfer?   Jeff Iddings
• [asterisk-users] IAXy behind NAT, unable to transfer?   Jeff Iddings
• [asterisk-users] res_sqlite problems   Michael Iedema
• [asterisk-users] AGI doesn't execute PHP5 script   Vahram Igityan
• [asterisk-users] wanpipe1:w1g1: Rx Error: 'Retry' exceeds maximum (64k): pci fatal error! (0x0007C03C)   Giorgio Incantalupo
• [asterisk-users] sangoma a102: "Rx Error: 'Retry' exceeds maximum (64k): pci fatal error"   Giorgio Incantalupo
• [asterisk-users] using asterisk + sangoma a102 to simulate telco PRI: is possible?   Giorgio Incantalupo
• [asterisk-users] using asterisk + sangoma a102 to simulate telco PRI: is possible?   Giorgio Incantalupo
• [asterisk-users] quadBRI beronet card: how to specify which ISDN channel to use to make calls   Giorgio Incantalupo
• [asterisk-users] misdn-init.conf card parameter for a monoBRI   Giorgio Incantalupo
• [asterisk-users] misdn-init.conf card parameter for a monoBRI   Giorgio Incantalupo
• [asterisk-users] misdn-init.conf card parameter for a monoBRI   Giorgio Incantalupo
• [asterisk-users] does misdn-mqueue work if compiled with gcc 4?   Giorgio Incantalupo
• [asterisk-users] does misdn-mqueue work if compiled with gcc 4?   Giorgio Incantalupo
• [asterisk-users] does misdn-mqueue work if compiled with gcc 4?   Giorgio Incantalupo
• [asterisk-users] does misdn-mqueue work if compiled with gcc 4?   Giorgio Incantalupo
• [asterisk-users] Asterisk 1.2.4 I hear other party's voice only when I speack need help   Giorgio Incantalupo
• [asterisk-users] "best" BRI card ?   Giorgio Incantalupo
• [asterisk-users] Extend analog phone via SIP (OT)   Ira
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Ira
• [asterisk-users] Variables sip redirects and call forward   Ira
• [asterisk-users] can i detect a voice with asterisk ?   Ira
• [asterisk-users] Auto retry on Busy   Ira
• [asterisk-users] CallerID is not displaying for my incoming calls   Ira
• [asterisk-users] Recent additions to the Digium Asteriskdevelopment team   Ira
• [asterisk-users] can not get ${LEN(VAR)} and greater than ">" to work for me   Ira
• [asterisk-users] Asterisk with PABX   Ira
• [asterisk-users] does anyone offer truly unlimited voip in the US   Ira
• [asterisk-users] Asterisk with PABX   Ira
• [asterisk-users] Asterisk and T1 Extensions.   Morten Isaksen
• [asterisk-users] voicemail as email and attachment   Benjamin Jacob
• [asterisk-users] editing configs thru web/ apps   Benjamin Jacob
• [asterisk-users] Help in dailplan in asterisk   Benjamin Jacob
• [asterisk-users] voicemail as email and attachment   Benjamin Jacob
• [asterisk-users] SpanDSP Error   Christian Jensen
• [asterisk-users] SpanDSP Error   Christian Jensen
• [asterisk-users] SpanDSP Error   Christian Jensen
• [asterisk-users] speech gaps with iax2   Pavel Jezek
• [asterisk-users] speech gaps with iax2   Pavel Jezek
• [asterisk-users] SIP/Qualify   Pavel Jezek
• [SPAM] Re: [asterisk-users] SIP/Qualify   Pavel Jezek
• [asterisk-users] speech gaps with iax2   Pavel Jezek
• [asterisk-users] set minimum iax jitterbuffer   Pavel Jezek
• [asterisk-users] set minimum iax jitterbuffer   Pavel Jezek
• [asterisk-users] jitterbuffer SIP-IAX possible?   Pavel Jezek
• [asterisk-users] jitterbuffer SIP-IAX possible?   Pavel Jezek
• [asterisk-users] SIP Qualify   Pavel Jezek
• [asterisk-users] SendDTMF - how to use   Pavel Jezek
• [asterisk-users] chan_skinny - in trunk r40360 - error "unsupported format '0'"   Pavel Jezek
• [asterisk-users] chan_skinny - in trunk r40360 - error "unsupported format '0'"   Pavel Jezek
• [asterisk-users] monmp3thread: Request to schedule in the past?!?!   Pavel Jezek
• [asterisk-users] Intertex IX68 GW2 AIR 802.11G ADSL2+ ?   Jan Johansson
• [asterisk-users] Video Conferencing over Asterisk   Olle E Johansson
• [SPAM] Re: [asterisk-users] SIP/Qualify   Olle E Johansson
• [asterisk-users] Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk   Olle E Johansson
• [asterisk-users] Video Conferencing over Asterisk   Olle E Johansson
• [asterisk-users] Music On Hold Class Not Makin' Sense   Olle E Johansson
• [asterisk-users] Asterisk Training - Boston, US and Malaga, Spain   Olle E Johansson
• [asterisk-users] Asterisk@von - Von Fall, Boston Sept 11-14   Olle E Johansson
• [asterisk-users] Sending SIP 183 Session Progressing   Olle E Johansson
• [asterisk-users] Force immediate re-registration on sip reload   Olle E Johansson
• [asterisk-users] Manager Interface API's   Olle E Johansson
• [asterisk-users] SIP-NAT failure on dynamic IP   Olle E Johansson
• [asterisk-users] Registration Error   Olle E Johansson
• [asterisk-users] Realtime include   Olle E Johansson
• [asterisk-users] Accessing SIP URI (not ${SIPURI})   Olle E Johansson
• [asterisk-users] SIP_HEADER function; what names are available?   Olle E Johansson
• [asterisk-users] Presence SUBSCRIBE/NOTIFY behaviour   Olle E Johansson
• [asterisk-users] Snakes On A Plane using Asterisk?   Hugh L. Johnson
• [asterisk-users] Polycom 501 : How to make it ring when alreadyona call   Jerry Jones
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Jerry Jones
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Jerry Jones
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Jerry Jones
• [asterisk-users] Asterisk and Siemens Legacy PBX   Jerry Jones
• [asterisk-users] Polycom upgrade issue   Jerry Jones
• [asterisk-users] What to use beyond T1's?   Jerry Jones
• [asterisk-users] Asterisk - SIP client latency   Jerry Jones
• [asterisk-users] Polycom 501 config questions   Jerry Jones
• [asterisk-users] Re: Operator Console(s)/Shared Call Appearances   Mr. Jones
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Mr. Jones
• [asterisk-users] SIP header challenge   Mr. Jones
• [asterisk-users] SIP header challenge   Mr. Jones
• [asterisk-users] SIP header challenge   Mr. Jones
• [asterisk-users] getting SIP to listen on multiple ports   Mr. Jones
• [asterisk-users] Re: getting SIP to listen on multiple ports   Mr. Jones
• [asterisk-users] RE VoipNow 1.2.0 Beta   Senad Jordanovic
• [asterisk-users] High Availability with PRI failover   Senad Jordanovic
• [asterisk-users] creidt card processing sripts for asterisk   Joseph
• [asterisk-users] creidt card processing sripts for asterisk   Joseph
• [asterisk-users] creidt card processing sripts for asterisk   Joseph
• [asterisk-users] Calls over VPN   Joseph
• [asterisk-users] Calls over VPN   Joseph
• [asterisk-users] Sipura 3000 and Asterisk   Joseph
• [asterisk-users] Clipped audio at beginning of SIP calls.   Martin Joseph
• [asterisk-users] Re: SIP Connection Problems   Martin Joseph
• [asterisk-users] Re: SoftPhones: Bad, or just bad QoS?   Martin Joseph
• [asterisk-users] Re: So you all think VoIP sypply is warm and fuzzy   Martin Joseph
• [asterisk-users] Re: Crazy stuff in latest CVS HEAD   Martin Joseph
• [asterisk-users] Re: Asterisk load testing   Martin Joseph
• [asterisk-users] Re: G729   Martin Joseph
• [asterisk-users] Sorry! My Bad!   Martin Joseph
• [asterisk-users] Re: New Device   Martin Joseph
• [asterisk-users] Re: Equivalent of channel switching?   Martin Joseph
• [asterisk-users] Re: Frustration cubed   Martin Joseph
• [asterisk-users] Basic Asterisk Setup   Martin Joseph
• [asterisk-users] Re: Idiot questions   Martin Joseph
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Martin Joseph
• [asterisk-users] Re: SV: E61   Martin Joseph
• [asterisk-users] Re: IP phone with 2 ethernet jacks   Martin Joseph
• [asterisk-users] Re: GSM gateway and FXO ATA   Martin Joseph
• [asterisk-users] Re: GSM gateway and FXO ATA   Martin Joseph
• [asterisk-users] Re: Wellgate 3804a   Martin Joseph
• [asterisk-users] Re: Wellgate 3804a   Martin Joseph
• [asterisk-users] Re: GSM gateway and FXO ATA   Martin Joseph
• [asterisk-users] Re: GSM gateway and FXO ATA   Martin Joseph
• [asterisk-users] Re: Wellgate 3804a   Martin Joseph
• [asterisk-users] Re: Adit 3104 randomly reboot   Martin Joseph
• [asterisk-users] Polycom just disconnects   Bartosz Jozwiak
• [asterisk-users] Polycom just disconnects   Bartosz Jozwiak
• [asterisk-users] ANNOUNCEMENT : Asterisk2Billing V1.2.3 (BrainCoral)   Areski K
• [asterisk-users] SuSE 10.1 zaptel init script   Manolis Kaliotis
• [asterisk-users] Dialing out using SIP terminal   Allan Kamau
• [asterisk-users] running agi application in the background   Allan Kamau
• [asterisk-users] How To NOT Generate A CDR For A Call?   Nate Kapi
• [asterisk-users] Ignoring PRI call?   Roy Sigurd Karlsbakk
• [asterisk-users] Ignoring PRI call?   Roy Sigurd Karlsbakk
• [asterisk-users] Connecting an cellphone to asterisk   Roy Sigurd Karlsbakk
• [asterisk-users] Ignoring PRI call?   Roy Sigurd Karlsbakk
• [asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found   Zen Kato
• [asterisk-users] Static in Monitor recordings   Adam Kavan
• [asterisk-users] Static in Monitor recordings   Adam Kavan
• [asterisk-users] New UK prompts   Steve Kennedy
• [asterisk-users] PROBLEM MUSIC ON HOLD   Steve Kennedy
• [asterisk-users] Connecting an cellphone to asterisk   Steve Kennedy
• [asterisk-users] GSM gateway and FXO ATA   Steve Kennedy
• [asterisk-users] Controllable hold music   Thomas Kenyon
• [asterisk-users] Codec selection / IAX tunnels   Thomas Kenyon
• [asterisk-users] IAX and Accountcode   Thomas Kenyon
• [asterisk-users] set minimum iax jitterbuffer   Thomas Kenyon
• [asterisk-users] set minimum iax jitterbuffer   Thomas Kenyon
• [asterisk-users] Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk   Thomas Kenyon
• [asterisk-users] t.38 bounty   Thomas Kenyon
• [asterisk-users] Adding/Removing Prefixes   Thomas Kenyon
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Thomas Kenyon
• [asterisk-users] IAX2 extn not registering on 4569   Thomas Kenyon
• [asterisk-users] SIP Error message   Thomas Kenyon
• [asterisk-users] Re: Asterisk Development and Release Cycle   Thomas Kenyon
• [asterisk-users] Asterisk => Master and Slave ?   Thomas Kenyon
• [asterisk-users] Per DID Codec Negotiation   Thomas Kenyon
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Mindaugas Kezys
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Mindaugas Kezys
• [asterisk-users] MOR and MCC - billing solutions for Asterisk released   Mindaugas Kezys
• [asterisk-users] Detecting sound before answer   Mindaugas Kezys
• [asterisk-users] any experiece with SPHINX   Abubakar A. Khaliq
• [asterisk-users] monitor a hangup in AGI application?   Abubakar A. Khaliq
• [asterisk-users] polycom soundstation 501 crash   Stas Khromoy
• [asterisk-users] polycom soundstation 501 crash   Stas Khromoy
• [asterisk-users] polycom soundstation 501 crash   Stas Khromoy
• [asterisk-users] Help with perl AGI script   Roy Kidder
• [asterisk-users] Help with perl AGI script   Roy Kidder
• [asterisk-users] AGI transfer question   Roy Kidder
• [asterisk-users] Announce caller-id   Roy Kidder
• [asterisk-users] Hotels...   Kristian Kielhofner
• [asterisk-users] Ringtone/gentone/busy and g729   Kristian Kielhofner
• [asterisk-users] Re: Ringtone/gentone/busy and g729   Kristian Kielhofner
• [asterisk-users] AstLinux 0.4.3 Released!   Kristian Kielhofner
• [asterisk-users] Will Asterisk work with Exchange 2007 UM?   Kristian Kielhofner
• [asterisk-users] SIP T1 timer and qualify=yes   Kristian Kielhofner
• [asterisk-users] SER Dispatcher Load Balance How-To?   Kristian Kielhofner
• [asterisk-users] 911 Testing   Kevin Kiely
• [asterisk-users] Handytone 286 T.38 SDP parameters   Jason Kim
• [asterisk-users] Connecting Asterisk to Avaya Definity over H.323   Matt King
• [asterisk-users] SJPhone and Asterisk over H323   Matt King
• [asterisk-users] RE: [asterisk-dev] Phone status   Andrew Kirch
• Autoreply: [asterisk-users] Snom MWI   Andrew D Kirch
• Autoreply: [asterisk-users] Snom MWI   Andrew D Kirch
• [asterisk-users] Cisco 7970 8.0.4 SIP firmware   Richard Klingler
• [asterisk-users] Auto retry on Busy   Philipp von Klitzing
• [asterisk-users] sms callback?   Rene Kluwen
• [asterisk-users] Re: sms callback?   Rene Kluwen
• [asterisk-users] volume adjustment?   Kohler, Jeffrey
• [asterisk-users] DTMF problems   Kohler, Jeffrey
• [asterisk-users] DTMF problems   Kohler, Jeffrey
• [asterisk-users] DTMF problems   Kohler, Jeffrey
• [asterisk-users] MWI from Asterisk to Meridian   Andrew Kohlsmith
• [asterisk-users] MWI from Asterisk to Meridian   Andrew Kohlsmith
• [asterisk-users] FXO module burn out !?   Andrew Kohlsmith
• [asterisk-users] Limitations of IAX   Andrew Kohlsmith
• [asterisk-users] Limitations of IAX   Andrew Kohlsmith
• [asterisk-users] Clocking Multiple T1 Cards   Andrew Kohlsmith
• [asterisk-users] How to check if channel varaible have been set/notempty?   Andrew Kohlsmith
• [asterisk-users] What to use beyond T1's?   Andrew Kohlsmith
• [asterisk-users] Asterisk 'Hosting'   Andrew Kohlsmith
• [asterisk-users] PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units   Andrew Kohlsmith
• [asterisk-users] Ignoring PRI call?   Andrew Kohlsmith
• [asterisk-users] Hint extension issue - bug?   Andrew Kohlsmith
• [asterisk-users] No retry after DNS failure   Andrew Kohlsmith
• [asterisk-users] Hint extension issue - bug?   Andrew Kohlsmith
• [asterisk-users] IP phone with 2 ethernet jacks   Andrew Kohlsmith
• [asterisk-users] [RESOLVED] One way audion on Sangoma   Andrew Kohlsmith
• [asterisk-users] Asterisk Performance without RTP?   Andrew Kohlsmith
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] BRIDGEPEER and DIALEDPEERNAME empty   Koopmann, Jan-Peter
• [asterisk-users] Echo cancell   Koopmann, Jan-Peter
• [asterisk-users] transfer call von D-channel   Koopmann, Jan-Peter
• [asterisk-users] Annoying Bristuff   Stelios Koroneos
• [asterisk-users] Annoying Bristuff   Stelios Koroneos
• [asterisk-users] AEL2 patch issues   Stephen Kratzer
• [asterisk-users] RTP Proxy   Ranjeet Kumar
• [asterisk-users] Re: need a pointer regarding scripting asterisk   Andy Kuo
• [asterisk-users] FAX questions   Andy Kuo
• [asterisk-users] Asterisk t38passthrough   Andy Kuo
• [asterisk-users] REQ: BATM gw-232 sip firmware   Mindaugas Kuprys
• [asterisk-users] spanDSP + rxfax   Mindaugas Kuprys
• [asterisk-users] IP phone with 2 ethernet jacks   Mindaugas Kuprys
• [asterisk-users] Sangoma A101 problem   Marcin Kwiatkowski
• [asterisk-users] (solved) Sangoma A101 problem   Marcin Kwiatkowski
• [asterisk-users] Asterisk and failover   Alex Lake
• [asterisk-users] Asterisk and failover   Alex Lake
• [asterisk-users] cannot received calls in pstn line   Lito Lampitoc
• [asterisk-users] asterisk and hipath 3750   Lito Lampitoc
• [asterisk-users] recommended hardware specs   Lito Lampitoc
• [asterisk-users] Page Groups   Steve Langstaff
• [asterisk-users] IP interface "box" for Meridian type digital phone   Steve Langstaff
• [asterisk-users] AMI initiate call probs   Niklas Larsson
• [asterisk-users] AMI initiate call probs   Niklas Larsson
• [asterisk-users] Hotels...   Andrew Latham
• [asterisk-users] Snom360 with 6.2.2 firmware   Andrew Latham
• [asterisk-users] Snom360 with 6.2.2 firmware   Andrew Latham
• [asterisk-users] Active Directory Listing Feauture   Andrew Latham
• [asterisk-users] Call Parking Ring Back (Snoms)   Andrew Latham
• [asterisk-users] Snom phones locking up   Andrew Latham
• [asterisk-users] Call Parking Ring Back (Snoms)   Andrew Latham
• [asterisk-users] Re: [asterisk-biz] Asterisk Tools   Andrew Latham
• [asterisk-users] problem with queues   Muhammad Zeeshan Latif
• [asterisk-users] FW: problem with queues   Muhammad Zeeshan Latif
• [asterisk-users] transfer call von D-channel   Christophorus Laube
• [asterisk-users] working chan_bluetooth enviroment   Matthias Laug
• [asterisk-users] Trunk with multiple IPs?   Benjamin Lawetz
• [asterisk-users] Trunk with multiple IPs?   Benjamin Lawetz
• [asterisk-users] Weird compile problem   Benjamin Lawetz
• [asterisk-users] DTMF between cisco and sipura going through asterisk   Benjamin Lawetz
• [asterisk-users] DTMF between cisco and sipura going throughasterisk   Benjamin Lawetz
• [asterisk-users] DTMF between cisco and sipura going throughasterisk   Benjamin Lawetz
• [asterisk-users] Unknown CLI output   Carlos Leal
• [asterisk-users] Unknown CLI output   Carlos Leal
• [asterisk-users] capi (divas4linux) bearer setting   Farkas Levente
• [asterisk-users] how to set 'transfercapability'   Farkas Levente
• [asterisk-users] Zap channel-modem/fax conn. problems   Tamas Levente
• [asterisk-users] Strange behavior with SIP registration/connectivity   Ronald Lewis
• [asterisk-users] Adit 3104 randomly reboot   Xue Liangliang
• [asterisk-users] Can not hear the telco System Announcement   Xue Liangliang
• [asterisk-users] DNS   Ola Lidholm
• [asterisk-users] Stopping Queue after nobody picked up the call . .   Ralph Liebessohn
• [asterisk-users] wildcard always busy   Ralph Liebessohn
• [asterisk-users] Circuit/channel Congestion   Ralph Liebessohn
• [asterisk-users] Circuit/channel Congestion   Ralph Liebessohn
• [asterisk-users] Asterisk 'Hosting'   Ralph Liebessohn
• [asterisk-users] ANI agi   Sharon Lim
• [asterisk-users] problem- 0:10 long message   Sharon Lim
• [asterisk-users] realtime+mysql   Sharon Lim
• [asterisk-users] Re: realtime+mysql   Sharon Lim
• [asterisk-users] Asterisk Billing   Sharon Lim
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Sharon Lim
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   Sharon Lim
• [asterisk-users] TrixBox install   Sharon Lim
• [asterisk-users] e&m wink, TE110P, * answers too soon   Steve Linabery
• [asterisk-users] SER Dispatcher Load Balance How-To?   Adam Linford
• [asterisk-users] Cisco Phone Configuration Tool cannot find Files   Nate List
• [asterisk-users] How to transfer call without getting dropped   Matt Arnilo S. Baluyos (Mailing Lists)
• [asterisk-users] Asterisk not parking calls - causes? how to fix?   Matt Arnilo S. Baluyos (Mailing Lists)
• [asterisk-users] SIP Termination Questions   Duracom Lists
• [asterisk-users] Recent additions to the Digium Asterisk development team   Rob Lith
• [asterisk-users] Re: SV: E61   Rob Lith
• [asterisk-users] Cisco 7960 password reset   Maxx Lobo
• [asterisk-users] Cisco 7960 password reset   Maxx Lobo
• [asterisk-users] Softphone for Windows Mobile 5?   Nilesh Londhe
• [asterisk-users] Connecting an cellphone to asterisk   Nilesh Londhe
• [asterisk-users] Idiot questions   Nilesh Londhe
• [asterisk-users] How to run a batch file on the asterisk CLI   Nilesh Londhe
• [asterisk-users] polycom soundstation 501 crash   Alexander Lopez
• [asterisk-users] how to check the status of a channel   Alexander Lopez
• [asterisk-users] By week extension dialing   Alexander Lopez
• [asterisk-users] By week extension dialing   Alexander Lopez
• [asterisk-users] agi script runs even if no answer   Alexander Lopez
• Define PMS was: [asterisk-users] Hotels...   Alexander Lopez
• [asterisk-users] Ever donate Software to Digium? If you did your afool.   Alexander Lopez
• [asterisk-users] Config quesiton: all inbound on PRI   Alexander Lopez
• [asterisk-users] SIP Qualify   Alexander Lopez
• [asterisk-users] 1 way audio. Dual NIC's.   Alexander Lopez
• [asterisk-users] Iaxy and SendDTMF??   Alexander Lopez
• [asterisk-users] detecting a users number using the dialplan orAGI   Alexander Lopez
• [asterisk-users] Possible to To Have Different Outgoing VM Messages, but One Mailbox?   Luki
• [asterisk-users] SPA3000 dialplan coding...   Luki
• [asterisk-users] app_rxfax and T.38   Luki
• [asterisk-users] Setting the contact header on outbound INVITE   Michael Lunsford
• OT: Re: [asterisk-users] Asterisk 'Hosting'   Richard Lyman
• [asterisk-users] Jabber questions   Julian Lyndon-Smith
• [asterisk-users] Problems with monitor / mixmonitor stopping if using Local channels   Julian Lyndon-Smith
• [asterisk-users] Problems with monitor / mixmonitor stopping if using Local channels   Julian Lyndon-Smith
• [asterisk-users] Help - call recording being cut short if transferred   Julian Lyndon-Smith
• [asterisk-users] Help - call recording being cut short if transferred   Julian Lyndon-Smith
• [asterisk-users] UK mobile "reject" codes   Julian Lyndon-Smith
• [asterisk-users] Jabber: Difference between client and component   Julian Lyndon-Smith
• [asterisk-users] OT: Changing Cisco tftp root directory   Julian Lyndon-Smith
• [asterisk-users] Can budgetone 101 display name part of cid?   Julian Lyndon-Smith
• [asterisk-users] Unable to match on CallerID in an include block   Julian Lyndon-Smith
• [asterisk-users] Unable to match on CallerID in an include block   Julian Lyndon-Smith
• [asterisk-users] dtmf during a call   Julian Lyndon-Smith
• [asterisk-users] app_txfax / app_rxfax   Julian Lyndon-Smith
• [asterisk-users] Which BRI Card ?   Julian Lyndon-Smith
• [asterisk-users] "best" BRI card ?   Julian Lyndon-Smith
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] for some of my users, VoiceMail is being cutoff when leaving message   Doug Lytle
• [asterisk-users] Conditional branching   Doug Lytle
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] Problems with Hangup   Doug Lytle
• [asterisk-users] Can budgetone 101 display name part of cid?   Doug Lytle
• [asterisk-users] Can budgetone 101 display name part of cid?   Doug Lytle
• [asterisk-users] Page()   Doug Lytle
• [asterisk-users] MaxRetries:1 - Problems Dialout Call files   Doug Lytle
• [asterisk-users] 1.2.10 and 1.2.9.1   Doug Lytle
• [asterisk-users] Direct to Voicemail   Doug Lytle
• [asterisk-users] VM - advanced options?   Doug Lytle
• [asterisk-users] Direct to Voicemail   Doug Lytle
• [Asterisk-Users] Help need to reset Adit 600 for Asterisk install   Doug Lytle
• [asterisk-users] NoCDR()   Doug Lytle
• [asterisk-users] Active Directory Listing Feauture   Doug Lytle
• [asterisk-users] voicemailmain   Doug Lytle
• [asterisk-users] voicemailmain   Doug Lytle
• [asterisk-users] How to set MWI   Doug Lytle
• [asterisk-users] How to set MWI   Doug Lytle
• [asterisk-users] Missing number 2 in "advanced options" of VM   Doug Lytle
• [asterisk-users] How to set MWI   Doug Lytle
• [asterisk-users] Missing number 2 in "advanced options" of VM   Doug Lytle
• [asterisk-users] Digium makes the list!   Doug Lytle
• [asterisk-users] New Parrot application, repeats what you say and more!   Doug Lytle
• [asterisk-users] Parked call, park-dail context   Doug Lytle
• [asterisk-users] Polycom 501 config questions   Doug Lytle
• [asterisk-users] Help - call recording being cut short if transferred   Julian J. M.
• [asterisk-users] Re: Abstraction for a newbie   M.Hockings
• [asterisk-users] I can´t set to work two tdm2400p and one TE205p on same machine, please help   MF
• [asterisk-users] PAP2T (or PAP2-NA or GrandStream 286) always busy on incoming calls with zaptel   Olivier MONNET
• [asterisk-users] Phone Newbie Questions   Colin MacMillan
• [asterisk-users] Abstraction for a newbie   Colin MacMillan
• [asterisk-users] Ringing after answered on zaptel   Brodie Macleod
• [asterisk-users] Snom360 with 6.2.2 firmware   Brodie Macleod
• R: [asterisk-users] Snom360 with 6.2.2 firmware   Brodie Macleod
• [asterisk-users] SendText Queue Notification   Brodie Macleod
• [asterisk-users] E61   Brodie Macleod
• [asterisk-users] DTMF codes in feature.conf not comming through   Henrik Ostergaard Madsen
• [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration   Angelito Manansala
• [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration   Angelito Manansala
• [asterisk-users] Festival through AGI can't handle strings longer than 15 chars   Mario
• [asterisk-users] Festival through AGI can't handle strings longer than 15 chars   Mario
• [asterisk-users] Festival through AGI can't handle strings longer than 15 chars   Mario
• [asterisk-users] Festival through AGI can't handle strings longer than 15 chars   Mario
• [asterisk-users] How to find which queue member answered a call?   Mario
• [asterisk-users] How to find which queue member answered a call?   Mario
• [asterisk-users] How to find which queue member answered a call?   Mario
• [asterisk-users] Connecting an cellphone to asterisk   Mario
• [asterisk-users] IP phone with 2 ethernet jacks   Mario
• [asterisk-users] No retry after DNS failure   John Marvin
• [asterisk-users] No retry after DNS failure   John Marvin
• [asterisk-users] No retry after DNS failure   John Marvin
• [asterisk-users] How to set externip in sip.conf automatically?   John Marvin
• [asterisk-users] IP phone with 2 ethernet jacks   John Marvin
• [asterisk-users] IP phone with 2 ethernet jacks   John Marvin
• [asterisk-users] * during voicemail greeting to access mailbox   Marty Mastera
• [asterisk-users] Remote CAPI - ISDN over TCP/IP   Daniel Matos
• [asterisk-users] Remote CAPI - ISDN over TCP/IP   Daniel Matos
• [asterisk-users] Remote CAPI - ISDN over TCP/IP   Daniel Matos
• [asterisk-users] Question on iax2 show netstats   Matt
• [asterisk-users] can i detect a voice with asterisk ?   Matt
• [asterisk-users] Max number of SIP devices registered to an extension   Matt
• [asterisk-users] Multiple Queue Problem   Matt
• [asterisk-users] FAX questions   Artifex Maximus
• [asterisk-users] compile problems with app_rxfax.c and asterisk 1.2.11   Artifex Maximus
• [asterisk-users] FAX questions   Artifex Maximus
• [asterisk-users] Agent solution w/o id/password   Artifex Maximus
• [asterisk-users] agent autologoff   Artifex Maximus
• [asterisk-users] Multiple site multi server setup   Ron McCarthy
• [asterisk-users] Idiot questions   Ron McCarthy
• [asterisk-users] Using Flite in a call file.   Joey McDonald
• [asterisk-users] Run a script at certain CLI writes   Joey McDonald
• [asterisk-users] Using Flite in a call file.   Joey McDonald
• [asterisk-users] Server Hardware   Raymond McKay
• [asterisk-users] SIP asterisk over Linksys VPN   Raymond McKay
• [asterisk-users] SIP asterisk over Linksys VPN   Raymond McKay
• [asterisk-users] Host 207.174.202.2 failed to authenticate as teliax   Jeremy McNamara
• [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)   Jeremy McNamara
• [asterisk-users] Asterisk 'Hosting'   Jeremy McNamara
• [asterisk-users] Asterisk 'Hosting'   Jeremy McNamara
• [asterisk-users] Manager Interface API's   Jeremy McNamara
• [asterisk-users] Asterisk 'Hosting'   Jeremy McNamara
• [asterisk-users] Asterisk 'Hosting'   Jeremy McNamara
• [asterisk-users] Security   Jeremy McNamara
• [asterisk-users] Asterisk 'Hosting'   Jeremy McNamara
• [asterisk-users] Asterisk 'Hosting'   Jeremy McNamara
• [asterisk-users] Asterisk 'Hosting'   Jeremy McNamara
• [asterisk-users] Asterisk 'Hosting'   Jeremy McNamara
• [asterisk-users] TE207P   Jeremy McNamara
• [asterisk-users] Realtime Extensions -- Comments?   Jeremy McNamara
• [asterisk-users] problem with asterisk billing time...   Jeremy McNamara
• [asterisk-users] SSH connection hangs on logout?   Jeremy McNamara
• [asterisk-users] SER Dispatcher Load Balance How-To?   Jeremy McNamara
• [asterisk-users] SER Dispatcher Load Balance How-To?   Jeremy McNamara
• [asterisk-users] US Toll-Free DID Providers with Caller ID NAME?   Jeremy McNamara
• [asterisk-users] 3COM NBX and Digium Cards...   Antoine Megalla
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Jorge Mendoza
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Jorge Mendoza
• [asterisk-users] PRI and Asterisk   Jorge Mendoza
• [asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS   Phil Menico
• [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS   Phil Menico
• [asterisk-users] [RESOLUTION] Polycom microbrowser issue Error HTTP 406 withIIS   Phil Menico
• [asterisk-users] RE: Ever donate Software to Digium? If you did your afool.   Yaakov Menken
• [asterisk-users] linuxdevices.com: >>Trolltech woos developers with "open" Linux phone<< Who'll be the first with * on a mobile?   Robert Michel
• [asterisk-users] Problems with Codecs in Asterisk   Chan Kwang Mien
• [asterisk-users] Problems with Codecs in Asterisk   Chan Kwang Mien
• [asterisk-users] Problems with Hangup   Chan Kwang Mien
• [asterisk-users] Hangup Problem with PSTN and ISDN   Chan Kwang Mien
• [asterisk-users] Extension for Incoming Call through Zap Channel   Chan Kwang Mien
• [asterisk-users] "hint" for Hold   Chan Kwang Mien
• [asterisk-users] Polycom 501 : How to make it ring when already on a call   Mike
• [asterisk-users] Polycom 501 : How to make it ring when alreadyona call   Mike
• [asterisk-users] Polycom 501 : How to make it ring whenalreadyona call   Mike
• [asterisk-users] Meetme echo in recording   Mike
• [asterisk-users] Polycom 501 vs 601 provisioning   Mike
• [asterisk-users] Polycom 501 config questions   Mike
• [asterisk-users] Polycom 501 config questions   Mike
• [asterisk-users] Polycom 501 config questions   Mike
• [asterisk-users] Polycom 501 config questions   Mike
• [asterisk-users] polycom soundstation 501 crash   Sebastian Milioto
• [asterisk-users] astbill white screen!!   Sebastian Milioto
• [asterisk-users] astbill white screen!!   Sebastian Milioto
• [asterisk-users] USB GSM gateway for Asterisk?   Jay Milk
• [asterisk-users] Detect if cell phone or users   Jay Milk
• [asterisk-users] SPA-942 TFTP Provisioning   Jeremiah Millay
• [asterisk-users] SPA-942 Sound Quality   Jeremiah Millay
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Avi Miller
• [asterisk-users] Setting CALLERID on a residential telco line   Avi Miller
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Avi Miller
• [asterisk-users] Polycom 1.6.7 firmware?   Avi Miller
• [asterisk-users] Polycom 1.6.7 firmware?   Avi Miller
• [asterisk-users] Polycom 1.6.7 firmware?   Avi Miller
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Avi Miller
• [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)   Avi Miller
• [asterisk-users] Call to a queue killing Asterisk?   Avi Miller
• [asterisk-users] Polycom IP430 won't finish boot   Avi Miller
• [asterisk-users] Tracing audio problems   Avi Miller
• [asterisk-users] Tracing audio problems   Avi Miller
• [asterisk-users] Tracing audio problems   Avi Miller
• [asterisk-users] Tracing audio problems   Avi Miller
• [asterisk-users] Tracing audio problems   Avi Miller
• [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes   Avi Miller
• [asterisk-users] Polycom 601 Issues   Noah Miller
• [asterisk-users] Polycom IP430 won't finish boot   Noah Miller
• [asterisk-users] Cepstral and Asterisk   John Millican
• [asterisk-users] Sipura 3000 dialplan strings.   John Millican
• [asterisk-users] can not get ${LEN(VAR)} and greater than ">" to work for me   John Millican
• [asterisk-users] can not get ${LEN(VAR)} and greater than ">" to work for me   John Millican
• [asterisk-users] Connecting an cellphone to asterisk   Danko Miocevic
• [asterisk-users] Sending signals to asterisk   Danko Miocevic
• [asterisk-users] Phone status   Mir
• [asterisk-users] RE: [asterisk-dev] Phone status   Mir
• [asterisk-users] Phone status   Mir
• [asterisk-users] RE: [asterisk-dev] Phone status   Mir
• [asterisk-users] Polycom 1.6.7 firmware?   Louis-David Mitterrand
• [asterisk-users] Re: Polycom 1.6.7 firmware?   Louis-David Mitterrand
• [asterisk-users] Help debugging strange asterisk behaviour   Mojo with Horan & Company, LLC
• [asterisk-users] sip phone networking question [possibly OT]   Mojo with Horan & Company, LLC
• [asterisk-users] creidt card processing sripts for asterisk   Mojo with Horan & Company, LLC
• [asterisk-users] Quick One - PHP Script to restart Asterisk   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch   Mojo with Horan & Company, LLC
• [asterisk-users] Abstraction for a newbie   Mojo with Horan & Company, LLC
• [asterisk-users] Idiot questions   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk codec strangeness   Mojo with Horan & Company, LLC
• [asterisk-users] Copying a recording to a voice mail box   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk codec strangeness   Mojo with Horan & Company, LLC
• [asterisk-users] Can anyone recommend a large button sip phone for the elderly.   Mojo with Horan & Company, LLC
• [asterisk-users] Polycom 501 config questions   Mojo with Horan & Company, LLC
• [asterisk-users] Sipura 3000 and Asterisk   Ariel Monaco
• [asterisk-users] Sipura 3000 and Asterisk   Ariel Monaco
• [asterisk-users] File structure question   Jay Moore
• [asterisk-users] Dialplan "or" matching   William Moore
• [asterisk-users] No CLID from PSTN using X100P FXO Card   William Moore
• [asterisk-users] quadBRI beronet card: how to specify which ISDN channel to use to make calls   William Moore
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Mac Address Authentication Methods   Daniel Botelho P. Moraes
• [asterisk-users] A2Billing - destination   Luciano Moreira
• [asterisk-users] Comfort noise support incomplete in Asterisk (RFC 3389).   Luciano Moreira
• [asterisk-users] No outbound with A2Billing   Luciano Moreira
• [asterisk-users] Missing Fast AGI calling 'h' exten without hanging up   Tony Mountifield
• [asterisk-users] Re: Missing Fast AGI calling 'h' exten without hanging up   Tony Mountifield
• [asterisk-users] Re: DUNDi with SIP   Tony Mountifield
• [asterisk-users] Re: DUNDi with SIP   Tony Mountifield
• [asterisk-users] Re: Configuring meetme recording quality (8kHz to 32kHz or higher)   Tony Mountifield
• [asterisk-users] Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users?   Tony Mountifield
• [asterisk-users] Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users?   Tony Mountifield
• [asterisk-users] Re: agi script runs even if no answer   Tony Mountifield
• [asterisk-users] Re: agi script runs even if no answer   Tony Mountifield
• [asterisk-users] Re: A question about AGI and RECORD FILE   Tony Mountifield
• [asterisk-users] Re: A question about AGI and RECORD FILE   Tony Mountifield
• [asterisk-users] Re: AgentcallbackLogin()   Tony Mountifield
• [asterisk-users] Re: Associating an Originate Request to a Channel before the call is answered   Tony Mountifield
• [asterisk-users] Re: Associating an Originate Request to a Channel before the call is answered   Tony Mountifield
• [asterisk-users] Re: Return data from Fast AGI   Tony Mountifield
• [asterisk-users] Re: Extension presedence.   Tony Mountifield
• [asterisk-users] Re: Ringtone/gentone/busy and g729   Tony Mountifield
• [asterisk-users] Re: Manager API: matching an Originate to the Newchannel event   Tony Mountifield
• [asterisk-users] Re: Dialling from extension to extension with Manager   Tony Mountifield
• [asterisk-users] Re: channel variable   Tony Mountifield
• [asterisk-users] Re: channel variable   Tony Mountifield
• [asterisk-users] Re: Digium makes the list!   Tony Mountifield
• [asterisk-users] Re: IAX call drops, recent instability   Tony Mountifield
• [asterisk-users] Is there a smarter way to ban expensive calls in dial plan?   Marco Mouta
• [asterisk-users] nat and qualify questions   Marco Mouta
• [asterisk-users] Rookie question, trying to learn   Marco Mouta
• [asterisk-users] Simple incomming fax solution ...   Marco Mouta
• [asterisk-users] SIP trunks: order or type   Marco Mouta
• [asterisk-users] Set DID?   Marco Mouta
• [asterisk-users] safe_asterisk to start latest version from SVN - trying asterisk with googletalk   Marco Mouta
• [asterisk-users] FAX questions   Marco Mouta
• [asterisk-users] FAX questions   Marco Mouta
• [asterisk-users] Multiple registrations to the same asterisk server   Marco Mouta
• [asterisk-users] AsteriskSpeaksGoogleTalk - User is always disconnected - Problems   Marco Mouta
• [asterisk-users] Multiple registrations to the same asterisk server   Marco Mouta
• [asterisk-users] Problems with outgoing calls on a TE410P   Marco Mouta
• [asterisk-users] PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units   Marco Mouta
• [asterisk-users] MaxRetries:1 - Problems Dialout Call files   Marco Mouta
• [asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!   Marco Mouta
• [asterisk-users] help me!!Problem on incoming calls   Marco Mouta
• [asterisk-users] help me!!Problem on incoming calls   Marco Mouta
• [asterisk-users] Asterisk Sending Data to a Web Page   Marco Mouta
• [asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!   Marco Mouta
• [asterisk-users] Multiple registrations to the same asterisk server   Juan Luis Moyano
• [asterisk-users] Multiple registrations to the same asterisk server   Juan Luis Moyano
• [asterisk-users] Multiple lines in body of UserEvent   Florian Muellner
• [asterisk-users] Problem with a TDM400P   Mark Muffett
• [asterisk-users] Problem with a TDM400P   Mark Muffett
• [asterisk-users] Problem with a TDM400P   Mark Muffett
• [asterisk-users] SIP ATA Channels for outbound calls - How to select in dialplan   Devraj Mukherjee
• [asterisk-users] Selecting outbound trunk   Devraj Mukherjee
• [asterisk-users] Selecting outbound trunk   Devraj Mukherjee
• [asterisk-users] Asterisk with BT's broadband voice service.   Ronan Mullally
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Stephen Murphy
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Stephen Murphy
• [asterisk-users] Polycom 1.6.7 firmware?   Stephen Murphy
• [asterisk-users] Polycom 1.6.7 firmware?   Stephen Murphy
• [asterisk-users] Re: AEL2 Looping   Steve Murphy
• [asterisk-users] Re: what is the real use of AEL?   Steve Murphy
• [asterisk-users] Re: asterisk-users Digest, Vol 25, Issue 139   Steve Murphy
• [asterisk-users] creidt card processing sripts for asterisk   Dennis Nacino
• [asterisk-users] Sending SIP 183 Session Progressing   Dinesh Nair
• [asterisk-users] Sending SIP 183 Session Progressing   Dinesh Nair
• [asterisk-users] Sending SIP 183 Session Progressing   Dinesh Nair
• [asterisk-users] linuxdevices.com: >>Trolltech woos developers with "open" Linux phone<< Who'll be the first with * on a mobile?   Dinesh Nair
• [asterisk-users] Nokia E60/61/70 and SIP   Dinesh Nair
• [asterisk-users] Nobody is responding. Why? (Implement music on transfer)   Dinesh Nair
• [asterisk-users] Call Max Time   Dinesh Nair
• [asterisk-users] asterisk gui   Rajeev Natarajan
• [asterisk-users] asterisk gui   Rajeev Natarajan
• [asterisk-users] Softphone for Windows Mobile 5?   Rajeev Natarajan
• [asterisk-users] 911 Testing   Leif Neland
• [asterisk-users] Asterisk Configuration   Jordi Nelissen
• [asterisk-users] Re: ESCAUX net.PBX registration and boot sequence (was Re: [asterisk-biz]ESCAUX releases net.PBX Free Edition)   Jordi Nelissen
• [asterisk-users] ESCAUX net.PBX, new template with autoconfig of all major IP Phones   Jordi Nelissen
• [asterisk-users] ESCAUX releases net.PBX Free Edition   Jordi Nelissen_
• [asterisk-users] New Voicemail Client for Win32, Linux x86, Mac OS X released   Jordi Nelissen_
• [asterisk-users] New Voicemail Client for Win32, Linux x86, Mac OS X released   Jordi Nelissen_
• [asterisk-users] SIP/Qualify   Peder at NetworkOblivion
• [asterisk-users] ARA & Regseconds   Peder at NetworkOblivion
• [asterisk-users] Cisco Buddies   Peder at NetworkOblivion
• [asterisk-users] Cisco Buddies   Peder at NetworkOblivion
• [asterisk-users] Cisco Buddies   Peder at NetworkOblivion
• [asterisk-users] t.38 bounty   Peder at NetworkOblivion
• [asterisk-users] Cisco PIX firewall and nat=yes   Peder at NetworkOblivion
• [asterisk-users] Uptime Record?   Peder at NetworkOblivion
• [asterisk-users] Uptime Record?   Peder at NetworkOblivion
• [asterisk-users] Max number of SIP devices registered to anextension   Peder at NetworkOblivion
• [asterisk-users] RTP Proxy   Peder at NetworkOblivion
• [asterisk-users] Macro inside macro   Michael Neuhauser
• [asterisk-users] Polycom HD Voice   Eldon Neustaeter
• [asterisk-users] Japanese Sound Files   Nhadie
• [asterisk-users] USA Toll Free   Marnus van Niekerk
• [asterisk-users] Large Asterisk system   Marnus van Niekerk
• [asterisk-users] GSM gateway and FXO ATA   Marnus van Niekerk
• [asterisk-users] Re: GSM gateway and FXO ATA   Marnus van Niekerk
• [asterisk-users] Speex Problemz   Ninneman, Tj
• [asterisk-users] Voicemail Platform   John Novack
• [asterisk-users] Auto retry on Busy   John Novack
• [asterisk-users] Auto retry on Busy   John Novack
• [asterisk-users] 911 Testing   John Novack
• [asterisk-users] Zap difficulties   John Novack
• [asterisk-users] Manager Interface API's   John Novack
• [asterisk-users] New Device   John Novack
• [asterisk-users] Digium TDM400P Vs Sangoma A200   John Novack
• [asterisk-users] Digium TDM400P Vs Sangoma A200   John Novack
• [asterisk-users] Manager Interface API's (OSS doc discussion)   John Novack
• [asterisk-users] Connecting an cellphone to asterisk   John Novack
• [asterisk-users] Connecting an cellphone to asterisk   John Novack
• [asterisk-users] Hint extension issue - bug?   John Novack
• [asterisk-users] Advice needed - asterisk & Mitel 200SX   John Novack
• [asterisk-users] Sangoma Problems - A104d not detected   John Novack
• [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration   John Novack
• [asterisk-users] reloading agents and queues   Jordan Novak
• [asterisk-users] Annoying Bristuff   Andrew Nowrot
• [asterisk-users] Annoying Bristuff   Andrew Nowrot
• [asterisk-users] Annoying Bristuff   Andrew Nowrot
• [asterisk-users] Annoying Bristuff   Andrew Nowrot
• [asterisk-users] Annoying Bristuff   Andrew Nowrot
• [asterisk-users] Annoying Bristuff   Andrew Nowrot
• [asterisk-users] Annoying Bristuff   Andrew Nowrot
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Massimo Nuvoli
• [asterisk-users] MixMonitor and g729 licenses   Massimo Nuvoli
• [asterisk-users] Return data from Fast AGI   Anders Nygren
• [asterisk-users] Apache for FastAGI   Anders Nygren
• [asterisk-users] Apache for FastAGI   Anders Nygren
• [asterisk-users] Joining calls via manager.api or AGI   Obelix
• [asterisk-users] cmd DIAL - Who picked up the call?   Kai Ober
• [asterisk-users] cmd DIAL - Who picked up the call?   Kai Ober
• [asterisk-users] where/when to set__TRANSFER_CONTEXT ?   Kai Ober
• [asterisk-users] Has anybody a usefull example for the DIAL-option G(context|exten|prio)   Kai Ober
• [asterisk-users] Annoying Bristuff   Kai Ober
• [asterisk-users] misdn-init.conf card parameter for a monoBRI   Kai Ober
• [asterisk-users] does misdn-mqueue work if compiled with gcc 4?   Kai Ober
• [asterisk-users] does misdn-mqueue work if compiled with gcc 4?   Kai Ober
• [asterisk-users] transform bridged call into a conference   Kai Ober
• [asterisk-users] transform bridged call into a conference   Kai Ober
• [asterisk-users] does misdn-mqueue work if compiled with gcc 4?   Kai Ober
• [asterisk-users] SER Dispatcher Load Balance How-To?   Natambu Obleton
• [asterisk-users] A good price for FXS 48 ports?   Javier Matos Odut
• [asterisk-users] Handling inbound and outbound calls passed from a proxy   Fran Oliveira
• [asterisk-users] SIP trunks: order or type   Fran Oliveira
• [Asterisk-Users] ISDN Y cable   Olivier
• [asterisk-users] Line drops   J. Oquendo
• [asterisk-users] Ringing all extensions   J. Oquendo
• [asterisk-users] Snom MWI   J. Oquendo
• [asterisk-users] Fast busy signals... Satisfying my curiousity   J. Oquendo
• [asterisk-users] More SNOM, Message Indicator/Retrieval issues   J. Oquendo
• [asterisk-users] More SNOM, Message Indicator/Retrieval issues   J. Oquendo
• [asterisk-users] Re: Asterisk load testing   J. Oquendo
• [asterisk-users] Call Parking initiator cannot retrieve parked calls   J. Oquendo
• [asterisk-users] Call Parking Ring Back (Snoms)   J. Oquendo
• [Asterisk-Users] Extension Ring on Multiple Phones   J. Oquendo
• [asterisk-users] Snom Function keys   J. Oquendo
• [asterisk-users] Load balancing of IAX2   Florian Overkamp
• [asterisk-users] Re: Load balancing of IAX2   Florian Overkamp
• [asterisk-users] Samsung Prostar DCS   Florian Overkamp
• [asterisk-users] Asterisk in Xen 3.0   Florian Overkamp
• [asterisk-users] IAXMODEM   From PH
• [asterisk-users] 911 versus 9.911   George Pajari
• [asterisk-users] SIP Redirect   Benoit Panizzon
• [asterisk-users] Voicemail, how to localize date in email notifications?   Benoit Panizzon
• [asterisk-users] IAX over two T1 connections bad quality   Tim Panton
• [asterisk-users] IAX and Accountcode   Tim Panton
• [asterisk-users] IAX2 Trunking CPU usage   Tim Panton
• [asterisk-users] About IVR and Oracle   Tim Panton
• [asterisk-users] Standard for transfer via IAX provider?   Tim Panton
• [asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!   Tim Panton
• [asterisk-users] Rookie question, trying to learn   Randy Paries
• [asterisk-users] Rookie question, trying to learn   Randy Paries
• [asterisk-users] VoiceMail being cutoff when leaving message   Randy Paries
• [asterisk-users] Rookie question, trying to learn   Randy Paries
• [asterisk-users] for some of my users, VoiceMail is being cutoff when leaving message   Randy Paries
• [asterisk-users] for some of my users, VoiceMail is being cutoff when leaving message   Randy Paries
• [asterisk-users] looking to pay a consultant to help with my asterisk installation   Randy Paries
• [asterisk-users] A question about AGI and RECORD FILE   Randy Paries
• [asterisk-users] Re: A question about AGI and RECORD FILE   Randy Paries
• [asterisk-users] SIP Qualify   Jason Parker
• [asterisk-users] Manager Interface API's   Jason Parker
• [asterisk-users] chan_skinny - in trunk r40360 - error "unsupported format '0'"   Jason Parker
• [asterisk-users] Realtime Extensions -- Comments?   Jason Parker
• [asterisk-users] Realtime Extensions -- Comments?   Jason Parker
• [asterisk-users] Problem dialing out with a TDB400P   Dante Passalacqua
• [asterisk-users] Jitterbuffer on SIP   Patrick
• [asterisk-users] USA Toll Free   Patrick
• [asterisk-users] Server Hardware   Patrick
• [asterisk-users] What to use beyond T1's?   Patrick
• [asterisk-users] Server Hardware   Patrick
• [asterisk-users] SpanDSP Error   Patrick
• [asterisk-users] misdn-init.conf card parameter for a monoBRI   Patrick
• [asterisk-users] Will Asterisk work with Exchange 2007 UM?   Patrick
• [asterisk-users] misdn-init.conf card parameter for a monoBRI   Patrick
• [asterisk-users] ISDN BRI, and Trixbox   Patrick
• [asterisk-users] help me!!Problem on incoming calls   Patrick
• [asterisk-users] Deployment for less than 10 phones   Erick Perez
• [asterisk-users] Asterisk load testing   Erick Perez
• [asterisk-users] OT: Bandwidth calculations and PCI/PCIX/PCIE   Erick Perez
• [asterisk-users] CPU configuration for 250 calls SIP to SIP to IAX and fonebridge and two asterisk servers   Erick Perez
• [asterisk-users] About IVR and Oracle   Infobox Peru
• [asterisk-users] Compatibility INTEL E7520   Infobox Peru
• [asterisk-users] Hotels...   Storm D. J. Petersen
• [asterisk-users] Help please ==> Wrong password   Noc Phibee
• [asterisk-users] Asterisk => Master and Slave ?   Noc Phibee
• [asterisk-users] Problems compil 1.2.11   Noc Phibee
• [asterisk-users] Problems compil 1.2.11   Noc Phibee
• [asterisk-users] Abstraction for a newbie   Mark Phillips
• [asterisk-users] zap channel media volume   Wolfgang Pichler
• [asterisk-users] question of CLI   Tim St. Pierre
• [asterisk-users] * during voicemail greeting to access mailbox   Tim St. Pierre
• [asterisk-users] voicemail as email and attachment   Tim St. Pierre
• [asterisk-users] Problems with recording   Tim St. Pierre
• [asterisk-users] Sipura 3000 and Asterisk   Tim St. Pierre
• [asterisk-users] ticks in the pstn side audio   Rosario Pingaro
• [asterisk-users] Adding/Removing Prefixes   Scott Pinhorne
• [asterisk-users] Call Handoff   Scott Pinhorne
• [asterisk-users] Cisco PIX firewall and nat=yes   Scott Pinhorne
• [asterisk-users] Monitoring/Listening In   Scott Pinhorne
• [asterisk-users] Is there a smarter way to ban expensive calls in dial plan?   William Piper
• [asterisk-users] create custom cdr's   William Piper
• [asterisk-users] SIP/Qualify   William Piper
• [asterisk-users] 1 way audio. Dual NIC's.   William Piper
• [asterisk-users] 1 way audio. Dual NIC's.   William Piper
• [asterisk-users] 1 way audio. Dual NIC's.   William Piper
• [asterisk-users] Asterisk t38passthrough   William Piper
• [Asterisk-Users] Extension Ring on Multiple Phones   William Piper
• [Asterisk-Users] Extension Ring on Multiple Phones   William Piper
• [Asterisk-Users] Extension Ring on Multiple Phones   William Piper
• [asterisk-users] asterisk presence (from manager API)   William Piper
• [asterisk-users] asterisk presence (from manager API)   William Piper
• [asterisk-users] question of CLI   William Piper
• [asterisk-users] How to run a batch file on the asterisk CLI   William Piper
• [Asterisk-Users] Extension Ring on Multiple Phones   William Piper
• [Asterisk-Users] Extension Ring on Multiple Phones   William Piper
• [asterisk-users] Support a malformed SIP INVITE   Tom Playford
• [asterisk-users] Assigning specific RTP ports to SIP clients   Tom Playford
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Joe Pokupec
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Joe Pokupec
• [asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?   Martin Polainer
• [asterisk-users] No audio from IVR of Cisco Call Manager   Alessio Pompigna
• [asterisk-users] Voicemail/Email Integration   Porier, Jeremy M.
• [asterisk-users] REGISTER attempt   Juanjo Portela
• [asterisk-users] creidt card processing sripts for asterisk   Jon Pounder
• [asterisk-users] creidt card processing sripts for asterisk   Jon Pounder
• [asterisk-users] "best" BRI card ?   Jon Pounder
• [asterisk-users] Fax with asterisk?   Jon Pounder
• [asterisk-users] No CLID from PSTN using X100P FXO Card   Nathan E. Pralle
• [asterisk-users] CentOS4.3 or Debian 3.1r2?   Russ Price
• [asterisk-users] PRI Connection in Lima, Peru   Carlos Prieto
• [asterisk-users] PRI Connection in Lima, Peru   Carlos Prieto
• [asterisk-users] IP JitterBuffer for 1.2.5   Thierry Querette
• [asterisk-users] Jitterbuffer on SIP   Thierry Querette
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Thierry Querette
• [asterisk-users] Jitterbuffer on SIP   Thierry Querette
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Thierry Querette
• [asterisk-users] A good price for FXS 48 ports?   Thierry Querette
• [asterisk-users] Dapper Drake, Asterisk, and Faxing   David R.
• [asterisk-users] Trixbox Fax to PDF   David R.
• [asterisk-users] Asterisk Sending Data to a Web Page   David R.
• [asterisk-users] app_conference   RR
• [asterisk-users] app_conference   RR
• [asterisk-users] 1.2.10 and 1.2.9.1   RR
• [asterisk-users] How to set externip in sip.conf automatically?   RR
• [asterisk-users] Prompts recording for Asterisk   RR
• [asterisk-users] How to set externip in sip.conf automatically?   RR
• [asterisk-users] Cisco Router QOS and IAX2   RR
• [asterisk-users] Can the codec/format for name/greeting in voicemail be changed?   RR
• [asterisk-users] Can the codec/format for name/greeting in voicemail be changed?   RR
• [asterisk-users] REGISTER attempt   RR
• [asterisk-users] Prompts playback changing tempo in SMP kernel   RR
• [asterisk-users] OT: Call For Papers -- 2007 Southern California Linux Expo   Ilan Rabinovitch
• [asterisk-users] Will Asterisk work with Exchange 2007 UM?   Jon Radon
• [asterisk-users] PROBLEM MUSIC ON HOLD   Elpidio Ramos
• [asterisk-users] SIP ActiveX?   Elpidio Ramos
• [asterisk-users] SER local as an Asterisk Trunk   Nhadie Ramos
• [asterisk-users] Asterisk with VoIP phone   J Rangi
• [asterisk-users] RE: Asterisk with VoIP phone (shadowym)   J Rangi
• [asterisk-users] Problems with outgoing calls on a TE410P   Nils Rasmussen
• [asterisk-users] "Off-circuits are busy now. Please try your call again later"   Wolfgang Paul Rauchholz
• [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)   Wolfgang Paul Rauchholz
• [asterisk-users] Asterisk Configuration   R.Linga Reddy
• [asterisk-users] Asterisk Configuration   R.Linga Reddy
• [asterisk-users] IAX2 Bandwidth setting   R.Linga Reddy
• [asterisk-users] codec_g729a.so coredump in SVN trunk   Walt Reed
• [asterisk-users] Asterisk with VoIP phone   Bruce Reeves
• [asterisk-users] Run a script at certain CLI writes   Bruce Reeves
• [asterisk-users] polycom headset question   Bruce Reeves
• [asterisk-users] Asterisk Configuration   Bruce Reeves
• [asterisk-users] 911 Testing   Bruce Reeves
• [asterisk-users] Server Hardware   Bruce Reeves
• [asterisk-users] Cisco Router QOS and IAX2   Bruce Reeves
• [asterisk-users] Cisco Router QOS and IAX2   Bruce Reeves
• [asterisk-users] Voicemail/Email Integration   Bruce Reeves
• [asterisk-users] Polycom 501 config questions   Bruce Reeves
• [asterisk-users] SIP_HEADER() read-only   Vincent Regnard
• [asterisk-users] SIP_HEADER() read-only   Vincent Regnard
• [asterisk-users] SIP_HEADER() read-only   Vincent Regnard
• [asterisk-users] Associating an Originate Request to a Channel before the call is answered   Stefan Reuter
• [asterisk-users] Manager Interface API's   Stefan Reuter
• [asterisk-users] Manager Interface API's   Stefan Reuter
• [asterisk-users] Re: Manager API: matching an Originate to the Newchannel event   Stefan Reuter
• [asterisk-users] ANNOUNCEMENT: Asterisk-Java 0.3-m1 released   Stefan Reuter
• [asterisk-users] polycom soundstation 501 crash   Jim Rice
• [asterisk-users] polycom config error 0x4020: possibly related to RE:Polycom upgrade issue?   Jim Rice
• [asterisk-users] Ring Groups   Hadley Rich
• [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)   Hadley Rich
• [asterisk-users] safe_asterisk to start latest version from SVN - trying asterisk with googletalk   Hadley Rich
• [asterisk-users] STRFTIME dialplan function not picking up system timezone   Hadley Rich
• [asterisk-users] STRFTIME dialplan function not picking up system timezone   Hadley Rich
• [asterisk-users] voicemailmain   Hadley Rich
• [asterisk-users] Asterisk server crashes after two years   Hadley Rich
• [asterisk-users] Status of Monitor   Richard
• [asterisk-users] canreinvite=yes and RTP dropping in and out   Gary Richardson
• [asterisk-users] canreinvite=yes and RTP dropping in and out   Gary Richardson
• [asterisk-users] Bind Ounbound SIP Trunk to second virtual IP on server   JR Richardson
• [asterisk-users] Setting CALLERID on a residential telco line   Steven Ringwald
• [asterisk-users] Linksys and Call Park   Steven Ringwald
• [asterisk-users] More SNOM, Message Indicator/Retrieval issues   Steven Ringwald
• [asterisk-users] SPA-942 TFTP Provisioning   Steven Ringwald
• [asterisk-users] TE207P   Steven Ringwald
• [asterisk-users] Realtime Peers Disappearing   Steven Ringwald
• [asterisk-users] TE207P   Steven Ringwald
• [asterisk-users] HP ProLiant and Digium 24xxp   Robert Roach
• [asterisk-users] asterisk gui   Alex Robar
• [asterisk-users] asterisk gui   Alex Robar
• [asterisk-users] FXO module burn out !?   Alex Robar
• [asterisk-users] DUNDi with SIP   Alex Robar
• [asterisk-users] DUNDi with SIP   Alex Robar
• [asterisk-users] Sangoma A200 and Disconnected Cables   Alex Robar
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Alex Robar
• [asterisk-users] Ignoring the # key on a call   Alex Robar
• [asterisk-users] Asterisk Voicemail Setup   Alex Robar
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Alex Robar
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Alex Robar
• [asterisk-users] USA Toll Free   Alex Robar
• [asterisk-users] Port Forwarding SIP rtp   Alex Robar
• [asterisk-users] Anyone know a DID provider in Panama (country code 507)?   Alex Robar
• [asterisk-users] Asterisk And Java?   Alex Robar
• [asterisk-users] Set DID   Alex Robar
• [asterisk-users] Set DID   Alex Robar
• [asterisk-users] Turn Off chan_sip Debug Messages   Alex Robar
• [asterisk-users] About Digium cards and HP DL servers   Anthony Rodgers
• [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects   Anthony Rodgers
• [asterisk-users] Run As User Asterisk   Anthony Rodgers
• [asterisk-users] Speed dials on Polycom IP601?   Anthony Rodgers
• [asterisk-users] Agent solution w/o id/password   Anthony Rodgers
• [asterisk-users] Asterisk speaks Russian!   Anthony Rodgers
• [asterisk-users] Park / ParkAndAnnounce   Guillermo Roditi
• [asterisk-users] SendText() & displaying text messages onaSIPhandset's screen   Guillermo Roditi
• [asterisk-users] Mailcall question   Gabriel Rosca
• [asterisk-users] High Availability   Eric Rousse
• [asterisk-users] Queue Management   Eric Rousse
• [asterisk-users] Problems with Authorization and INVITE   Ken Rozinsky
• [asterisk-users] Linksys SPA-3000 Administration Guide   Marcos Rubino
• [asterisk-users] MySQL 5.0+ and the MySQL addon - Can use stored procedures?   Rushowr
• [asterisk-users] How to check if channel varaible have been set/notempty?   Rushowr
• [asterisk-users] SIP/Qualify   Rushowr
• [asterisk-users] Realtime SIP Authentication   Rushowr
• [asterisk-users] In CDR record not what I want   Rushowr
• [asterisk-users] Auto retry on Busy   Rushowr
• [asterisk-users] Auto retry on Busy   Rushowr
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Rushowr
• [asterisk-users] Problems with Hangup   Rushowr
• [asterisk-users] Problems with Hangup   Rushowr
• [asterisk-users] Macro inside macro   Rushowr
• [asterisk-users] Macro inside macro   Rushowr
• [asterisk-users] Asterisk and PHP?   Rushowr
• [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time   Rushowr
• [asterisk-users] Recent additions to the DigiumAsterisk development team   Rushowr
• [asterisk-users] astbill white screen!!   Rushowr
• [asterisk-users] Asterisk 'Hosting'   Rushowr
• MySQL Addon and MySQL5 Stored Procs (WAS: RE: [asterisk-users] Asterisk 'Hosting')   Rushowr
• [asterisk-users] Sending Email From A Dial Plan   Rushowr
• [asterisk-users] Asterisk 'Hosting'   Rushowr
• [asterisk-users] Asterisk Real Time and sip.conf file used at   Rushowr
• [asterisk-users] CallerID is not displaying for my incoming calls   Rushowr
• [Asterisk-Users] OPENSER / SER and Asterisk   Rushowr
• [asterisk-users] CallerID is not displaying for my incoming calls   Rushowr
• [asterisk-users] Re: what is the real use of AEL?   Rushowr
• [asterisk-users] Dialplan "or" matching   Rushowr
• [asterisk-users] Asterisk 'Hosting'   Rushowr
• [asterisk-users] Variable to show caller id for a current call?   Rushowr
• [asterisk-users] Variable to show caller id for a current call?   Rushowr
• [asterisk-users] Variable to show caller id for a current call?   Rushowr
• [asterisk-users] Quick, hopefully easy, question   Rushowr
• [asterisk-users] re-writing the dial plan - some hints please   Rushowr
• [asterisk-users] if command for or missing callerid?   Rushowr
• [asterisk-users] Setting RPID privacy?   Rushowr
• [asterisk-users] Setting RPID privacy?   Rushowr
• [asterisk-users] Strange SIP response   Rushowr
• [asterisk-users] Strange SIP response   Rushowr
• [asterisk-users] Setting the contact header on outbound INVITE   Rushowr
• [asterisk-users] SSH connection hangs on logout?   Rushowr
• [asterisk-users] Quiet on the list today?   Rushowr
• [asterisk-users] SSH connection hangs on logout?   Rushowr
• [asterisk-users] Adding/Removing Prefixes   Rushowr
• [asterisk-users] Re: column width in CLI   Rushowr
• [asterisk-users] How to set externip in sip.conf automatically?   Rushowr
• [asterisk-users] Trunk with multiple IPs?   Rushowr
• [asterisk-users] MySQL CDR   Rushowr
• [asterisk-users] Help compiling asterisk-addons on Debian?   Rushowr
• [asterisk-users] Call Max Time   Rushowr
• [asterisk-users] Call Max Time   Rushowr
• [asterisk-users] Call Max Time   Rushowr
• [asterisk-users] Shared NFS or Shared MySQL for redundant secondaryserver?   Rushowr
• [asterisk-users] Max number of SIP devices registered to anextension   Rushowr
• [asterisk-users] Max number of SIP devices registered to anextension   Rushowr
• [asterisk-users] RE: [asterisk-dev] Phone status   Rushowr
• [asterisk-users] Max number of SIP devices registered toanextension   Rushowr
• [asterisk-users] Asterisk with PABX   Rushowr
• [asterisk-users] manual mods with GUI in place   Rushowr
• [asterisk-users] Asterisk with PABX   Rushowr
• [asterisk-users] Connecting two asterisk servers   Rushowr
• [Asterisk-Users] Extension Ring on Multiple Phones   Rushowr
• [Asterisk-Users] Extension Ring on Multiple Phones   Rushowr
• [asterisk-users] Linksys SPA-3000 Administration Guide   Alberto Sagredo
• [asterisk-users] VoipNow 1.2.0 Beta   Daniel Salama
• [asterisk-users] GXP-2000 Call Transfer Problem   Daniel Salama
• [asterisk-users] Linksys PAP2 Ring Settings   Daniel Salama
• [asterisk-users] Linksys PAP2 Ring Settings   Daniel Salama
• [asterisk-users] Voicemail's mail formate   Mohammad Salaque
• [asterisk-users] H323   Mohammad Salaque
• [asterisk-users] Server Hardware   David Sampson
• [asterisk-users] need help with error code   Michael Sampson
• [asterisk-users] How to set MWI   Michael Sampson
• [asterisk-users] How to set MWI   Michael Sampson
• [asterisk-users] How to set MWI   Michael Sampson
• [asterisk-users] About IVR and Oracle   Javier Lara Sanchez
• [asterisk-users] read more than 2 digits on festival   Javier Lara Sanchez
• [asterisk-users] Silent Calls (Ghost Calls) When Picking Up Queue Calls   Sascha
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   Kevin Savoy
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   Kevin Savoy
• [asterisk-users] Auto retry on Busy   Kevin Savoy
• [asterisk-users] Text to Speech   Kevin Savoy
• [asterisk-users] Text to Speech   Kevin Savoy
• [asterisk-users] PRI and Asterisk   Kevin Savoy
• [asterisk-users] Unable to start special tone   Kevin Savoy
• [asterisk-users] does anyone offer truly unlimited voip in the US   Steven M. Sawczyn
• [asterisk-users] reloading agents and queues   Dave Schardin
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Armin Schindler
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Armin Schindler
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Armin Schindler
• [asterisk-users] New people in this world and his problem with ISDN   Armin Schindler
• [asterisk-users] Calls over VPN   Armin Schindler
• [asterisk-users] Calls over VPN   Armin Schindler
• [asterisk-users] Remote CAPI - ISDN over TCP/IP   Armin Schindler
• [asterisk-users] Remote CAPI - ISDN over TCP/IP   Armin Schindler
• [asterisk-users] SIP T1 timer and qualify=yes   Christian Schlatter
• [asterisk-users] Asterisk with AVM B1 and HFC   Peer Oliver Schmidt
• [asterisk-users] Asterisk installations in Germany   Peer Oliver Schmidt
• [asterisk-users] Can anyone recommend a large button sip phone for the elderley.   Peer Oliver Schmidt
• [asterisk-users] Cisco 7960 password reset   David Schmitt
• [asterisk-users] Return data from Fast AGI   Roger Schreiter
• [asterisk-users] Asterisk t38passthrough   Roger Schreiter
• [asterisk-users] Fax with asterisk?   Roger Schreiter
• [asterisk-users] Fax with asterisk?   Roger Schreiter
• [asterisk-users] Festival Not Working   Jon Scottorn
• [asterisk-users] RE VoipNow 1.2.0 Beta   Miles Scruggs
• [asterisk-users] Aastra VLAN issues   Kris Seraphine
• [asterisk-users] Sipura 3000 and Asterisk   Francisco Seratti
• [asterisk-users] Sipura 3000 and Asterisk   Francisco Seratti
• [asterisk-users] Sipura 3000 and Asterisk   Francisco Seratti
• [asterisk-users] Sipura 3000 and Asterisk   Francisco Seratti
• [asterisk-users] Re: Ever donate Software to Digium? If you did your afool.   Gonzalo Servat
• [asterisk-users] Macro inside macro   Gonzalo Servat
• [asterisk-users] Asterisk - SIP client latency   Freddy Setiawan
• [asterisk-users] Asterisk - SIP client latency   Freddy Setiawan
• [asterisk-users] zap fxo to sip fxs intermitently not connecting to each other   David Sfiligoi
• [asterisk-users] Samsung Prostar DCS   Curt Shaffer
• [asterisk-users] Odd IAX stats   Curt Shaffer
• [asterisk-users] Zap difficulties   Curt Shaffer
• [asterisk-users] Zap difficulties   Curt Shaffer
• [asterisk-users] Page Groups   Curt Shaffer
• [asterisk-users] IAX unstable with large number of calls?   Curt Shaffer
• [asterisk-users] Polycom upgrade issue   Curt Shaffer
• [asterisk-users] polycom config error 0x4020: possibly related to RE:Polycom upgrade issue?   Curt Shaffer
• [asterisk-users] hotel teledex integration anyone?   Curt Shaffer
• [asterisk-users] manual mods with GUI in place   Curt Shaffer
• [asterisk-users] manual mods with GUI in place   Curt Shaffer
• [asterisk-users] w as pause dialing issue   Curt Shaffer
• [asterisk-users] (no subject)   Sony Veri Shandy
• [asterisk-users] Polycom IP600 HTTP Provisioning problem   VaibhaV Sharma
• [asterisk-users] Dundi and Dial Arguments   Mitch Sharp
• [asterisk-users] Anybody using Eicon SoftIP with Asterisk   Tim Sharp
• [asterisk-users] realtime+mysql   Shaun
• [asterisk-users] Re: realtime+mysql   Shaun
• [asterisk-users] Re: Re: realtime+mysql   Shaun
• [asterisk-users] Problem dialing out with a TDB400P   T. Shaw
• [asterisk-users] sip incoming stop working, what to look for in logs?   T. Shaw
• [asterisk-users] sip incoming stop working, what to look for in logs?   T. Shaw
• [asterisk-users] sip incoming stop working, what to look for in logs?   T. Shaw
• [asterisk-users] need dialout help in python script   Shidan
• [asterisk-users] need dialout help in python script   Shidan
• [asterisk-users] Apache for FastAGI   Shidan
• [asterisk-users] Apache for FastAGI   Shidan
• [asterisk-users] Video Conferencing over Asterisk   Siqhamo Sifo
• [asterisk-users] Iptables ,rtp   Siqhamo Sifo
• [asterisk-users] Port Forwarding SIP rtp   Siqhamo Sifo
• [asterisk-users] Re: no audio issue (asterisk@teladdict.com)   Siqhamo Sifo
• [asterisk-users] SER+iptables+Asterisk   Siqhamo Sifo
• [asterisk-users] E61   Andreas Sikkema
• [asterisk-users] E61   Andreas Sikkema
• [asterisk-users] Unicall stack, right versions?   Moises Silva
• [asterisk-users] Unicall stack, right versions?   Moises Silva
• [asterisk-users] IAXy behind NAT, unable to transfer?   Moises Silva
• [asterisk-users] Ragi without rails possible ?   Moises Silva
• [asterisk-users] DTMF problems   Moises Silva
• [asterisk-users] DTMF problems   Moises Silva
• [asterisk-users] E1 for Voice and Data with MFC/R2   Moises Silva
• [asterisk-users] looking to pay a consultant to help with my asterisk installation   Moises Silva
• [asterisk-users] E1 for Voice and Data with MFC/R2   Moises Silva
• [asterisk-users] How to link 2 existing calls   Moises Silva
• [asterisk-users] PRI Connection in Lima, Peru   Moises Silva
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Moises Silva
• [asterisk-users] Digit timeout on Asterisk Assisted Transfers   Moises Silva
• [asterisk-users] Asterisk And Java?   Moises Silva
• [asterisk-users] Asterisk and PHP?   Moises Silva
• [asterisk-users] Asterisk and PHP?   Moises Silva
• [asterisk-users] Manager Interface API's   Moises Silva
• [asterisk-users] Manager Interface API's   Moises Silva
• [asterisk-users] MFCR2 and Unicall PDF   Moises Silva
• [asterisk-users] Festival through AGI can't handle strings longer than 15 chars   Moises Silva
• [asterisk-users] Recent additions to the Digium Asterisk development team   Moises Silva
• [asterisk-users] About IVR and Oracle   Moises Silva
• [asterisk-users] how to enable REACHABLE/UNREACHABLE messages in logs   Moises Silva
• [asterisk-users] Dial C option   Moises Silva
• [asterisk-users] Dial C option   Moises Silva
• [asterisk-users] Dial C option   Moises Silva
• [asterisk-users] Asterisk Manager Interface Question   Moises Silva
• [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration   Moises Silva
• [asterisk-users] Auto retry on Busy   Noah Silverman
• [asterisk-users] Auto retry on Busy   Noah Silverman
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Nir Simionovich
• [asterisk-users] VoiceMail and Fax on same extension   Adi Simon
• [asterisk-users] GSM analogue router   Garth van Sittert
• [asterisk-users] Asterisk queues and dynamic members   Garth van Sittert
• [asterisk-users] Call transfer issues   Kevin Smith
• [asterisk-users] Auto retry on Busy   Kevin Smith
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Kevin Smith
• [asterisk-users] Auto retry on Busy   Kevin Smith
• [asterisk-users] Call transfer issues   Kevin Smith
• [asterisk-users] Polycom upgrade issue   Kevin Smith
• [asterisk-users] Dialplan "or" matching   Kevin Smith
• [asterisk-users] Dialplan "or" matching   Kevin Smith
• [asterisk-users] Help with newbie: D-link admin setup   Dominic Son
• [asterisk-users] Abstraction for a newbie   Dominic Son
• [asterisk-users] Abstraction for a newbie   Dominic Son
• [asterisk-users] Is there a smarter way to ban expensive calls in dial plan?   Andrea Spadaccini
• [asterisk-users] Setting CALLERID on a residential telco line   Andrea Spadaccini
• [asterisk-users] Size of realtime appdata field under MySQL   Peter Spikings
• [asterisk-users] Unicall stack, right versions?   Barzilai Spinak
• [asterisk-users] MWI from Asterisk to Meridian   Johann Steinwendtner
• [asterisk-users] MWI from Asterisk to Meridian   Johann Steinwendtner
• [asterisk-users] Re: Re: FYI - first release of alarm response code.   Steven
• [asterisk-users] Re: How to forward a call to an outside line   Steven
• [asterisk-users] Re: Re: How to forward a call to an outside line   Steven
• [asterisk-users] Re: Run a script at certain CLI writes   Steven
• [asterisk-users] Re: Using Flite in a call file.   Steven
• [asterisk-users] Re: By week extension dialing   Steven
• [asterisk-users] Re: can Digium FXS channels support been half mileto 1 mile length away from phone?   Steven
• [asterisk-users] Re: Page Groups   Steven
• [asterisk-users] Re: 1 way audio. Dual NIC's.   Steven
• [asterisk-users] Re: Softphone for Windows Mobile 5?   Steven
• [asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5   Steven
• [asterisk-users] Re: Re: modprobe wctdm fails in /etc/rc.local on FC5   Steven
• [asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5   Steven
• [asterisk-users] Re: Zapand SendDTMF??   Steven
• [asterisk-users] Re: New Voicemail Client for Win32, Linux x86, MacOS X released   Steven
• [asterisk-users] Re: Re: New Voicemail Client for Win32, Linux x86, MacOS X released   Steven
• [asterisk-users] Re: Zap and SendDTMF??   Steven
• [asterisk-users] Re: column width in CLI   Steven
• [asterisk-users] Re: problems with wevbmail   Steven
• [asterisk-users] Re: Simple CDR parser to print to webpage   Steven
• [asterisk-users] Re: Attempt to setup paging and intercom   Steven
• [asterisk-users] Re: Re: Attempt to setup paging and intercom   Steven
• [asterisk-users] Re: manual mods with GUI in place   Steven
• [asterisk-users] Re: manual mods with GUI in place   Steven
• [asterisk-users] Re: IP interface "box" for Meridian type digital phone   Steven
• [asterisk-users] Re: 911 versus 9.911   Steven
• [asterisk-users] Re: visual indication of temp. closed mode   Steven
• [asterisk-users] Re: voicemail as email and attachment   Steven
• [asterisk-users] Re: need a pointer regarding scripting asterisk   Benjamin Stocker
• [asterisk-users] queue in realtime   Benjamin Stocker
• [asterisk-users] dynamic queue problem   Roi Stork
• [asterisk-users] queuememberstatus overwhelms manager socket connection to asterisk   Roi Stork
• [asterisk-users] client socket to asterisk manager gets disconnected   Roi Stork
• [asterisk-users] Asterisk Manager Interface Question   Roi Stork
• [asterisk-users] IP phone with 2 ethernet jacks   Christian Stredicke
• [asterisk-users] Setting CALLERID on a residential telco line   VoIP Street
• [asterisk-users] Macro inside macro   Michael Strelnikov
• [asterisk-users] No retry after DNS failure   Michael Strelnikov
• [asterisk-users] Sipura SPA3000   Michael Strelnikov
• [asterisk-users] Asterisk speaks Russian!   Stuart
• [asterisk-users] Cron Job to Drop a Call File When the Hard Drive Gets over 50% Full   Massimiliano Stucchi
• [asterisk-users] Indonesian MFC/R2   Danang Suharno
• [asterisk-users] Indonesian MFC-R2   Danang Suharno
• [asterisk-users] Problem with Cisco7970 SIP load / call transfer   Juha Suhonen
• [asterisk-users] Problems with Codecs in Asterisk   Rosli Sukri
• [asterisk-users] Problems with Codecs in Asterisk   Rosli Sukri
• [asterisk-users] Problems with Codecs in Asterisk   Rosli Sukri
• [asterisk-users] DTMF problems   Rosli Sukri
• [asterisk-users] Problems with Codecs in Asterisk   Rosli Sukri
• [asterisk-users] Port Forwarding SIP rtp   Rosli Sukri
• [asterisk-users] H323   Rosli Sukri
• [asterisk-users] How to link 2 existing calls   Leon Sun
• [asterisk-users] SIP-NAT failure on dynamic IP   Technical Support
• [asterisk-users] Linksys PAP2 Ring Settings   Shanon Swafford
• [asterisk-users] Zaptel trunk failed to compile - Still but another error   Administrator TOOTAI
• [asterisk-users] Softphone for Windows Mobile 5?   Administrator TOOTAI
• [asterisk-users] Doubled digits on vm pasword   Administrator TOOTAI
• [asterisk-users] Record phone call using asterisk   Sam Tam
• [asterisk-users] call barge   Sam Tam
• [asterisk-users] GSM gateway and FXO ATA   Sam Tam
• [asterisk-users] Problem with distortion of initial voicemail prompt   Frank Tarczynski
• [asterisk-users] Garbled initial voicemail prompt   Frank Tarczynski
• [asterisk-users] Anyone know a DID provider in Panama (country code 507)?   Frank Tarczynski
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   Don Tasker
• [asterisk-users] A good price for FXS 48 ports?   Don Tasker
• [asterisk-users] Asterisk 1.2.11, Asterisk-Addons 1.2.4 and Zaptel 1.2.8 Released   Asterisk Development Team
• [asterisk-users] Ever donate Software to Digium? If you did youra fool.   MBIT Technologies
• [asterisk-users] PRI Dropouts (Solved)   MBIT Technologies
• [asterisk-users] Calls over VPN   Chris Teesdale
• [asterisk-users] G729 Softphone   Julio Tejera
• [asterisk-users] 1 way audio. Dual NIC's.   Earl Terwilliger
• [asterisk-users] Phone status   Earl Terwilliger
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   Shaw Terwilliger
• [asterisk-users] IP phone with 2 ethernet jacks   Joseph Tesmer
• [asterisk-users] Re: 911 versus 9.911   James Texter
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Thczv F. Thczv
• [asterisk-users] Is there a smarter way to ban expensive calls indial plan?   Martin Schrott - Thinking-Systems
• [asterisk-users] g.711 Codec Question   David Thomas
• [asterisk-users] Softphone for Windows Mobile 5?   David Thomas
• [asterisk-users] Softphone for Windows Mobile 5?   David Thomas
• [asterisk-users] Softphone for Windows Mobile 5?   David Thomas
• [asterisk-users] Question about 7940s and call forwarding   Joshua M Thompson
• [asterisk-users] IAXy can't connect to analog phone   Martti Tienhaara
• [asterisk-users] modprobe wctdm fails in /etc/rc.local on FC5   Martti Tienhaara
• [asterisk-users] modprobe wctdm fails in /etc/rc.local on FC5   Martti Tienhaara
• [asterisk-users] H323   Mark Tinka
• [asterisk-users] How to check if channel varaible have been set/not empty?   Jan du Toit
• [asterisk-users] Configuring meetme recording quality (8kHz to 32kHz or higher)   Jan du Toit
• [asterisk-users] Extension presedence.   Jan du Toit
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   Tom
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Tom
• [asterisk-users] What to use beyond T1's?   Tom
• [asterisk-users] iax vs. sip?   Tom
• [asterisk-users] Forbidden - wrong password on authentication for INVITE   Brent Torrenga
• [asterisk-users] RE: By week extension dialing   Brent Torrenga
• [asterisk-users] Re: DUNDi with SIP   Steve Totaro
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Steve Totaro
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Steve Totaro
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Steve Totaro
• [asterisk-users] Steve Totaro I am trying to reach you.   Steve Totaro
• [asterisk-users] Cron Job to Drop a Call File When the Hard Drive Gets over 50% Full   Steve Totaro
• [asterisk-users] Detecting voicemail from CO on FXO port andpassing to H.323 phone. Possible?   Steven Totaro
• [asterisk-users] channel.c: Avoided initial deadlock for '0x8de2dc0', 10 retries!   Tristan
• [asterisk-users] New asterisk jukebox needs testing   Justin Tunney
• [asterisk-users] DMTF issues on voicemail on Zap   Justin Tunney
• [asterisk-users] Return data from Fast AGI   Justin Tunney
• [asterisk-users] linuxdevices.com: >>Trolltech woos developers with "open" Linux phone<< Who'll be the first with * on a mobile?   Justin Tunney
• [asterisk-users] recommended hardware specs   Justin Tunney
• [asterisk-users] Callback in within voicemail broken   Justin Tunney
• [asterisk-users] Asterisk not parking calls - causes? how to fix?   Justin Tunney
• [asterisk-users] New Asterisk Voice Changer 0.4   Justin Tunney
• [asterisk-users] Uptime Record?   Justin Tunney
• [asterisk-users] Re: New Asterisk Voice Changer 0.4   Justin Tunney
• [asterisk-users] CDR Function - Asterisk-1.2.10   Justin Tunney
• [asterisk-users] SEXY WOMAN wants to know about =>Callback in within voicemail broken   Justin Tunney
• [asterisk-users] New Parrot application, repeats what you say and more!   Justin Tunney
• [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects   Jeff Turner
• [asterisk-users] nat and qualify questions   Alyed Tzompa
• [asterisk-users] channel.c: Avoided initial deadlock for '0x8de2dc0', 10 retries!   Alyed Tzompa
• [asterisk-users] Unicall stack, right versions?   Steve Underwood
• [asterisk-users] Unicall stack, right versions?   Steve Underwood
• [asterisk-users] Unicall stack, right versions?   Steve Underwood
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Steve Underwood
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Steve Underwood
• [asterisk-users] MFC/R2 country and carrier specific protocol variants   Steve Underwood
• [asterisk-users] Ever donate Software to Digium? If you did youra fool.   Steve Underwood
• [asterisk-users] Indonesian MFC/R2   Steve Underwood
• [asterisk-users] t.38 bounty   Steve Underwood
• [asterisk-users] SpanDSP Error   Steve Underwood
• [asterisk-users] SpanDSP Error   Steve Underwood
• [asterisk-users] Can I increase DTMF sensitivity?   Steve Underwood
• [asterisk-users] Fax with asterisk?   Steve Underwood
• [asterisk-users] queue announcements when using ringback   Urban
• [asterisk-users] About Digium cards and HP DL servers   VOICEIN at aol.com
• [asterisk-users] Re: SV: E61   Jens Vagelpohl
• [asterisk-users] Re: SV: E61   Jens Vagelpohl
• [asterisk-users] Problem with Tycho Voicemail   Race Vanderdecken
• [asterisk-users] ESCAUX releases net.PBX Free Edition   Gary Vanhoute
• [asterisk-users] Reboot Mediatrix   Julian Varanini
• [asterisk-users] Mediatrix 1204 and Asterisk 1.2.10   Julian Varanini
• [asterisk-users] PRI Ethernet Bridge   Julian Varanini
• [asterisk-users] PRI and Asterisk   Julian Varanini
• [asterisk-users] New Voicemail Client for Win32, Linux x86, Mac OS X released   Arnd Vehling
• [asterisk-users] New Voicemail Client for Win32, Linux x86, Mac OS X released   Arnd Vehling
• [asterisk-users] Re: New Voicemail Client for Win32, Linux x86, MacOS X released   Arnd Vehling
• [asterisk-users] Problem with Tycho Voicemail   Arnd Vehling
• [asterisk-users] Problem with Tycho Voicemail   Arnd Vehling
• [asterisk-users] Problem with Tycho Voicemail   Arnd Vehling
• [asterisk-users] Problem with Tycho Voicemail   Arnd Vehling
• [asterisk-users] Problem with Tycho Voicemail   Arnd Vehling
• [asterisk-users] Voicemail/Email Integration   Arnd Vehling
• [asterisk-users] SER+iptables+Asterisk   Arnd Vehling
• [asterisk-users] Re: asterisk-users Digest, Vol 25, Issue 35   Miguel Ruiz Velasco
• [asterisk-users] DMTF issues on voicemail on Zap   Miguel Ruiz Velasco
• [asterisk-users] Re: DMTF issues on voicemail on Zap   Miguel Ruiz Velasco
• [asterisk-users] Callback and Asterisks   Vic
• [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?   Tom Vile
• [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?   Tom Vile
• [asterisk-users] Phone Newbie Questions   Tom Vile
• [asterisk-users] Snakes On A Plane using Asterisk?   Tom Vile
• [asterisk-users] Can budgetone 101 display name part of cid?   Tom Vile
• [asterisk-users] Nobody is responding. Why? (Implement music on transfer)   Tom Vile
• [asterisk-users] does anyone offer truly unlimited voip in the US   Tom Vile
• [asterisk-users] does anyone offer truly unlimited voip in the US   Tom Vile
• [asterisk-users] Asterisk And Java?   Lennie De Villiers
• [asterisk-users] Asterisk and PHP?   Lennie De Villiers
• [asterisk-users] Asterisk and PHP?   Lennie De Villiers
• [asterisk-users] Asterisk and PHP?   Lennie De Villiers
• [asterisk-users] Asterisk & Gizmo?   Lennie De Villiers
• [asterisk-users] SIP ActiveX?   Lennie De Villiers
• [asterisk-users] Associating an Originate Request to a Channel before the call is answered   Janahan Vivekanandan
• [asterisk-users] Call to a queue killing Asterisk?   Terry Wade
• [asterisk-users] Port Forwarding SIP rtp   Evalyn Wafula
• [asterisk-users] oddity with TDM400P / Asterisk setup   Ted Wallingford
• [asterisk-users] oddity with TDM400P / Asterisk setup   Ted Wallingford
• [asterisk-users] PRI Clock Signal Problem   Nico van der Walt
• [asterisk-users] Connecting to another server   Dennis Wambugu
• [asterisk-users] RE VoipNow 1.2.0 Beta   Matthew Warren
• [asterisk-users] codec conversion   Wasif
• [asterisk-users] Asterisk Voicemail Setup   Wasif
• [asterisk-users] Asterisk Billing   Wasif
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] SIP Qualify   Watkins, Bradley
• [asterisk-users] Hint extension issue - bug?   Watkins, Bradley
• [asterisk-users] How to set externip in sip.conf automatically?   Watkins, Bradley
• [asterisk-users] Hint extension issue - bug?   Watkins, Bradley
• [asterisk-users] Hint extension issue - bug?   Watkins, Bradley
• [asterisk-users] Hint extension issue - bug?   Watkins, Bradley
• [asterisk-users] Asterisk and Speakeasy VOIP   Mike Weaver
• [asterisk-users] Manager Interface API's   Mike Weaver
• [asterisk-users] Trying to build a hybrid Asterisk system w/ one POTS and 2 Speakeasy VOIPs - anyone done something similar w/ Speakeasy?   Mike Weaver
• [asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?   Mike Weaver
• [asterisk-users] Japanese Sound Files   Hermann Wecke
• [asterisk-users] Setting CALLERID on a residential telco line   Hermann Wecke
• [asterisk-users] Check call duration on active call in CLI?   Hermann Wecke
• [asterisk-users] HFC-S Cards in the UK   Hermann Wecke
• [asterisk-users] Callback and Asterisks   Hermann Wecke
• [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found   Hermann Wecke
• [asterisk-users] CDR Variable   Hermann Wecke
• [asterisk-users] OT: Changing Cisco tftp root directory   Hermann Wecke
• [asterisk-users] Asterisk time not the same as unit time ?   Hermann Wecke
• [asterisk-users] Portuguese sound files available?   Hermann Wecke
• [asterisk-users] Cisco 7960G SIP firmware 8.4   Hermann Wecke
• [asterisk-users] Sipura SPA3000   Hermann Wecke
• [asterisk-users] asterisk optimizing   Jack Wei
• [asterisk-users] HFC-S Cards in the UK   Ron Wellsted
• [asterisk-users] HFC-S Cards in the UK   Ron Wellsted
• [asterisk-users] HFC-S Cards in the UK   Ron Wellsted
• [asterisk-users] Working Sipura 3000 or Linksys 3102 configuration?   Ron Wellsted
• [asterisk-users] IP phone with 2 ethernet jacks   Ron Wellsted
• [asterisk-users] Using asterisk to simulate ISDN BRI line   Ron Wellsted
• [asterisk-users] polycom config error 0x4020: possibly related to RE:Polycom upgrade issue?   Michael Welter
• [asterisk-users] Asterisk server crashes after two years   Michael Welter
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   BJ Weschke
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   BJ Weschke
• [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch   BJ Weschke
• [asterisk-users] reloading agents and queues   BJ Weschke
• [asterisk-users] Turn Off chan_sip Debug Messages   BJ Weschke
• [asterisk-users] Recent additions to the Digium Asterisk development team   BJ Weschke
• [asterisk-users] How to find which queue member answered a call?   BJ Weschke
• [asterisk-users] How to find which queue member answered a call?   BJ Weschke
• [asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?   BJ Weschke
• [asterisk-users] Cisco Router QOS and IAX2   BJ Weschke
• [asterisk-users] What are my logs telling me here?   BJ Weschke
• [asterisk-users] Make Asterisk server initiate a Call   BJ Weschke
• [asterisk-users] pri rdnis found as Facility but not set   Henrik Westerberg
• [asterisk-users] Idiot questions   Derek Whitten
• [asterisk-users] Asterisk with PABX   Tux Wi-FI
• [asterisk-users] Asterisk with PABX   Tux Wi-FI
• [asterisk-users] 1.2.10 - g726 Issues   Cullin J. Wible
• [asterisk-users] 1.2.10 - g726 Issues   Cullin J. Wible
• [asterisk-users] 1.2.10 - g726 Issues   Cullin J. Wible
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk= same in 1.2.10   WideVOIP
• [asterisk-users] cmd DIAL - Who picked up the call?   Eric "ManxPower" Wieling
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Eric "ManxPower" Wieling
• [asterisk-users] SIP_HEADER() read-only   Eric "ManxPower" Wieling
• [asterisk-users] Run a script at certain CLI writes   Eric "ManxPower" Wieling
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Eric "ManxPower" Wieling
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Eric "ManxPower" Wieling
• [asterisk-users] Run a script at certain CLI writes   Eric "ManxPower" Wieling
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Eric "ManxPower" Wieling
• [asterisk-users] Setting CALLERID on a residential telco line   Eric "ManxPower" Wieling
• [asterisk-users] Setting CALLERID on a residential telco line   Eric "ManxPower" Wieling
• [asterisk-users] how to check the status of a channel   Eric "ManxPower" Wieling
• [asterisk-users] Variables sip redirects and call forward   Eric "ManxPower" Wieling
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Eric "ManxPower" Wieling
• [asterisk-users] FXS gateway/Channel Bank   Eric "ManxPower" Wieling
• [asterisk-users] Polycom 1.6.7 firmware?   Eric "ManxPower" Wieling
• [asterisk-users] PRI Connection in Lima, Peru   Eric "ManxPower" Wieling
• [asterisk-users] Polycom 1.6.7 firmware?   Eric "ManxPower" Wieling
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Eric "ManxPower" Wieling
• [asterisk-users] can i detect a voice with asterisk ?   Eric "ManxPower" Wieling
• [asterisk-users] Set DID?   Eric "ManxPower" Wieling
• [asterisk-users] Set DID?   Eric "ManxPower" Wieling
• [asterisk-users] MailboxExists not branching to n+101   Eric "ManxPower" Wieling
• [asterisk-users] Auto retry on Busy   Eric "ManxPower" Wieling
• [asterisk-users] Auto retry on Busy   Eric "ManxPower" Wieling
• [asterisk-users] Fast busy signals... Satisfying my curiousity   Eric "ManxPower" Wieling
• [asterisk-users] Macro inside macro   Eric "ManxPower" Wieling
• [asterisk-users] Macro inside macro   Eric "ManxPower" Wieling
• [asterisk-users] SIP Qualify   Eric "ManxPower" Wieling
• [asterisk-users] SIP Qualify   Eric "ManxPower" Wieling
• [asterisk-users] Zap difficulties   Eric "ManxPower" Wieling
• [asterisk-users] Ringing after answered on zaptel   Eric "ManxPower" Wieling
• [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)   Eric "ManxPower" Wieling
• [asterisk-users] Comfort noise support incomplete in Asterisk (RFC 3389).   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk 'Hosting'   Eric "ManxPower" Wieling
• [asterisk-users] No "zap" command?   Eric "ManxPower" Wieling
• [asterisk-users] Sending Email From A Dial Plan   Eric "ManxPower" Wieling
• [asterisk-users] NAT problems   Eric "ManxPower" Wieling
• [asterisk-users] Max number of SIP devices registered to anextension   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk with PABX   Eric "ManxPower" Wieling
• [asterisk-users] does anyone offer truly unlimited voip in the US   Eric "ManxPower" Wieling
• [asterisk-users] trinary expression   John Williams
• [asterisk-users] Running AGI in background   John Williams
• [asterisk-users] Simple config question   John Williams
• [asterisk-users] RE: By week extension dialing   John Williams
• [asterisk-users] Asterisk Performance without RTP?   Kelvin Williams
• [asterisk-users] Sipura 3000 and Asterisk   Mark Willis
• [asterisk-users] g729 and trafic   Walter Willis
• [asterisk-users] Multi Asterisk Server to relay call request   Stephen Wingfield
• [asterisk-users] sox &gsm   Ronald Wiplinger
• [asterisk-users] if command for or missing callerid?   Ronald Wiplinger
• [asterisk-users] re-writing the dial plan - some hints please   Ronald Wiplinger
• [asterisk-users] Wellgate 3804a   Ronald Wiplinger
• [asterisk-users] GIZMO and Asterisk, "Failed to authenticate"   Ronald Wiplinger
• [asterisk-users] Wellgate 3804a: Got SIP response 486 "Busy Here"   Ronald Wiplinger
• [asterisk-users] Am I looking for automon?   Ronald Wiplinger
• [asterisk-users] Asterisk server crashes after two years   Ronald Wiplinger
• [asterisk-users] Problems with recording   Ronald Wiplinger
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Wireless
• [asterisk-users] Re: New Device   Wireless
• [asterisk-users] PRI Dropouts (Solved)   Kevin Withnall
• [asterisk-users] Re: IP phone with 2 ethernet jacks   Hans Witvliet
• [asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?   Henrik Woffinden
• [asterisk-users] Cannot dial out through SIP provider   Henrik Woffinden
• [asterisk-users] Cannot dial out through SIP provider   Henrik Woffinden
• [asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?   Henrik Woffinden
• [asterisk-users] Ascom Eurit 133 cordless ISDN phone   Henrik Woffinden
• [asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?   Henrik Woffinden
• [asterisk-users] running agi application in the background   Tobias Wolf
• [Asterisk-Users] Using HINT with Cisco 7940/SIP   Conrad Wood
• [asterisk-users] Is there a Blue tooth wireless headset that will work with asterisk?   Conrad Wood
• [asterisk-users] IAX unstable with large number of calls?   Simon Woodhead
• [asterisk-users] Apache for FastAGI   Simon Woodhead
• [asterisk-users] Calls over VPN   Simon Woodhead
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Simon Woodhead
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Simon Woodhead
• [asterisk-users] E61   Simon Woodhead
• [asterisk-users] E61   Simon Woodhead
• [asterisk-users] Re: Nokia E60/61/70 and SIP   Simon Woodhead
• [asterisk-users] iax vs. sip?   Simon Woodhead
• [asterisk-users] FXS gateway/Channel Bank   Roger Workman
• [asterisk-users] Voicemail Platform   Roger Workman
• [asterisk-users] Voicemail Platform   Roger Workman
• [asterisk-users] Missing Extension   Roger Workman
• [asterisk-users] Missing Extension   Roger Workman
• [asterisk-users] Apache for FastAGI   Tielin Xu
• [asterisk-users] Apache for FastAGI   Tielin Xu
• [asterisk-users] SPA-942 TFTP Provisioning   Vahan Yerkanian
• [asterisk-users] Selecting outbound trunk   Iain Young
• [asterisk-users] 911 Testing   Shane Young
• [asterisk-users] Text to Speech   Shane Young
• [asterisk-users] Choppy calls on IAX trunk but no problems on internal calls   Carl Youngblood
• [asterisk-users] Choppy calls on IAX trunk but no problems on internal calls   Carl Youngblood
• [asterisk-users] Asterisk Real Time and sip.conf file used at   Yusuf
• [asterisk-users] 0005162: RTP Packetization : Few questions   Yusuf
• [asterisk-users] Help in dailplan in asterisk   Sylvain ZUCCA
• [asterisk-users] PROBLEM MUSIC ON HOLD   Zeeshan Zakaria
• [asterisk-users] Festival Not Working   Zeeshan Zakaria
• [Asterisk-Users] trixbox 1.1 download   Zeeshan Zakaria
• [asterisk-users] Is anybody moderating this list?   Zeeshan Zakaria
• [asterisk-users] voicemail in mp3 format   Ever Zalazar
• [asterisk-users] Dapper Drake, Asterisk, and Faxing   Warrick Zedi
• [asterisk-users] Asterfax and Gentoo   Warrick Zedi
• [asterisk-users] Asterisk and spandsp   Warrick Zedi
• [asterisk-users] Re: Asterisk IAXmodem HylaFax?   Warrick Zedi
• [asterisk-users] Re: Asterisk IAXmodem HylaFax?   Warrick Zedi
• [asterisk-users] Re: Asterisk IAXmodem HylaFax?   Warrick Zedi
• [asterisk-users] Newzealand zaptel DTMF problem   Ma Zhiyong
• [asterisk-users] newbie - suggestions on installing Asterisk for SOHO   Pele Zico
• [asterisk-users] Arrays ???   Pele Zico
• [asterisk-users] Re: Arrays ???   Pele Zico
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Paul Zimm
• [asterisk-users] How to modify incoming DNIS?   Chris Ziomkowski
• [asterisk-users] Asterisk forum - forum.globalvoicenet.com   Zoa
• [asterisk-users] monmp3thread: Request to schedule in the past?!?!   Zoa
• [asterisk-users] USB GSM gateway for Asterisk?   Zoa
• [asterisk-users] help with app_sms and chan_capi   Patrick Zwahlen
• [asterisk-users] Asterisk and Siemens Legacy PBX   Wolfgang Zweimueller
• [asterisk-users] Asterisk and Siemens Legacy PBX   Wolfgang Zweimueller
• [asterisk-users] Asterisk and Siemens Legacy PBX   Wolfgang Zweimueller
• [asterisk-users] HFC-S Cards in the UK   Wolfgang Zweimueller
• [asterisk-users] Modems dialing over sangoma a104d   Wolfgang Zweimueller
• [asterisk-users] abhishek invites you to join Zorpia   abhishek
• [asterisk-users] New Device   ahester at galacticltd.com
• [asterisk-users] Voicemail/Email Integration   ahester at galacticltd.com
• [asterisk-users] Voicemail/Email Integration   ahester at galacticltd.com
• [asterisk-users] Using Thunderbird (mail client) to call Contacts from Address Book   ajmcello
• [asterisk-users] NAT problems   andrutto
• [asterisk-users] Re: NAT problems   andrutto
• [asterisk-users] H323   andrutto
• [asterisk-users] About Digium cards and HP DL servers   angom at telnor.net
• [asterisk-users] Zaptel install - Fedora Core 5   anto
• [asterisk-users] personal address progress pri   antonio
• [asterisk-users] CallerID and call progress pri   antonio
• [asterisk-users] newbie request   arthurvdmolen
• [asterisk-users] codec_g729a.so coredump in SVN trunk   ast-lists at delvar.com
• [asterisk-users] FXO module burn out !?   asterisk
• [asterisk-users] Asterisk installations in Germany   asterisk-robert
• [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?   asterisk at anime.net
• [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?   asterisk at anime.net
• [asterisk-users] question about oh323 and ring tone   asterisk at frameweb.it
• [asterisk-users] Asterisk and T1 Extensions.   asterisk at key.com.br
• [asterisk-users] No-audio problem   asterisk at teladdict.com
• [asterisk-users] no audio issue   asterisk at teladdict.com
• [asterisk-users] Re: no audio issue   asterisk at teladdict.com
• [asterisk-users] CDR Function - Asterisk-1.2.10   asterisk at teladdict.com
• [asterisk-users] CDR Function - Asterisk-1.2.10   asterisk at teladdict.com
• [asterisk-users] CDR Function - Asterisk-1.2.10   asterisk at teladdict.com
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group - Saturday August 12th - 11:30am   asterisk_help at iwishi.nu
• [asterisk-users] Conditional branching   bails
• [asterisk-users] Conditional branching   bails
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   bails
• [asterisk-users] More SNOM, Message Indicator/Retrieval issues   bails
• [asterisk-users] Zaptel install - Fedora Core 5   simon elliston ball
• [asterisk-users] Re: Zaptel install - Fedora Core 5   simon elliston ball
• [asterisk-users] Realtime Extensions -- Comments?   simon elliston ball
• [asterisk-users] Realtime Extensions -- Comments?   simon elliston ball
• [asterisk-users] Realtime Extensions -- Comments?   simon elliston ball
• [asterisk-users] Will Asterisk work with Exchange 2007 UM?   blawrence at qwest.net
• [asterisk-users] need dialout help in python script   shawn bright
• [asterisk-users] need dialout help in python script   shawn bright
• [asterisk-users] need dialout help in python script   shawn bright
• [asterisk-users] Re: need a pointer regarding scripting asterisk   shawn bright
• [asterisk-users] Ragi without rails possible ?   shawn bright
• [asterisk-users] Ragi without rails possible ?   shawn bright
• [asterisk-users] agi script runs even if no answer   shawn bright
• [asterisk-users] agi script runs even if no answer   shawn bright
• [asterisk-users] Re: agi script runs even if no answer   shawn bright
• [asterisk-users] Re: agi script runs even if no answer   shawn bright
• [asterisk-users] need dialout help in python script   shawn bright
• [asterisk-users] can i detect a voice with asterisk ?   shawn bright
• [asterisk-users] can i detect a voice with asterisk ?   shawn bright
• [asterisk-users] can i detect a voice with asterisk ?   shawn bright
• [asterisk-users] prob with star input agi-bin   shawn bright
• [asterisk-users] prob with star input agi-bin   shawn bright
• [asterisk-users] pyAst   shawn bright
• [asterisk-users] Unicall stack, right versions?   leonimar cape
• [asterisk-users] Unicall stack, right versions?   leonimar cape
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   leonimar cape
• [asterisk-users] t.38 asterisk-trunk   marek cervenka
• [asterisk-users] t.38 bounty   marek cervenka
• [asterisk-users] Announce caller-id   coolbreeze
• [asterisk-users] previous reload of asterisk did not finish   John covici
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   John covici
• [asterisk-users] Text to Speech   John covici
• [asterisk-users] asterisk registering as extension to another asterisk server problem   John covici
• [asterisk-users] upgrade problem on IP phone 9133i   Jean-Louis curty
• [asterisk-users] upgrade problem on IP phone 9133i   Jean-Louis curty
• [asterisk-users] SendText Queue Notification   Jean-Louis curty
• [asterisk-users] upgrade problem on IP phone 9133i   Jean-Louis curty
• [asterisk-users] Debian and Asterisk IAX2 channel driver   dawson at lunaazulgallery.com
• [asterisk-users] originate from group + dialplan   dmb
• [asterisk-users] voicemailmain   existx
• [asterisk-users] voicemailmain   existx
• [asterisk-users] voicemailmain   existx
• [asterisk-users] Problem with a TDM400P   f6hqz-m at hamwlan.net
• [asterisk-users] Snom phones locking up   garth at bitco.co.za
• [asterisk-users] Question about context for incoming calls   gc
• [asterisk-users] Got error when compiling asterisk 1.2.11   gc
• [asterisk-users] Administrator Forum Email   bilal ghayyad
• Autoreply: [asterisk-users] Deployment for less than 10 phones   gparlato at tnet.it
• Autoreply: [asterisk-users] e&m wink, TE110P, * answers too soon   gparlato at tnet.it
• Autoreply: [asterisk-users] Tri-Link Technologies?   gparlato at tnet.it
• Autoreply: [asterisk-users] How to, astcc and virtuemart   gparlato at tnet.it
• Autoreply: [asterisk-users] Snom MWI   gparlato at tnet.it
• Autoreply: Re: Autoreply: [asterisk-users] Tri-Link Technologies?   gparlato at tnet.it
• [asterisk-users] Unknown CLI output   wendell hamilton
• [asterisk-users] SRTP   harrygaillac-sip at yahoo.fr
• [asterisk-users] IAX2 Peer   hernany.ce
• [asterisk-users] IAX2 Turnks   hernany.ce
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   dorn hetzel
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] Setting CALLERID on a residential telco line   hugolivude
• [asterisk-users] Setting CALLERID on a residential telco line   hugolivude
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   hugolivude
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   hugolivude
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   hugolivude
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   hugolivude
• [asterisk-users] Cannot dial out through SIP provider   hugolivude
• [asterisk-users] Cannot dial out through SIP provider   hugolivude
• [asterisk-users] VoiceMail being cutoff when leaving message   hugolivude
• [asterisk-users] Need help   hugolivude
• [asterisk-users] caller id problem   hugolivude
• [asterisk-users] help me!!Problem on incoming calls   Andrea infoteam
• [asterisk-users] FXO module burn out !?   jacobso1
• [asterisk-users] how to add prefix 0 (zero) when coming from ISDN trunk   kitti jaisong
• [asterisk-users] Help debugging strange asterisk behaviour   jan.sarin at securia.se
• [asterisk-users] SV: Help debugging strange asterisk behaviour   jan.sarin at securia.se
• SV: [asterisk-users] Help debugging strange asterisk behaviour   jan.sarin at securia.se
• SV: [asterisk-users] Help debugging strange asterisk behaviour   jan.sarin at securia.se
• [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument   jan.sarin at securia.se
• [asterisk-users] Encoding recorded queue calls to mp3   jan.sarin at securia.se
• SV: [asterisk-users] Help debugging strange asterisk behaviour   jan.sarin at securia.se
• SV: [asterisk-users] Help debugging strange asterisk behaviour (update)   jan.sarin at securia.se
• [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?   jan.sarin at securia.se
• SV: [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?   jan.sarin at securia.se
• [asterisk-users] Idiot questions   joea, j4computers
• [asterisk-users] Idiot questions   joea, j4computers
• [asterisk-users] Idiot questions   joea, j4computers
• [asterisk-users] Idiot questions   joea, j4computers
• [asterisk-users] MixMonitor and g729 licenses   jurgen
• [asterisk-users] video call monitor   atik khan
• [asterisk-users] Nokia E60/61/70 and SIP   atik khan
• [asterisk-users] H323   atik khan
• [asterisk-users] Handling inbound and outbound calls passed from a proxy   kjcsb
• [Asterisk-Users] CDRTool   kjcsb
• [Asterisk-Users] OPENSER / SER and Asterisk   kjcsb
• [asterisk-users] Force immediate re-registration on sip reload   kjcsb
• [asterisk-users] SPA-942 TFTP Provisioning   kjcsb
• [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time   kjcsb
• [asterisk-users] Dial out based on SIP invite   kjcsb
• [asterisk-users] Accessing SIP URI (not ${SIPURI})   kjcsb
• [asterisk-users] SIP_HEADER function; what names are available?   kjcsb
• [asterisk-users] SIP_HEADER function; what names are available?   kjcsb
• [asterisk-users] About IVR and Oracle (Tim Panton)   leoarellan at hotmail.com
• [asterisk-users] Monitoring/Listening In (Scott Pinhorne)   leoarellan at hotmail.com
• [asterisk-users] Speed dials on Polycom IP601?   Warren (mailing lists)
• [asterisk-users] Variable to show caller id for a current call?   Warren (mailing lists)
• [asterisk-users] Variable to show caller id for a current call?   Warren (mailing lists)
• [asterisk-users] Trunk with multiple IPs?   Warren (mailing lists)
• [asterisk-users] Trunk with multiple IPs?   Warren (mailing lists)
• [asterisk-users] quadbri & TDM400P on same pbx ?   mike
• [asterisk-users] VoipNow 1.2.0 Beta   mitcheloc
• [asterisk-users] linuxdevices.com: >>Trolltech woos developers with "open" Linux phone<< Who'll be the first with * on a mobile?   mitcheloc
• [asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?   mogorman
• [asterisk-users] Asterisk config with Analouge Audio Codec model number MP108FXS   Mr shobhit nirala
• [asterisk-users] playback() breaks audio in zap->iax->iax->zap channel   jeff oconnell
• [asterisk-users] RTP Stream not set up correct at outgoing call   ontae
• [asterisk-users] Need help: RTP Stream not set up correct only when calling out   ontae
• [asterisk-users] Ignoring the # key on a call   patrick
• [asterisk-users] Simple incomming fax solution ...   phil.dawson at marnock.com
• [asterisk-users] Stable asterisk version   phil.dawson at marnock.com
• [asterisk-users] compile problems with app_rxfax.c and asterisk 1.2.11   phil.dawson at marnock.com
• [asterisk-users] Re: [Serusers] SER local as an Asterisk Trunk   ram
• [asterisk-users] Asterisk config with Analouge Audio Codec model number MP108FXS   ram
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   ram
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   ram
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution   ram
• [Asterisk-Users] OPENSER / SER and Asterisk   ram
• [asterisk-users] Asterisk Real Time and sip.conf file used at the same time   rcarvalho at iric.up.pt
• [asterisk-users] Asterisk t38passthrough   rcarvalho at iric.up.pt
• [asterisk-users] Mix Monitor call quality   robb at boardman.me.uk
• [asterisk-users] Realtime SIP Authentication   ronn100200 at aim.com
• [asterisk-users] howto install asterisk on freebsd release 4.11   mansour safai
• [asterisk-users] problem- 0:10 long message   ismir saljic
• [asterisk-users] Line detection with TDM400P   levy samuel
• [asterisk-users] How to use a Half E1 with Asterisk?   levy samuel
• [asterisk-users] Follow ON calling on DISA   sdcharly at gmail.com
• [asterisk-users] AG-168V not registering.   sdcharly at gmail.com
• [asterisk-users] Re: AgentcallbackLogin()   sdcharly at gmail.com
• [asterisk-users] SPA3000 dialplan coding...   sdcharly at gmail.com
• [asterisk-users] IAX2 extn not registering on 4569   sdcharly at gmail.com
• [asterisk-users] Sipura 3000 and Asterisk   sdcharly at gmail.com
• [asterisk-users] Asterisk with VoIP phone   shadowym
• [asterisk-users] VOIP phone for Receptionist use   shadowym
• [asterisk-users] Opinions on Rhino PCI FXO cards   shadowym
• [asterisk-users] Ever donate Software to Digium? If you did youra fool.   shadowym
• [asterisk-users] Digium TDM400P Vs Sangoma A200   shadowym
• [asterisk-users] SLA.conf   shadowym
• [asterisk-users] SLA.conf   shadowym
• [asterisk-users] CentOS4.3 or Debian 3.1r2?   shadowym
• [asterisk-users] Help compiling asterisk-addons on Debian?   shadowym
• [asterisk-users] IP phone with 2 ethernet jacks   shadowym
• [asterisk-users] Asterisk with PABX   shadowym
• [asterisk-users] upgrade problem on IP phone 9133i   shadowym
• [asterisk-users] upgrade problem on IP phone 9133i   shadowym
• [asterisk-users] How is GXP2000 with latest firmware   shadowym
• [asterisk-users] Issues compiling addons on Fedora Core 3   sip
• [asterisk-users] Issues compiling addons on Fedora Core 3   sip
• [asterisk-users] Issues compiling addons on Fedora Core 3   sip
• [asterisk-users] Maximum length of CID using SET CALLERID in AGI?   sip
• [asterisk-users] Snom phones locking up   sip
• [asterisk-users] Grabbing authenticated mailbox value from VoicemailMain()   sip
• [asterisk-users] Grabbing authenticated mailbox value from VoicemailMain()   sip
• [asterisk-users] What I can use with ASTERISK to call clients to remind them about their appointments   dmitri smirnoff
• [asterisk-users] Queue problem - autofill option   equis software
• [asterisk-users] Analyze core file prodeced after safe_asterisk crashh   equis software
• [asterisk-users] Analyze core file prodeced after safe_asterisk crashh   equis software
• [asterisk-users] Problems using Queues with Autofill option   equis software
• [asterisk-users] New to Asterisk...   raviprakash sunkara
• [asterisk-users] Help in dailplan in asterisk   raviprakash sunkara
• [asterisk-users] call barge   support
• [asterisk-users] Ztdummy - No audio in BackGround function   support_list
• [asterisk-users] Two card NT-TE mode   support_list
• [asterisk-users] Two card NT-TE mode   support_list
• [asterisk-users] IAX2 TRUNK CPU consumption   support_list
• [asterisk-users] H323 can not register to remote openh323gk?   tengulre
• [asterisk-users] asterisk optimizing   tracinet
• [asterisk-users] queue in realtime   unplug
• [asterisk-users] queue in realtime   unplug
• [asterisk-users] Declined to talk, Call rejected: 603 Declined   unplug
• [asterisk-users] Question about queue   unplug
• [asterisk-users] Re: Question about queue   unplug
• [asterisk-users] channel variable   unplug
• [asterisk-users] Re: channel variable   unplug
• [asterisk-users] about MusicOnHold / Playback   unplug
• [asterisk-users] caller display problem   unplug
• [asterisk-users] question of CLI   unplug
• [asterisk-users] caller id problem   unplug
• [asterisk-users] asterisk gui   vivek at staff.ownmail.com
• [asterisk-users] asterisk dosenot compile   vivek at staff.ownmail.com
• [asterisk-users] sip giving problems, please help.   vivek at staff.ownmail.com
• [asterisk-users] dialplan help   vivek at staff.ownmail.com
• [asterisk-users] dialplan help   vivek at staff.ownmail.com
• [asterisk-users] Check call duration on active call in CLI?   voiplist
• [asterisk-users] Linksys SPA-941 Message Waiting Indicator   voiplist
• [asterisk-users] placing a call with the Manager interface   whois wes
• [asterisk-users] US Toll-Free DID Providers with Caller ID NAME?   www.IPKall.com
• [asterisk-users] Registering IP Phone To Asterisk   wyatt.wmvg at comcast.net
• [asterisk-users] Asterisk suddenly die   yusuf
• [asterisk-users] valgrind + Asterisk   yusuf
• [asterisk-users] valgrind + Asterisk   yusuf