August 2003 Archives by author

• Messages sorted by: [ thread ] [ subject ] [ date ]
• More info on this list...

Starting: Fri Aug 1 00:07:45 MST 2003
Ending: Sun Aug 31 23:16:37 MST 2003
Messages: 2187

• [Asterisk-Users] call center - operators not using phone keys   Miguel Bettencourt Dias (Netopia)
• [Asterisk-Users] call center - operators not using phone keys   Miguel Bettencourt Dias (Netopia)
• [Asterisk-Users] call center - operators not using phone keys   Miguel Bettencourt Dias (Netopia)
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   WipeOut .
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   WipeOut .
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   WipeOut .
• [Asterisk-Users] SendDtmf   WipeOut .
• [Asterisk-Users] SendDtmf   WipeOut .
• [Asterisk-Users] SendDtmf   WipeOut .
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   WipeOut .
• [Asterisk-Users] chan_capi: Hanging channels - again   WipeOut .
• [Asterisk-Users] Call Monitor   WipeOut .
• [Asterisk-Users] Newbie just starting out with *   WipeOut .
• [Asterisk-Users] X-Lite <-> Snom200   WipeOut .
• [Asterisk-Users] X-Lite <-> Snom200   WipeOut .
• [Asterisk-Users] X-Lite <-> Snom200   WipeOut .
• [Asterisk-Users] Behind Firewalls, SonicWalls, etc..   WipeOut .
• [Asterisk-Users] Budgettone Newbie   WipeOut .
• [Asterisk-Users] Leftover Budgettone issues   WipeOut .
• [Asterisk-Users] MWI bug ?   WipeOut .
• [Asterisk-Users] Call Waiting and Call Parking Together??   WipeOut .
• [Asterisk-Users] Call Waiting and Call Parking Together??   WipeOut .
• [Asterisk-Users] X-Lite - No sound + chan_sip issue   WipeOut .
• [Asterisk-Users] X-Lite - No sound + chan_sip issue   WipeOut .
• [Asterisk-Users] G.729 licensing -- an opinion   WipeOut .
• [Asterisk-Users] ATT: marrandy - Re: Grandstream Budgettone 102   WipeOut .
• [Asterisk-Users] list proposal   WipeOut .
• [Asterisk-Users] Chan_Capi questions??   WipeOut .
• [Asterisk-Users] SNOM200 firmware roll back!!   WipeOut .
• [Asterisk-Users] Windows Messenger   WipeOut .
• [Asterisk-Users] OT: Grandstream power supplies..   WipeOut .
• [Asterisk-Users] OT: Grandstream power supplies..   WipeOut .
• [Asterisk-Users] New gastman clone + what else?   WipeOut .
• [Asterisk-Users] Future Grandstream codecs??   WipeOut .
• [Asterisk-Users] IP phone recommendation   WipeOut .
• [Asterisk-Users] Codec?   WipeOut .
• [Asterisk-Users] IP phone recommendation   WipeOut .
• [Asterisk-Users] IP phone recommendation   WipeOut .
• [Asterisk-Users] Extension and phone management best practices??   WipeOut .
• [Asterisk-Users] Extension and phone management bestpractices??   WipeOut .
• [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?   WipeOut .
• [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?   WipeOut .
• [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?   WipeOut .
• [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?   WipeOut .
• [Asterisk-Users] Configuring iptables to allow sip and dynamically allocate rtp ports   WipeOut .
• [Asterisk-Users] Recomendations for an ISDN-PBX to use with asterisk   WipeOut .
• [Asterisk-Users] Grandstream, SIP encryption   WipeOut .
• [Asterisk-Users] Compile problems   WipeOut .
• [Asterisk-Users] SIP using which codec?   WipeOut .
• [Asterisk-Users] Is Asterisk ready for "real" use?   WipeOut .
• [Asterisk-Users] VAD (silence suppression) on Asterisk   WipeOut .
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   WipeOut .
• [Asterisk-Users] Xphone Lite Cannot make work on Asterisk   WipeOut .
• [Asterisk-Users] Asterisk + SNOM + Pound and star keys   WipeOut .
• [Asterisk-Users] Grandstream Budgetone Defective Units   WipeOut .
• [Asterisk-Users] Best SIP phone?   WipeOut .
• [Asterisk-Users] Asterisk + SNOM + Pound and star keys   WipeOut .
• [Asterisk-Users] Webmin and incoming call recording   WipeOut .
• [Asterisk-Users] Grandstream firmware update. {HTTP error}   WipeOut .
• [Asterisk-Users] SIP phones   WipeOut .
• [Asterisk-Users] SNOM 200 bugs   WipeOut .
• [Asterisk-Users] X100P not hanging up.   WipeOut .
• [Asterisk-Users] Conference without zaptel??   WipeOut .
• [Asterisk-Users] Conference without zaptel??   WipeOut .
• [Asterisk-Users] Newbie Question / ISDN   Rémi Letot
• [Asterisk-Users] cdr_csv actual duaration   Голышев Алексей
• [Asterisk-Users] Echo cancellation problem from SIP to PSTN   Daniel ANDRE
• [Asterisk-Users] SIP and ECHO   Daniel ANDRE
• [Asterisk-Users] Call transfer ATA186   ASN
• [Asterisk-Users] Re: Zaptel cards, working FXS and SIP, no audio?   Adams, Gavin
• [Asterisk-Users] Overlap on PRI to PSTN   Adams, Gavin
• [Asterisk-Users] Seting up TDM40B   Adams, Gavin
• [Asterisk-Users] Any pointers for setting up PRI for incoming and outgoing calls?   Adams, Gavin
• [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal   Adams, Gavin
• [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal   Adams, Gavin
• [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal   Adams, Gavin
• [Asterisk-Users] Voicemail2 - auto fill the dialing extension?   Adams, Gavin
• [Asterisk-Users] Voicemail2 - auto fill the dialing extension?   Adams, Gavin
• [Asterisk-Users] Voicemail2 - auto fill the dialing extension?   Adams, Gavin
• [Asterisk-Users] help please with single t1 configuration   Adams, Gavin
• [Asterisk-Users] Fax Handled   Adams, Gavin
• [Asterisk-Users] SIP NAT question   Adams, Gavin
• [Asterisk-Users] Alogirthm Used for Extension Matching ?   Adams, Gavin
• [Asterisk-Users] Can I runAsterisk remotely from telnet session?   Adams, Gavin
• [Asterisk-Users] Can I runAsterisk remotely from telnetsession?   Adams, Gavin
• [Asterisk-Users] T100P/ TSU 600 installation problem   Adams, Gavin
• [Asterisk-Users] list of sip phones?   Rich Adamson
• [Asterisk-Users] Sip and One Way Audio   Rich Adamson
• [Asterisk-Users] FXO gateway experience?   Rich Adamson
• [Asterisk-Users] Asterix Newbie   Dayo Adeyeye
• [Asterisk-Users] Ansering machine/voicemail detect?   Garry Adkins
• [Asterisk-Users] reload not working   Marcus Adolfsson
• [Asterisk-Users] reload not working   Marcus Adolfsson
• [Asterisk-Users] reload not working   Marcus Adolfsson
• [Asterisk-Users] Asterisk SIP bug with Net2Phone   Kostyantyn Ahafontsev
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Tan Aks
• [Asterisk-Users] Bad sound quality with G729A on SNOMs   Tan Aks
• [Asterisk-Users] Snome-200 with Asterisk   Tan Aks
• [Asterisk-Users] OT: Grandstream power supplies..   Tan Aks
• [Asterisk-Users] Running Asterisk behind NAT?   Tan Aks
• [Asterisk-Users] Grandstream Budgetone   Tan Aks
• [Asterisk-Users] Configuring iptables to allow sip and dynamically allocate rtp ports   Tan Aks
• [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)   Tan Aks
• [Asterisk-Users] Is Asterisk ready for "real" use?   Chris Albertson
• [Asterisk-Users] IAX <> IAX trunking... DP cache?   Chris Albertson
• [Asterisk-Users] X100P CallerID issue solved for my PSTN connection   Richard Alexander
• [Asterisk-Users] Dial out modem via * -> VOIP (Real world experience needed)   Richard Alexander
• [Asterisk-Users] .:. .: .. .Stottering audio ??   Richard Alexander
• [Asterisk-Users] Question About BRI Cards   Holger von Ameln
• [Asterisk-Users] call routing based on dnis   Azher Amin
• [Asterisk-Users] h323 over NAT   Erik Anderson
• [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c) or unixODBC (cdr_unixodbc.c) ?   Erik Anderson
• [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c)or unixODBC (cdr_unixodbc.c) ?   Erik Anderson
• [Asterisk-Users] Mysql CDR   Erik Anderson
• [Asterisk-Users] HP300 phone   Micke Andersson
• [Asterisk-Users] CLASS feature syntax   Micke Andersson
• [Asterisk-Users] SIP phones   Micke Andersson
• [Asterisk-Users] Grandstream Budgetone Defective Units   Andres
• [Asterisk-Users] Voicemail2 and RFC2833 DTMF   Andres
• [Asterisk-Users] CVS Question   Andres
• [Asterisk-Users] CVS Question   Andres
• [Asterisk-Users] GS on ebay...   Andres
• [Asterisk-Users] Asterisk stops responding   Andres
• [Asterisk-Users] RE: Asterisk stops responding   Andres
• [Asterisk-Users] bugs.digium.com   Joe Antkowiak
• [Asterisk-Users] Someone used ADIT 600 Channel Bank.   Joe Antkowiak
• [Asterisk-Users] Asterisk and AT&T 964 phones...   Joe Antkowiak
• [Asterisk-Users] FXO mode   Joe Antkowiak
• [Asterisk-Users] queue / agent documentation   CallTrex Personal Assistant
• [Asterisk-Users] VoIP dialtone?   Dan Austin
• [Asterisk-Users] VoIP dialtone?   Dan Austin
• [Asterisk-Users] VoIP dialtone?   Dan Austin
• [Asterisk-Users] segmentation fault with asterisk and OH323   Rattana BIV
• [Asterisk-Users] segmentation fault with asterisk and OH323   Rattana BIV
• [Asterisk-Users] segmentation fault with asterisk and OH323   Rattana BIV
• [Asterisk-Users] CDR   Rattana BIV
• [Asterisk-Users] H323 CallerID   Rattana BIV
• [Asterisk-Users] H323 CallerID   Rattana BIV
• [Asterisk-Users] H323 CallerID   Rattana BIV
• [Asterisk-Users] H323 CallerID   Rattana BIV
• [Asterisk-Users] Alias limitation in asterisk-oh323.0.5.5   Rattana BIV
• [Asterisk-Users] H323 caller ID   Rattana BIV
• [Asterisk-Users] include context   Rattana BIV
• [Asterisk-Users] Include context   Rattana BIV
• [Asterisk-Users] H323 and SIP   Adelino Baena
• [Asterisk-Users] Sip and One Way Audio   Adelino Baena
• [Asterisk-Users] Extension handling.   Michael Baird
• [Asterisk-Users] AGI accountcode.   Michael Baird
• [Asterisk-Users] AGI accountcode.   Michael Baird
• [Asterisk-Users] unsubscribe   Jean-Emmanuel Baleba
• [Asterisk-Users] Hardware Requirement for Asterisk PBX   Tarun Banka
• [Asterisk-Users] TE401P driver warning   Derek Barber
• [Asterisk-Users] Ordering ISDN PRI   Derek Barber
• [Asterisk-Users] Voicemail2 patches   Brad Bergman
• [Asterisk-Users] 0 out of voicemail to different secretaries   Brad Bergman
• [Asterisk-Users] 0 out of voicemail to different secretaries   Brad Bergman
• [Asterisk-Users] 0 out of voicemail to different secretaries   Brad Bergman
• [Asterisk-Users] indications.conf settings for Belgium   Emmanuel Bergmans
• [Asterisk-Users] problem with Wildcard 100XP and hangup signal   Emmanuel Bergmans
• [Asterisk-Users] problem with Wildcard 100XP and hangup signal   Emmanuel Bergmans
• [Asterisk-Users] FXO mode   Emmanuel Bergmans
• [Asterisk-Users] FXO mode   Emmanuel Bergmans
• [Asterisk-Users] Dialed Number Identification in analog hunt group   Stephen R. Besch
• [Asterisk-Users] Dialed Number Identification in analog hunt group   Stephen R. Besch
• [Asterisk-Users] zaptel sync   Michiel Betel
• [Asterisk-Users] chan_capi compile errors with latest CVS   Michiel Betel
• [Asterisk-Users] chan_zap.c zt_rec: Unknown error 500   Michiel Betel
• [Asterisk-Users] Why doesnt anyone reply me ?   Michiel Betel
• [Asterisk-Users] chan_zap.c zt_rec: Unknown error 500   Michiel Betel
• [Asterisk-Users] chan_zap.c zt_rec: Unknown error 500   Michiel Betel
• [Asterisk-Users] DBSaveTree & DBLoadTree   Michiel Betel
• [Asterisk-Users] DBSaveTree & DBLoadTree   Michiel Betel
• [Asterisk-Users] DBSaveTree & DBLoadTree   Michiel Betel
• [Asterisk-Users] Multiple E1 configuration question   Michael Bielicki
• [Asterisk-Users] problem with Wildcard 100XP and hangup signal   Michael Bielicki
• [Asterisk-Users] New-ish list of hardware phone vendors   Michael Bielicki
• [Asterisk-Users] asterisk-oh323 v0.5.5   Michael Bielicki
• [Asterisk-Users] Using OH323 and Gatekeeper   John Bigelow
• [Asterisk-Users] Music on hold - multiple formats   Sam Bingner
• [Asterisk-Users] CVS troubles   Felix Bizaoui
• [Asterisk-Users] Working with FWD, IPTel, SIPPhone?   Ian Blenke
• [Asterisk-Users] Asterisk SIP calls failing - not a proxy? What of RTP codec transcoding?   Ian Blenke
• [Asterisk-Users] Receiving iaxtel calls   Ian Blenke
• [Asterisk-Users] Asterisk SIP calls failing - not a proxy? What of RTP codec transcoding?   Ian Blenke
• [Asterisk-Users] Grandstream, SIP encryption   Ian Blenke
• [Asterisk-Users] IAX <> IAX trunking... DP cache?   Ian Blenke
• [Asterisk-Users] IAX <> IAX trunking... DP cache?   Ian Blenke
• [Asterisk-Users] IAX <> IAX trunking... DP cache?   Ian Blenke
• [Asterisk-Users] IAX <> IAX trunking... DP cache?   Ian Blenke
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   Ian Blenke
• [Asterisk-Users] Working example of "switch"?   Ian Blenke
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   Ian Blenke
• [Asterisk-Users] Grandstream firmware update DMTF Payload Type   Ian Blenke
• [Asterisk-Users] Grandstream firmware update DMTF Payload Type   Ian Blenke
• [Asterisk-Users] limiting out going calls to a maximum duration   Robert Boardman
• [Asterisk-Users] OT: Grandstream power supplies..   Robert Boardman
• [Asterisk-Users] Re: Open source IP phone, maybe?   Leo Ann Boon
• [Asterisk-Users] RTP header compression?   Leo Ann Boon
• [Asterisk-Users] PRI CallerID problem   Steve Bourg
• [Asterisk-Users] Recomendations for an ISDN-PBX to use with asterisk   Oliver Brandt
• [Asterisk-Users] Installation Problem   Phillip Britt
• [Asterisk-Users] Installation Problem   Phillip Britt
• [Asterisk-Users] music on hold help   John Brown
• [Asterisk-Users] Voicemail cliping digits via sip   John Brown
• [Asterisk-Users] music on hold help   John Brown
• [Asterisk-Users] Voicemail cliping digits via sip   John Brown
• [Asterisk-Users] LAN switches with PoE? PoE phones?   John Brown
• [Asterisk-Users] Grandstream Budgetone   John Brown
• [Asterisk-Users] Monitor application temporary hack   John Brown
• [Asterisk-Users] Monitor application temporary hack   John Brown
• [Asterisk-Users] dumb x100p question   John Brown
• [Asterisk-Users] trying to make a X100P work   John Brown
• [Asterisk-Users] trying to make a X100P work   John Brown
• [Asterisk-Users] trying to make a X100P work   John Brown
• [Asterisk-Users] echo on the sip side   John Brown
• [Asterisk-Users] Is Asterisk ready for "real" use?   John Brown
• [Asterisk-Users] Strange happenings   John Brown
• [Asterisk-Users] Caller ID problem   John Brown
• Qwest CallerID question re:[Asterisk-Users] Caller ID problem   John Brown
• Qwest CallerID question re:[Asterisk-Users] Caller ID problem   John Brown
• [Asterisk-Users] zaptel compile problems   John Brown
• [Asterisk-Users] zaptel compile problems   John Brown
• [Asterisk-Users] Grandstream and CallerID not working   John Brown
• [Asterisk-Users] Grandstream and CallerID not working   John Brown
• [Asterisk-Users] DIAL via CLI missing   John Brown
• [Asterisk-Users] Grandstream and CallerID not working   John Brown
• [Asterisk-Users] DIAL via CLI missing   John Brown
• list police strike again Re: [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   John Brown
• [Asterisk-Users] fatal embrace control in menus ?   John Brown
• [Asterisk-Users] fatal embrace control in menus ?   John Brown
• [Asterisk-Users] line numbering and gosub   John Brown
• [Asterisk-Users] GS geek info   John Brown
• [Asterisk-Users] GS on ebay...   John Brown
• [Asterisk-Users] Call Center RFP   Ray Burkholder
• [Asterisk-Users] LAN switches with PoE? PoE phones?   Ray Burkholder
• [Asterisk-Users] Warning message in /var/log/asterisk/messages   Ray Burkholder
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Ray Burkholder
• [Asterisk-Users] Is Asterisk ready for "real" use?   Ray Burkholder
• [Asterisk-Users] Intercom with Cisco SIP 796x phones?   Ray Burkholder
• [Asterisk-Users] SIP vs SCCP vs XML   Ray Burkholder
• [Asterisk-Users] SIP vs SCCP vs XML   Ray Burkholder
• [Asterisk-Users] SIP vs SCCP vs XML   Ray Burkholder
• [Asterisk-Users] SIP vs SCCP vs XML   Ray Burkholder
• [Asterisk-Users] Call transfer ATA186   CW_ASN
• [Asterisk-Users] Call transfer ATA186   CW_ASN
• [Asterisk-Users] Asterisk Outbound Calling Warning: Unable To Forward Voice   CW_ASN
• [Asterisk-Users] Asterisk Outbound Calling Warning: Unable To Forward Voice   CW_ASN
• [Asterisk-Users] PrePaid and IVR   CW_ASN
• [Asterisk-Users] ATA 186 & DynExtenDB (query extensions vía sql)   CW_ASN
• [Asterisk-Users] Re: [Asterisk-Users] ATA 186 & DynExtenDB (query extensions vía sql)   CW_ASN
• [Asterisk-Users] Re: [Asterisk-Users] Re: [Asterisk-Users] ATA 186 & DynExtenDB (query extensions vía sql)   CW_ASN
• [Asterisk-Users] Asterisks integration with pre-existing PBXs   Kim C. Callis
• [Asterisk-Users] Open G.729A codec   Kim C. Callis
• [Asterisk-Users] Fair comparison   Kim C. Callis
• [Asterisk-Users] Video conference apps/appliances?   Kim C. Callis
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   Brian Capouch
• [Asterisk-Users] Budgettone Newbie   Brian Capouch
• [Asterisk-Users] Leftover Budgettone issues   Brian Capouch
• [Asterisk-Users] How to determine line signalling?   Brian Capouch
• [Asterisk-Users] Workaround for BudgeTone "ringing in your ear?"   Brian Capouch
• [Asterisk-Users] "Double" transfer?   Brian Capouch
• [Asterisk-Users] BudgeTone NAT issues   Brian Capouch
• [Asterisk-Users] Call transfer ATA186   Brian Capouch
• [Asterisk-Users] VoIP dialtone?   Brian Capouch
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   Brian Capouch
• [Asterisk-Users] Re: ATAs   Brian Capouch
• [Asterisk-Users] Problems with TDM400P & X100P   Brian Capouch
• [Asterisk-Users] GS on ebay...   Brian Capouch
• [Asterisk-Users] X-Lite <-> Snom200   Jamie Carl
• [Asterisk-Users] X-Lite <-> Snom200   Jamie Carl
• [Asterisk-Users] SNOM200 firmware roll back!!   Jamie Carl
• [Asterisk-Users] SNOM200 firmware roll back!!   Jamie Carl
• [Asterisk-Users] Windows Messenger   Jamie Carl
• [Asterisk-Users] Get faxed you faxing faxer!   Jamie Carl
• [Asterisk-Users] Get faxed you faxing faxer!   Jamie Carl
• [Asterisk-Users] Get faxed you faxing faxer!   Jamie Carl
• [Asterisk-Users] How to Asterisk   Jamie Carl
• [Asterisk-Users] usrobotics modem and pstn   Jamie Carl
• [Asterisk-Users] Open G.729A codec   Jamie Carl
• [Asterisk-Users] CODEC & DTMF   Jamie Carl
• [Asterisk-Users] SIP Transfer   Jamie Carl
• [Asterisk-Users] no incoming packets & Sound: Recording overrun   Jamie Carl
• [Asterisk-Users] no incoming packets & Sound: Recording overrun   Jamie Carl
• [Asterisk-Users] Recomendations for an ISDN-PBX to use with asterisk   Jamie Carl
• [Asterisk-Users] SIP agent logging into queue?   Jamie Carl
• [Asterisk-Users] SIP QUESTION   Jamie Carl
• [Asterisk-Users] SIP QUESTION   Jamie Carl
• [Asterisk-Users] VIRUS ALERT   Jamie Carl
• [Asterisk-Users] VIRUS ALERT   Jamie Carl
• [Asterisk-Users] VIRUS ALERT   Jamie Carl
• [Asterisk-Users] VIRUS ALERT   Jamie Carl
• [Asterisk-Users] SIP vs SCCP vs XML   Jamie Carl
• [Asterisk-Users] SIP vs SCCP vs XML   Jamie Carl
• [Asterisk-Users] Dialed Number Identification in analog hunt group   Jamie Carl
• [Asterisk-Users] * behind ISDN pbx - Forwarding to extensions with in primary pbx   Justin Carlson
• [Asterisk-Users] IP phone recommendation   David Carr
• [Asterisk-Users] Provisioning CO lines   David Carr
• [Asterisk-Users] Provisioning CO lines   David Carr
• [Asterisk-Users] Provisioning CO lines   David Carr
• [Asterisk-Users] Syncronize Monitored Calls   David Carr
• [Asterisk-Users] E-mail (still version 1) is not being Delivered   Uriel Carrasquilla
• [Asterisk-Users] E-mail (version 1) is not being Delivered   Uriel Carrasquilla
• [Asterisk-Users] IP phone recommendation   Uriel Carrasquilla
• [Asterisk-Users] Cordless SIP phones   Uriel Carrasquilla
• [Asterisk-Users] Asterisk's configuration : Which signalling in France with an E1 ?   Nicolas Cartron
• [Asterisk-Users] Zaptel.conf & digium E100P   Nicolas Cartron
• [Asterisk-Users] SIP calls cause Segmentation Fault   Dave Alan Caruana
• [Asterisk-Users] SIP calls cause Segmentation Fault   Dave Alan Caruana
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   Dave Alan Caruana
• [Asterisk-Users] SIP calls cause segmentation fault   Dave Alan Caruana
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   Dave Alan Caruana
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   Dave Alan Caruana
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   Dave Alan Caruana
• [Asterisk-Users] Newbie Issue   Dave Alan Caruana
• [Asterisk-Users] OT: Grandstream power supplies..   Dave Alan Caruana
• [Asterisk-Users] Why doesnt anyone reply me ?   Dave Alan Caruana
• [Asterisk-Users] Registering SIP with FWD and ICONNECTHERE   Terence Chan
• [Asterisk-Users] Running Asterisk behind NAT?   Terence Chan
• [Asterisk-Users] Running Asterisk behind NAT?   Terence Chan
• [Asterisk-Users] Using H323 and Gatekeeper   Paul Cheng
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Paul Cheng
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Paul Cheng
• [Asterisk-Users] Inbound SIP DTMF detection   Paul Cheng
• [Asterisk-Users] SIP NAT question   Paul Cheng
• Fwd: FW: [Asterisk-Users] SIP NAT question   Paul Cheng
• [Asterisk-Users] make: warning: Clock skew detected. Your build may be incomplete.   Paul Cheng
• [Asterisk-Users] Voicemail2 patches   Paul Cheng
• [Asterisk-Users] App Directory issues-again?   Paul Cheng
• [Asterisk-Users] ENUM, iax,iax2 and h323?   Paul Cheng
• [Asterisk-Users] H323 and SIP   Lubomir Christov
• [Asterisk-Users] PrePaid and IVR   Lubomir Christov
• [Asterisk-Users] PrePaid and IVR   Lubomir Christov
• [Asterisk-Users] Restricting concurrent SIP calls   Lubomir Christov
• [Asterisk-Users] sample configs   Lubomir Christov
• [Asterisk-Users] h323 and cvs one way audio   Kelvin Chua
• [Asterisk-Users] h323 and cvs one way audio   Kelvin Chua
• [Asterisk-Users] Chan_h323 one way audio   Kelvin Chua
• [Asterisk-Users] Chan_h323 one way audio   Kelvin Chua
• [Asterisk-Users] LAN switches with PoE? PoE phones?   Mike Ciholas
• [Asterisk-Users] Re: LAN switches with PoE? PoE phones?   Mike Ciholas
• [Asterisk-Users] Is Asterisk ready for "real" use?   Mike Ciholas
• [Asterisk-Users] Is Asterisk ready for "real" use?   Mike Ciholas
• [Asterisk-Users] Is Asterisk ready for "real" use?   Mike Ciholas
• [Asterisk-Users] VoIP dialtone?   Mike Ciholas
• [Asterisk-Users] VoIP dialtone?   Mike Ciholas
• [Asterisk-Users] VoIP dialtone?   Mike Ciholas
• [Asterisk-Users] VoIP dialtone?   Mike Ciholas
• [Asterisk-Users] VoIP dialtone?   Mike Ciholas
• [Asterisk-Users] VoIP dialtone?   Mike Ciholas
• [Asterisk-Users] Provisioning CO lines   Mike Ciholas
• [Asterisk-Users] Provisioning CO lines   Mike Ciholas
• [Asterisk-Users] Dial in modem speeds over VoIP?   Mike Ciholas
• [Asterisk-Users] T1 to T1 on asterisk?   Mike Ciholas
• [Asterisk-Users] T1 to T1 on asterisk?   Mike Ciholas
• [Asterisk-Users] T1 to T1 on asterisk?   Mike Ciholas
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Mike Ciholas
• [Asterisk-Users] QOH (quiet on hold)?   Mike Ciholas
• [Asterisk-Users] Difference between Cisco 7940/7940G, 7960/7960G   Mike Ciholas
• [Asterisk-Users] Queue and Agents in CVS   John Congdon
• [Asterisk-Users] Error loading latest CVS   John Congdon
• [Asterisk-Users] Error loading latest CVS   John Congdon
• [Asterisk-Users] Upgrading Queue App   John Congdon
• [Asterisk-Users] Data calls through *   John Congdon
• [Asterisk-Users] Warning from chan_zap ring requested   John Congdon
• Fwd: [Asterisk-Users] Data calls through *   John Congdon
• [Asterisk-Users] Warning from chan_zap ring requested   John Congdon
• [Asterisk-Users] Asterisk stops responding   John Congdon
• [Asterisk-Users] Asterisk stops responding   John Congdon
• [Asterisk-Users] Asterisk stops responding   John Congdon
• [Asterisk-Users] Data calls through *   John Congdon
• [Asterisk-Users] Which GS IP product to buy??   Joe Cooke
• [Asterisk-Users] SIP with an iptables fiewall   Dave Cotton
• [Asterisk-Users] Asterisk + SER   Dave Cotton
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Dave Cotton
• [Asterisk-Users] Newbie just starting out with *   Dave Cotton
• [Asterisk-Users] Push to Talk   Dave Cotton
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK? [now] outside the US?   Dave Cotton
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID inUK? [now] outside the US?   Dave Cotton
• [Asterisk-Users] Does Wildcard x100p support Caller ID outside the US?   Dave Cotton
• [Asterisk-Users] Does Wildcard x100p support Caller ID outside the US? (fwd)   Dave Cotton
• [Asterisk-Users] How to Asterisk   Dave Cotton
• [Asterisk-Users] OT: Grandstream power supplies..   Dave Cotton
• [Asterisk-Users] Sip and One Way Audio   Dave Cotton
• [Asterisk-Users] Gnophone   Dave Cotton
• [Asterisk-Users] Sip and One Way Audio   Dave Cotton
• [Asterisk-Users] Sip and One Way Audio   Dave Cotton
• [Asterisk-Users] Weird DTMF issue   Dave Cotton
• [Asterisk-Users] FXO mode   Dave Cotton
• [Asterisk-Users] FXO mode   Dave Cotton
• [Asterisk-Users] FXO mode   Dave Cotton
• [Asterisk-Users] FXO mode   Dave Cotton
• [Asterisk-Users] Running Asterisk behind NAT?   Dave Cotton
• [Asterisk-Users] FXO mode   Dave Cotton
• [Asterisk-Users] Running Asterisk behind NAT?   Dave Cotton
• [Asterisk-Users] Problems to install X100P   Dave Cotton
• [Asterisk-Users] Problems to install X100P   Dave Cotton
• [Asterisk-Users] FXO mode   Dave Cotton
• [Asterisk-Users] Grandstream Budgetone   Dave Cotton
• [Asterisk-Users] SIP agent logging into queue?   Dave Cotton
• [Asterisk-Users] PrePaid and IVR   Dave Cotton
• [Asterisk-Users] [OT] Virus propagation by asterisk user member.   Dave Cotton
• [Asterisk-Users] [OT] Virus propagation by asterisk user member.   Dave Cotton
• [Asterisk-Users] Strange happenings   Dave Cotton
• [Asterisk-Users] Strange happenings   Dave Cotton
• [Asterisk-Users] Re: Some questions about Asterisk and reliability   Dave Cotton
• [Asterisk-Users] Intresting.. hrm   Dave Cotton
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Dave Cotton
• [Asterisk-Users] Non Asterisk - Apology to the list   Dave Cotton
• [Asterisk-Users] Grandstream firmware update DMTF Payload Type   Dave Cotton
• [Asterisk-Users] IAXtel + NAT   Dave Cotton
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Dave Cotton
• [Asterisk-Users] Asterisk -> Intertex   Dave Cotton
• [Asterisk-Users] Installation Problem   Dave Cotton
• [Asterisk-Users] Voicemail2 - auto fill the dialing extension?   Steve Creel
• [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank   Steve Creel
• [Asterisk-Users] Forward but wait for acknowledgement   Steve Creel
• [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c) or unixODBC (cdr_unixodbc.c) ?   Steven Critchfield
• [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c)or unixODBC (cdr_unixodbc.c) ?   Steven Critchfield
• [Asterisk-Users] Asterisk agi interface leaveszombie processes?   Steven Critchfield
• [Asterisk-Users] Zhone Zplex 10 units   Steven Critchfield
• [Asterisk-Users] Zhone Zplex 10 units   Steven Critchfield
• [Asterisk-Users] Zhone Zplex 10 units   Steven Critchfield
• [Asterisk-Users] Why are FXO so expensive?   Steven Critchfield
• [Asterisk-Users] Ansering machine/voicemail detect?   Steven Critchfield
• [Asterisk-Users] Call routing question   Steven Critchfield
• [Asterisk-Users] Fax Handled   Steven Critchfield
• [Asterisk-Users] list proposal   Steven Critchfield
• [Asterisk-Users] list proposal   Steven Critchfield
• [Asterisk-Users] X100P Ringing/Answering   Steven Critchfield
• [Asterisk-Users] FWD SIP phone format=2, FWD call format=4, why?   Steven Critchfield
• [Asterisk-Users] Asterisk and AT&T 964 phones...   Steven Critchfield
• [Asterisk-Users] Extension and phone management best practices??   Steven Critchfield
• [Asterisk-Users] Extension and phone management best practices??   Steven Critchfield
• [Asterisk-Users] Can't compile cdr_mysql   Steven Critchfield
• [Asterisk-Users] Extension and phone management bestpractices??   Steven Critchfield
• [Asterisk-Users] Extension and phone management bestpractices??   Steven Critchfield
• [Asterisk-Users] Problems to install X100P   Steven Critchfield
• [Asterisk-Users] Extension and phone management bestpractices??   Steven Critchfield
• [Asterisk-Users] Can I runAsterisk remotely from telnet session?   Steven Critchfield
• [Asterisk-Users] LAN switches with PoE? PoE phones?   Steven Critchfield
• [Asterisk-Users] Re: LAN switches with PoE? PoE phones?   Steven Critchfield
• [Asterisk-Users] Brooktrout PRI-ISA48 card... info..   Steven Critchfield
• [Asterisk-Users] [OT] Virus propagation by asterisk user member.   Steven Critchfield
• [Asterisk-Users] Brooktrout PRI-ISA48 card... info..   Steven Critchfield
• [Asterisk-Users] [OT] Virus propagation by asterisk user member.   Steven Critchfield
• [Asterisk-Users] Re: Open source IP phone, maybe?   Steven Critchfield
• [Asterisk-Users] trying to make a X100P work   Steven Critchfield
• [Asterisk-Users] trying to make a X100P work   Steven Critchfield
• [Asterisk-Users] Limit Number of user in Conference   Steven Critchfield
• [Asterisk-Users] weird error message with zaptel   Steven Critchfield
• [Asterisk-Users] [OT] Virus propagation by asterisk user member.   Steven Critchfield
• [Asterisk-Users] VIRUS ALERT   Steven Critchfield
• [Asterisk-Users] VIRUS ALERT   Steven Critchfield
• [Asterisk-Users] Limit Number of user in Conference   Steven Critchfield
• [Asterisk-Users] Configurable auto forward in Asterisk   Steven Critchfield
• [Asterisk-Users] Conference + time limit   Steven Critchfield
• [Asterisk-Users] Sending dtmf over an ougoing call from asterisk   Steven Critchfield
• [Asterisk-Users] Provisioning CO lines   Steven Critchfield
• [Asterisk-Users] Conference + time limit   Steven Critchfield
• [Asterisk-Users] Provisioning CO lines   Steven Critchfield
• [Asterisk-Users] Asterisk + SNOM + Pound and star keys   Steven Critchfield
• [Asterisk-Users] Conference + time limit   Steven Critchfield
• [Asterisk-Users] Provisioning CO lines   Steven Critchfield
• [Asterisk-Users] line numbering and gosub   Steven Critchfield
• [Asterisk-Users] unsubscribe   Steven Critchfield
• [Asterisk-Users] Configuration Adtran TA 750   Steven Critchfield
• [Asterisk-Users] Data calls through *   Steven Critchfield
• [Asterisk-Users] Incomming call issue   Steven Critchfield
• [Asterisk-Users] Newbie IVR question   Steven Critchfield
• [Asterisk-Users] Newbie IVR question   Steven Critchfield
• [Asterisk-Users] Newbie IVR question   Steven Critchfield
• [Asterisk-Users] X-Lite Build 1059 problems   Ed Dack
• [Asterisk-Users] Voicemail group, adding to the vm when forwarding   Yifang Dai
• [Asterisk-Users] 'System' application exit with error even if it performs the job as expected   Dan
• [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected   Dan
• [Asterisk-Users] 'System' application exit with error evenifit performs the job as expected   Dan
• [Asterisk-Users] Monitor app   Dan
• [Asterisk-Users] phone rings while already on a call   Dan
• [Asterisk-Users] 'System' application exit with error evenifitperforms the job as expected   Dan
• [Asterisk-Users] Musiconhold interrupted sound   Dan
• [Asterisk-Users] Windows IAX soft phone   Dan
• [Asterisk-Users] Windows IAX soft phone   Dan
• [Asterisk-Users] iax.conf / Registration rejected   Dan
• [Asterisk-Users] X100P and Caller ID (again and again...)   Dan
• [Asterisk-Users] X100P and Caller ID (again and again...)   Dan
• [Asterisk-Users] X100P CallerID issue solved for my PSTN connection   Dan
• [Asterisk-Users] Windows IAX soft phone   Dan
• [Asterisk-Users] Asterisk launch on boot   Dan
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   Dan
• [Asterisk-Users] X100P delivery   Dan
• [Asterisk-Users] Minimum system requirement for ....   Dan
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID inUK? [now] outside the US?   Dan
• [Asterisk-Users] CallerID, DECT phones and ATA   Dan
• [Asterisk-Users] CallerID, DECT phones and ATA   Dan
• [Asterisk-Users] Play Music On Hold just for a fixed period of time   Dan
• [Asterisk-Users] Play Music On Hold just for a fixed period of time   Dan
• [Asterisk-Users] CallerID, DECT phones and ATA   Dan
• [Asterisk-Users] IAX protocol description   Dan
• [Asterisk-Users] callerid, british and french type DECTs   Dan
• [Asterisk-Users] This is how to set ATA186 for different standards of CallerID format   Dan
• [Asterisk-Users] callerid (Bell type) in Europe   Dan
• [Asterisk-Users] Something very strange regarding callerid   Dan
• [Asterisk-Users] InternetPhoneWiazard   Dan
• [Asterisk-Users] New-ish list of hardware phone vendors   Dan
• [Asterisk-Users] Open G.729A codec   Dan
• [Asterisk-Users] Open G.729A codec   Dan
• [Asterisk-Users] How can I know if a user is busy or not connected?   Dan
• [Asterisk-Users] '#' doesn't work for me   Dan
• [Asterisk-Users] Virtual extension as local modem   Dan
• [Asterisk-Users] '#' doesn't work for me   Dan
• [Asterisk-Users] '#' doesn't work for me   Dan
• [Asterisk-Users] '#' doesn't work for me   Dan
• [Asterisk-Users] How can I know if a user is busy or not connected?   Dan
• [Asterisk-Users] '#' doesn't work for me   Dan
• [Asterisk-Users] How can I know if a user is busy or not connected?   Dan
• [Asterisk-Users] Cordless SIP phones   Dan
• [Asterisk-Users] Cordless SIP phones   Dan
• [Asterisk-Users] Cordless SIP phones   Dan
• [Asterisk-Users] Call transfer ATA186   Dan
• [Asterisk-Users] Cisco 7940 7960   Dan
• [Asterisk-Users] Voicemail2 vs. Voicemail   Dan
• [Asterisk-Users] Call transfer ATA186   Dan
• [Asterisk-Users] Call transfer ATA186   Dan
• [Asterisk-Users] Cisco 7940 7960   Dan
• [Asterisk-Users] Configurable auto forward in Asterisk   Dan
• [Asterisk-Users] Configurable auto forward in Asterisk   Dan
• [Asterisk-Users] sox and wav to gsm conversion quality issue   Dan
• [Asterisk-Users] sox and wav to gsm conversion quality issue   Dan
• [Asterisk-Users] sox and wav to gsm conversion quality issue   Dan
• [Asterisk-Users] Caller ID problem   Dan
• [Asterisk-Users] callerid, callwaiting callerid, Asterisk and ATA   Dan
• [Asterisk-Users] There is any cache for sound files?   Dan
• [Asterisk-Users] There is any cache for sound files?   Dan
• [Asterisk-Users] X100P disconnecting without any reason   Dan
• [Asterisk-Users] Sound files for internal functions in another place??   Dan
• [Asterisk-Users] Sound files for internal functions in another place??   Dan
• [Asterisk-Users] Decent DECT cordless compatible with Asterisk/ATA?   Dan
• [Asterisk-Users] * server based Phonebook   Dan
• [Asterisk-Users] * server based Phonebook   Dan
• [Asterisk-Users] * server based Phonebook   Dan
• [Asterisk-Users] * server based Phonebook   Dan
• [Asterisk-Users] Asterisk internal database access   Dan
• [Asterisk-Users] Asterisk internal database access   Dan
• [Asterisk-Users] SIP and ECHO   Dan
• [Asterisk-Users] X100P not hanging up.   Dan
• [Asterisk-Users] chan_capi ptp mode   Marian Danisek
• [Asterisk-Users] avm fritz pci   Marian Danisek
• [Asterisk-Users] How to Asterisk   Dave
• [Asterisk-Users] SIP clients not sending audio   Adam Donnison
• [Asterisk-Users] SIP clients not sending audio   Adam Donnison
• [Asterisk-Users] SIP calls cause segmentation fault   Adam Donnison
• [Asterisk-Users] Get faxed you faxing faxer!   Adam Donnison
• [Asterisk-Users] Newbie Question / ISDN   Peter Eckhardt
• [Asterisk-Users] Newbie Question / ISDN   Peter Eckhardt
• [Asterisk-Users] Crash using alsa   Peter Eckhardt
• [Asterisk-Users] Slowly get it ... Hardware   Peter Eckhardt
• [Asterisk-Users] Newbie IVR question   Josh Edwards
• [Asterisk-Users] Newbie IVR question   Josh Edwards
• [Asterisk-Users] Newbie IVR question   Josh Edwards
• [Asterisk-Users] Asterisk introductory talk: Portland, OR USA   Steve Edwards
• [Asterisk-Users] Asterisk Newbie ...   Fabia
• [Asterisk-Users] Can't compile cdr_mysql   Jerk Face
• [Asterisk-Users] Can't compile cdr_mysql   Jerk Face
• [Asterisk-Users] Can't compile cdr_mysql   Jerk Face
• [Asterisk-Users] Can't compile cdr_mysql   Jerk Face
• [Asterisk-Users] Problem with latest cdr Makefile???   Jerk Face
• [Asterisk-Users] Which version of MySQL are you running?   Jerk Face
• [Asterisk-Users] Intermittant IAX Call Failures   Matthew Farley
• [Asterisk-Users] X100P Ringing/Answering   Mark Farver
• [Asterisk-Users] CDR-Event on AstManager   Dan Fernandez
• [Asterisk-Users] problem with manager: Response error, Missing action in request   Dan Fernandez
• [Asterisk-Users] x100P: Ring/off-hook in strange state 6 on channel1   Dan Fernandez
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   Bruce Ferrell
• [Asterisk-Users] Semi-newbie question "Softswitch" and Asterisk - Is there a difference?   Bruce Ferrell
• [Asterisk-Users] Asterisk, quicknet and codecs (G729, 7231 Question)   Bruce Ferrell
• [Asterisk-Users] Hardware question   Bruce Ferrell
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Bruce Ferrell
• [Asterisk-Users] VoIP dialtone?   Bruce Ferrell
• [Asterisk-Users] VoIP dialtone?   Bruce Ferrell
• [Asterisk-Users] VoIP dialtone?   Bruce Ferrell
• [Asterisk-Users] Game time is over gang   Bruce Ferrell
• [Asterisk-Users] h323 compile error   Sean Figgins
• [Asterisk-Users] Voicemail2 - auto fill the dialing extension?   Sean Figgins
• [Asterisk-Users] chan_h323.c   Sean Figgins
• [Asterisk-Users] chan_h323.c   Sean Figgins
• [Asterisk-Users] Ring when leaveing queue?   Sebastian Filzek
• [Asterisk-Users] SIP agent logging into queue?   Sebastian Filzek
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Stefano Finetti
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Stefano Finetti
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Stefano Finetti
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Stefano Finetti
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Stefano Finetti
• [Asterisk-Users] Asterisk launch on boot   Stefano Finetti
• [Asterisk-Users] Play Music On Hold just for a fixed period of time   Stefano Finetti
• [Asterisk-Users] Running Asterisk behind NAT?   Jim Flagg
• [Asterisk-Users] Windows IAX soft phone   William Flanagan
• [Asterisk-Users] Windows IAX soft phone   William Flanagan
• [Asterisk-Users] Windows IAX soft phone   William Flanagan
• [Asterisk-Users] Test   Jorge Cisneros Flores
• [Asterisk-Users] SIP QUESTION   Jorge Cisneros Flores
• [Asterisk-Users] Mysql CDR   Chee Foong
• [Asterisk-Users] SendDtmf   Chee Foong
• [Asterisk-Users] SendDtmf   Chee Foong
• [Asterisk-Users] SendDtmf + chan_h323   Chee Foong
• [Asterisk-Users] chan_OH323   Chee Foong
• [Asterisk-Users] WipeOut - gateway access with pin solution   Chee Foong
• [Asterisk-Users] chan_oh323 + dtmf   Chee Foong
• [Asterisk-Users] chan_oh323 + dtmf   Chee Foong
• [Asterisk-Users] chan_oh323 + dtmf   Chee Foong
• [Asterisk-Users] H323 + DTMF detection   Chee Foong
• [Asterisk-Users] Conference + E100P + H323   Chee Foong
• [Asterisk-Users] Conference + E100P + H323   Chee Foong
• [Asterisk-Users] Conference + E100P + H323   Chee Foong
• [Asterisk-Users] reload   Chee Foong
• [Asterisk-Users] Conference Number + CDR   Chee Foong
• [Asterisk-Users] Limit Number of user in Conference   Chee Foong
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Chee Foong
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Chee Foong
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Chee Foong
• [Asterisk-Users] Limit Number of user in Conference   Chee Foong
• [Asterisk-Users] Limit Number of user in Conference   Chee Foong
• [Asterisk-Users] Conference + time limit   Chee Foong
• [Asterisk-Users] Conference + time limit   Chee Foong
• [Asterisk-Users] conference authorization   Chee Foong
• [Asterisk-Users] Chan_h323 + amaflag   Chee Foong
• [Asterisk-Users] Asterisk and authentication   Tom Forbes
• [Asterisk-Users] Asterisk Outbound Calling Warning: Unable To Forward Voice   Lee Forkenbrock
• [Asterisk-Users] Cisco 7940 SIP   Lee Forkenbrock
• [Asterisk-Users] Call Transfer problem   John Fortman
• [Asterisk-Users] Call transfer problem   John Fortman
• [Asterisk-Users] ast_channel_alloc() losing pvt struct   John Fortman
• [Asterisk-Users] chan_h323.c   John Fortman
• [Asterisk-Users] # Transfer context problem   John Fortman
• [Asterisk-Users] trying to make a X100P work   John Fortman
• [Asterisk-Users] chan_h323.c   John Fortman
• [Asterisk-Users] Question on setting up MeetMe conference bridge   John Fortman
• [Asterisk-Users] GS on ebay...   John Fortman
• [Asterisk-Users] GS on ebay...   John Fortman
• [Asterisk-Users] Problem with SIP Native Bridging and UPnP   Layton Freeman
• [Asterisk-Users] Monitor app   Jim Friedeck
• [Asterisk-Users] AgentCallbackLogin   Jim Friedeck
• [Asterisk-Users] AgentCallbackLogin   Jim Friedeck
• [Asterisk-Users] AgentCallbackLogin   Jim Friedeck
• [Asterisk-Users] AgentCallbackLogin   Jim Friedeck
• [Asterisk-Users] AgentCallbackLogin   Jim Friedeck
• [Asterisk-Users] AgentCallbackLogin   Jim Friedeck
• [Asterisk-Users] segfaults with queue   Jim Friedeck
• [Asterisk-Users] segfaults with queue   Jim Friedeck
• [Asterisk-Users] Killing runaway PBX   Jim Friedeck
• [Asterisk-Users] Killing runaway PBX   Jim Friedeck
• [Asterisk-Users] Killing runaway PBX   Jim Friedeck
• [Asterisk-Users] Killing runaway PBX   Jim Friedeck
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Jim Friedeck
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Jim Friedeck
• [Asterisk-Users] Ring while on phone   Jim Friedeck
• [Asterisk-Users] Manager interface & Event:Leave   Jim Friedeck
• [Asterisk-Users] RTP / SIP routing issues   Matthew M. Gamble
• [Asterisk-Users] Call routing question   Matthew M. Gamble
• [Asterisk-Users] SCO/Linux concerns   Gary Gapinski
• [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)   Ing. Angel Gomez Garcia
• [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)   Ing. Angel Gomez Garcia
• [Asterisk-Users] Segmentation fault   Gary
• [Asterisk-Users] Get faxed you faxing faxer!   Gary
• [Asterisk-Users] Get faxed you faxing faxer!   Gary
• [Asterisk-Users] ATA-186 locking: implausible unlock method   Gary
• [Asterisk-Users] T1 to T1 on asterisk?   Gary
• [Asterisk-Users] Video conference apps/appliances?   Gary
• [Asterisk-Users] Newbie - setup help   Gavin
• [Asterisk-Users] SIP and MD5 logins.   Deon George
• [Asterisk-Users] Free World Dialup.   Deon George
• [Asterisk-Users] iax.conf order important?   Deon George
• [Asterisk-Users] ip phones and intercom/paging   Jerry Gibson
• [Asterisk-Users] sample configs   Andrew Gillham
• [Asterisk-Users] Asterisk and Cisco 7960   Andrew Gillham
• [Asterisk-Users] AgentCallbackLogin   James Golovich
• [Asterisk-Users] h extension seems to wipe variables?   James Golovich
• [Asterisk-Users] Fax Handled   James Golovich
• [Asterisk-Users] AGI Channel Status   James Golovich
• [Asterisk-Users] Festival and Asterisk   James Golovich
• [Asterisk-Users] Seting up TDM40B   Eduardo Goncalves
• [Asterisk-Users] Seting up TDM40B   Eduardo Goncalves
• [Asterisk-Users] Seting up TDM40B   Eduardo Goncalves
• [Asterisk-Users] Seting up TDM40B   Eduardo Goncalves
• [Asterisk-Users] Seting up TDM40B   Eduardo Goncalves
• [Asterisk-Users] Seting up TDM40B   Eduardo Goncalves
• [Asterisk-Users] Fax Handled   Eduardo Goncalves
• [Asterisk-Users] Fax Handled   Eduardo Goncalves
• [Asterisk-Users] Fax Handled   Eduardo Goncalves
• [Asterisk-Users] Fax Handled   Eduardo Goncalves
• [Asterisk-Users] Fax Handled   Eduardo Goncalves
• [Asterisk-Users] Fax Handled   Eduardo Goncalves
• [Asterisk-Users] Fax Handled   Eduardo Goncalves
• [Asterisk-Users] Fax Handled   Eduardo Goncalves
• [Asterisk-Users] VAD (silence suppression) on Asterisk   Eduardo Goncalves
• [Asterisk-Users] VAD (silence suppression) on Asterisk   Eduardo Goncalves
• [Asterisk-Users] Accountcode and cdr-csv   Eduardo Goncalves
• [Asterisk-Users] How to setup a VOIP Service provider if your * is behind a NAT   Lee Goodman
• [Asterisk-Users] MWI bug ?   Lee Goodman
• [Asterisk-Users] MWI bug ?   Lee Goodman
• [Asterisk-Users] MWI bug ?   Lee Goodman
• Fw: [Asterisk-Users] Fax Handled   Lee Goodman
• [Asterisk-Users] Weird DTMF issue   Lee Goodman
• [Asterisk-Users] Weird DTMF issue   Lee Goodman
• [Asterisk-Users] Weird DTMF issue   Lee Goodman
• [Asterisk-Users] MOH with SIP   Lee Goodman
• [Asterisk-Users] Cisco 7940 7960   Lee Goodman
• [Asterisk-Users] echo on the sip side   Lee Goodman
• [Asterisk-Users] VAD (silence suppression) on Asterisk   Lee Goodman
• [Asterisk-Users] VAD (silence suppression) on Asterisk   Lee Goodman
• [Asterisk-Users] IAX to zaptel echo   Lee Goodman
• [Asterisk-Users] VAD (silence suppression) on Asterisk   Lee Goodman
• [Asterisk-Users] Question on setting up MeetMe conference bridge   Lee Goodman
• [Asterisk-Users] SIP vs SCCP vs XML   Lee Goodman
• [Asterisk-Users] Intercom with Cisco SIP 796x phones?   Lee Goodman
• [Asterisk-Users] Problem starting Asterisk after abnormal shutdown   Lee Goodman
• [Asterisk-Users] Ansering machine/voicemail detect?   Adam Goryachev
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Adam Goryachev
• [Asterisk-Users] New gastman clone + what else?   Adam Goryachev
• [Asterisk-Users] Can't compile cdr_mysql   Adam Goryachev
• [Asterisk-Users] Is Asterisk ready for "real" use?   Adam Goryachev
• [Asterisk-Users] IAXtel + NAT   Adam Goryachev
• [Asterisk-Users] I4L CallerID not working   Adam Goryachev
• [Asterisk-Users] IAXtel + NAT   Adam Goryachev
• [Asterisk-Users] preventing timeout while transferring   Manoj K Gupta
• [Asterisk-Users] CODEC & DTMF   Manoj K Gupta
• [Asterisk-Users] cdr_mysql   Manoj K Gupta
• [Asterisk-Users] Sending dtmf over an ougoing call from asterisk   Manoj K Gupta
• AW: [Asterisk-Users] CDR-Event on AstManager   Thomas Haeger
• AW: [Asterisk-Users] ADSI Programs   Thomas Haeger
• [Asterisk-Users] additional digit in front of the dialed extenesion by outgoing pri/E1 call   Thomas Haeger
• AW: [Asterisk-Users] additional digit in front of the dialedextenesion by outgoing pri/E1 call   Thomas Haeger
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   Steve Haehnichen
• [Asterisk-Users] New SIP Phone   Steve Haehnichen
• [Asterisk-Users] New SIP Phone   Steve Haehnichen
• [Asterisk-Users] BRI newbie queries.   Steve Haehnichen
• [Asterisk-Users] New SIP Phone   Steve Haehnichen
• [Asterisk-Users] Feedback for Asterisk Handbook?   Steve Haehnichen
• [Asterisk-Users] Grandstream Budgetone Defective Units   Steve Haehnichen
• [Asterisk-Users] Grandstream firmware update. {HTTP error}   Steve Haehnichen
• [Asterisk-Users] Question About BRI Cards   Steve Haehnichen
• [Asterisk-Users] Looking for pricing on complete setup...   Chris Hale
• [Asterisk-Users] Asterisk and AT&T 964 phones...   Chris Hale
• [Asterisk-Users] Codecs   Tais M. Hansen
• [Asterisk-Users] Congestion   Tais M. Hansen
• [Asterisk-Users] Pops   Tais M. Hansen
• [Asterisk-Users] Pops   Tais M. Hansen
• [Asterisk-Users] cdr_mysql   Tais M. Hansen
• [Asterisk-Users] cdr_mysql   Tais M. Hansen
• [Asterisk-Users] cdr_mysql   Tais M. Hansen
• [Asterisk-Users] cdr_mysql   Tais M. Hansen
• [Asterisk-Users] cdr_mysql   Tais M. Hansen
• [Asterisk-Users] cdr_mysql   Tais M. Hansen
• [Asterisk-Users] cdr_mysql   Tais M. Hansen
• [Asterisk-Users] Pops   Tais M. Hansen
• [Asterisk-Users] cdr_mysql   Tais M. Hansen
• [Asterisk-Users] Pops   Tais M. Hansen
• [Asterisk-Users] (no subject)   Matthew Hardeman
• [Asterisk-Users] Open G.729A codec   Matthew Hardeman
• [Asterisk-Users] Open G.729A codec   Matthew Hardeman
• [Asterisk-Users] Open G.729A codec   Matthew Hardeman
• [Asterisk-Users] IP phone recommendation   Matthew Hardeman
• [Asterisk-Users] Open G.729A codec   Matthew Hardeman
• [Asterisk-Users] Bridged trunks stuck off hook.   John Harragin
• [Asterisk-Users] Setting a minimum 'on-hook' interval?   John Harragin
• [Asterisk-Users] Syncronize Monitored Calls   David Harris
• [Asterisk-Users] Asterisk stops responding   David Harris
• [Asterisk-Users] RE: Asterisk stops responding   David Harris
• [Asterisk-Users] RE: Asterisk stops responding   David Harris
• [Asterisk-Users] RE: Asterisk stops responding   David Harris
• [Asterisk-Users] RE: Asterisk stops responding   David Harris
• [Asterisk-Users] Restricting concurrent SIP calls   David Harris
• [Asterisk-Users] Speex & openh323   Adam Hart
• [Asterisk-Users] Speex & openh323   Adam Hart
• [Asterisk-Users] Fix for Redhat 9 zombie AGI processes   Fredrik Hedberg
• [Asterisk-Users] SIP Lines   Fredrik Hedberg
• [Asterisk-Users] IAX/SIP (Was: Windows IAX soft phone)   Fredrik Hedberg
• [Asterisk-Users] Sip and One Way Audio   Fredrik Hedberg
• [Asterisk-Users] Call transfer ATA186   Fredrik Hedberg
• [Asterisk-Users] AgentLogin and Huntgroups   Jason Helmich
• [Asterisk-Users] List proposal - Asterick-Proessional   Devon Henderson
• [Asterisk-Users] Extension and phone management best practices??   Devon Henderson
• [Asterisk-Users] Extension and phone management best practices??   Devon Henderson
• [Asterisk-Users] Extension and phone management bestpractices??   Devon Henderson
• [Asterisk-Users] Extension and phone management bestpractices??   Devon Henderson
• [Asterisk-Users] Extension and phone management bestpractices??   Devon Henderson
• [Asterisk-Users] Asterix Newbie   Devon Henderson
• [Asterisk-Users] Re: asterisk-u] Can I runAsterisk remotely from telnetsession?   R P Herrold
• [Asterisk-Users] list proposal   Andy Hester
• [Asterisk-Users] list proposal   Andy Hester
• [Asterisk-Users] IP phone recommendation   Andy Hester
• [Asterisk-Users] ADSI Phones   Andy Hester
• [Asterisk-Users] DTMF Tone length   Andy Hester
• [Asterisk-Users] Provisioning CO lines   Andy Hester
• [Asterisk-Users] Default Flash Time   Andy Hester
• [Asterisk-Users] PCI X100P card interrupt problems   Andy Hester
• [Asterisk-Users] Quintum A400 sip registration problem   David Hindmarsh
• [Asterisk-Users] Sip Trunk config   David Hindmarsh
• [Asterisk-Users] Sip Trunk config   David Hindmarsh
• [Asterisk-Users] Newbie just starting out with *   Chris Hirsch
• [Asterisk-Users] Newbie just starting out with *   Chris Hirsch
• [Asterisk-Users] Newbie just starting out with *   Chris Hirsch
• [Asterisk-Users] Newbie just starting out with *   Chris Hirsch
• [Asterisk-Users] list proposal   Chris Hirsch
• [Asterisk-Users] How to Asterisk   Chris Hirsch
• [Asterisk-Users] New gastman clone + what else?   Stuart Hirst
• [Asterisk-Users] Mixing audio from Music on Hold and IVR   Stuart Hirst
• [Asterisk-Users] Java SIP Client   Stuart Hirst
• [Asterisk-Users] Receptionist Console   Stuart Hirst
• [Asterisk-Users] MOH with SIP   Stuart Hirst
• [Asterisk-Users] X-Lite Build 1059 problems   Stuart Hirst
• [Asterisk-Users] SNOM 200 bugs   Stuart Hirst
• [Asterisk-Users] X100P in Spain & Busy Detect   Stuart Hirst
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Gavin Hollinger
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Gavin Hollinger
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Gavin Hollinger
• [Asterisk-Users] pardon the newbie question   Mike Hollis
• [Asterisk-Users] Documentation   Mike Hollis
• [Asterisk-Users] Documentation   Mike Hollis
• [Asterisk-Users] Grandstream Budgettone 100 & 102   Steven Honson
• [Asterisk-Users] Libpri and asterisk fail to install   Rhys Hopkins
• [Asterisk-Users] Libpri and asterisk fail to install   Rhys Hopkins
• [Asterisk-Users] Libpri and asterisk fail to install   Rhys Hopkins
• [Asterisk-Users] No audio in either direction, sip channels hanging, asterisk will not shut down.   Rhys Hopkins
• [Asterisk-Users] chan_capi in the US   Justin Huff
• [Asterisk-Users] Standard Analoge modem - can it be used?   Asterisk - linux - JVB
• [Asterisk-Users] .:. .: .. .Stottering audio ??   Asterisk - linux - JVB
• [Asterisk-Users] .:. .: .. .Stottering audio ??   Asterisk - linux - JVB
• [Asterisk-Users] .:. .: .. .Stottering audio ??   Asterisk - linux - JVB
• [Asterisk-Users] MusicOnHold   Asterisk - linux - JVB
• [Asterisk-Users] .:. .: .. .Stottering audio ??   Asterisk - linux - JVB
• [Asterisk-Users] MusicOnHold   Asterisk - linux - JVB
• [Asterisk-Users] MusicOnHold   Asterisk - linux - JVB
• [Asterisk-Users] MusicOnHold   Asterisk - linux - JVB
• [Asterisk-Users] Conference call   Asterisk - linux - JVB
• [Asterisk-Users] Conference call   Asterisk - linux - JVB
• [Asterisk-Users] .:. .: .. .Stottering audio ??   Asterisk - linux - JVB
• [Asterisk-Users] GSM codec   Tamas Jalsovszky
• [Asterisk-Users] SIP Lines   Andrew Joakimsen
• [Asterisk-Users] SIP Lines   Andrew Joakimsen
• [Asterisk-Users] Iconnecthere   Andrew Joakimsen
• [Asterisk-Users] Iconnecthere   Andrew Joakimsen
• [Asterisk-Users] Iconnecthere   Andrew Joakimsen
• [Asterisk-Users] Iconnecthere   Andrew Joakimsen
• [Asterisk-Users] Great concept but a few issues unresolved   Andrew Joakimsen
• [Asterisk-Users] Grandstream Budgetone   Andrew Joakimsen
• [Asterisk-Users] LAN switches with PoE? PoE phones?   Andrew Joakimsen
• [Asterisk-Users] Voicemail cliping digits via sip   Andrew Joakimsen
• [Asterisk-Users] Voicemail cliping digits via sip   Andrew Joakimsen
• [Asterisk-Users] Cordless SIP phones   Andrew Joakimsen
• [Asterisk-Users] MusicOnHold   Andrew Joakimsen
• [Asterisk-Users] MusicOnHold   Andrew Joakimsen
• [Asterisk-Users] Vonage locked ATA-186 question   Andrew Joakimsen
• [Asterisk-Users] SIP using which codec?   Andrew Joakimsen
• [Asterisk-Users] SIP using which codec?   Andrew Joakimsen
• [Asterisk-Users] G723 (was SIP using which codec?)   Andrew Joakimsen
• [Asterisk-Users] G723 (was SIP using which codec?)   Andrew Joakimsen
• [Asterisk-Users] Grandstream Budgetone Defective Units   Andrew Joakimsen
• [Asterisk-Users] pardon the newbie question   Andrew Joakimsen
• [Asterisk-Users] Intresting.. hrm   Andrew Joakimsen
• [Asterisk-Users] Grandstream and CallerID not working   Andrew Joakimsen
• [Asterisk-Users] gnophone connection   Andrew Joakimsen
• [Asterisk-Users] (no subject)   Andrew Joakimsen
• [Asterisk-Users] Packet8 DTA310   Andrew Joakimsen
• [Asterisk-Users] Asterisk and Cisco 7960   Andrew Joakimsen
• [Asterisk-Users] Asterisk and Cisco 7960   Andrew Joakimsen
• [Asterisk-Users] Conference without zaptel??   Andrew Joakimsen
• [Asterisk-Users] Asterisk as a stand alone voice mail server (fwd)   Rainer Jochem
• [Asterisk-Users] Fair comparison   Rainer Jochem
• [Asterisk-Users] Newbie - setup help   Rainer Jochem
• [Asterisk-Users] ISDN Examples   Olle E. Johansson
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   Olle E. Johansson
• [Asterisk-Users] sample configs   Travis Johnson
• [Asterisk-Users] Looking for pricing on complete setup...   Ashley Jones
• [Asterisk-Users] dtmf detection from AS5350 over SIP   Brian Jones
• [Asterisk-Users] Asterisk agi interface leaves zombie processes?   Daryl Jones
• [Asterisk-Users] Asterisk agi interface leaves zombie processes?   Daryl Jones
• [Asterisk-Users] pcphoneline producs   Senad Jordanovic
• [Asterisk-Users] SCO/Linux concerns   Senad Jordanovic
• [Asterisk-Users] newbie question - devices   Senad Jordanovic
• [Asterisk-Users] newbie question - devices   Senad Jordanovic
• [Asterisk-Users] X-Lite - No sound + chan_sip issue   Senad Jordanovic
• [Asterisk-Users] X-Lite - No sound + chan_sip issue   Senad Jordanovic
• [Asterisk-Users] queue / agent documentation   Senad Jordanovic
• [Asterisk-Users] X-Lite - No sound + chan_sip issue   Senad Jordanovic
• [Asterisk-Users] X-Lite - No sound + chan_sip issue   Senad Jordanovic
• [Asterisk-Users] list proposal   Senad Jordanovic
• [Asterisk-Users] H323 and SIP   Senad Jordanovic
• [Asterisk-Users] Registering SIP with FWD and ICONNECTHERE   Senad Jordanovic
• [Asterisk-Users] 403 FORBIDDEN Help!   Bartosz Jozwiak
• [Asterisk-Users] 403 FORBIDDEN Help!   Bartosz Jozwiak
• [Asterisk-Users] 403 FORBIDDEN Help!   Bartosz Jozwiak
• [Asterisk-Users] 403 FORBIDDEN Help!   Bartosz Jozwiak
• [Asterisk-Users] PrePaid and IVR   Bartosz Jozwiak
• [Asterisk-Users] PrePaid and IVR   Bartosz Jozwiak
• [Asterisk-Users] PrePaid and IVR   Bartosz Jozwiak
• [Asterisk-Users] Analog lines   Bartosz Jozwiak
• [Asterisk-Users] PrePaid and IVR   Bartosz Jozwiak
• [Asterisk-Users] Queue   Bartosz Jozwiak
• [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank   Bartosz Jozwiak
• [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank   Bartosz Jozwiak
• [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank   Bartosz Jozwiak
• [Asterisk-Users] Hardware question   Bartosz Jozwiak
• [Asterisk-Users] Adtran TA 750   Bartosz Jozwiak
• [Asterisk-Users] Dialogic Hardware   Bartosz Jozwiak
• [Asterisk-Users] Xphone Lite Cannot make work on Asterisk   Bartosz Jozwiak
• [Asterisk-Users] ATA 186 - X-Lite and Asterisk   Bartosz Jozwiak
• [Asterisk-Users] Xphone Lite Cannot make work on Asterisk   Bartosz Jozwiak
• [Asterisk-Users] Asterisk + SNOM + Pound and star keys   Bartosz Jozwiak
• [Asterisk-Users] Configuration Adtran TA 750   Bartosz Jozwiak
• [Asterisk-Users] Festival and Asterisk   Bartosz Jozwiak
• [Asterisk-Users] Festival and Asterisk   Bartosz Jozwiak
• [Asterisk-Users] Festival and Asterisk   Bartosz Jozwiak
• [Asterisk-Users] Asterisk Newbie ...   Julien
• [Asterisk-Users] Asterisk Newbie ...   Julien
• [Asterisk-Users] Asterisk Newbie ...   Julien
• [Asterisk-Users] Asterisk Newbie ...   Julien
• [Asterisk-Users] chan_capi: Hanging channels - again   Klaus-Peter Junghanns
• [Asterisk-Users] chan_capi: Hanging channels - again   Klaus-Peter Junghanns
• [Asterisk-Users] Chan_Capi questions??   Klaus-Peter Junghanns
• [Asterisk-Users] avm fritz pci   Klaus-Peter Junghanns
• [Asterisk-Users] chan_capi compile errors with latest CVS   Klaus-Peter Junghanns
• [Asterisk-Users] fatal embrace control in menus ?   Klaus-Peter Junghanns
• [Asterisk-Users] I4L CallerID not working   Klaus-Peter Junghanns
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Klaus-Peter Junghanns
• SPAMWARNING:216.207.245.21:RE: [Asterisk-Users] Re: LAN switches with PoE? PoE phones?   Daryl G. Jurbala
• [Asterisk-Users] sip and pix   Daryl G. Jurbala
• [Asterisk-Users] Installation Problem   Daryl G. Jurbala
• [Asterisk-Users] Newbie - setup help   Daryl G. Jurbala
• [Asterisk-Users] Newbie - setup help   Daryl G. Jurbala
• [Asterisk-Users] RTP header compression?   Kevin K
• [Asterisk-Users] RTP header compression?   Kevin K
• [Asterisk-Users] zaptel compile problems   Ajit Kallingal
• [Asterisk-Users] PCI X100P card interrupt problems   Ajit M Kallingal
• [Asterisk-Users] New SIP Phone   Michael Kane
• [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)   Michael Kane
• [Asterisk-Users] No Audio on SIP Phone Connection   Kang.ChenJi at c3smail.monmouth.army.mil
• [Asterisk-Users] cdr_mysql uncompress   Johanna Kangas
• [Asterisk-Users] weird error message with zaptel   Johanna Kangas
• [Asterisk-Users] weird error message with zaptel   Johanna Kangas
• [Asterisk-Users] memory leak?   Roy Sigurd Karlsbakk
• [Asterisk-Users] small fix in chan_mgcp.c   Roy Sigurd Karlsbakk
• [Asterisk-Users] mem leak in logger.c?   Roy Sigurd Karlsbakk
• [Asterisk-Users] memory leak?   Roy Sigurd Karlsbakk
• [Asterisk-Users] chan_capi: Hanging channels - again   Roy Sigurd Karlsbakk
• [Asterisk-Users] chan_capi: Hanging channels - again   Roy Sigurd Karlsbakk
• [Asterisk-Users] Re: Asterisk diskless server, a web page with more info?   Ben Klang
• [Asterisk-Users] IAX to zaptel echo   Claude Klimos
• [Asterisk-Users] DTMF onto the reall world   Nick Knight
• [Asterisk-Users] Why are FXO so expensive?   Gene Kochanowsky
• [Asterisk-Users] Re: Open source IP phone, maybe?   Gene Kochanowsky
• [Asterisk-Users] Is Asterisk ready for "real" use?   Gene Kochanowsky
• [Asterisk-Users] Is Asterisk ready for "real" use?   Gene Kochanowsky
• [Asterisk-Users] ISDN BRI outgoing call instant hangup   Haris Koutsouris
• [Asterisk-Users] Delete voicemail files on disk   Tjardick van der Kraan
• [Asterisk-Users] SIP agent logging into queue?   Tjardick van der Kraan
• [Asterisk-Users] * server based Phonebook   Tjardick van der Kraan
• [Asterisk-Users] bug report: whitespaces in uris   Jiri Kuthan
• [Asterisk-Users] * server based Phonebook   Jiri Kuthan
• [Asterisk-Users] * server based Phonebook   Jiri Kuthan
• [Asterisk-Users] Re: The Almighty X-Lite DTMF Problem (patch tested)   Erik Lagerway
• [Asterisk-Users] Codec?   Paul Lambert
• [Asterisk-Users] Codec?   Paul Lambert
• [Asterisk-Users] newbie question - devices   Scott Lambert
• [Asterisk-Users] Is Asterisk ready for "real" use?   Scott Lambert
• [Asterisk-Users] Problems with TDM400P & X100P   Scott Lambert
• [Asterisk-Users] Sip and One Way Audio   Steve Lane
• [Asterisk-Users] Fair comparison   Steve Lane
• [Asterisk-Users] Xten-Lite and Asterisk.   Steve Lane
• [Asterisk-Users] Working with FWD, IPTel, SIPPhone?   Steve Lane
• [Asterisk-Users] New Asterisk user.   Steve Lane
• [Asterisk-Users] Can I runAsterisk remotely from telnet session?   Steve Lane
• [Asterisk-Users] Can I runAsterisk remotely from telnet session?   Steve Lane
• [Asterisk-Users] Registring soft phones in Asterisk   Steve Lane
• [Asterisk-Users] VoIP dialtone?   Steve Lane
• [Asterisk-Users] VoIP dialtone?   Steve Lane
• Subject: [Asterisk-Users] Provisioning CO lines   Steve Lane
• [Asterisk-Users] Unified messaging.   Steve Lane
• [Asterisk-Users] Is Asterisk ready for "real" use?   Steve Lane
• [Asterisk-Users] GSM codec   Siggi Langauf
• [Asterisk-Users] Asterisk as a stand alone voice mail server   Siggi Langauf
• [Asterisk-Users] Asterisk as a stand alone voice mail server (fwd)   Siggi Langauf
• [Asterisk-Users] Semi-newbie question "Softswitch" and Asterisk - Is there a difference?   Siggi Langauf
• [Asterisk-Users] Does Wildcard x100p support Caller ID outside the US? (fwd)   Siggi Langauf
• [Asterisk-Users] ip phones and intercom/paging   Siggi Langauf
• [Asterisk-Users] Asterisk as a stand alone voice mail server   Siggi Langauf
• [Asterisk-Users] ip phones and intercom/paging   Siggi Langauf
• [Asterisk-Users] Asterisk as a stand alone voice mail server (fwd)   Siggi Langauf
• [Asterisk-Users] Asterisk as a stand alone voice mail server   Siggi Langauf
• [Asterisk-Users] Extension and phone management bestpractices??   Siggi Langauf
• [Asterisk-Users] SIP QUESTION   Siggi Langauf
• [Asterisk-Users] Cisco 79xx XML carriage returns/line feeds   Siggi Langauf
• [Asterisk-Users] Using OH323 and Gatekeeper   Langley, Sean
• [Asterisk-Users] Using OH323 and Gatekeeper   Langley, Sean
• [Asterisk-Users] Using OH323 and Gatekeeper   Langley, Sean
• [Asterisk-Users] Using OH323 and Gatekeeper   Langley, Sean
• [Asterisk-Users] Using OH323 and Gatekeeper   Langley, Sean
• [Asterisk-Users] Gatekeeper   Langley, Sean
• [Asterisk-Users] *-openh323 & faststart   Langley, Sean
• [Asterisk-Users] *-openh323 faststart   Langley, Sean
• [Asterisk-Users] How Do I disable faststart?   Langley, Sean
• [Asterisk-Users] Intercom with Cisco SIP 796x phones?   John Laur
• [Asterisk-Users] CDR MySQL   Tim Leeland
• [Asterisk-Users] callerid (Bell type) in Europe   Julien Lesaint
• [Asterisk-Users] bugs.digium.com   Tilghman Lesher
• [Asterisk-Users] iax.conf / Registration rejected   Tilghman Lesher
• [Asterisk-Users] Windows IAX soft phone   Tilghman Lesher
• [Asterisk-Users] [OT] unsubscribe   Tilghman Lesher
• [Asterisk-Users] Killing runaway PBX   Tilghman Lesher
• [Asterisk-Users] list proposal   Tilghman Lesher
• [Asterisk-Users] Fax Handled   Tilghman Lesher
• [Asterisk-Users] Fax Handled   Tilghman Lesher
• [Asterisk-Users] Fax Handled   Tilghman Lesher
• [Asterisk-Users] Codec?   Tilghman Lesher
• [Asterisk-Users] URI for dialing   Tilghman Lesher
• [Asterisk-Users] Can't compile cdr_mysql   Tilghman Lesher
• [Asterisk-Users] Can't compile cdr_mysql   Tilghman Lesher
• [Asterisk-Users] Fax Handled   Tilghman Lesher
• [Asterisk-Users] Fax Handled   Tilghman Lesher
• [Asterisk-Users] Can't compile cdr_mysql   Tilghman Lesher
• [Asterisk-Users] Receiving iaxtel calls   Tilghman Lesher
• [Asterisk-Users] make: warning: Clock skew detected. Your build may be incomplete.   Tilghman Lesher
• [Asterisk-Users] Voicemail2 patches   Tilghman Lesher
• [Asterisk-Users] MORE Questions regarding CDR's   Tilghman Lesher
• [Asterisk-Users] MORE Questions regarding CDR's   Tilghman Lesher
• [Asterisk-Users] cdr_mysql   Tilghman Lesher
• [Asterisk-Users] cdr_mysql   Tilghman Lesher
• [Asterisk-Users] cdr_mysql   Tilghman Lesher
• [Asterisk-Users] cdr_mysql   Tilghman Lesher
• [Asterisk-Users] Voicemail2 vs. Voicemail   Tilghman Lesher
• [Asterisk-Users] Newbie - setup help   Tilghman Lesher
• [Asterisk-Users] DBSaveTree & DBLoadTree   Tilghman Lesher
• [Asterisk-Users] Voicemail cliping digits via sip   Ernest W. Lessenger
• [Asterisk-Users] Voicemail cliping digits via sip   Ernest W. Lessenger
• [Asterisk-Users] Is Asterisk ready for "real" use?   Ernest W. Lessenger
• [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)   Ernest W. Lessenger
• [Asterisk-Users] VoIP dialtone?   Ernest W. Lessenger
• [Asterisk-Users] VoIP dialtone?   Ernest W. Lessenger
• [Asterisk-Users] Asterisk + SNOM + Pound and star keys   Ernest W. Lessenger
• [Asterisk-Users] Provisioning CO lines   Ernest W. Lessenger
• [Asterisk-Users] Asterisk + SNOM + Pound and star keys   Ernest W. Lessenger
• [Asterisk-Users] call center - operators not using phone keys   Ernest W. Lessenger
• [Asterisk-Users] Killing runaway PBX   Todd Lieberman
• [Asterisk-Users] reload not working   Todd Lieberman
• [Asterisk-Users] reload not working   Todd Lieberman
• [Asterisk-Users] fatal embrace control in menus ?   Todd Lieberman
• [Asterisk-Users] SIP NAT question   George Lin
• [Asterisk-Users] SIP NAT question   George Lin
• [Asterisk-Users] CODEC & DTMF   George Lin
• [Asterisk-Users] DTMF SIP   George Lin
• [Asterisk-Users] H323/SIP gatekeeper   George Lin
• [Asterisk-Users] Incomming call issue   Lists
• [Asterisk-Users] Caller Id Issues   Lists
• [Asterisk-Users] New SIP Phone   Nathan Littlepage
• [Asterisk-Users] Processor Consuption   Nathan Littlepage
• [Asterisk-Users] X-Lite - No sound + chan_sip issue   Nathan Littlepage
• [Asterisk-Users] X-Lite - No sound + chan_sip issue   Nathan Littlepage
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Nathan Littlepage
• [Asterisk-Users] Sip and One Way Audio   Nathan Littlepage
• [Asterisk-Users] Sip and One Way Audio   Nathan Littlepage
• [Asterisk-Users] IP phone recommendation   Nathan Littlepage
• [Asterisk-Users] IP phone recommendation   Nathan Littlepage
• [Asterisk-Users] IP phone recommendation   Nathan Littlepage
• [Asterisk-Users] IP phone recommendation   Nathan Littlepage
• [Asterisk-Users] Cisco 7940 7960   Nathan Littlepage
• [Asterisk-Users] Re: LAN switches with PoE? PoE phones?   Nathan Littlepage
• [Asterisk-Users] Re: LAN switches with PoE? PoE phones?   Nathan Littlepage
• [Asterisk-Users] Cisco 7940 7960   Nathan Littlepage
• [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank   Nathan Littlepage
• [Asterisk-Users] Cisco 7940 SIP   Nathan Littlepage
• [Asterisk-Users] Cisco 7940 SIP   Nathan Littlepage
• [Asterisk-Users] Why doesnt anyone reply me ?   Pavel Litvinenko
• [Asterisk-Users] cdr_mysql   Thorsten Lockert
• [Asterisk-Users] ENUM on Asterisk   Dickson Loh
• [Asterisk-Users] Pager support   Alex Lopez
• [Asterisk-Users] RE: T100P/ TSU 600 installation problem   Alex Lopez
• [Asterisk-Users] Malicious Call Trace   Low, Adam
• [Asterisk-Users] cdr_mysql   Low, Adam
• [Asterisk-Users] Malicious Call Trace   Low, Adam
• [Asterisk-Users] Cisco 79xx XML carriage returns/line feeds   Low, Adam
• [Asterisk-Users] Is Asterisk ready for "real" use?   Low, Adam
• [Asterisk-Users] DTMF tones not long enough on out going call s   Low, Adam
• [Asterisk-Users] DTMF tones not long enough on out going call s   Low, Adam
• [Asterisk-Users] Why doesnt anyone reply me ?   Low, Adam
• [Asterisk-Users] Why doesnt anyone reply me ?   Low, Adam
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Richard Lyman
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Richard Lyman
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Richard Lyman
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Richard Lyman
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Richard Lyman
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Richard Lyman
• [Asterisk-Users] Extension and phone management best practices??   Richard Lyman
• [Asterisk-Users] SetVar on sample.call   Richard Lyman
• [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal   Leif Madsen
• [Asterisk-Users] I can't get a two way conversation going?   Leif Madsen
• [Asterisk-Users] I can't get a two way conversation going?   Leif Madsen
• [Asterisk-Users] Need help with installation of H323 chanel driver   Serge Mankovski
• [Asterisk-Users] Need help with installation of H323 chanel driver   Serge Mankovski
• [Asterisk-Users] zaptel does not compile anymore   Serge Mankovski
• [Asterisk-Users] RE: Asterisk stops responding   Serge Mankovski
• [Asterisk-Users] R2 support   Paulo Mannheimer
• [Asterisk-Users] new on E100P   Paulo Mannheimer
• [Asterisk-Users] new on E100P   Paulo Mannheimer
• [Asterisk-Users] * and IAX as a gateway to video conferencing   Paulo Mannheimer
• [Asterisk-Users] Why doesnt anyone reply me ?   Paulo Mannheimer
• [Asterisk-Users] segmentation fault with asterisk and OH323   Michael Manousos
• [Asterisk-Users] HELP!!!!   Michael Manousos
• [Asterisk-Users] CDR   Michael Manousos
• [Asterisk-Users] Musiconhold interrupted sound   Michael Manousos
• [Asterisk-Users] Musiconhold interrupted sound   Michael Manousos
• [Asterisk-Users] SendDtmf   Michael Manousos
• [Asterisk-Users] Musiconhold interrupted sound   Michael Manousos
• [Asterisk-Users] Wierd Message   Michael Manousos
• [Asterisk-Users] chan_oh323 + dtmf   Michael Manousos
• [Asterisk-Users] chan_oh323 + dtmf   Michael Manousos
• [Asterisk-Users] Wierd Message   Michael Manousos
• [Asterisk-Users] Warning Messages   Michael Manousos
• [Asterisk-Users] H323 CallerID   Michael Manousos
• [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk   Michael Manousos
• Re2: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]   Michael Manousos
• [Asterisk-Users] H323 CallerID   Michael Manousos
• Re2: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]   Michael Manousos
• [Asterisk-Users] .:. .: .. .Stottering audio ??   Michael Manousos
• [Asterisk-Users] Pops   Michael Manousos
• [Asterisk-Users] cdr_mysql   Michael Manousos
• [Asterisk-Users] asterisk-oh323 v0.5.5   Michael Manousos
• [Asterisk-Users] Asterisk and RTP flow   Michael Manousos
• [Asterisk-Users] asterisk-oh323 v0.5.5   Michael Manousos
• [Asterisk-Users] Little patch for packetization in G729 in chan_h323   Michael Manousos
• [Asterisk-Users] Alias limitation in asterisk-oh323.0.5.5   Michael Manousos
• [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers   Michael Manousos
• [Asterisk-Users] Need help with installation of H323 chanel driver   Michalis Manousos
• [Asterisk-Users] Registering via IAX2 succeeds, but bridging to the registered peer fails   Manuel
• [Asterisk-Users] Prepaid calling card   Asad Manzur
• [Asterisk-Users] Asterisk and RTP flow   Tebaldi Marco
• [Asterisk-Users] RTP channel   Tebaldi Marco
• [Asterisk-Users] Filling PHP Variable from EXTENSION in AGI   Brancaleoni Matteo
• [Asterisk-Users] reload   Alastair Maw
• [Asterisk-Users] How do i configure so an incoming call triggers an http request?   Alastair Maw
• [Asterisk-Users] How do i configure so an incoming call triggers an http request?   Alastair Maw
• [Asterisk-Users] How do i configure so an incoming call triggers an http request?   Alastair Maw
• [Asterisk-Users] h extension seems to wipe variables?   Alastair Maw
• [Asterisk-Users] h extension seems to wipe variables?   Alastair Maw
• [Asterisk-Users] SIP agent logging into queue?   Alastair Maw
• [Asterisk-Users] Java SIP Client   Alastair Maw
• [Asterisk-Users] Conference call   Alastair Maw
• [Asterisk-Users] Dialogic Hardware   Alastair Maw
• [Asterisk-Users] Conference without zaptel??   Alastair Maw
• [Asterisk-Users] Vonage ATA-186 password recovery   Scrotus Maximus
• [Asterisk-Users] Vonage ATA-186 password recovery   Scrotus Maximus
• [Asterisk-Users] Seting up TDM40B   McAughan, Matt
• [Asterisk-Users] Seting up TDM40B   McAughan, Matt
• [Asterisk-Users] Seting up TDM40B   McAughan, Matt
• [Asterisk-Users] Some questions about a potential usage scena rio for asterisk   McAughan, Matt
• [Asterisk-Users] newbie question - devices   McAughan, Matt
• [Asterisk-Users] newbie question - devices   McAughan, Matt
• [Asterisk-Users] Newbie just starting out with *   McAughan, Matt
• [Asterisk-Users] Newbie just starting out with *   McAughan, Matt
• [Asterisk-Users] (no subject)   McAughan, Matt
• [Asterisk-Users] queue / agent documentation   McAughan, Matt
• [Asterisk-Users] queue / agent documentation   McAughan, Matt
• [Asterisk-Users] ANI/DNIS call routing   McAughan, Matt
• [Asterisk-Users] ANI/DNIS call routing   McAughan, Matt
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   McAughan, Matt
• [Asterisk-Users] I can't get a two way conversation going?   McAughan, Matt
• [Asterisk-Users] Background Noise   McAughan, Matt
• [Asterisk-Users] call center - operators not using phone keys   McAughan, Matt
• [Asterisk-Users] FXO/FXS hotline   Darren McIntosh
• [Asterisk-Users] New SIP Phone   Fearghas McKay
• [Asterisk-Users] h323 over NAT   Jeremy McNamara
• [Asterisk-Users] Using OH323 and Gatekeeper   Jeremy McNamara
• [Asterisk-Users] Using OH323 and Gatekeeper   Jeremy McNamara
• [Asterisk-Users] Using H323 and Gatekeeper   Jeremy McNamara
• [Asterisk-Users] H323 CallerID   Jeremy McNamara
• [Asterisk-Users] limiting out going calls to a maximum duration   Jeremy McNamara
• [Asterisk-Users] Using OH323 and Gatekeeper   Jeremy McNamara
• [Asterisk-Users] h323 and cvs one way audio   Jeremy McNamara
• [Asterisk-Users] segfaults with queue   Jeremy McNamara
• [Asterisk-Users] G.729 licensing -- an opinion   Jeremy McNamara
• [Asterisk-Users] Using OH323 and Gatekeeper   Jeremy McNamara
• [Asterisk-Users] Known problem?   Jeremy McNamara
• [Asterisk-Users] Fair comparison   Jeremy McNamara
• [Asterisk-Users] chan_h323, Asterisk and DTMF issue   Jeremy McNamara
• [Asterisk-Users] Conference + E100P + H323   Jeremy McNamara
• [Asterisk-Users] ast_channel_alloc() losing pvt struct   Jeremy McNamara
• [Asterisk-Users] Asterisk H323 Trunk   Jeremy McNamara
• [Asterisk-Users] Chan_h323 one way audio   Jeremy McNamara
• [Asterisk-Users] No voice call from H.323-phone to SIP-phone   Jeremy McNamara
• [Asterisk-Users] Grandstream, SIP encryption   Jeremy McNamara
• [Asterisk-Users] PrePaid and IVR   Jeremy McNamara
• [Asterisk-Users] PrePaid and IVR   Jeremy McNamara
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Jeremy McNamara
• [Asterisk-Users] VoIP dialtone?   Jeremy McNamara
• [Asterisk-Users] Conference + time limit   Jeremy McNamara
• [Asterisk-Users] VoIP dialtone?   Jeremy McNamara
• [Asterisk-Users] Working example of "switch"?   Jeremy McNamara
• [Asterisk-Users] Little patch for packetization in G729 in chan_h323   Jeremy McNamara
• [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers   Jeremy McNamara
• [Asterisk-Users] H.323 channel problems   Jeremy McNamara
• [Asterisk-Users] conference authorization   Jeremy McNamara
• [Asterisk-Users] conference authorization   Jeremy McNamara
• [Asterisk-Users] Chan_h323 + amaflag   Jeremy McNamara
• [Asterisk-Users] Re: [Asterisk-Users] ATA 186 & DynExtenDB (query extensions vía sql)   Jeremy McNamara
• [Asterisk-Users] Newbie IVR question   Jeremy McNamara
• [Asterisk-Users] Speex Problem   San Mehat
• [Asterisk-Users] TDM10M && Siemens Euroset 2015   Olaf Menzel
• [Asterisk-Users] Libpri and asterisk fail to install   Jim Mercer
• [Asterisk-Users] Gatekeeper   Wayne Methorst
• [Asterisk-Users] phone rings while already on a call   Steve Meyers
• [Asterisk-Users] phone rings while already on a call   Steve Meyers
• [Asterisk-Users] callwaiting in sip can't be disabled   Steve Meyers
• [Asterisk-Users] callwaiting in sip can't be disabled   Steve Meyers
• [Asterisk-Users] Channel banks, etc.   Steve Meyers
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   Steve Meyers
• [Asterisk-Users] New SIP Phone   Steve Meyers
• [Asterisk-Users] Leftover Budgettone issues   Steve Meyers
• [Asterisk-Users] [OT] unsubscribe   Steve Meyers
• [Asterisk-Users] [OT] unsubscribe   Steve Meyers
• [Asterisk-Users] list proposal   Steve Meyers
• [Asterisk-Users] list proposal   Steve Meyers
• [Asterisk-Users] Ring while on phone   Steve Meyers
• [Asterisk-Users] Ring while on phone   Steve Meyers
• [Asterisk-Users] IP phone recommendation   Steve Meyers
• [Asterisk-Users] FXO mode   Steve Meyers
• [Asterisk-Users] FXO mode <2147483647.1060797007@[192.168.1.210]> <1060794258.27544.62.camel@RobinHood.LinuxAutrement.com>   Steve Meyers
• [Asterisk-Users] Cordless SIP phones   Steve Meyers
• [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank   Steve Meyers
• [Asterisk-Users] IAX <> IAX trunking... DP cache?   Steve Meyers
• [Asterisk-Users] VoIP dialtone?   Steve Meyers
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   Steve Meyers
• [Asterisk-Users] VIRUS ALERT   Steve Meyers
• [Asterisk-Users] Conference + time limit   Steve Meyers
• [Asterisk-Users] Restricting concurrent SIP calls   Steve Meyers
• [Asterisk-Users] PRI CallerID problem   Michael
• [Asterisk-Users] no incoming packets & Sound: Recording overrun   Miernik
• [Asterisk-Users] ip phones and intercom/paging   Benjamin Miller
• [Asterisk-Users] UNIX command-line interaction with astdb   Benjamin Miller
• [Asterisk-Users] * and IAX as a gateway to video conferencing   Masakazu Nakano
• [Asterisk-Users] Newbie Question / ISDN   Masakazu Nakano
• [Asterisk-Users] CVS version build error   Nguyen Nam
• [Asterisk-Users] Stable versions of Asterisk (Was: Re: Fair comparison (John Todd))   Nguyen Nam
• [Asterisk-Users] Fwd: Stable versions of Asterisk (Was: Re: Fair comparison (John Todd))   Nguyen Nam
• [Asterisk-Users] Cordless SIP phones   Nathan
• [Asterisk-Users] Cordless SIP phones   Nathan
• [Asterisk-Users] Unsubscribe   Nathan
• [Asterisk-Users] Musiconhold interrupted sound   Jamie Neil
• [Asterisk-Users] Musiconhold interrupted sound   Jamie Neil
• [Asterisk-Users] X-Lite <-> Snom200   Jamie Neil
• [Asterisk-Users] Inbound SIP DTMF detection   Jamie Neil
• [Asterisk-Users] Recomendations for an ISDN-PBX to use with asterisk   Jamie Neil
• [Asterisk-Users] MOH with SIP   Jamie Neil
• [Asterisk-Users] MOH with SIP   Jamie Neil
• [Asterisk-Users] MOH with SIP   Jamie Neil
• [Asterisk-Users] Cisco 7940 7960   Jamie Neil
• [Asterisk-Users] sox and wav to gsm conversion quality issue   Jamie Neil
• [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?   Fats Neutron
• [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?   Fats Neutron
• [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?   Fats Neutron
• [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?   Fats Neutron
• [Asterisk-Users] Has anyone got sip/IAX working behind a firewall?   Fats Neutron
• [Asterisk-Users] Monitor application temporary hack   Grzegorz Nosek
• [Asterisk-Users] weird error message with zaptel   Grzegorz Nosek
• [Asterisk-Users] Three way calling on outgoing FXO line   Carlton J. O'Riley
• [Asterisk-Users] Re: Three way calling on outgoing FXO line (Martin Pycko)   Carlton J. O'Riley
• [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?   Florian Overkamp
• [Asterisk-Users] Asterisk Newbie ...   Florian Overkamp
• [Asterisk-Users] IP phone recommendation   Florian Overkamp
• [Asterisk-Users] h extension seems to wipe variables?   Florian Overkamp
• [Asterisk-Users] Limit Number of user in Conference   Florian Overkamp
• [Asterisk-Users] Limit Number of user in Conference   Florian Overkamp
• [Asterisk-Users] Unified messaging.   Florian Overkamp
• [Asterisk-Users] SIP vs SCCP vs XML   Florian Overkamp
• [Asterisk-Users] unsubscribe   FRANCISCO PEREZ-LANDAETA
• [Asterisk-Users] Repost MS Messenger 4.7 docs?   Dave Packham
• [Asterisk-Users] re: Web GUI   Dave Packham
• [Asterisk-Users] PHP Web Interface helpers   Dave Packham
• [Asterisk-Users] SIP change...   Dave Packham
• [Asterisk-Dev] Re: [Asterisk-Users] SIP change...   Dave Packham
• [Asterisk-Users] Syncronize Monitored Calls   Dave Packham
• [Asterisk-Users] Syncronize Monitored Calls   Dave Packham
• [Asterisk-Users] Syncronize Monitored Calls   Dave Packham
• [Asterisk-Users] Grandstream Budgettone 100 & 102   John Paine
• [Asterisk-Users] New SIP Phone   George Pajari
• [Asterisk-Users] Re: FW: Fax from 925 603 5512 (18 pages)   George Pajari
• [Asterisk-Users] Re: FW: Fax from 925 603 5512 (18 pages)   George Pajari
• [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?   Lorenzo Pallara
• [Asterisk-Users] Hardware for a Big PBX   Alvaro Parres
• [Asterisk-Users] Sip Trunk config   Patrick
• [Asterisk-Users] SIP Lines   Patrick
• [Asterisk-Users] Prepaid calling card   Francisco Perez-Landaeta
• [Asterisk-Users] unsubscribe   Francisco Perez-Landaeta
• [Asterisk-Users] Warning file format_pcm_alaw.c Anyone knows where to look?   Ivar van de Pieterman
• [Asterisk-Users] SNOM 200 bugs   Pertti Pikkarainen
• [Asterisk-Users] PRI Question   Don Pobanz
• [Asterisk-Users] Provisioning CO lines   Don Pobanz
• [Asterisk-Users] 0 out of voicemail to different secretaries   Don Pobanz
• [Asterisk-Users] 0 out of voicemail to different secretaries   Don Pobanz
• [Asterisk-Users] Dialed Number Identification in analog hunt group   Don Pobanz
• [Asterisk-Users] SIP and ECHO   Don Pobanz
• [Asterisk-Users] help please with single t1 configuration   Barry Porch
• [Asterisk-Users] help please with single t1 configuration   Barry Porch
• [Asterisk-Users] PRI Question   Barry Porch
• [Asterisk-Users] PRI Question   Barry Porch
• [Asterisk-Users] PRI Question   Barry Porch
• [Asterisk-Users] PRI Question   Barry Porch
• [Asterisk-Users] VIRUS ALERT   Jon Pounder
• [Asterisk-Users] Sending dtmf over an ougoing call from asterisk   Jon Pounder
• [Asterisk-Users] New SIP Phone   Andy Powell
• [Asterisk-Users] Problem with the Internet LineJACK ISA card...   Andy Powell
• [Asterisk-Users] New SIP Phone   Andy Powell
• [Asterisk-Users] Ansering machine/voicemail detect?   Andy Powell
• [Asterisk-Users] [OT] unsubscribe   Andy Powell
• [Asterisk-Users] CallerID, DECT phones and ATA   Andy Powell
• [Asterisk-Users] CallerID, DECT phones and ATA   Andy Powell
• [Asterisk-Users] list proposal   Andy Powell
• [Asterisk-Users] To Switch or not to Switch... that is the question....   Andy Powell
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Andy Powell
• [Asterisk-Users] Asterisk Newbie ...   Andy Powell
• [Asterisk-Users] Asterisk Newbie ...   Andy Powell
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Andy Powell
• [Asterisk-Users] FXO mode   Andy Powell
• [Asterisk-Users] FXO mode   Andy Powell
• [Asterisk-Users] FXO mode   Andy Powell
• [Asterisk-Users] Running Asterisk behind NAT?   Andy Powell
• [Asterisk-Users] FXO mode   Andy Powell
• [Asterisk-Users] Running Asterisk behind NAT?   Andy Powell
• [Asterisk-Users] FXO mode   Andy Powell
• [Asterisk-Users] FXO mode   Andy Powell
• [Asterisk-Users] Can I runAsterisk remotely from telnet session?   Andy Powell
• [Asterisk-Users] Intresting.. hrm   Andy Powell
• [Asterisk-Users] SetVar on sample.call   Andy Powell
• [Asterisk-Users] Zhone Zplex   Marcel Prisi
• [Asterisk-Users] Polycom SoundPoint 500 with Asterisk   Karl Putland
• [Asterisk-Users] Mutex problem in sip?   Martin Pycko
• [Asterisk-Users] Seting up TDM40B   Martin Pycko
• [Asterisk-Users] Seting up TDM40B   Martin Pycko
• [Asterisk-Users] Hangup after a Timeout   Martin Pycko
• [Asterisk-Users] call waiting   Martin Pycko
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Martin Pycko
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Martin Pycko
• [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?   Martin Pycko
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   Martin Pycko
• [Asterisk-Users] Wierd Message   Martin Pycko
• [Asterisk-Users] (no subject)   Martin Pycko
• [Asterisk-Users] Wierd Message   Martin Pycko
• [Asterisk-Users] Wierd Message   Martin Pycko
• [Asterisk-Users] X100P and Caller ID (again and again...)   Martin Pycko
• [Asterisk-Users] Libpri and asterisk fail to install   Martin Pycko
• [Asterisk-Users] Libpri and asterisk fail to install   Martin Pycko
• [Asterisk-Users] Libpri and asterisk fail to install   Martin Pycko
• [Asterisk-Users] Seting up TDM40B   Martin Pycko
• [Asterisk-Users] Asterisk launch on boot   Martin Pycko
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   Martin Pycko
• [Asterisk-Users] Sip Trunk config   Martin Pycko
• [Asterisk-Users] Unregister SIP connection?   Martin Pycko
• [Asterisk-Users] cdr_mysql uncompress   Martin Pycko
• [Asterisk-Users] Busy detect options   Martin Pycko
• [Asterisk-Users] g729 problems   Martin Pycko
• [Asterisk-Users] Killing runaway PBX   Martin Pycko
• [Asterisk-Users] zaptel sync   Martin Pycko
• [Asterisk-Users] Fax Handled   Martin Pycko
• [Asterisk-Users] How to Asterisk   Martin Pycko
• [Asterisk-Users] Fax Handled   Martin Pycko
• [Asterisk-Users] Weird DTMF issue   Martin Pycko
• [Asterisk-Users] Conference + E100P + H323   Martin Pycko
• [Asterisk-Users] X100P Ringing/Answering   Martin Pycko
• [Asterisk-Users] '#' doesn't work for me   Martin Pycko
• [Asterisk-Users] FXO mode   Martin Pycko
• [Asterisk-Users] Can't compile cdr_mysql   Martin Pycko
• [Asterisk-Users] Virtual extension as local modem   Martin Pycko
• [Asterisk-Users] '#' doesn't work for me   Martin Pycko
• [Asterisk-Users] Problem with latest cdr Makefile???   Martin Pycko
• [Asterisk-Users] preventing timeout while transferring   Martin Pycko
• [Asterisk-Users] Fax Handled   Martin Pycko
• [Asterisk-Users] '#' doesn't work for me   Martin Pycko
• [Asterisk-Users] chan_capi compile errors with latest CVS   Martin Pycko
• [Asterisk-Users] Asterisk's configuration : Which signalling in France with an E1 ?   Martin Pycko
• [Asterisk-Users] PRI Question   Martin Pycko
• [Asterisk-Users] Vonage locked ATA-186 question   Martin Pycko
• [Asterisk-Users] Vonage locked ATA-186 question   Martin Pycko
• [Asterisk-Users] PRI Question   Martin Pycko
• [Asterisk-Users] IAXtel + NAT   Martin Pycko
• [Asterisk-Users] how to change Contact info?   Martin Pycko
• [Asterisk-Users] Warning from chan_zap ring requested   Martin Pycko
• [Asterisk-Users] include context   Martin Pycko
• [Asterisk-Users] Default Flash Time   Martin Pycko
• [Asterisk-Users] Three way calling on outgoing FXO line   Martin Pycko
• [Asterisk-Users] Asterisk stops responding   Martin Pycko
• [Asterisk-Users] X100P in Spain & Busy Detect   Martin Pycko
• [Asterisk-Users] Asterisk stops responding   Martin Pycko
• [Asterisk-Users] RE: Asterisk stops responding   Martin Pycko
• [Asterisk-Users] G729 error messages   Martin Pycko
• [Asterisk-Users] G729 error messages   Martin Pycko
• [Asterisk-Users] Packet8 DTA310   Martin Pycko
• [Asterisk-Users] Cisco AS5300 -- Not hearing anything   Luciano Ramos
• [Asterisk-Users] GSM codec   Luciano Ramos
• [Asterisk-Users] GSM codec   Luciano Ramos
• [Asterisk-Users] GSM codec   Luciano Ramos
• [Asterisk-Users] RTP session traversing Asterisk server...   Andrew Reich
• [Asterisk-Users] Grandstream Budgetone Defective Units   Greg Renouf
• [Asterisk-Users] Grandstream Budgetone Defective Units   Greg Renouf
• [Asterisk-Users] Sound files for internal functions in another place??   Eris Riswanto
• [Asterisk-Users] VoIP dialtone?   Adam Roach
• [Asterisk-Users] VoIP dialtone?   Adam Roach
• [Asterisk-Users] VoIP dialtone?   Adam Roach
• [Asterisk-Users] VoIP dialtone?   Adam Roach
• [Asterisk-Users] DTMF tones not long enough on out going call s   Adam Roach
• [Asterisk-Users] DTMF tones not long enough on out going call s   Adam Roach
• [Asterisk-Users] line numbering and gosub   Adam Roach
• [Asterisk-Users] SIP phones   Adam Roach
• [Asterisk-Users] SIP vs SCCP vs XML   Adam Roach
• [Asterisk-Users] 403 FORBIDDEN Help!   Josh Roberson
• [Asterisk-Users] 403 FORBIDDEN Help!   Josh Roberson
• [Asterisk-Users] Brooktrout PRI-ISA48 card... info..   Josh Roberson
• [Asterisk-Users] Brooktrout PRI-ISA48 card... info..   Josh Roberson
• [Asterisk-Users] Dialogic cards...   Josh Roberson
• [Asterisk-Users] OT: My congestion music.   Josh Roberson
• [Asterisk-Users] Asterisk (g729) termination on CISCO   Roman
• [Asterisk-Users] PRI CallerID problem   Michael Rose
• [Asterisk-Users] Documentation   Jason Ross
• [Asterisk-Users] Difference between Cisco 7940/7940G, 7960/7960G   Jason Ross
• [Asterisk-Users] OT: Grandstream power supplies..   Brenton D. Rothchild
• [Asterisk-Users] Is the DTMF bug in bugs.digium.com what number.   Brenton D. Rothchild
• [Asterisk-Users] Transitioning from existing PBX   Peter Rowell
• [Asterisk-Users] 'System' application exit with error even if it performs the job as expected   Sip Rtp
• [Asterisk-Users] 'System' application exit with error even if it performs the job as expected   Sip Rtp
• [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk   Sip Rtp
• [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk   Sip Rtp
• Re2: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]   Sip Rtp
• Re2: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]   Sip Rtp
• [Asterisk-Users] G.729 licensing -- an opinion   Jan Rychter
• [Asterisk-Users] G.729 licensing -- an opinion   Jan Rychter
• [Asterisk-Users] "Out of area" displayed as caller-id   Jan Rychter
• [Asterisk-Users] Open G.729A codec   Jan Rychter
• [Asterisk-Users] Open G.729A codec   Jan Rychter
• [Asterisk-Users] One-way audio using console   Jan Rychter
• [Asterisk-Users] H.323 channel problems   Jan Rychter
• [Asterisk-Users] Message-waiting-indicator thru ZAP interfaces - how to?   Sam S
• [Asterisk-Users] Config files - examples   SMorys
• [Asterisk-Users] chan_h323, Asterisk and DTMF issue   Jay Sakata
• [Asterisk-Users] chan_h323, Asterisk and DTMF issue   Jay Sakata
• [Asterisk-Users] Problems reloading   Jay Sakata
• [Asterisk-Users] G729 error messages   Jay Sakata
• [Asterisk-Users] G729 error messages   Jay Sakata
• [Asterisk-Users] G729 error messages   Jay Sakata
• [Asterisk-Users] Outdial digits - non TDM trunk   Roger De Salis
• [Asterisk-Users] Asterisk H323 Trunk   Roger De Salis
• [Asterisk-Users] Cisco 7920 phone   Roger De Salis
• [Asterisk-Users] Structured release, Maillists   Roger De Salis
• [Asterisk-Users] Re: ATAs   Roger De Salis
• [Asterisk-Users] retrieving dialed number when overlap dialing?   Thilo Salmon
• [Asterisk-Users] usable/affordable usb phone?   Thilo Salmon
• [Asterisk-Users] Need a "trick" to generate calls   Thilo Salmon
• [Asterisk-Users] Slowly get it ... Hardware   Thilo Salmon
• [Asterisk-Users] additional digit in front of the dialed extenesion by outgoing pri/E1 call   Thilo Salmon
• [Asterisk-Users] Buffering DTMF input   Thilo Salmon
• [Asterisk-Users] Buffering DTMF input   Thilo Salmon
• [Asterisk-Users] Problem with * server and FWD   Yehiel Samson
• [Asterisk-Users] how to change Contact info?   Yehiel Samson
• [Asterisk-Users] Connecting to FWD behind a NAT.   Yehiel Samson
• [Asterisk-Users] Grandstream, SIP encryption   Michael Sandee
• [Asterisk-Users] Grandstream, SIP encryption   Michael Sandee
• [Asterisk-Users] Oen source IP phone, maybe?   Michael Sandee
• [Asterisk-Users] Zhone Zplex 10 units   John Schmerold
• [Asterisk-Users] Zhone Zplex 10 units   John Schmerold
• [Asterisk-Users] Zhone Zplex 10 units   John Schmerold
• [Asterisk-Users] Newbie just starting out with *   John Schmerold
• [Asterisk-Users] Why are FXO so expensive?   John Schmerold
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   John Schmerold
• [Asterisk-Users] Dial out modem via * -> VOIP (Real world experience needed)   John Schmerold
• [Asterisk-Users] List proposal - Asterick-Proessional   John Schmerold
• [Asterisk-Users] list proposal   John Schmerold
• [Asterisk-Users] Is Asterisk ready for "real" use?   John Schmerold
• [Asterisk-Users] SIP phone review   John Schmerold
• [Asterisk-Users] Asterisk as a stand alone voice mail server   Maik Schmitt
• [Asterisk-Users] Asterisk as a stand alone voice mail server   Maik Schmitt
• [Asterisk-Users] Asterisk as a stand alone voice mail server (fwd)   Maik Schmitt
• [Asterisk-Users] SIP and ECHO   Brian J. Schrock
• [Asterisk-Users] SIP and ECHO   Brian J. Schrock
• [Asterisk-Users] MWI question   Bill Schultz
• Subject: [Asterisk-Users] Provisioning CO lines   Bill Schultz
• [Asterisk-Users] BRI newbie queries.   Richard Scobie
• [Asterisk-Users] Busy detect options   Richard Scobie
• [Asterisk-Users] FXO mode   Richard Scobie
• [Asterisk-Users] FXO mode   Richard Scobie
• [Asterisk-Users] FXO mode   Richard Scobie
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Richard Scobie
• [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)   Rob Scott
• [Asterisk-Users] Using Asterisk with FWD through NAT   Borut Senicar
• [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #1033 - 7 msgs   Borut Senicar
• [Asterisk-Users] Twisted Idea   Troy Settle
• [Asterisk-Users] Is Asterisk ready for "real" use?   Troy Settle
• [Asterisk-Users] ISDN Sub Address   Troy Settle
• [Asterisk-Users] RE: Asterisk stops responding   Troy Settle
• [Asterisk-Users] RTP session traversing Asterisk server ...   Artur C. Severo
• [Asterisk-Users] Seting up TDM40B   James Sharp
• [Asterisk-Users] Seting up TDM40B   James Sharp
• [Asterisk-Users] Some questions about a potential usage scenario for asterisk   James Sharp
• [Asterisk-Users] newbie question - devices   James Sharp
• [Asterisk-Users] bugs.digium.com   James Sharp
• [Asterisk-Users] Get faxed you faxing faxer!   James Sharp
• [Asterisk-Users] Can't compile cdr_mysql   James Sharp
• [Asterisk-Users] Oen source IP phone, maybe?   James Sharp
• [Asterisk-Users] Oen source IP phone, maybe?   James Sharp
• [Asterisk-Users] VoIP dialtone?   James Sharp
• [Asterisk-Users] Provisioning CO lines   James Sharp
• Qwest CallerID question re:[Asterisk-Users] Caller ID problem   James Sharp
• [Asterisk-Users] T1 to T1 on asterisk?   James Sharp
• [Asterisk-Users] Dialed Number Identification in analog hunt group   James Sharp
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   James Sizemore
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   James Sizemore
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   James Sizemore
• [Asterisk-Users] Fair comparison   James Sizemore
• [Asterisk-Users] Asterisk as a stand alone voice mail server   James Sizemore
• [Asterisk-Users] IP phone recommendation   James Sizemore
• [Asterisk-Users] Park and out-going trunk calls.   James Sizemore
• [Asterisk-Users] SIP Transfer   James Sizemore
• [Asterisk-Users] '#' doesn't work for me   James Sizemore
• [Asterisk-Users] Grandstream Budgetone Defective Units   James Sizemore
• [Asterisk-Users] DTMF tones not long enough on out going calls   James Sizemore
• [Asterisk-Users] DTMF tones not long enough on out going call s   James Sizemore
• [Asterisk-Users] DTMF tones not long enough on out going call s   James Sizemore
• [Asterisk-Users] Is the DTMF bug in bugs.digium.com what number.   James Sizemore
• [Asterisk-Users] Is the DTMF bug in bugs.digium.com what number.   James Sizemore
• [Asterisk-Users] Is the DTMF bug in bugs.digium.com what number.   James Sizemore
• [Asterisk-Users] Is the DTMF bug in bugs.digium.com what number.   James Sizemore
• [Asterisk-Users] Asterisk stops responding   James Sizemore
• [Asterisk-Users] Asterisk agi interface leaves zombie processes?   Jared Smith
• [Asterisk-Users] Asterisk agi interface leaves zombie processes?   Jared Smith
• [Asterisk-Users] dumb x100p question   Jared Smith
• [Asterisk-Users] Monitor application temporary hack   Jared Smith
• [Asterisk-Users] Monitor application temporary hack   Jared Smith
• [Asterisk-Users] CVS Question   Jared Smith
• [Asterisk-Users] Is Asterisk ready for "real" use?   Jared Smith
• [Asterisk-Users] SIP vs SCCP vs XML   Jared Smith
• [Asterisk-Users] Pickup groups with SIP   Jared Smith
• [Asterisk-Users] Configuration Adtran TA 750   Jared Smith
• [Asterisk-Users] RE: Asterisk stops responding   Jared Smith
• [Asterisk-Users] Version number not updating after CVS   Jared Smith
• [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c) or unixODBC (cdr_unixodbc.c) ?   Steven J. Sobol
• [Asterisk-Users] (no subject)   Steven J. Sobol
• [Asterisk-Users] So now I'm playing around with Queues....   Steven J. Sobol
• [Asterisk-Users] Unregister SIP connection?   Steven J. Sobol
• [Asterisk-Users] So now I'm playing around with Queues....   Steven J. Sobol
• [Asterisk-Users] VoIP dialtone?   Steven J. Sobol
• [Asterisk-Users] Minnesota PUC: Phone rules apply to VoIP   Steven J. Sobol
• Qwest CallerID question re:[Asterisk-Users] Caller ID problem   Steven J. Sobol
• [Asterisk-Users] Any way to distinguish between...   Steven J. Sobol
• [Asterisk-Users] More questions. Call Waiting and Threeway   Steven J. Sobol
• [Asterisk-Users] Best SIP phone?   Timothy Soos
• [Asterisk-Users] Polycom SoundPoint 500 with Asterisk   Timothy Soos
• [Asterisk-Users] Newbie question   Timothy Soos
• [Asterisk-Users] Newbie question   Timothy Soos
• [Asterisk-Users] VoIP (H.323) -> PSTN gateway functionality   Andrei Sosnin
• [Asterisk-Users] retrieving dialed number when overlap dialing?   Mark Spencer
• [Asterisk-Users] ADSI and SoftKeys   Mark Spencer
• [Asterisk-Users] Queue and Agents in CVS   Mark Spencer
• [Asterisk-Users] SIP calls cause Segmentation Fault   Mark Spencer
• [Asterisk-Users] Mutex problem in sip?   Mark Spencer
• [Asterisk-Users] SIP calls cause Segmentation Fault   Mark Spencer
• [Asterisk-Users] memory leak?   Mark Spencer
• [Asterisk-Users] DTMF modes and external IVR systems over ISDN   Mark Spencer
• [Asterisk-Users] Queue and Agents in CVS   Mark Spencer
• [Asterisk-Users] Asterisk agi interface leaveszombie processes?   Mark Spencer
• [Asterisk-Users] SIP app_queue   Mark Spencer
• [Asterisk-Users] T-shirt ideas   Mark Spencer
• [Asterisk-Users] 3xx SIP messages   Mark Spencer
• [Asterisk-Users] TE401P driver warning   Mark Spencer
• [Asterisk-Users] 3xx SIP messages   Mark Spencer
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Mark Spencer
• [Asterisk-Users] Does Wildcard x100p support Caller ID outside the US? (fwd)   Mark Spencer
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Mark Spencer
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Mark Spencer
• [Asterisk-Users] Sip and One Way Audio   Mark Spencer
• [Asterisk-Users] Open G.729A codec   Mark Spencer
• [Asterisk-Users] Open G.729A codec   Mark Spencer
• [Asterisk-Users] Conference + E100P + H323   Mark Spencer
• [Asterisk-Users] Open G.729A codec   Mark Spencer
• [Asterisk-Users] "Double" transfer?   Mark Spencer
• [Asterisk-Users] Don't know how to calculate timelen   Mark Spencer
• [Asterisk-Users] Don't know how to calculate timelen   Mark Spencer
• [Asterisk-Users] Voicemail2 vs. Voicemail   Mark Spencer
• [Asterisk-Users] chan_h323.c   Mark Spencer
• [Asterisk-Users] Voicemail2 vs. Voicemail   Mark Spencer
• [Asterisk-Users] Call transfer ATA186   Mark Spencer
• [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank   Mark Spencer
• [Asterisk-Users] reload problem fixed   Mark Spencer
• [Asterisk-Users] VoIP dialtone?   Mark Spencer
• [Asterisk-Users] Structured release, Maillists   Mark Spencer
• [Asterisk-Users] SIP change...   Mark Spencer
• [Asterisk-Users] SIP change...   Mark Spencer
• [Asterisk-Users] DIAL via CLI missing   Mark Spencer
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Mark Spencer
• [Asterisk-Users] fatal embrace control in menus ?   Mark Spencer
• [Asterisk-Users] fatal embrace control in menus ?   Mark Spencer
• [Asterisk-Users] T1 to T1 on asterisk?   Mark Spencer
• [Asterisk-Users] T1 to T1 on asterisk?   Mark Spencer
• [Asterisk-Users] chan_zap.c zt_rec: Unknown error 500   Mark Spencer
• [Asterisk-Users] Is Asterisk ready for "real" use?   Mark Spencer
• [Asterisk-Users] call center - operators not using phone keys   Mark Spencer
• [Asterisk-Users] T1 to T1 on asterisk?   Mark Spencer
• [Asterisk-Users] chan_zap.c zt_rec: Unknown error 500   Mark Spencer
• [Asterisk-Users] T1 to T1 on asterisk?   Mark Spencer
• [Asterisk-Users] te410p with serial console fails with error: TE410P: Double/missed interrupt detected   Mark Spencer
• [Asterisk-Users] call center - operators not using phone keys   Mark Spencer
• [Asterisk-Users] Killing runaway PBX   Sergey S. Stasyuk
• [Asterisk-Users] Digium & PCI-X   Victor Stevanovic
• [Asterisk-Users] Channel banks, etc.   Steve
• [Asterisk-Users] Budgettone 100 phone Configuration   Steve
• [Asterisk-Users] Budgettone 100 phone Configuration   Steve
• [Asterisk-Users] invalid argument 22 when modprobe wcfxs and wcfxo   Steve
• [Asterisk-Users] ztdummy & usb-ohci?   Iain Stevenson
• [Asterisk-Users] Patch - transfer with two rather than one #   Iain Stevenson
• [Asterisk-Users] Leftover Budgettone issues   Iain Stevenson
• [Asterisk-Users] list proposal   Iain Stevenson
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Iain Stevenson
• [Asterisk-Users] problem with Wildcard 100XP and hangup signal   Iain Stevenson
• [Asterisk-Users] FXO mode   Iain Stevenson
• [Asterisk-Users] Park and out-going trunk calls.   Iain Stevenson
• [Asterisk-Users] FXO mode   Iain Stevenson
• [Asterisk-Users] Cisco 7920 phone   Iain Stevenson
• [Asterisk-Users] Asterisk agi interface leaves zombie processes?   Scott Stingel
• [Asterisk-Users] Asterisk agi interface leaves zombie processes?   Scott Stingel
• [Asterisk-Users] Asterisk agi interface leaveszombie processes?   Scott Stingel
• [Asterisk-Users] Asterisk agi interface leaves zombie processes?   Scott Stingel
• [Asterisk-Users] Asterisk agi interface leaves zombie processes?   Scott Stingel
• [Asterisk-Users] limiting out going calls to a maximum duration   Scott Stingel
• [Asterisk-Users] Some questions about a potential usage scenario for asterisk   Scott Stingel
• [Asterisk-Users] Fix for Redhat 9 zombie AGI processes   Scott Stingel
• [Asterisk-Users] Fix for Redhat 9 zombie AGI processes   Scott Stingel
• [Asterisk-Users] Multiple E1 configuration question   Scott Stingel
• [Asterisk-Users] Autodialer / bulk dialer application   Scott Stingel
• [Asterisk-Users] Questions regarding CDR's   Scott Stingel
• [Asterisk-Users] MORE Questions regarding CDR's   Scott Stingel
• [Asterisk-Users] MORE Questions regarding CDR's   Scott Stingel
• [Asterisk-Users] Need a "trick" to generate calls   Scott Stingel
• [Asterisk-Users] Need a "trick" to generate calls   Scott Stingel
• [Asterisk-Users] "Frame rejections" on E1 trucks   Scott Stingel
• [Asterisk-Users] Need a "trick" to generate calls   Scott Stingel
• [Asterisk-Users] zaptel compile problems   Scott Stingel
• [Asterisk-Users] Asterisk stops responding   Scott Stingel
• [Asterisk-Users] Newbie - setup help   Scott Stingel
• [Asterisk-Users] Newbie - setup help   Scott Stingel
• [Asterisk-Users] Fix for Redhat 9 zombie AGI processes   Mark Street
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Martin Stubbs
• [Asterisk-Users] New firmware release for Grandstream Budgetone 100 series.   Grandstream Customer Support
• [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?   Linus Surguy
• [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?   Linus Surguy
• [Asterisk-Users] caller display in a telco environment.   Linus Surguy
• [Asterisk-Users] Known problem?   Linus Surguy
• [Asterisk-Users] Known problem?   Linus Surguy
• [Asterisk-Users] T1 to T1 on asterisk?   Linus Surguy
• [Asterisk-Users] D-link 102s and g723 parameters   John Sutter
• [Asterisk-Users] Help with linejack as a trunk?   John Sutter
• [Asterisk-Users] FAQs and Helpful URLs?   John Sutter
• [Asterisk-Users] Behind Firewalls, SonicWalls, etc..   John Sutter
• [Asterisk-Users] AgentCallbackLogin   TC
• [Asterisk-Users] Hardware for a Big PBX   TC
• [Asterisk-Users] ADSI and SoftKeys   TC
• [Asterisk-Users] segfaults with queue   TC
• [Asterisk-Users] segfaults with queue   TC
• [Asterisk-Users] ip phones and intercom/paging   TC
• [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank   TC
• [Asterisk-Users] Is Asterisk ready for "real" use?   TC
• [Asterisk-Users] Data calls through *   TC
• [Asterisk-Users] list proposal   James Taylor
• [Asterisk-Users] IP phone recommendation   Fabrice Tereszkiewicz
• [Asterisk-Users] Problems with TDM400P & X100P   Mark C. Thomas
• [Asterisk-Users] Chan_h323.so native?   Steven Thomas
• [Asterisk-Users] Chan_h323 one way audio   Steven Thomas
• [Asterisk-Users] No voice call from H.323-phone to SIP-phone   Steven Thomas
• [Asterisk-Users] Chan_h323 one way audio   Steven Thomas
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Steven Thomas
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Steven Thomas
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Steven Thomas
• [Asterisk-Users] Chan_h323 and a Cisco Gateway   Steven Thomas
• [Asterisk-Users] Chan_h323 and a Cisco Gateway   Steven Thomas
• [Asterisk-Users] Chan_h323 support for phone numbers via gateway?   Steven Thomas
• [Asterisk-Users] Chan_h323 does not seem to send the destimation number to gateway   Steven Thomas
• [Asterisk-Users] list proposal   James H. Thompson
• [Asterisk-Users] Is Asterisk ready for "real" use?   James H. Thompson
• [Asterisk-Users] Is Asterisk ready for "real" use?   James H. Thompson
• [Asterisk-Users] Is Asterisk ready for "real" use?   James H. Thompson
• [Asterisk-Users] Is Asterisk ready for "real" use?   James H. Thompson
• [Asterisk-Users] Asterisk SIP bug with Net2Phone   Mark Thompson
• [Asterisk-Users] Someone used ADIT 600 Channel Bank.   Anton Tinchev
• [Asterisk-Users] Is Asterisk ready for "real" use?   Anton Tinchev
• [Asterisk-Users] Re: Some questions about Asterisk and reliability   Anton Tinchev
• [Asterisk-Users] Re: Some questions about Asterisk and reliability   Anton Tinchev
• [Asterisk-Users] Conference + time limit   Anton Tinchev
• [Asterisk-Users] Which linux soft phone is best with asterisk.   Anton Tinchev
• [Asterisk-Users] Re: Some questions about Asterisk and reliability   Anton Tinchev
• [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?   John Todd
• [Asterisk-Users] bugs.digium.com   John Todd
• [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal   John Todd
• [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?   John Todd
• [Asterisk-Users] Newbie just starting out with *   John Todd
• [Asterisk-Users] Call Monitor   John Todd
• [Asterisk-Users] Syntax for hiding caller ID but still passing ANI?   John Todd
• [Asterisk-Users] Newbie just starting out with *   John Todd
• [Asterisk-Users] T-shirt ideas   John Todd
• [Asterisk-Users] R2 support   John Todd
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   John Todd
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   John Todd
• [Asterisk-Users] New SIP Phone   John Todd
• [Asterisk-Users] New SIP Phone   John Todd
• [Asterisk-Users] Feedback for Asterisk Handbook?   John Todd
• [Asterisk-Users] Sip Trunk config   John Todd
• [Asterisk-Users] Sip Trunk config / Least Cost Routing   John Todd
• [Asterisk-Users] Sip Trunk config / Least Cost Routing   John Todd
• [Asterisk-Users] ip phones and intercom/paging   John Todd
• [Asterisk-Users] UNIX command-line interaction with astdb   John Todd
• [Asterisk-Users] ip phones and intercom/paging   John Todd
• [Asterisk-Users] UNIX command-line interaction with astdb   John Todd
• [Asterisk-Users] To Switch or not to Switch... that is the question....   John Todd
• [Asterisk-Users] ANI/DNIS call routing   John Todd
• [Asterisk-Users] Fair comparison   John Todd
• [Asterisk-Users] Codec?   John Todd
• [Asterisk-Users] New-ish list of hardware phone vendors   John Todd
• [Asterisk-Users] Codec?   John Todd
• [Asterisk-Users] Running Asterisk behind NAT?   John Todd
• [Asterisk-Users] CLASS feature syntax   John Todd
• [Asterisk-Users] h extension seems to wipe variables?   John Todd
• [Asterisk-Users] Running Asterisk behind NAT?   John Todd
• [Asterisk-Users] How can I know if a user is busy or not connected?   John Todd
• [Asterisk-Users] How can I know if a user is busy or not connected?   John Todd
• [Asterisk-Users] BudgeTone NAT issues   John Todd
• [Asterisk-Users] pre-newbie - some basic questions...   John Todd
• [Asterisk-Users] Monitor application temporary hack   John Todd
• [Asterisk-Users] Cordless SIP phones   John Todd
• [Asterisk-Users] Cisco 7920 phone   John Todd
• [Asterisk-Users] 403 FORBIDDEN Help!   John Todd
• [Asterisk-Users] Malicious Call Trace   John Todd
• [Asterisk-Users] Monitor application temporary hack   John Todd
• [Asterisk-Users] MOH with SIP   John Todd
• [Asterisk-Users] Grandstream, SIP encryption   John Todd
• [Asterisk-Users] Monitor application temporary hack   John Todd
• [Asterisk-Users] Call transfer ATA186   John Todd
• [Asterisk-Users] Monitor application temporary hack   John Todd
• [Asterisk-Users] Grandstream, SIP encryption   John Todd
• [Asterisk-Users] Call transfer ATA186   John Todd
• [Asterisk-Users] Grandstream, SIP encryption   John Todd
• [Asterisk-Users] Grandstream, SIP encryption   John Todd
• [Asterisk-Users] Hiding and Changing Caller ID   John Todd
• [Asterisk-Users] Vonage locked ATA-186 question   John Todd
• [Asterisk-Users] Vonage locked ATA-186 question   John Todd
• [Asterisk-Users] SIP using which codec?   John Todd
• [Asterisk-Users] ATA-186 locking: implausible unlock method   John Todd
• [Asterisk-Users] Asterisk introductory talk: Portland, OR USA   John Todd
• [Asterisk-Users] VoIP dialtone?   John Todd
• [Asterisk-Users] RTP header compression?   John Todd
• [Asterisk-Users] VAD (silence suppression) on Asterisk   John Todd
• [Asterisk-Users] Strange happenings   John Todd
• [Asterisk-Users] 911, networks of * servers, etc. (was: VOIP Dialtone?)   John Todd
• [Asterisk-Users] Asterisk introductory talk: Portland, OR USA   John Todd
• [Asterisk-Users] VoIP dialtone?   John Todd
• [Asterisk-Users] Asterisk BoF: Boston, Sept _2[2-4] - interested?   John Todd
• [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)   John Todd
• [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)   John Todd
• [Asterisk-Users] RTP header compression?   John Todd
• [Asterisk-Users] Game time is over gang   John Todd
• [Asterisk-Users] Private ENUM examples?   John Todd
• [Asterisk-Users] Any way to distinguish between...   John Todd
• [Asterisk-Users] Cisco 7940 SIP   John Todd
• [Asterisk-Users] Is Asterisk ready for "real" use?   John Todd
• [Asterisk-Users] Intercom with Cisco SIP 796x phones?   John Todd
• [Asterisk-Users] SIP vs SCCP vs XML   John Todd
• [Asterisk-Users] Syncronize Monitored Calls   John Todd
• [Asterisk-Users] Intercom with Cisco SIP 796x phones?   John Todd
• [Asterisk-Users] Asterisk internal database access   John Todd
• [Asterisk-Users] Video conference apps/appliances?   John Todd
• [Asterisk-Users] sip and pix   John Todd
• [Asterisk-Users] Restricting concurrent SIP calls   John Todd
• [Asterisk-Users] Asterisk and Cisco 7960   John Todd
• [Asterisk-Users] VoicemailMain2, inband digits detection, rcf2833 digits detection (rtp issue, I think)   Jose Ildefonso Camargo Tolosa
• [Asterisk-Users] FWD SIP phone format=2, FWD call format=4, why?   Jose Ildefonso Camargo Tolosa
• [Asterisk-Users] Re: The Almighty X-Lite DTMF Problem (patch tested)   Jose Ildefonso Camargo Tolosa
• [Asterisk-Users] Re: Open source IP phone, maybe?   Jose Ildefonso Camargo Tolosa
• [Asterisk-Users] Why are FXO so expensive?   Samy Touati
• [Asterisk-Users] AgentCallbackLogin   The Traveller
• [Asterisk-Users] FWD-gateway prefix   The Traveller
• [Asterisk-Users] Queue timeouts   David C. Troy
• [Asterisk-Users] Compile problems   Keith Tucker
• [Asterisk-Users] Compile problems   Keith Tucker
• [Asterisk-Users] Compile problems   Keith Tucker
• [Asterisk-Users] Installation Problem   Keith Tucker
• Qwest CallerID question re:[Asterisk-Users] Caller ID problem   Ryan Tucker
• [Asterisk-Users] Faking Ring tone   Jay Tyndall
• [Asterisk-Users] Re: ATAs   Andrey Tyushkin
• [Asterisk-Users] Musiconhold interrupted sound   Michael Ulitskiy
• [Asterisk-Users] HELP!!!! Ringback oh323   Michael Ulitskiy
• [Asterisk-Users] Musiconhold interrupted sound   Michael Ulitskiy
• [Asterisk-Users] Musiconhold interrupted sound   Michael Ulitskiy
• [Asterisk-Users] Musiconhold interrupted sound   Michael Ulitskiy
• [Asterisk-Users] Musiconhold interrupted sound   Michael Ulitskiy
• [Asterisk-Users] 3xx SIP messages   Michael Ulitskiy
• [Asterisk-Users] 3xx SIP messages   Michael Ulitskiy
• [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers   Michael Ulitskiy
• [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers   Michael Ulitskiy
• [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers   Michael Ulitskiy
• [Asterisk-Users] New SIP Phone   Steve Underwood
• [Asterisk-Users] Semi-newbie question "Softswitch" and Asterisk - Is there a difference?   Steve Underwood
• [Asterisk-Users] Multiple E1 configuration question   Steve Underwood
• [Asterisk-Users] Fair comparison   Steve Underwood
• [Asterisk-Users] Open G.729A codec   Steve Underwood
• [Asterisk-Users] Open G.729A codec   Steve Underwood
• [Asterisk-Users] Open G.729A codec   Steve Underwood
• [Asterisk-Users] Open G.729A codec   Steve Underwood
• [Asterisk-Users] Open G.729A codec   Steve Underwood
• [Asterisk-Users] Open G.729A codec   Steve Underwood
• [Asterisk-Users] Monitor application temporary hack   Steve Underwood
• [Asterisk-Users] ADSI and SoftKeys   Jayson Vantuyl
• [Asterisk-Users] ADSI and SoftKeys   Jayson Vantuyl
• [Asterisk-Users] ADSI and SoftKeys   Jayson Vantuyl
• [Asterisk-Users] Asterisk Problem   Cristian Vasiliu
• [Asterisk-Users] PrePaid and IVR   Cristian Vasiliu
• [Asterisk-Users] 'System' application exit with error even if it performs the job as expected   Armand A. Verstappen
• [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected   Armand A. Verstappen
• [Asterisk-Users] 'System' application exit with error evenifit performs the job as expected   Armand A. Verstappen
• [Asterisk-Users] callwaiting in sip can't be disabled   Armand A. Verstappen
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Armand A. Verstappen
• [Asterisk-Users] X100P CallerID issue solved for my PSTN connection   Armand A. Verstappen
• [Asterisk-Users] FWD-gateway prefix   Armand A. Verstappen
• [Asterisk-Users] X-Lite <-> Snom200   Armand A. Verstappen
• [Asterisk-Users] chan_capi: Hanging channels - again   Armand A. Verstappen
• [Asterisk-Users] FWD-gateway prefix   Armand A. Verstappen
• [Asterisk-Users] AgentCallbackLogin   Armand A. Verstappen
• [Asterisk-Users] ADSI and SoftKeys   Armand A. Verstappen
• [Asterisk-Users] ADSI and SoftKeys   Armand A. Verstappen
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Armand A. Verstappen
• [Asterisk-Users] Newbie Issue   Armand A. Verstappen
• [Asterisk-Users] ADSI and SoftKeys   Armand A. Verstappen
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Armand A. Verstappen
• [Asterisk-Users] Iconnecthere   Armand A. Verstappen
• [Asterisk-Users] Iconnecthere   Armand A. Verstappen
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Armand A. Verstappen
• [Asterisk-Users] Iconnecthere   Armand A. Verstappen
• [Asterisk-Users] Iconnecthere   Armand A. Verstappen
• [Asterisk-Users] Asterisk Newbie ...   Armand A. Verstappen
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Armand A. Verstappen
• [Asterisk-Users] How to Asterisk   Armand A. Verstappen
• [Asterisk-Users] unsubscribe   Armand A. Verstappen
• Ethics (was Re: [Asterisk-Users] VIRUS ALERT)   Armand A. Verstappen
• [Asterisk-Users] Newbie Question / ISDN   Armand A. Verstappen
• [Asterisk-Users] weird error message with zaptel   Armand A. Verstappen
• [Asterisk-Users] Status of ISDN && DTMF (AFAIK): Please add corrections and comments   Armand A. Verstappen
• [Asterisk-Users] Sound files for internal functions in another place??   Armand A. Verstappen
• [Asterisk-Users] Why doesnt anyone reply me ?   Armand A. Verstappen
• [Asterisk-Users] ADSI Programs   Armand A. Verstappen
• [Asterisk-Users] Problem starting Asterisk after abnormal shutdown   Armand A. Verstappen
• [Asterisk-Users] sample configs / load module failure   Armand A. Verstappen
• [Asterisk-Users] Question About BRI Cards   Armand A. Verstappen
• [Asterisk-Users] RTP session traversing Asterisk server...   Ricardo Villa
• [Asterisk-Users] g.729 licenses do not release when used in Voicemail   Ricardo Villa
• [Asterisk-Users] Wierd Message   Ricardo Villa
• [Asterisk-Users] Wierd Message   Ricardo Villa
• [Asterisk-Users] Wierd Message   Ricardo Villa
• [Asterisk-Users] Wierd Message   Ricardo Villa
• [Asterisk-Users] Sip Trunk config   Ricardo Villa
• [Asterisk-Users] list proposal   Ricardo Villa
• [Asterisk-Users] Question About BRI Cards   Gustavo Villaran
• [Asterisk-Users] Question About BRI Cards   Gustavo Villaran
• [Asterisk-Users] Grandstream firmware update.   John Vozza
• [Asterisk-Users] Grandstream firmware update DMTF Payload Type   John Vozza
• [Asterisk-Users] Grandstream firmware update DMTF Payload Type   John Vozza
• [Asterisk-Users] SetVar on sample.call   DUSTIN WILDES
• [Asterisk-Users] Is Asterisk ready for "real" use?   DUSTIN WILDES
• [Asterisk-Users] SetVar on sample.call   DUSTIN WILDES
• [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #1133 - 18 msgs   Wallingford, Ted
• [Asterisk-Users] voicemail.conf emailbody fromaddress   Mark Wehberg
• [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal   Dave Weis
• [Asterisk-Users] Is Asterisk ready for "real" use?   Dave Weis
• [Asterisk-Users] Is Asterisk ready for "real" use?   Dave Weis
• [Asterisk-Users] Multi-extension buttoned phones   Dave Weis
• [Asterisk-Users] Provisioning CO lines   Dave Weis
• [Asterisk-Users] DBSaveTree & DBLoadTree   Dave Weis
• [Asterisk-Users] DBSaveTree & DBLoadTree   Dave Weis
• [Asterisk-Users] Sip Trunk config   Wade Weppler
• [Asterisk-Users] Sip Trunk config / Least Cost Routing   Wade Weppler
• [Asterisk-Users] ADSI and SoftKeys   Wade Weppler
• [Asterisk-Users] Sip Trunk config / Least Cost Routing   Wade Weppler
• [Asterisk-Users] Sip Trunk config / Least Cost Routing   Wade Weppler
• [Asterisk-Users] Call routing question   Wade Weppler
• [Asterisk-Users] Pops   Wade Weppler
• [Asterisk-Users] sound problem   Wade Weppler
• [Asterisk-Users] 403 FORBIDDEN Help!   Wade Weppler
• [Asterisk-Users] zaptel does not compile anymore   Wade Weppler
• [Asterisk-Users] MWI question   Wade Weppler
• [Asterisk-Users] GS geek info   Wade Weppler
• [Asterisk-Users] T100P/ TSU 600 installation problem   Wade Weppler
• [Asterisk-Users] AGI accountcode.   Wade J. Weppler
• [Asterisk-Users] Compile problems   Wade J. Weppler
• [Asterisk-Users] Asterisk and Cisco 7960   Ben Wern
• [Asterisk-Users] Asterisk and Cisco 7960   Ben Wern
• [Asterisk-Users] Asterisk and Cisco 7960   Ben Wern
• [Asterisk-Users] Extension handling.   Brian West
• [Asterisk-Users] Grandstream Budgettone 100 & 102   Brian West
• [Asterisk-Users] Webalizer for CDR logs....   Brian West
• [Asterisk-Users] SIP app_queue   Brian West
• [Asterisk-Users] SIP app_queue   Brian West
• [Asterisk-Users] SIP app_queue   Brian West
• [Asterisk-Users] callwaiting in sip can't be disabled   Brian West
• [Asterisk-Users] GSM codec   Brian West
• [Asterisk-Users] Channel banks, etc.   Brian West
• [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?   Brian West
• [Asterisk-Users] Festival 1.4.3   Brian West
• [Asterisk-Users] MWI bug ?   Brian West
• [Asterisk-Users] h323 and cvs one way audio   Brian West
• [Asterisk-Users] FYI: G723.1 Licensing Prices   Brian West
• [Asterisk-Users] queue / agent documentation   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] help please with single t1 configuration   Brian West
• [Asterisk-Users] Need help with installation of H323 chanel driver   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] Ring while on phone   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] IP phone recommendation   Brian West
• [Asterisk-Users] Working with FWD, IPTel, SIPPhone?   Brian West
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Brian West
• [Asterisk-Users] Mixing audio from Music on Hold and IVR   Brian West
• [Asterisk-Users] h extension seems to wipe variables?   Brian West
• [Asterisk-Users] Extension and phone management best practices??   Brian West
• [Asterisk-Users] Extension and phone management best practices??   Brian West
• [Asterisk-Users] Receiving iaxtel calls   Brian West
• [Asterisk-Users] '#' doesn't work for me   Brian West
• [Asterisk-Users] '#' doesn't work for me   Brian West
• [Asterisk-Users] '#' doesn't work for me   Brian West
• [Asterisk-Users] DTMF SIP   Brian West
• [Asterisk-Users] Can I runAsterisk remotely from telnet session?   Brian West
• [Asterisk-Users] Asterisk, quicknet and codecs (G729, 7231 Question)   Brian West
• [Asterisk-Users] Voicemail2 patches   Brian West
• [Asterisk-Users] Voicemail cliping digits via sip   Brian West
• [Asterisk-Users] SIP agent logging into queue?   Brian West
• [Asterisk-Users] Monitor application temporary hack   Brian West
• [Asterisk-Users] Monitor application temporary hack   Brian West
• [Asterisk-Users] Voicemail2 vs. Voicemail   Brian West
• [Asterisk-Users] Monitor application temporary hack   Brian West
• [Asterisk-Users] Voicemail2 vs. Voicemail   Brian West
• [Asterisk-Users] Monitor application temporary hack   Brian West
• [Asterisk-Users] Call transfer ATA186   Brian West
• [Asterisk-Users] Monitor application temporary hack   Brian West
• [Asterisk-Users] How Do I disable faststart?   Brian West
• [Asterisk-Users] RE: IAX, Asterisk, GSM,SPEEX and ILBC (fwd)   Brian West
• [Asterisk-Users] MusicOnHold   Brian West
• [Asterisk-Users] Vonage locked ATA-186 question   Brian West
• [Asterisk-Users] SIP QUESTIO   Brian West
• [Asterisk-Users] SIP QUESTION   Brian West
• [Asterisk-Users] Problem with * server and FWD   Brian West
• [Asterisk-Users] PrePaid and IVR   Brian West
• [Asterisk-Users] PrePaid and IVR   Brian West
• [Asterisk-Users] reload not working   Brian West
• [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for Chan_h323   Brian West
• [Asterisk-Users] Is Asterisk ready for "real" use?   Brian West
• [Asterisk-Users] VAD (silence suppression) on Asterisk   Brian West
• [Asterisk-Users] IAX <> IAX trunking... DP cache?   Brian West
• [Asterisk-Users] Is Asterisk ready for "real" use?   Brian West
• [Asterisk-Users] VAD (silence suppression) on Asterisk   Brian West
• [Asterisk-Users] VoIP dialtone?   Brian West
• [Asterisk-Users] VoIP dialtone?   Brian West
• [Asterisk-Users] VoIP dialtone?   Brian West
• [Asterisk-Users] Hardware question   Brian West
• [Asterisk-Users] VIRUS ALERT   Brian West
• [Asterisk-Users] VoIP dialtone?   Brian West
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   Brian West
• [Asterisk-Users] Newbie Question / ISDN   Brian West
• [Asterisk-Users] VoIP dialtone?   Brian West
• [Asterisk-Users] Voicemail2 and RFC2833 DTMF   Brian West
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   Brian West
• [Asterisk-Users] VoIP dialtone?   Brian West
• [Asterisk-Users] sox and wav to gsm conversion quality issue   Brian West
• [Asterisk-Users] Game time is over gang   Brian West
• [Asterisk-Users] Intresting.. hrm   Brian West
• [Asterisk-Users] Intresting.. hrm   Brian West
• [Asterisk-Users] Intresting Vonage story...   Brian West
• [Asterisk-Users] Intresting.. hrm   Brian West
• [Asterisk-Users] Private ENUM examples?   Brian West
• [Asterisk-Users] Private ENUM examples?   Brian West
• [Asterisk-Users] Private ENUM examples?   Brian West
• [Asterisk-Users] GS on ebay...   Brian West
• [Asterisk-Users] Why doesnt anyone reply me ?   Brian West
• [Asterisk-Users] Syncronize Monitored Calls   Brian West
• [Asterisk-Users] 0 out of voicemail to different secretaries   Brian West
• [Asterisk-Users] Syncronize Monitored Calls   Brian West
• [Asterisk-Users] Private ENUM examples?   Brian West
• [Asterisk-Users] 0 out of voicemail to different secretaries   Brian West
• [Asterisk-Users] Syncronize Monitored Calls   Brian West
• [Asterisk-Users] Chan_h323 and a Cisco Gateway   Brian West
• [Asterisk-Users] H.323 channel problems   Brian West
• [Asterisk-Users] SIP and ECHO   Brian West
• [Asterisk-Users] Festival and Asterisk   Brian West
• [Asterisk-Users] Festival and Asterisk   Brian West
• [Asterisk-Users] sample configs   Brian West
• [Asterisk-Users] GS on ebay...   Brian West
• [Asterisk-Users] ENUM, iax,iax2 and h323?   Brian West
• [Asterisk-Users] FWD-gateway prefix   Chris Wetemans
• [Asterisk-Users] Some questions about a potential usage scenario for asterisk   Howard White
• [Asterisk-Users] Background Noise   Howard White
• [Asterisk-Users] SIP with an iptables fiewall   Eric Wieling
• [Asterisk-Users] Patch - transfer with two rather than one #   Eric Wieling
• [Asterisk-Users] Windows IAX soft phone   Eric Wieling
• [Asterisk-Users] FYI: G723.1 Licensing Prices   Eric Wieling
• [Asterisk-Users] How to determine line signalling?   Eric Wieling
• [Asterisk-Users] g729 problems   Eric Wieling
• [Asterisk-Users] Windows IAX soft phone   Eric Wieling
• [Asterisk-Users] Asterisk Newbie ...   Eric Wieling
• [Asterisk-Users] Fax Handled   Eric Wieling
• [Asterisk-Users] Get faxed you faxing faxer!   Eric Wieling
• [Asterisk-Users] Working with FWD, IPTel, SIPPhone?   Eric Wieling
• [Asterisk-Users] Using Asterisk with FWD through NAT   Eric Wieling
• [Asterisk-Users] Open G.729A codec   Eric Wieling
• [Asterisk-Users] Asterisk and AT&T 964 phones...   Eric Wieling
• [Asterisk-Users] CLASS feature syntax   Eric Wieling
• [Asterisk-Users] Receiving iaxtel calls   Eric Wieling
• [Asterisk-Users] Cordless SIP phones   Eric Wieling
• [Asterisk-Users] Voicemail2 vs. Voicemail   Eric Wieling
• [Asterisk-Users] G723 (was SIP using which codec?)   Eric Wieling
• [Asterisk-Users] G723 (was SIP using which codec?)   Eric Wieling
• [Asterisk-Users] Sending dtmf over an ougoing call from asterisk   Eric Wieling
• [Asterisk-Users] Re: Some questions about Asterisk and reliability   Eric Wieling
• [Asterisk-Users] Re: Some questions about Asterisk and reliability   Eric Wieling
• [Asterisk-Users] DTMF tones not long enough on out going call s   Eric Wieling
• [Asterisk-Users] Intresting.. hrm   Eric Wieling
• [Asterisk-Users] Why doesnt anyone reply me ?   Eric Wieling
• [Asterisk-Users] Is Asterisk ready for "real" use?   Eric Wieling
• [Asterisk-Users] Why doesnt anyone reply me ?   Eric Wieling
• [Asterisk-Users] SIP vs SCCP vs XML   Eric Wieling
• [Asterisk-Users] Zhone Zplex 10 units   Kent Williams
• [Asterisk-Users] Some questions about a potential usage scenario for asterisk   Dave Wilson
• [Asterisk-Users] Some questions about a potential usage scenario for asterisk   Dave Wilson
• [Asterisk-Users] Some questions about a potential usage scenario for asterisk   Dave Wilson
• [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?   Dave Wilson
• [Asterisk-Users] How do i configure so an incoming call triggers an http request?   Dave Wilson
• [Asterisk-Users] How do i configure so an incoming call triggers an http request?   Dave Wilson
• [Asterisk-Users] Don't know how to calculate timelen   Dave Wilson
• [Asterisk-Users] Don't know how to calculate timelen   Dave Wilson
• [Asterisk-Users] Why are FXO so expensive?   Anthony Wood
• [Asterisk-Users] Get faxed you faxing faxer!   Anthony Wood
• [Asterisk-Users] Hardware question   Anthony Wood
• [Asterisk-Users] PRI CallerID problem   Anthony Wood
• [Asterisk-Users] Some questions about a potential usage scenario for asterisk   Simon Woodhead
• [Asterisk-Users] GSM file format   Simon Woodhead
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Simon Woodhead
• [Asterisk-Users] E-mail (still version 1) is not being Delivered   Simon Woodhead
• [Asterisk-Users] app_queue, fewestcalls and leastrecent logic   Simon Woodhead
• [Asterisk-Users] snom100(with latest firmware) screeching noise when doing transfers,   Anton Yurchenko
• [Asterisk-Users] phonecore in gnophone?   Anton Yurchenko
• [Asterisk-Users] Mutex problem in sip?   Alex Zarubin
• [Asterisk-Users] dtmf/audio before going offhook   Alex Zarubin
• [Asterisk-Users] Why are FXO so expensive?   Peter Zeltins
• [Asterisk-Users] Unable to detect process 2 frames   Peter Zeltins
• [Asterisk-Users] Hangup after a Timeout   Lei Zhang
• [Asterisk-Users] Grandstream Budgetone Defective Units   William Zhang
• [Asterisk-Users] Grandstream and CallerID not working   William Zhang
• [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)   Tom Zimnicki
• [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)   Tom Zimnicki
• [Asterisk-Users] SIP and MD5 logins.   andrewg at felinemenace.org
• [Asterisk-Users] Voicetronix V4PCI   andrewg at felinemenace.org
• [Asterisk-Users] conference authorization   andrewg at felinemenace.org
• [Asterisk-Users] conference authorization   andrewg at felinemenace.org
• [Asterisk-Users] Installation Problem   andrewg at felinemenace.org
• [Asterisk-Users] gnophone connection   ashish
• [Asterisk-Users] (no subject)   ashishagrawal at hfcl.com
• [Asterisk-Users] Version number not updating after CVS   asterisk-users at sensecompute.com
• [Asterisk-Users] h323 over NAT   ayaz
• [Asterisk-Users] No voice call from H.323-phone to SIP-phone   shong ching
• [Asterisk-Users] Caller ID and Call Waiting.   countrdd at adelphia.net
• [Asterisk-Users] ip phones and intercom/paging   cwitte
• [Asterisk-Users] pre-newbie - some basic questions...   d.redmore at att.net
• [Asterisk-Users] Grandstream Budgetone   denon
• [Asterisk-Users] Analog lines   denon
• AW: [Asterisk-Users] ADSI Programs   denon
• [Asterisk-Users] Snome-200 with Asterisk   denzel-infotechs
• [Asterisk-Users] Snome-200 with Asterisk   denzel-infotechs
• [Asterisk-Users] Questions regarding CDR's   diana
• [Asterisk-Users] Chan_h323.so native?   diana
• [Asterisk-Users] Festival 1.4.3   digium.paluszak at spamex.com
• [Asterisk-Users] Dialed Number Identification in analog hunt group   dpobanz at hastingsutilities.com
• [Asterisk-Users] AGI accountcode.   duncan
• [Asterisk-Users] limiting out going calls to a maximum duration   duncan
• [Asterisk-Users] sip and pix   duncan
• [Asterisk-Users] AGI Channel Status   jerk face
• [Asterisk-Users] AGI Channel Status   jerk face
• [Asterisk-Users] T100P/ TSU 600 installation problem   jerk face
• [Asterisk-Users] T100P/ TSU 600 installation problem   jerk face
• [Asterisk-Users] T100P/ TSU 600 installation problem   jerk face
• [Asterisk-Users] RE: T100P/ TSU 600 installation problem   jerk face
• [Asterisk-Users] RE: T100P/ TSU 600 installation problem   jerk face
• [Asterisk-Users] ADSI Programs   jerk face
• AW: [Asterisk-Users] ADSI Programs   jerk face
• [Asterisk-Users] Call Waiting and Call Parking Together??   firedude at shorelinuxsolutions.com
• [Asterisk-Users] ip phones and intercom/paging   firedude at shorelinuxsolutions.com
• [Asterisk-Users] Call Waiting and Call Parking Together??   firedude at shorelinuxsolutions.com
• [Asterisk-Users] Call Waiting and Call Parking Together??   firedude at shorelinuxsolutions.com
• [Asterisk-Users] Killing runaway PBX   firedude at shorelinuxsolutions.com
• [Asterisk-Users] Windows Messenger   firedude at shorelinuxsolutions.com
• [Asterisk-Users] Grandstream Budgetone   firedude at shorelinuxsolutions.com
• [Asterisk-Users] Voicemail2 patches   firedude at shorelinuxsolutions.com
• [Asterisk-Users] Has anyone got sip/IAX working behind a firewall?   firedude at shorelinuxsolutions.com
• [Asterisk-Users] MOH with SIP   firedude at shorelinuxsolutions.com
• [Asterisk-Users] screeching and MOH bleed on PRI   firedude at shorelinuxsolutions.com
• [Asterisk-Users] [OT] Virus propagation by asterisk user member.   firedude at shorelinuxsolutions.com
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   frank.barthe at prescom.net
• [Asterisk-Users] current status of i4l and dtmf stuff   pedro bulach gapski
• [Asterisk-Users] Status of ISDN && DTMF (AFAIK): Please add corrections and comments   pedro bulach gapski
• [Asterisk-Users] X100P delivery   isamar at isamarmaia.org
• [Asterisk-Users] Problems to install X100P   isamar at isamarmaia.org
• [Asterisk-Users] Problems to install X100P   isamar at isamarmaia.org
• [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-Digium Partner   isamar at isamarmaia.org
• [Asterisk-Users] Newbie Issue   jeff.gunther at intalgent.com
• [Asterisk-Users] X100P Ringing/Answering   jeff.gunther at intalgent.com
• [Asterisk-Users] X100P Ringing/Answering   jeff.gunther at intalgent.com
• [Asterisk-Users] Background Noise   jeff.gunther at intalgent.com
• [Asterisk-Users] Background Noise   jeff.gunther at intalgent.com
• [Asterisk-Users] HELP!!!!   jorge at redenlaces.com.mx
• [Asterisk-Users] HELP!!!! Ringback oh323   jorge at redenlaces.com.mx
• [Asterisk-Users] Segmentation fault   jorge at redenlaces.com.mx
• [Asterisk-Users] ztdummy & usb-ohci?   justin at vergeworks.com
• [Asterisk-Users] Asterisk launch on boot   justin at vergeworks.com
• [Asterisk-Users] Minnesota PUC: Phone rules apply to VoIP   justin at vergeworks.com
• [Asterisk-Users] call waiting   lists
• [Asterisk-Users] call waiting   lists
• [Asterisk-Users] Fax Detection?   lists
• [Asterisk-Users] call waiting   lists
• [Asterisk-Users] VIRUS ALERT   marrandy
• Ethics (was Re: [Asterisk-Users] VIRUS ALERT)   marrandy
• [Asterisk-Users] Multi-extension buttoned phones   marrandy
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   marrandy
• [Asterisk-Users] BudgeTone Firmware 1.0.3.78?   marrandy
• [Asterisk-Users] Grandstream firmware update DMTF Payload Type   marrandy
• [Asterisk-Users] Astrisk admin and webmin question   mawali at news.icns.com
• [Asterisk-Users] Can I runAsterisk remotely from telnet session?   mawali at news.icns.com
• [Asterisk-Users] Asterisk, quicknet and codecs (G729, 7231 Question)   mawali at news.icns.com
• [Asterisk-Users] Can Asterisk use hardware codecs?   mawali at news.icns.com
• [Asterisk-Users] H323/SIP gatekeeper   mawali at news.icns.com
• [Asterisk-Users] [OT] Virus propagation by asterisk user member.   mawali at news.icns.com
• [Asterisk-Users] Oen source IP phone, maybe?   mawali at news.icns.com
• [Asterisk-Users] chan_h323 and packetization period   mawali at news.icns.com
• [Asterisk-Users] chan_h323 and packetization period   mawali at news.icns.com
• [Asterisk-Users] Intresting.. hrm   mawali at news.icns.com
• [Asterisk-Users] Intresting.. hrm   mawali at news.icns.com
• [Asterisk-Users] zaptel compile problems   mawali at news.icns.com
• [Asterisk-Users] Little patch for packetization in G729 in chan_h323   mawali at news.icns.com
• [Asterisk-Users] Little patch for packetization in G729 in chan_h323   mawali at news.icns.com
• [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-Digium Partner   mawali at news.icns.com
• [Asterisk-Users] New gastman clone + what else?   michiel at betel.nl
• [Asterisk-Users] cdr_mysql   pat munis
• [Asterisk-Users] Filling PHP Variable from EXTENSION in AGI   romsun p
• [Asterisk-Users] * newbie: overhead paging and nbsd   erik petersen
• [Asterisk-Users] How to Asterisk   prakashmodak_74
• [Asterisk-Users] Asterisk Problem   prakashmodak_74
• [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)   r6henso6 at swbell.net
• [Asterisk-Users] conference authorization   radan
• [Asterisk-Users] conference authorization   radan
• [Asterisk-Users] have everyone use asterisk to setup a network as iconnecthere or quicknet?   romulo eugenio ribeiro
• RES: [Asterisk-Users] Three way calling on outgoing FXO line   romulo eugenio ribeiro
• [Asterisk-Users] newbie question - devices   santiago
• [Asterisk-Users] newbie question - devices   santiago
• [Asterisk-Users] usrobotics modem and pstn   santiago
• [Asterisk-Users] sound problem   santiago
• [Asterisk-Users] Background messages while waiting for pick-up   Peer Oliver schmidt
• [Asterisk-Users] iax.conf / Registration rejected   Peer Oliver schmidt
• [Asterisk-Users] iax.conf / Registration rejected   Peer Oliver schmidt
• [Asterisk-Users] iax.conf / Registration rejected   Peer Oliver schmidt
• [Asterisk-Users] URI for dialing   Peer Oliver schmidt
• [Asterisk-Users] Hangup after a Timeout   surajee at infotechs.lk
• [Asterisk-Users] Warning Messages   surajee at infotechs.lk
• [Asterisk-Users] Hiding and Changing Caller ID   surajee at infotechs.lk
• [Asterisk-Users] unsubscribe   swarren at hexicom.com
• [Asterisk-Users] sample configs / load module failure   ted at indexc.com
• [Asterisk-Users] dialogic D/41 ESC with asterisk   kaku ustaad
• [Asterisk-Users] (ATTENTION ) Quicknet Lan jack and phone jack   kaku ustaad
• [Asterisk-Users] Webmin and incoming call recording   kaku ustaad
• [Asterisk-Users] Why doesnt anyone reply me ?   kaku ustaad
• [Asterisk-Users] Why doesnt anyone reply me ?   kaku ustaad
• [Asterisk-Users] Why doesnt anyone reply me ?   kaku ustaad
• [Asterisk-Users] SendDtmf   wasim at convergence.com.pk
• [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers   wasim at convergence.com.pk
• [Asterisk-Users] Windows IAX soft phone   wasim at convergence.com.pk
• [Asterisk-Users] Running Asterisk behind NAT?   wasim at convergence.com.pk
• [Asterisk-Users] call routing based on dnis   wasim at convergence.com.pk
• [Asterisk-Users] Video conference apps/appliances?   wasim at convergence.com.pk
• [Asterisk-Users] Newbie question   wasim at convergence.com.pk
• [Asterisk-Users] Unified Messaging Support ?   wasim at convergence.com.pk